diff options
| author | Chih-hung Hsieh <chh@google.com> | 2015-12-01 17:07:48 +0000 |
|---|---|---|
| committer | android-build-merger <android-build-merger@google.com> | 2015-12-01 17:07:48 +0000 |
| commit | a4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch) | |
| tree | 672a185b294789cf991f385c3e395dd63bea9063 /webrtc/modules/audio_coding/codecs/ilbc | |
| parent | 3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff) | |
| parent | fe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff) | |
| download | webrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz | |
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579
* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
WIP: Changes after merge commit 'cb3f9bd'
Make the nonlinear beamformer steerable
Utilize bitrate above codec max to protect video.
Enable VP9 internal resize by default.
Filter overlapping RTP header extensions.
Make VCMEncodedFrameCallback const.
MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
Remove redudant encoder rate calls.
Create isolate files for nonparallel tests.
Register header extensions in RtpRtcpObserver to avoid log spam.
Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
ACM: Move NACK functionality inside NetEq
Fix chromium-style warnings in webrtc/sound/.
Create a 'webrtc_nonparallel_tests' target.
Update scalability structure data according to updates in the RTP payload profile.
audio_coding: rename interface -> include
Rewrote perform_action_on_all_files to be parallell.
Update reference indices according to updates in the RTP payload profile.
Disable P2PTransport...TestFailoverControlledSide on Memcheck
pass clangcl compile options to ignore warnings in gflags.cc
...
Diffstat (limited to 'webrtc/modules/audio_coding/codecs/ilbc')
150 files changed, 12359 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/OWNERS b/webrtc/modules/audio_coding/codecs/ilbc/OWNERS new file mode 100644 index 0000000000..3ee6b4bf5f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/OWNERS @@ -0,0 +1,5 @@ + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* diff --git a/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c new file mode 100644 index 0000000000..263749ad2a --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c @@ -0,0 +1,80 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_AbsQuant.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "abs_quant_loop.h" + + +/*----------------------------------------------------------------* + * predictive noise shaping encoding of scaled start state + * (subrutine for WebRtcIlbcfix_StateSearch) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_AbsQuant( + IlbcEncoder *iLBCenc_inst, + /* (i) Encoder instance */ + iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax + and idxVec, uses state_first as + input) */ + int16_t *in, /* (i) vector to encode */ + int16_t *weightDenum /* (i) denominator of synthesis filter */ + ) { + int16_t *syntOut; + size_t quantLen[2]; + + /* Stack based */ + int16_t syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS]; + int16_t in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER]; + int16_t *in_weighted = &in_weightedVec[LPC_FILTERORDER]; + + /* Initialize the buffers */ + WebRtcSpl_MemSetW16(syntOutBuf, 0, LPC_FILTERORDER+STATE_SHORT_LEN_30MS); + syntOut = &syntOutBuf[LPC_FILTERORDER]; + /* Start with zero state */ + WebRtcSpl_MemSetW16(in_weightedVec, 0, LPC_FILTERORDER); + + /* Perform the quantization loop in two sections of length quantLen[i], + where the perceptual weighting filter is updated at the subframe + border */ + + if (iLBC_encbits->state_first) { + quantLen[0]=SUBL; + quantLen[1]=iLBCenc_inst->state_short_len-SUBL; + } else { + quantLen[0]=iLBCenc_inst->state_short_len-SUBL; + quantLen[1]=SUBL; + } + + /* Calculate the weighted residual, switch perceptual weighting + filter at the subframe border */ + WebRtcSpl_FilterARFastQ12( + in, in_weighted, + weightDenum, LPC_FILTERORDER+1, quantLen[0]); + WebRtcSpl_FilterARFastQ12( + &in[quantLen[0]], &in_weighted[quantLen[0]], + &weightDenum[LPC_FILTERORDER+1], LPC_FILTERORDER+1, quantLen[1]); + + WebRtcIlbcfix_AbsQuantLoop( + syntOut, + in_weighted, + weightDenum, + quantLen, + iLBC_encbits->idxVec); + +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h new file mode 100644 index 0000000000..5154534524 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h @@ -0,0 +1,39 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_AbsQuant.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * predictive noise shaping encoding of scaled start state + * (subrutine for WebRtcIlbcfix_StateSearch) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_AbsQuant( + IlbcEncoder *iLBCenc_inst, + /* (i) Encoder instance */ + iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax + and idxVec, uses state_first as + input) */ + int16_t *in, /* (i) vector to encode */ + int16_t *weightDenum /* (i) denominator of synthesis filter */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c new file mode 100644 index 0000000000..4b76453446 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c @@ -0,0 +1,87 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_AbsQuantLoop.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "sort_sq.h" + +void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN, + int16_t *weightDenumIN, size_t *quantLenIN, + int16_t *idxVecIN ) { + size_t k1, k2; + int16_t index; + int32_t toQW32; + int32_t toQ32; + int16_t tmp16a; + int16_t xq; + + int16_t *syntOut = syntOutIN; + int16_t *in_weighted = in_weightedIN; + int16_t *weightDenum = weightDenumIN; + size_t *quantLen = quantLenIN; + int16_t *idxVec = idxVecIN; + + for(k1=0;k1<2;k1++) { + for(k2=0;k2<quantLen[k1];k2++){ + + /* Filter to get the predicted value */ + WebRtcSpl_FilterARFastQ12( + syntOut, syntOut, + weightDenum, LPC_FILTERORDER+1, 1); + + /* the quantizer */ + toQW32 = (int32_t)(*in_weighted) - (int32_t)(*syntOut); + + toQ32 = (((int32_t)toQW32)<<2); + + if (toQ32 > 32767) { + toQ32 = (int32_t) 32767; + } else if (toQ32 < -32768) { + toQ32 = (int32_t) -32768; + } + + /* Quantize the state */ + if (toQW32<(-7577)) { + /* To prevent negative overflow */ + index=0; + } else if (toQW32>8151) { + /* To prevent positive overflow */ + index=7; + } else { + /* Find the best quantization index + (state_sq3Tbl is in Q13 and toQ is in Q11) + */ + WebRtcIlbcfix_SortSq(&xq, &index, + (int16_t)toQ32, + WebRtcIlbcfix_kStateSq3, 8); + } + + /* Store selected index */ + (*idxVec++) = index; + + /* Compute decoded sample and update of the prediction filter */ + tmp16a = ((WebRtcIlbcfix_kStateSq3[index] + 2 ) >> 2); + + *syntOut = (int16_t) (tmp16a + (int32_t)(*in_weighted) - toQW32); + + syntOut++; in_weighted++; + } + /* Update perceptual weighting filter at subframe border */ + weightDenum += 11; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h new file mode 100644 index 0000000000..c8bf67575f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_AbsQuantLoop.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * predictive noise shaping encoding of scaled start state + * (subrutine for WebRtcIlbcfix_StateSearch) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN, + int16_t *weightDenumIN, size_t *quantLenIN, + int16_t *idxVecIN); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc new file mode 100644 index 0000000000..ba6284f33d --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc @@ -0,0 +1,56 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h" + +#include "webrtc/base/checks.h" +#include "webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h" + +namespace webrtc { + +AudioDecoderIlbc::AudioDecoderIlbc() { + WebRtcIlbcfix_DecoderCreate(&dec_state_); + WebRtcIlbcfix_Decoderinit30Ms(dec_state_); +} + +AudioDecoderIlbc::~AudioDecoderIlbc() { + WebRtcIlbcfix_DecoderFree(dec_state_); +} + +bool AudioDecoderIlbc::HasDecodePlc() const { + return true; +} + +int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + RTC_DCHECK_EQ(sample_rate_hz, 8000); + int16_t temp_type = 1; // Default is speech. + int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) { + return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames); +} + +void AudioDecoderIlbc::Reset() { + WebRtcIlbcfix_Decoderinit30Ms(dec_state_); +} + +size_t AudioDecoderIlbc::Channels() const { + return 1; +} + +} // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc new file mode 100644 index 0000000000..065dc06817 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc @@ -0,0 +1,153 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h" + +#include <cstring> +#include <limits> +#include "webrtc/base/checks.h" +#include "webrtc/common_types.h" +#include "webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h" + +namespace webrtc { + +namespace { + +const int kSampleRateHz = 8000; + +AudioEncoderIlbc::Config CreateConfig(const CodecInst& codec_inst) { + AudioEncoderIlbc::Config config; + config.frame_size_ms = codec_inst.pacsize / 8; + config.payload_type = codec_inst.pltype; + return config; +} + +} // namespace + +// static +const size_t AudioEncoderIlbc::kMaxSamplesPerPacket; + +bool AudioEncoderIlbc::Config::IsOk() const { + return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 || + frame_size_ms == 60) && + static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <= + kMaxSamplesPerPacket; +} + +AudioEncoderIlbc::AudioEncoderIlbc(const Config& config) + : config_(config), + num_10ms_frames_per_packet_( + static_cast<size_t>(config.frame_size_ms / 10)), + encoder_(nullptr) { + Reset(); +} + +AudioEncoderIlbc::AudioEncoderIlbc(const CodecInst& codec_inst) + : AudioEncoderIlbc(CreateConfig(codec_inst)) {} + +AudioEncoderIlbc::~AudioEncoderIlbc() { + RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_)); +} + +size_t AudioEncoderIlbc::MaxEncodedBytes() const { + return RequiredOutputSizeBytes(); +} + +int AudioEncoderIlbc::SampleRateHz() const { + return kSampleRateHz; +} + +int AudioEncoderIlbc::NumChannels() const { + return 1; +} + +size_t AudioEncoderIlbc::Num10MsFramesInNextPacket() const { + return num_10ms_frames_per_packet_; +} + +size_t AudioEncoderIlbc::Max10MsFramesInAPacket() const { + return num_10ms_frames_per_packet_; +} + +int AudioEncoderIlbc::GetTargetBitrate() const { + switch (num_10ms_frames_per_packet_) { + case 2: case 4: + // 38 bytes per frame of 20 ms => 15200 bits/s. + return 15200; + case 3: case 6: + // 50 bytes per frame of 30 ms => (approx) 13333 bits/s. + return 13333; + default: + FATAL(); + } +} + +AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal( + uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) { + RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes()); + + // Save timestamp if starting a new packet. + if (num_10ms_frames_buffered_ == 0) + first_timestamp_in_buffer_ = rtp_timestamp; + + // Buffer input. + std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_, + audio, + kSampleRateHz / 100 * sizeof(audio[0])); + + // If we don't yet have enough buffered input for a whole packet, we're done + // for now. + if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) { + return EncodedInfo(); + } + + // Encode buffered input. + RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_); + num_10ms_frames_buffered_ = 0; + const int output_len = WebRtcIlbcfix_Encode( + encoder_, + input_buffer_, + kSampleRateHz / 100 * num_10ms_frames_per_packet_, + encoded); + RTC_CHECK_GE(output_len, 0); + EncodedInfo info; + info.encoded_bytes = static_cast<size_t>(output_len); + RTC_DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes()); + info.encoded_timestamp = first_timestamp_in_buffer_; + info.payload_type = config_.payload_type; + return info; +} + +void AudioEncoderIlbc::Reset() { + if (encoder_) + RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_)); + RTC_CHECK(config_.IsOk()); + RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_)); + const int encoder_frame_size_ms = config_.frame_size_ms > 30 + ? config_.frame_size_ms / 2 + : config_.frame_size_ms; + RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms)); + num_10ms_frames_buffered_ = 0; +} + +size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const { + switch (num_10ms_frames_per_packet_) { + case 2: return 38; + case 3: return 50; + case 4: return 2 * 38; + case 6: return 2 * 50; + default: FATAL(); + } +} + +} // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c new file mode 100644 index 0000000000..1a3735fc3d --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c @@ -0,0 +1,63 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_AugmentedCbCorr.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "augmented_cb_corr.h" + +void WebRtcIlbcfix_AugmentedCbCorr( + int16_t *target, /* (i) Target vector */ + int16_t *buffer, /* (i) Memory buffer */ + int16_t *interpSamples, /* (i) buffer with + interpolated samples */ + int32_t *crossDot, /* (o) The cross correlation between + the target and the Augmented + vector */ + size_t low, /* (i) Lag to start from (typically + 20) */ + size_t high, /* (i) Lag to end at (typically 39) */ + int scale) /* (i) Scale factor to use for + the crossDot */ +{ + size_t lagcount; + size_t ilow; + int16_t *targetPtr; + int32_t *crossDotPtr; + int16_t *iSPtr=interpSamples; + + /* Calculate the correlation between the target and the + interpolated codebook. The correlation is calculated in + 3 sections with the interpolated part in the middle */ + crossDotPtr=crossDot; + for (lagcount=low; lagcount<=high; lagcount++) { + + ilow = lagcount - 4; + + /* Compute dot product for the first (lagcount-4) samples */ + (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale); + + /* Compute dot product on the interpolated samples */ + (*crossDotPtr) += WebRtcSpl_DotProductWithScale(target+ilow, iSPtr, 4, scale); + targetPtr = target + lagcount; + iSPtr += lagcount-ilow; + + /* Compute dot product for the remaining samples */ + (*crossDotPtr) += WebRtcSpl_DotProductWithScale(targetPtr, buffer-lagcount, SUBL-lagcount, scale); + crossDotPtr++; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h new file mode 100644 index 0000000000..c5c408880e --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h @@ -0,0 +1,41 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_AugmentedCbCorr.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Calculate correlation between target and Augmented codebooks + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_AugmentedCbCorr( + int16_t *target, /* (i) Target vector */ + int16_t *buffer, /* (i) Memory buffer */ + int16_t *interpSamples, /* (i) buffer with + interpolated samples */ + int32_t *crossDot, /* (o) The cross correlation between + the target and the Augmented + vector */ + size_t low, /* (i) Lag to start from (typically + 20) */ + size_t high, /* (i) Lag to end at (typically 39 */ + int scale); /* (i) Scale factor to use for the crossDot */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c b/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c new file mode 100644 index 0000000000..b02abab3f2 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c @@ -0,0 +1,42 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_BwExpand.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * lpc bandwidth expansion + *---------------------------------------------------------------*/ + +/* The output is in the same domain as the input */ +void WebRtcIlbcfix_BwExpand( + int16_t *out, /* (o) the bandwidth expanded lpc coefficients */ + int16_t *in, /* (i) the lpc coefficients before bandwidth + expansion */ + int16_t *coef, /* (i) the bandwidth expansion factor Q15 */ + int16_t length /* (i) the length of lpc coefficient vectors */ + ) { + int i; + + out[0] = in[0]; + for (i = 1; i < length; i++) { + /* out[i] = coef[i] * in[i] with rounding. + in[] and out[] are in Q12 and coef[] is in Q15 + */ + out[i] = (int16_t)((coef[i] * in[i] + 16384) >> 15); + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.h b/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.h new file mode 100644 index 0000000000..b3b16d5d69 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_BwExpand.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * lpc bandwidth expansion + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_BwExpand( + int16_t *out, /* (o) the bandwidth expanded lpc coefficients */ + int16_t *in, /* (i) the lpc coefficients before bandwidth + expansion */ + int16_t *coef, /* (i) the bandwidth expansion factor Q15 */ + int16_t length /* (i) the length of lpc coefficient vectors */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c new file mode 100644 index 0000000000..cacf3ace28 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c @@ -0,0 +1,67 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbConstruct.c + +******************************************************************/ + +#include "defines.h" +#include "gain_dequant.h" +#include "get_cd_vec.h" + +/*----------------------------------------------------------------* + * Construct decoded vector from codebook and gains. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_CbConstruct( + int16_t *decvector, /* (o) Decoded vector */ + int16_t *index, /* (i) Codebook indices */ + int16_t *gain_index, /* (i) Gain quantization indices */ + int16_t *mem, /* (i) Buffer for codevector construction */ + size_t lMem, /* (i) Length of buffer */ + size_t veclen /* (i) Length of vector */ + ){ + size_t j; + int16_t gain[CB_NSTAGES]; + /* Stack based */ + int16_t cbvec0[SUBL]; + int16_t cbvec1[SUBL]; + int16_t cbvec2[SUBL]; + int32_t a32; + int16_t *gainPtr; + + /* gain de-quantization */ + + gain[0] = WebRtcIlbcfix_GainDequant(gain_index[0], 16384, 0); + gain[1] = WebRtcIlbcfix_GainDequant(gain_index[1], gain[0], 1); + gain[2] = WebRtcIlbcfix_GainDequant(gain_index[2], gain[1], 2); + + /* codebook vector construction and construction of total vector */ + + /* Stack based */ + WebRtcIlbcfix_GetCbVec(cbvec0, mem, (size_t)index[0], lMem, veclen); + WebRtcIlbcfix_GetCbVec(cbvec1, mem, (size_t)index[1], lMem, veclen); + WebRtcIlbcfix_GetCbVec(cbvec2, mem, (size_t)index[2], lMem, veclen); + + gainPtr = &gain[0]; + for (j=0;j<veclen;j++) { + a32 = (*gainPtr++) * cbvec0[j]; + a32 += (*gainPtr++) * cbvec1[j]; + a32 += (*gainPtr) * cbvec2[j]; + gainPtr -= 2; + decvector[j] = (int16_t)((a32 + 8192) >> 14); + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h new file mode 100644 index 0000000000..b676ef97ad --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbConstruct.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Construct decoded vector from codebook and gains. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_CbConstruct( + int16_t *decvector, /* (o) Decoded vector */ + int16_t *index, /* (i) Codebook indices */ + int16_t *gain_index, /* (i) Gain quantization indices */ + int16_t *mem, /* (i) Buffer for codevector construction */ + size_t lMem, /* (i) Length of buffer */ + size_t veclen /* (i) Length of vector */ + ); + + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c new file mode 100644 index 0000000000..6ad2f8eb7a --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c @@ -0,0 +1,79 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbMemEnergy.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "cb_mem_energy_calc.h" + +/*----------------------------------------------------------------* + * Function WebRtcIlbcfix_CbMemEnergy computes the energy of all + * the vectors in the codebook memory that will be used in the + * following search for the best match. + *----------------------------------------------------------------*/ + +void WebRtcIlbcfix_CbMemEnergy( + size_t range, + int16_t *CB, /* (i) The CB memory (1:st section) */ + int16_t *filteredCB, /* (i) The filtered CB memory (2:nd section) */ + size_t lMem, /* (i) Length of the CB memory */ + size_t lTarget, /* (i) Length of the target vector */ + int16_t *energyW16, /* (o) Energy in the CB vectors */ + int16_t *energyShifts, /* (o) Shift value of the energy */ + int scale, /* (i) The scaling of all energy values */ + size_t base_size /* (i) Index to where energy values should be stored */ + ) { + int16_t *ppi, *ppo, *pp; + int32_t energy, tmp32; + + /* Compute the energy and store it in a vector. Also the + * corresponding shift values are stored. The energy values + * are reused in all three stages. */ + + /* Calculate the energy in the first block of 'lTarget' sampels. */ + ppi = CB+lMem-lTarget-1; + ppo = CB+lMem-1; + + pp=CB+lMem-lTarget; + energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale); + + /* Normalize the energy and store the number of shifts */ + energyShifts[0] = (int16_t)WebRtcSpl_NormW32(energy); + tmp32 = energy << energyShifts[0]; + energyW16[0] = (int16_t)(tmp32 >> 16); + + /* Compute the energy of the rest of the cb memory + * by step wise adding and subtracting the next + * sample and the last sample respectively. */ + WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, 0); + + /* Next, precompute the energy values for the filtered cb section */ + energy=0; + pp=filteredCB+lMem-lTarget; + + energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale); + + /* Normalize the energy and store the number of shifts */ + energyShifts[base_size] = (int16_t)WebRtcSpl_NormW32(energy); + tmp32 = energy << energyShifts[base_size]; + energyW16[base_size] = (int16_t)(tmp32 >> 16); + + ppi = filteredCB + lMem - 1 - lTarget; + ppo = filteredCB + lMem - 1; + + WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, base_size); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h new file mode 100644 index 0000000000..6da2f43c0b --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbMemEnergy.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_ + +void WebRtcIlbcfix_CbMemEnergy( + size_t range, + int16_t *CB, /* (i) The CB memory (1:st section) */ + int16_t *filteredCB, /* (i) The filtered CB memory (2:nd section) */ + size_t lMem, /* (i) Length of the CB memory */ + size_t lTarget, /* (i) Length of the target vector */ + int16_t *energyW16, /* (o) Energy in the CB vectors */ + int16_t *energyShifts, /* (o) Shift value of the energy */ + int scale, /* (i) The scaling of all energy values */ + size_t base_size /* (i) Index to where energy values should be stored */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c new file mode 100644 index 0000000000..acd6b9c118 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c @@ -0,0 +1,67 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbMemEnergyAugmentation.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +void WebRtcIlbcfix_CbMemEnergyAugmentation( + int16_t *interpSamples, /* (i) The interpolated samples */ + int16_t *CBmem, /* (i) The CB memory */ + int scale, /* (i) The scaling of all energy values */ + size_t base_size, /* (i) Index to where energy values should be stored */ + int16_t *energyW16, /* (o) Energy in the CB vectors */ + int16_t *energyShifts /* (o) Shift value of the energy */ + ){ + int32_t energy, tmp32; + int16_t *ppe, *pp, *interpSamplesPtr; + int16_t *CBmemPtr; + size_t lagcount; + int16_t *enPtr=&energyW16[base_size-20]; + int16_t *enShPtr=&energyShifts[base_size-20]; + int32_t nrjRecursive; + + CBmemPtr = CBmem+147; + interpSamplesPtr = interpSamples; + + /* Compute the energy for the first (low-5) noninterpolated samples */ + nrjRecursive = WebRtcSpl_DotProductWithScale( CBmemPtr-19, CBmemPtr-19, 15, scale); + ppe = CBmemPtr - 20; + + for (lagcount=20; lagcount<=39; lagcount++) { + + /* Update the energy recursively to save complexity */ + nrjRecursive += (*ppe * *ppe) >> scale; + ppe--; + energy = nrjRecursive; + + /* interpolation */ + energy += WebRtcSpl_DotProductWithScale(interpSamplesPtr, interpSamplesPtr, 4, scale); + interpSamplesPtr += 4; + + /* Compute energy for the remaining samples */ + pp = CBmemPtr - lagcount; + energy += WebRtcSpl_DotProductWithScale(pp, pp, SUBL-lagcount, scale); + + /* Normalize the energy and store the number of shifts */ + (*enShPtr) = (int16_t)WebRtcSpl_NormW32(energy); + tmp32 = energy << *enShPtr; + *enPtr = (int16_t)(tmp32 >> 16); + enShPtr++; + enPtr++; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h new file mode 100644 index 0000000000..594ba5f01b --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h @@ -0,0 +1,31 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbMemEnergyAugmentation.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_ + +void WebRtcIlbcfix_CbMemEnergyAugmentation( + int16_t *interpSamples, /* (i) The interpolated samples */ + int16_t *CBmem, /* (i) The CB memory */ + int scale, /* (i) The scaling of all energy values */ + size_t base_size, /* (i) Index to where energy values should be stored */ + int16_t *energyW16, /* (o) Energy in the CB vectors */ + int16_t *energyShifts /* (o) Shift value of the energy */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c new file mode 100644 index 0000000000..f2415febc0 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c @@ -0,0 +1,65 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbMemEnergyCalc.c + +******************************************************************/ + +#include "defines.h" + +/* Compute the energy of the rest of the cb memory + * by step wise adding and subtracting the next + * sample and the last sample respectively */ +void WebRtcIlbcfix_CbMemEnergyCalc( + int32_t energy, /* (i) input start energy */ + size_t range, /* (i) number of iterations */ + int16_t *ppi, /* (i) input pointer 1 */ + int16_t *ppo, /* (i) input pointer 2 */ + int16_t *energyW16, /* (o) Energy in the CB vectors */ + int16_t *energyShifts, /* (o) Shift value of the energy */ + int scale, /* (i) The scaling of all energy values */ + size_t base_size /* (i) Index to where energy values should be stored */ + ) +{ + size_t j; + int16_t shft; + int32_t tmp; + int16_t *eSh_ptr; + int16_t *eW16_ptr; + + + eSh_ptr = &energyShifts[1+base_size]; + eW16_ptr = &energyW16[1+base_size]; + + for (j = 0; j + 1 < range; j++) { + + /* Calculate next energy by a +/- + operation on the edge samples */ + tmp = (*ppi) * (*ppi) - (*ppo) * (*ppo); + energy += tmp >> scale; + energy = WEBRTC_SPL_MAX(energy, 0); + + ppi--; + ppo--; + + /* Normalize the energy into a int16_t and store + the number of shifts */ + + shft = (int16_t)WebRtcSpl_NormW32(energy); + *eSh_ptr++ = shft; + + tmp = energy << shft; + *eW16_ptr++ = (int16_t)(tmp >> 16); + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h new file mode 100644 index 0000000000..2991869dd9 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbMemEnergyCalc.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_ + +void WebRtcIlbcfix_CbMemEnergyCalc( + int32_t energy, /* (i) input start energy */ + size_t range, /* (i) number of iterations */ + int16_t *ppi, /* (i) input pointer 1 */ + int16_t *ppo, /* (i) input pointer 2 */ + int16_t *energyW16, /* (o) Energy in the CB vectors */ + int16_t *energyShifts, /* (o) Shift value of the energy */ + int scale, /* (i) The scaling of all energy values */ + size_t base_size /* (i) Index to where energy values should be stored */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c new file mode 100644 index 0000000000..be94951449 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c @@ -0,0 +1,403 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbSearch.c + +******************************************************************/ + +#include "defines.h" +#include "gain_quant.h" +#include "filtered_cb_vecs.h" +#include "constants.h" +#include "cb_mem_energy.h" +#include "interpolate_samples.h" +#include "cb_mem_energy_augmentation.h" +#include "cb_search_core.h" +#include "energy_inverse.h" +#include "augmented_cb_corr.h" +#include "cb_update_best_index.h" +#include "create_augmented_vec.h" + +/*----------------------------------------------------------------* + * Search routine for codebook encoding and gain quantization. + *----------------------------------------------------------------*/ + +void WebRtcIlbcfix_CbSearch( + IlbcEncoder *iLBCenc_inst, + /* (i) the encoder state structure */ + int16_t *index, /* (o) Codebook indices */ + int16_t *gain_index, /* (o) Gain quantization indices */ + int16_t *intarget, /* (i) Target vector for encoding */ + int16_t *decResidual,/* (i) Decoded residual for codebook construction */ + size_t lMem, /* (i) Length of buffer */ + size_t lTarget, /* (i) Length of vector */ + int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */ + size_t block /* (i) the subblock number */ + ) { + size_t i, range; + int16_t ii, j, stage; + int16_t *pp; + int16_t tmp; + int scale; + int16_t bits, temp1, temp2; + size_t base_size; + int32_t codedEner, targetEner; + int16_t gains[CB_NSTAGES+1]; + int16_t *cb_vecPtr; + size_t indexOffset, sInd, eInd; + int32_t CritMax=0; + int16_t shTotMax=WEBRTC_SPL_WORD16_MIN; + size_t bestIndex=0; + int16_t bestGain=0; + size_t indexNew; + int16_t CritNewSh; + int32_t CritNew; + int32_t *cDotPtr; + size_t noOfZeros; + int16_t *gainPtr; + int32_t t32, tmpW32; + int16_t *WebRtcIlbcfix_kGainSq5_ptr; + /* Stack based */ + int16_t CBbuf[CB_MEML+LPC_FILTERORDER+CB_HALFFILTERLEN]; + int32_t cDot[128]; + int32_t Crit[128]; + int16_t targetVec[SUBL+LPC_FILTERORDER]; + int16_t cbvectors[CB_MEML + 1]; /* Adding one extra position for + Coverity warnings. */ + int16_t codedVec[SUBL]; + int16_t interpSamples[20*4]; + int16_t interpSamplesFilt[20*4]; + int16_t energyW16[CB_EXPAND*128]; + int16_t energyShifts[CB_EXPAND*128]; + int16_t *inverseEnergy=energyW16; /* Reuse memory */ + int16_t *inverseEnergyShifts=energyShifts; /* Reuse memory */ + int16_t *buf = &CBbuf[LPC_FILTERORDER]; + int16_t *target = &targetVec[LPC_FILTERORDER]; + int16_t *aug_vec = (int16_t*)cDot; /* length [SUBL], reuse memory */ + + /* Determine size of codebook sections */ + + base_size=lMem-lTarget+1; + if (lTarget==SUBL) { + base_size=lMem-19; + } + + /* weighting of the CB memory */ + noOfZeros=lMem-WebRtcIlbcfix_kFilterRange[block]; + WebRtcSpl_MemSetW16(&buf[-LPC_FILTERORDER], 0, noOfZeros+LPC_FILTERORDER); + WebRtcSpl_FilterARFastQ12( + decResidual+noOfZeros, buf+noOfZeros, + weightDenum, LPC_FILTERORDER+1, WebRtcIlbcfix_kFilterRange[block]); + + /* weighting of the target vector */ + WEBRTC_SPL_MEMCPY_W16(&target[-LPC_FILTERORDER], buf+noOfZeros+WebRtcIlbcfix_kFilterRange[block]-LPC_FILTERORDER, LPC_FILTERORDER); + WebRtcSpl_FilterARFastQ12( + intarget, target, + weightDenum, LPC_FILTERORDER+1, lTarget); + + /* Store target, towards the end codedVec is calculated as + the initial target minus the remaining target */ + WEBRTC_SPL_MEMCPY_W16(codedVec, target, lTarget); + + /* Find the highest absolute value to calculate proper + vector scale factor (so that it uses 12 bits) */ + temp1 = WebRtcSpl_MaxAbsValueW16(buf, lMem); + temp2 = WebRtcSpl_MaxAbsValueW16(target, lTarget); + + if ((temp1>0)&&(temp2>0)) { + temp1 = WEBRTC_SPL_MAX(temp1, temp2); + scale = WebRtcSpl_GetSizeInBits((uint32_t)(temp1 * temp1)); + } else { + /* temp1 or temp2 is negative (maximum was -32768) */ + scale = 30; + } + + /* Scale to so that a mul-add 40 times does not overflow */ + scale = scale - 25; + scale = WEBRTC_SPL_MAX(0, scale); + + /* Compute energy of the original target */ + targetEner = WebRtcSpl_DotProductWithScale(target, target, lTarget, scale); + + /* Prepare search over one more codebook section. This section + is created by filtering the original buffer with a filter. */ + WebRtcIlbcfix_FilteredCbVecs(cbvectors, buf, lMem, WebRtcIlbcfix_kFilterRange[block]); + + range = WebRtcIlbcfix_kSearchRange[block][0]; + + if(lTarget == SUBL) { + /* Create the interpolated samples and store them for use in all stages */ + + /* First section, non-filtered half of the cb */ + WebRtcIlbcfix_InterpolateSamples(interpSamples, buf, lMem); + + /* Second section, filtered half of the cb */ + WebRtcIlbcfix_InterpolateSamples(interpSamplesFilt, cbvectors, lMem); + + /* Compute the CB vectors' energies for the first cb section (non-filtered) */ + WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamples, buf, + scale, 20, energyW16, energyShifts); + + /* Compute the CB vectors' energies for the second cb section (filtered cb) */ + WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors, scale, + base_size + 20, energyW16, + energyShifts); + + /* Compute the CB vectors' energies and store them in the vector + * energyW16. Also the corresponding shift values are stored. The + * energy values are used in all three stages. */ + WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem, + lTarget, energyW16+20, energyShifts+20, scale, base_size); + + } else { + /* Compute the CB vectors' energies and store them in the vector + * energyW16. Also the corresponding shift values are stored. The + * energy values are used in all three stages. */ + WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem, + lTarget, energyW16, energyShifts, scale, base_size); + + /* Set the energy positions 58-63 and 122-127 to zero + (otherwise they are uninitialized) */ + WebRtcSpl_MemSetW16(energyW16+range, 0, (base_size-range)); + WebRtcSpl_MemSetW16(energyW16+range+base_size, 0, (base_size-range)); + } + + /* Calculate Inverse Energy (energyW16 is already normalized + and will contain the inverse energy in Q29 after this call */ + WebRtcIlbcfix_EnergyInverse(energyW16, base_size*CB_EXPAND); + + /* The gain value computed in the previous stage is used + * as an upper limit to what the next stage gain value + * is allowed to be. In stage 0, 16384 (1.0 in Q14) is used as + * the upper limit. */ + gains[0] = 16384; + + for (stage=0; stage<CB_NSTAGES; stage++) { + + /* Set up memories */ + range = WebRtcIlbcfix_kSearchRange[block][stage]; + + /* initialize search measures */ + CritMax=0; + shTotMax=-100; + bestIndex=0; + bestGain=0; + + /* loop over lags 40+ in the first codebook section, full search */ + cb_vecPtr = buf+lMem-lTarget; + + /* Calculate all the cross correlations (augmented part of CB) */ + if (lTarget==SUBL) { + WebRtcIlbcfix_AugmentedCbCorr(target, buf+lMem, + interpSamples, cDot, + 20, 39, scale); + cDotPtr=&cDot[20]; + } else { + cDotPtr=cDot; + } + /* Calculate all the cross correlations (main part of CB) */ + WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, range, scale, -1); + + /* Adjust the search range for the augmented vectors */ + if (lTarget==SUBL) { + range=WebRtcIlbcfix_kSearchRange[block][stage]+20; + } else { + range=WebRtcIlbcfix_kSearchRange[block][stage]; + } + + indexOffset=0; + + /* Search for best index in this part of the vector */ + WebRtcIlbcfix_CbSearchCore( + cDot, range, stage, inverseEnergy, + inverseEnergyShifts, Crit, + &indexNew, &CritNew, &CritNewSh); + + /* Update the global best index and the corresponding gain */ + WebRtcIlbcfix_CbUpdateBestIndex( + CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew+indexOffset], + inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset], + &CritMax, &shTotMax, &bestIndex, &bestGain); + + sInd = ((CB_RESRANGE >> 1) > bestIndex) ? + 0 : (bestIndex - (CB_RESRANGE >> 1)); + eInd=sInd+CB_RESRANGE; + if (eInd>=range) { + eInd=range-1; + sInd=eInd-CB_RESRANGE; + } + + range = WebRtcIlbcfix_kSearchRange[block][stage]; + + if (lTarget==SUBL) { + i=sInd; + if (sInd<20) { + WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors + lMem, + interpSamplesFilt, cDot, sInd + 20, + WEBRTC_SPL_MIN(39, (eInd + 20)), scale); + i=20; + cDotPtr = &cDot[20 - sInd]; + } else { + cDotPtr = cDot; + } + + cb_vecPtr = cbvectors+lMem-20-i; + + /* Calculate the cross correlations (main part of the filtered CB) */ + WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, + eInd - i + 1, scale, -1); + + } else { + cDotPtr = cDot; + cb_vecPtr = cbvectors+lMem-lTarget-sInd; + + /* Calculate the cross correlations (main part of the filtered CB) */ + WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, + eInd - sInd + 1, scale, -1); + + } + + /* Adjust the search range for the augmented vectors */ + indexOffset=base_size+sInd; + + /* Search for best index in this part of the vector */ + WebRtcIlbcfix_CbSearchCore( + cDot, eInd-sInd+1, stage, inverseEnergy+indexOffset, + inverseEnergyShifts+indexOffset, Crit, + &indexNew, &CritNew, &CritNewSh); + + /* Update the global best index and the corresponding gain */ + WebRtcIlbcfix_CbUpdateBestIndex( + CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew], + inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset], + &CritMax, &shTotMax, &bestIndex, &bestGain); + + index[stage] = (int16_t)bestIndex; + + + bestGain = WebRtcIlbcfix_GainQuant(bestGain, + (int16_t)WEBRTC_SPL_ABS_W16(gains[stage]), stage, &gain_index[stage]); + + /* Extract the best (according to measure) codebook vector + Also adjust the index, so that the augmented vectors are last. + Above these vectors were first... + */ + + if(lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) { + + if((size_t)index[stage]<base_size) { + pp=buf+lMem-lTarget-index[stage]; + } else { + pp=cbvectors+lMem-lTarget- + index[stage]+base_size; + } + + } else { + + if ((size_t)index[stage]<base_size) { + if (index[stage]>=20) { + /* Adjust index and extract vector */ + index[stage]-=20; + pp=buf+lMem-lTarget-index[stage]; + } else { + /* Adjust index and extract vector */ + index[stage]+=(int16_t)(base_size-20); + + WebRtcIlbcfix_CreateAugmentedVec(index[stage]-base_size+40, + buf+lMem, aug_vec); + pp = aug_vec; + + } + } else { + + if ((index[stage] - base_size) >= 20) { + /* Adjust index and extract vector */ + index[stage]-=20; + pp=cbvectors+lMem-lTarget- + index[stage]+base_size; + } else { + /* Adjust index and extract vector */ + index[stage]+=(int16_t)(base_size-20); + WebRtcIlbcfix_CreateAugmentedVec(index[stage]-2*base_size+40, + cbvectors+lMem, aug_vec); + pp = aug_vec; + } + } + } + + /* Subtract the best codebook vector, according + to measure, from the target vector */ + + WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain), + (int32_t)8192, (int16_t)14, lTarget); + + /* record quantized gain */ + gains[stage+1] = bestGain; + + } /* end of Main Loop. for (stage=0;... */ + + /* Calculte the coded vector (original target - what's left) */ + for (i=0;i<lTarget;i++) { + codedVec[i]-=target[i]; + } + + /* Gain adjustment for energy matching */ + codedEner = WebRtcSpl_DotProductWithScale(codedVec, codedVec, lTarget, scale); + + j=gain_index[0]; + + temp1 = (int16_t)WebRtcSpl_NormW32(codedEner); + temp2 = (int16_t)WebRtcSpl_NormW32(targetEner); + + if(temp1 < temp2) { + bits = 16 - temp1; + } else { + bits = 16 - temp2; + } + + tmp = (int16_t)((gains[1] * gains[1]) >> 14); + + targetEner = (int16_t)WEBRTC_SPL_SHIFT_W32(targetEner, -bits) * tmp; + + tmpW32 = ((int32_t)(gains[1]-1))<<1; + + /* Pointer to the table that contains + gain_sq5TblFIX * gain_sq5TblFIX in Q14 */ + gainPtr=(int16_t*)WebRtcIlbcfix_kGainSq5Sq+gain_index[0]; + temp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(codedEner, -bits); + + WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[j]; + + /* targetEner and codedEner are in Q(-2*scale) */ + for (ii=gain_index[0];ii<32;ii++) { + + /* Change the index if + (codedEnergy*gainTbl[i]*gainTbl[i])<(targetEn*gain[0]*gain[0]) AND + gainTbl[i] < 2*gain[0] + */ + + t32 = temp1 * *gainPtr; + t32 = t32 - targetEner; + if (t32 < 0) { + if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) { + j=ii; + WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[ii]; + } + } + gainPtr++; + } + gain_index[0]=j; + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h new file mode 100644 index 0000000000..ed1580c09d --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbSearch.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_ + +void WebRtcIlbcfix_CbSearch( + IlbcEncoder *iLBCenc_inst, + /* (i) the encoder state structure */ + int16_t *index, /* (o) Codebook indices */ + int16_t *gain_index, /* (o) Gain quantization indices */ + int16_t *intarget, /* (i) Target vector for encoding */ + int16_t *decResidual,/* (i) Decoded residual for codebook construction */ + size_t lMem, /* (i) Length of buffer */ + size_t lTarget, /* (i) Length of vector */ + int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */ + size_t block /* (i) the subblock number */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c new file mode 100644 index 0000000000..fafa39f69b --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c @@ -0,0 +1,113 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbSearchCore.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +void WebRtcIlbcfix_CbSearchCore( + int32_t *cDot, /* (i) Cross Correlation */ + size_t range, /* (i) Search range */ + int16_t stage, /* (i) Stage of this search */ + int16_t *inverseEnergy, /* (i) Inversed energy */ + int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy + with the offset 2*16-29 */ + int32_t *Crit, /* (o) The criteria */ + size_t *bestIndex, /* (o) Index that corresponds to + maximum criteria (in this + vector) */ + int32_t *bestCrit, /* (o) Value of critera for the + chosen index */ + int16_t *bestCritSh) /* (o) The domain of the chosen + criteria */ +{ + int32_t maxW32, tmp32; + int16_t max, sh, tmp16; + size_t i; + int32_t *cDotPtr; + int16_t cDotSqW16; + int16_t *inverseEnergyPtr; + int32_t *critPtr; + int16_t *inverseEnergyShiftPtr; + + /* Don't allow negative values for stage 0 */ + if (stage==0) { + cDotPtr=cDot; + for (i=0;i<range;i++) { + *cDotPtr=WEBRTC_SPL_MAX(0, (*cDotPtr)); + cDotPtr++; + } + } + + /* Normalize cDot to int16_t, calculate the square of cDot and store the upper int16_t */ + maxW32 = WebRtcSpl_MaxAbsValueW32(cDot, range); + + sh = (int16_t)WebRtcSpl_NormW32(maxW32); + cDotPtr = cDot; + inverseEnergyPtr = inverseEnergy; + critPtr = Crit; + inverseEnergyShiftPtr=inverseEnergyShift; + max=WEBRTC_SPL_WORD16_MIN; + + for (i=0;i<range;i++) { + /* Calculate cDot*cDot and put the result in a int16_t */ + tmp32 = *cDotPtr << sh; + tmp16 = (int16_t)(tmp32 >> 16); + cDotSqW16 = (int16_t)(((int32_t)(tmp16)*(tmp16))>>16); + + /* Calculate the criteria (cDot*cDot/energy) */ + *critPtr = cDotSqW16 * *inverseEnergyPtr; + + /* Extract the maximum shift value under the constraint + that the criteria is not zero */ + if ((*critPtr)!=0) { + max = WEBRTC_SPL_MAX((*inverseEnergyShiftPtr), max); + } + + inverseEnergyPtr++; + inverseEnergyShiftPtr++; + critPtr++; + cDotPtr++; + } + + /* If no max shifts still at initialization value, set shift to zero */ + if (max==WEBRTC_SPL_WORD16_MIN) { + max = 0; + } + + /* Modify the criterias, so that all of them use the same Q domain */ + critPtr=Crit; + inverseEnergyShiftPtr=inverseEnergyShift; + for (i=0;i<range;i++) { + /* Guarantee that the shift value is less than 16 + in order to simplify for DSP's (and guard against >31) */ + tmp16 = WEBRTC_SPL_MIN(16, max-(*inverseEnergyShiftPtr)); + + (*critPtr)=WEBRTC_SPL_SHIFT_W32((*critPtr),-tmp16); + critPtr++; + inverseEnergyShiftPtr++; + } + + /* Find the index of the best value */ + *bestIndex = WebRtcSpl_MaxIndexW32(Crit, range); + *bestCrit = Crit[*bestIndex]; + + /* Calculate total shifts of this criteria */ + *bestCritSh = 32 - 2*sh + max; + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h new file mode 100644 index 0000000000..9648cf29d3 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h @@ -0,0 +1,40 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbSearchCore.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_ + +#include "defines.h" + +void WebRtcIlbcfix_CbSearchCore( + int32_t *cDot, /* (i) Cross Correlation */ + size_t range, /* (i) Search range */ + int16_t stage, /* (i) Stage of this search */ + int16_t *inverseEnergy, /* (i) Inversed energy */ + int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy + with the offset 2*16-29 */ + int32_t *Crit, /* (o) The criteria */ + size_t *bestIndex, /* (o) Index that corresponds to + maximum criteria (in this + vector) */ + int32_t *bestCrit, /* (o) Value of critera for the + chosen index */ + int16_t *bestCritSh); /* (o) The domain of the chosen + criteria */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c new file mode 100644 index 0000000000..fc27ea9f6c --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c @@ -0,0 +1,88 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbUpdateBestIndex.c + +******************************************************************/ + +#include "defines.h" +#include "cb_update_best_index.h" +#include "constants.h" + +void WebRtcIlbcfix_CbUpdateBestIndex( + int32_t CritNew, /* (i) New Potentially best Criteria */ + int16_t CritNewSh, /* (i) Shift value of above Criteria */ + size_t IndexNew, /* (i) Index of new Criteria */ + int32_t cDotNew, /* (i) Cross dot of new index */ + int16_t invEnergyNew, /* (i) Inversed energy new index */ + int16_t energyShiftNew, /* (i) Energy shifts of new index */ + int32_t *CritMax, /* (i/o) Maximum Criteria (so far) */ + int16_t *shTotMax, /* (i/o) Shifts of maximum criteria */ + size_t *bestIndex, /* (i/o) Index that corresponds to + maximum criteria */ + int16_t *bestGain) /* (i/o) Gain in Q14 that corresponds + to maximum criteria */ +{ + int16_t shOld, shNew, tmp16; + int16_t scaleTmp; + int32_t gainW32; + + /* Normalize the new and old Criteria to the same domain */ + if (CritNewSh>(*shTotMax)) { + shOld=WEBRTC_SPL_MIN(31,CritNewSh-(*shTotMax)); + shNew=0; + } else { + shOld=0; + shNew=WEBRTC_SPL_MIN(31,(*shTotMax)-CritNewSh); + } + + /* Compare the two criterias. If the new one is better, + calculate the gain and store this index as the new best one + */ + + if ((CritNew >> shNew) > (*CritMax >> shOld)) { + + tmp16 = (int16_t)WebRtcSpl_NormW32(cDotNew); + tmp16 = 16 - tmp16; + + /* Calculate the gain in Q14 + Compensate for inverseEnergyshift in Q29 and that the energy + value was stored in a int16_t (shifted down 16 steps) + => 29-14+16 = 31 */ + + scaleTmp = -energyShiftNew-tmp16+31; + scaleTmp = WEBRTC_SPL_MIN(31, scaleTmp); + + gainW32 = ((int16_t)WEBRTC_SPL_SHIFT_W32(cDotNew, -tmp16) * invEnergyNew) >> + scaleTmp; + + /* Check if criteria satisfies Gain criteria (max 1.3) + if it is larger set the gain to 1.3 + (slightly different from FLP version) + */ + if (gainW32>21299) { + *bestGain=21299; + } else if (gainW32<-21299) { + *bestGain=-21299; + } else { + *bestGain=(int16_t)gainW32; + } + + *CritMax=CritNew; + *shTotMax=CritNewSh; + *bestIndex = IndexNew; + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h new file mode 100644 index 0000000000..a20fa38b2e --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CbUpdateBestIndex.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_ + +#include "defines.h" + +void WebRtcIlbcfix_CbUpdateBestIndex( + int32_t CritNew, /* (i) New Potentially best Criteria */ + int16_t CritNewSh, /* (i) Shift value of above Criteria */ + size_t IndexNew, /* (i) Index of new Criteria */ + int32_t cDotNew, /* (i) Cross dot of new index */ + int16_t invEnergyNew, /* (i) Inversed energy new index */ + int16_t energyShiftNew, /* (i) Energy shifts of new index */ + int32_t *CritMax, /* (i/o) Maximum Criteria (so far) */ + int16_t *shTotMax, /* (i/o) Shifts of maximum criteria */ + size_t *bestIndex, /* (i/o) Index that corresponds to + maximum criteria */ + int16_t *bestGain); /* (i/o) Gain in Q14 that corresponds + to maximum criteria */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c b/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c new file mode 100644 index 0000000000..21a8f40df6 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c @@ -0,0 +1,74 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Chebyshev.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*------------------------------------------------------------------* + * Calculate the Chevyshev polynomial series + * F(w) = 2*exp(-j5w)*C(x) + * C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2) + * T_i(x) is the i:th order Chebyshev polynomial + *------------------------------------------------------------------*/ + +int16_t WebRtcIlbcfix_Chebyshev( + /* (o) Result of C(x) */ + int16_t x, /* (i) Value to the Chevyshev polynomial */ + int16_t *f /* (i) The coefficients in the polynomial */ + ) { + int16_t b1_high, b1_low; /* Use the high, low format to increase the accuracy */ + int32_t b2; + int32_t tmp1W32; + int32_t tmp2W32; + int i; + + b2 = (int32_t)0x1000000; /* b2 = 1.0 (Q23) */ + /* Calculate b1 = 2*x + f[1] */ + tmp1W32 = (x << 10) + (f[1] << 14); + + for (i = 2; i < 5; i++) { + tmp2W32 = tmp1W32; + + /* Split b1 (in tmp1W32) into a high and low part */ + b1_high = (int16_t)(tmp1W32 >> 16); + b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1); + + /* Calculate 2*x*b1-b2+f[i] */ + tmp1W32 = ((b1_high * x + ((b1_low * x) >> 15)) << 2) - b2 + (f[i] << 14); + + /* Update b2 for next round */ + b2 = tmp2W32; + } + + /* Split b1 (in tmp1W32) into a high and low part */ + b1_high = (int16_t)(tmp1W32 >> 16); + b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1); + + /* tmp1W32 = x*b1 - b2 + f[i]/2 */ + tmp1W32 = ((b1_high * x) << 1) + (((b1_low * x) >> 15) << 1) - + b2 + (f[i] << 13); + + /* Handle overflows and set to maximum or minimum int16_t instead */ + if (tmp1W32>((int32_t)33553408)) { + return(WEBRTC_SPL_WORD16_MAX); + } else if (tmp1W32<((int32_t)-33554432)) { + return(WEBRTC_SPL_WORD16_MIN); + } else { + return (int16_t)(tmp1W32 >> 10); + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.h b/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.h new file mode 100644 index 0000000000..bf10132da1 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.h @@ -0,0 +1,37 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Chebyshev.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_ + +#include "defines.h" + +/*------------------------------------------------------------------* + * Calculate the Chevyshev polynomial series + * F(w) = 2*exp(-j5w)*C(x) + * C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2) + * T_i(x) is the i:th order Chebyshev polynomial + *------------------------------------------------------------------*/ + +int16_t WebRtcIlbcfix_Chebyshev( + /* (o) Result of C(x) */ + int16_t x, /* (i) Value to the Chevyshev polynomial */ + int16_t *f /* (i) The coefficients in the polynomial */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c new file mode 100644 index 0000000000..7653cb0c25 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c @@ -0,0 +1,49 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CompCorr.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Compute cross correlation and pitch gain for pitch prediction + * of last subframe at given lag. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_CompCorr( + int32_t *corr, /* (o) cross correlation */ + int32_t *ener, /* (o) energy */ + int16_t *buffer, /* (i) signal buffer */ + size_t lag, /* (i) pitch lag */ + size_t bLen, /* (i) length of buffer */ + size_t sRange, /* (i) correlation search length */ + int16_t scale /* (i) number of rightshifts to use */ + ){ + int16_t *w16ptr; + + w16ptr=&buffer[bLen-sRange-lag]; + + /* Calculate correlation and energy */ + (*corr)=WebRtcSpl_DotProductWithScale(&buffer[bLen-sRange], w16ptr, sRange, scale); + (*ener)=WebRtcSpl_DotProductWithScale(w16ptr, w16ptr, sRange, scale); + + /* For zero energy set the energy to 0 in order to avoid potential + problems for coming divisions */ + if (*ener == 0) { + *corr = 0; + *ener = 1; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h b/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h new file mode 100644 index 0000000000..ab78c72b3e --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h @@ -0,0 +1,39 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CompCorr.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Compute cross correlation and pitch gain for pitch prediction + * of last subframe at given lag. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_CompCorr( + int32_t *corr, /* (o) cross correlation */ + int32_t *ener, /* (o) energy */ + int16_t *buffer, /* (i) signal buffer */ + size_t lag, /* (i) pitch lag */ + size_t bLen, /* (i) length of buffer */ + size_t sRange, /* (i) correlation search length */ + int16_t scale /* (i) number of rightshifts to use */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m b/webrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m new file mode 100644 index 0000000000..f768194544 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m @@ -0,0 +1,49 @@ +clear; +pack; +% +% Enter the path to YOUR executable and remember to define the perprocessor +% variable PRINT_MIPS te get the instructions printed to the screen. +% +command = '!iLBCtest.exe 30 speechAndBGnoise.pcm out1.bit out1.pcm tlm10_30ms.dat'; +cout=' > st.txt'; %saves to matlab variable 'st' +eval(strcat(command,cout)); +if(length(cout)>3) + load st.txt +else + disp('No cout file to load') +end + +% initialize vector to zero +index = find(st(1:end,1)==-1); +indexnonzero = find(st(1:end,1)>0); +frames = length(index)-indexnonzero(1)+1; +start = indexnonzero(1) - 1; +functionOrder=max(st(:,2)); +new=zeros(frames,functionOrder); + +for i = 1:frames, + for j = index(start-1+i)+1:(index(start+i)-1), + new(i,st(j,2)) = new(i,st(j,2)) + st(j,1); + end +end + +result=zeros(functionOrder,3); +for i=1:functionOrder + nonzeroelements = find(new(1:end,i)>0); + result(i,1)=i; + + % Compute each function's mean complexity + % result(i,2)=(sum(new(nonzeroelements,i))/(length(nonzeroelements)*0.03))/1000000; + + % Compute each function's maximum complexity in encoding + % and decoding respectively and then add it together: + % result(i,3)=(max(new(1:end,i))/0.03)/1000000; + result(i,3)=(max(new(1:size(new,1)/2,i))/0.03)/1000000 + (max(new(size(new,1)/2+1:end,i))/0.03)/1000000; +end + +result + +% Compute maximum complexity for a single frame (enc/dec separately and together) +maxEncComplexityInAFrame = (max(sum(new(1:size(new,1)/2,:),2))/0.03)/1000000 +maxDecComplexityInAFrame = (max(sum(new(size(new,1)/2+1:end,:),2))/0.03)/1000000 +totalComplexity = maxEncComplexityInAFrame + maxDecComplexityInAFrame
\ No newline at end of file diff --git a/webrtc/modules/audio_coding/codecs/ilbc/constants.c b/webrtc/modules/audio_coding/codecs/ilbc/constants.c new file mode 100644 index 0000000000..9e341942e6 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/constants.c @@ -0,0 +1,666 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + constants.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/* HP Filters {b[0] b[1] b[2] -a[1] -a[2]} */ + +const int16_t WebRtcIlbcfix_kHpInCoefs[5] = {3798, -7596, 3798, 7807, -3733}; +const int16_t WebRtcIlbcfix_kHpOutCoefs[5] = {3849, -7699, 3849, 7918, -3833}; + +/* Window in Q11 to window the energies of the 5 choises (3 for 20ms) in the choise for + the 80 sample start state +*/ +const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[NSUB_MAX-1]= { + 1638, 1843, 2048, 1843, 1638 +}; + +/* LP Filter coeffs used for downsampling */ +const int16_t WebRtcIlbcfix_kLpFiltCoefs[FILTERORDER_DS_PLUS1]= { + -273, 512, 1297, 1696, 1297, 512, -273 +}; + +/* Constants used in the LPC calculations */ + +/* Hanning LPC window (in Q15) */ +const int16_t WebRtcIlbcfix_kLpcWin[BLOCKL_MAX] = { + 6, 22, 50, 89, 139, 200, 272, 355, 449, 554, 669, 795, + 932, 1079, 1237, 1405, 1583, 1771, 1969, 2177, 2395, 2622, 2858, 3104, + 3359, 3622, 3894, 4175, 4464, 4761, 5066, 5379, 5699, 6026, 6361, 6702, + 7050, 7404, 7764, 8130, 8502, 8879, 9262, 9649, 10040, 10436, 10836, 11240, + 11647, 12058, 12471, 12887, 13306, 13726, 14148, 14572, 14997, 15423, 15850, 16277, + 16704, 17131, 17558, 17983, 18408, 18831, 19252, 19672, 20089, 20504, 20916, 21325, + 21730, 22132, 22530, 22924, 23314, 23698, 24078, 24452, 24821, 25185, 25542, 25893, + 26238, 26575, 26906, 27230, 27547, 27855, 28156, 28450, 28734, 29011, 29279, 29538, + 29788, 30029, 30261, 30483, 30696, 30899, 31092, 31275, 31448, 31611, 31764, 31906, + 32037, 32158, 32268, 32367, 32456, 32533, 32600, 32655, 32700, 32733, 32755, 32767, + 32767, 32755, 32733, 32700, 32655, 32600, 32533, 32456, 32367, 32268, 32158, 32037, + 31906, 31764, 31611, 31448, 31275, 31092, 30899, 30696, 30483, 30261, 30029, 29788, + 29538, 29279, 29011, 28734, 28450, 28156, 27855, 27547, 27230, 26906, 26575, 26238, + 25893, 25542, 25185, 24821, 24452, 24078, 23698, 23314, 22924, 22530, 22132, 21730, + 21325, 20916, 20504, 20089, 19672, 19252, 18831, 18408, 17983, 17558, 17131, 16704, + 16277, 15850, 15423, 14997, 14572, 14148, 13726, 13306, 12887, 12471, 12058, 11647, + 11240, 10836, 10436, 10040, 9649, 9262, 8879, 8502, 8130, 7764, 7404, 7050, + 6702, 6361, 6026, 5699, 5379, 5066, 4761, 4464, 4175, 3894, 3622, 3359, + 3104, 2858, 2622, 2395, 2177, 1969, 1771, 1583, 1405, 1237, 1079, 932, + 795, 669, 554, 449, 355, 272, 200, 139, 89, 50, 22, 6 +}; + +/* Asymmetric LPC window (in Q15)*/ +const int16_t WebRtcIlbcfix_kLpcAsymWin[BLOCKL_MAX] = { + 2, 7, 15, 27, 42, 60, 81, 106, 135, 166, 201, 239, + 280, 325, 373, 424, 478, 536, 597, 661, 728, 798, 872, 949, + 1028, 1111, 1197, 1287, 1379, 1474, 1572, 1674, 1778, 1885, 1995, 2108, + 2224, 2343, 2465, 2589, 2717, 2847, 2980, 3115, 3254, 3395, 3538, 3684, + 3833, 3984, 4138, 4295, 4453, 4615, 4778, 4944, 5112, 5283, 5456, 5631, + 5808, 5987, 6169, 6352, 6538, 6725, 6915, 7106, 7300, 7495, 7692, 7891, + 8091, 8293, 8497, 8702, 8909, 9118, 9328, 9539, 9752, 9966, 10182, 10398, + 10616, 10835, 11055, 11277, 11499, 11722, 11947, 12172, 12398, 12625, 12852, 13080, + 13309, 13539, 13769, 14000, 14231, 14463, 14695, 14927, 15160, 15393, 15626, 15859, + 16092, 16326, 16559, 16792, 17026, 17259, 17492, 17725, 17957, 18189, 18421, 18653, + 18884, 19114, 19344, 19573, 19802, 20030, 20257, 20483, 20709, 20934, 21157, 21380, + 21602, 21823, 22042, 22261, 22478, 22694, 22909, 23123, 23335, 23545, 23755, 23962, + 24168, 24373, 24576, 24777, 24977, 25175, 25371, 25565, 25758, 25948, 26137, 26323, + 26508, 26690, 26871, 27049, 27225, 27399, 27571, 27740, 27907, 28072, 28234, 28394, + 28552, 28707, 28860, 29010, 29157, 29302, 29444, 29584, 29721, 29855, 29987, 30115, + 30241, 30364, 30485, 30602, 30717, 30828, 30937, 31043, 31145, 31245, 31342, 31436, + 31526, 31614, 31699, 31780, 31858, 31933, 32005, 32074, 32140, 32202, 32261, 32317, + 32370, 32420, 32466, 32509, 32549, 32585, 32618, 32648, 32675, 32698, 32718, 32734, + 32748, 32758, 32764, 32767, 32767, 32667, 32365, 31863, 31164, 30274, 29197, 27939, + 26510, 24917, 23170, 21281, 19261, 17121, 14876, 12540, 10126, 7650, 5126, 2571 +}; + +/* Lag window for LPC (Q31) */ +const int32_t WebRtcIlbcfix_kLpcLagWin[LPC_FILTERORDER + 1]={ + 2147483647, 2144885453, 2137754373, 2125918626, 2109459810, + 2088483140, 2063130336, 2033564590, 1999977009, 1962580174, + 1921610283}; + +/* WebRtcIlbcfix_kLpcChirpSyntDenum vector in Q15 corresponding + * floating point vector {1 0.9025 0.9025^2 0.9025^3 ...} + */ +const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[LPC_FILTERORDER + 1] = { + 32767, 29573, 26690, 24087, + 21739, 19619, 17707, 15980, + 14422, 13016, 11747}; + +/* WebRtcIlbcfix_kLpcChirpWeightDenum in Q15 corresponding to + * floating point vector {1 0.4222 0.4222^2... } + */ +const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[LPC_FILTERORDER + 1] = { + 32767, 13835, 5841, 2466, 1041, 440, + 186, 78, 33, 14, 6}; + +/* LSF quantization Q13 domain */ +const int16_t WebRtcIlbcfix_kLsfCb[64 * 3 + 128 * 3 + 128 * 4] = { + 1273, 2238, 3696, + 3199, 5309, 8209, + 3606, 5671, 7829, + 2815, 5262, 8778, + 2608, 4027, 5493, + 1582, 3076, 5945, + 2983, 4181, 5396, + 2437, 4322, 6902, + 1861, 2998, 4613, + 2007, 3250, 5214, + 1388, 2459, 4262, + 2563, 3805, 5269, + 2036, 3522, 5129, + 1935, 4025, 6694, + 2744, 5121, 7338, + 2810, 4248, 5723, + 3054, 5405, 7745, + 1449, 2593, 4763, + 3411, 5128, 6596, + 2484, 4659, 7496, + 1668, 2879, 4818, + 1812, 3072, 5036, + 1638, 2649, 3900, + 2464, 3550, 4644, + 1853, 2900, 4158, + 2458, 4163, 5830, + 2556, 4036, 6254, + 2703, 4432, 6519, + 3062, 4953, 7609, + 1725, 3703, 6187, + 2221, 3877, 5427, + 2339, 3579, 5197, + 2021, 4633, 7037, + 2216, 3328, 4535, + 2961, 4739, 6667, + 2807, 3955, 5099, + 2788, 4501, 6088, + 1642, 2755, 4431, + 3341, 5282, 7333, + 2414, 3726, 5727, + 1582, 2822, 5269, + 2259, 3447, 4905, + 3117, 4986, 7054, + 1825, 3491, 5542, + 3338, 5736, 8627, + 1789, 3090, 5488, + 2566, 3720, 4923, + 2846, 4682, 7161, + 1950, 3321, 5976, + 1834, 3383, 6734, + 3238, 4769, 6094, + 2031, 3978, 5903, + 1877, 4068, 7436, + 2131, 4644, 8296, + 2764, 5010, 8013, + 2194, 3667, 6302, + 2053, 3127, 4342, + 3523, 6595, 10010, + 3134, 4457, 5748, + 3142, 5819, 9414, + 2223, 4334, 6353, + 2022, 3224, 4822, + 2186, 3458, 5544, + 2552, 4757, 6870, + 10905, 12917, 14578, + 9503, 11485, 14485, + 9518, 12494, 14052, + 6222, 7487, 9174, + 7759, 9186, 10506, + 8315, 12755, 14786, + 9609, 11486, 13866, + 8909, 12077, 13643, + 7369, 9054, 11520, + 9408, 12163, 14715, + 6436, 9911, 12843, + 7109, 9556, 11884, + 7557, 10075, 11640, + 6482, 9202, 11547, + 6463, 7914, 10980, + 8611, 10427, 12752, + 7101, 9676, 12606, + 7428, 11252, 13172, + 10197, 12955, 15842, + 7487, 10955, 12613, + 5575, 7858, 13621, + 7268, 11719, 14752, + 7476, 11744, 13795, + 7049, 8686, 11922, + 8234, 11314, 13983, + 6560, 11173, 14984, + 6405, 9211, 12337, + 8222, 12054, 13801, + 8039, 10728, 13255, + 10066, 12733, 14389, + 6016, 7338, 10040, + 6896, 8648, 10234, + 7538, 9170, 12175, + 7327, 12608, 14983, + 10516, 12643, 15223, + 5538, 7644, 12213, + 6728, 12221, 14253, + 7563, 9377, 12948, + 8661, 11023, 13401, + 7280, 8806, 11085, + 7723, 9793, 12333, + 12225, 14648, 16709, + 8768, 13389, 15245, + 10267, 12197, 13812, + 5301, 7078, 11484, + 7100, 10280, 11906, + 8716, 12555, 14183, + 9567, 12464, 15434, + 7832, 12305, 14300, + 7608, 10556, 12121, + 8913, 11311, 12868, + 7414, 9722, 11239, + 8666, 11641, 13250, + 9079, 10752, 12300, + 8024, 11608, 13306, + 10453, 13607, 16449, + 8135, 9573, 10909, + 6375, 7741, 10125, + 10025, 12217, 14874, + 6985, 11063, 14109, + 9296, 13051, 14642, + 8613, 10975, 12542, + 6583, 10414, 13534, + 6191, 9368, 13430, + 5742, 6859, 9260, + 7723, 9813, 13679, + 8137, 11291, 12833, + 6562, 8973, 10641, + 6062, 8462, 11335, + 6928, 8784, 12647, + 7501, 8784, 10031, + 8372, 10045, 12135, + 8191, 9864, 12746, + 5917, 7487, 10979, + 5516, 6848, 10318, + 6819, 9899, 11421, + 7882, 12912, 15670, + 9558, 11230, 12753, + 7752, 9327, 11472, + 8479, 9980, 11358, + 11418, 14072, 16386, + 7968, 10330, 14423, + 8423, 10555, 12162, + 6337, 10306, 14391, + 8850, 10879, 14276, + 6750, 11885, 15710, + 7037, 8328, 9764, + 6914, 9266, 13476, + 9746, 13949, 15519, + 11032, 14444, 16925, + 8032, 10271, 11810, + 10962, 13451, 15833, + 10021, 11667, 13324, + 6273, 8226, 12936, + 8543, 10397, 13496, + 7936, 10302, 12745, + 6769, 8138, 10446, + 6081, 7786, 11719, + 8637, 11795, 14975, + 8790, 10336, 11812, + 7040, 8490, 10771, + 7338, 10381, 13153, + 6598, 7888, 9358, + 6518, 8237, 12030, + 9055, 10763, 12983, + 6490, 10009, 12007, + 9589, 12023, 13632, + 6867, 9447, 10995, + 7930, 9816, 11397, + 10241, 13300, 14939, + 5830, 8670, 12387, + 9870, 11915, 14247, + 9318, 11647, 13272, + 6721, 10836, 12929, + 6543, 8233, 9944, + 8034, 10854, 12394, + 9112, 11787, 14218, + 9302, 11114, 13400, + 9022, 11366, 13816, + 6962, 10461, 12480, + 11288, 13333, 15222, + 7249, 8974, 10547, + 10566, 12336, 14390, + 6697, 11339, 13521, + 11851, 13944, 15826, + 6847, 8381, 11349, + 7509, 9331, 10939, + 8029, 9618, 11909, + 13973, 17644, 19647, 22474, + 14722, 16522, 20035, 22134, + 16305, 18179, 21106, 23048, + 15150, 17948, 21394, 23225, + 13582, 15191, 17687, 22333, + 11778, 15546, 18458, 21753, + 16619, 18410, 20827, 23559, + 14229, 15746, 17907, 22474, + 12465, 15327, 20700, 22831, + 15085, 16799, 20182, 23410, + 13026, 16935, 19890, 22892, + 14310, 16854, 19007, 22944, + 14210, 15897, 18891, 23154, + 14633, 18059, 20132, 22899, + 15246, 17781, 19780, 22640, + 16396, 18904, 20912, 23035, + 14618, 17401, 19510, 21672, + 15473, 17497, 19813, 23439, + 18851, 20736, 22323, 23864, + 15055, 16804, 18530, 20916, + 16490, 18196, 19990, 21939, + 11711, 15223, 21154, 23312, + 13294, 15546, 19393, 21472, + 12956, 16060, 20610, 22417, + 11628, 15843, 19617, 22501, + 14106, 16872, 19839, 22689, + 15655, 18192, 20161, 22452, + 12953, 15244, 20619, 23549, + 15322, 17193, 19926, 21762, + 16873, 18676, 20444, 22359, + 14874, 17871, 20083, 21959, + 11534, 14486, 19194, 21857, + 17766, 19617, 21338, 23178, + 13404, 15284, 19080, 23136, + 15392, 17527, 19470, 21953, + 14462, 16153, 17985, 21192, + 17734, 19750, 21903, 23783, + 16973, 19096, 21675, 23815, + 16597, 18936, 21257, 23461, + 15966, 17865, 20602, 22920, + 15416, 17456, 20301, 22972, + 18335, 20093, 21732, 23497, + 15548, 17217, 20679, 23594, + 15208, 16995, 20816, 22870, + 13890, 18015, 20531, 22468, + 13211, 15377, 19951, 22388, + 12852, 14635, 17978, 22680, + 16002, 17732, 20373, 23544, + 11373, 14134, 19534, 22707, + 17329, 19151, 21241, 23462, + 15612, 17296, 19362, 22850, + 15422, 19104, 21285, 23164, + 13792, 17111, 19349, 21370, + 15352, 17876, 20776, 22667, + 15253, 16961, 18921, 22123, + 14108, 17264, 20294, 23246, + 15785, 17897, 20010, 21822, + 17399, 19147, 20915, 22753, + 13010, 15659, 18127, 20840, + 16826, 19422, 22218, 24084, + 18108, 20641, 22695, 24237, + 18018, 20273, 22268, 23920, + 16057, 17821, 21365, 23665, + 16005, 17901, 19892, 23016, + 13232, 16683, 21107, 23221, + 13280, 16615, 19915, 21829, + 14950, 18575, 20599, 22511, + 16337, 18261, 20277, 23216, + 14306, 16477, 21203, 23158, + 12803, 17498, 20248, 22014, + 14327, 17068, 20160, 22006, + 14402, 17461, 21599, 23688, + 16968, 18834, 20896, 23055, + 15070, 17157, 20451, 22315, + 15419, 17107, 21601, 23946, + 16039, 17639, 19533, 21424, + 16326, 19261, 21745, 23673, + 16489, 18534, 21658, 23782, + 16594, 18471, 20549, 22807, + 18973, 21212, 22890, 24278, + 14264, 18674, 21123, 23071, + 15117, 16841, 19239, 23118, + 13762, 15782, 20478, 23230, + 14111, 15949, 20058, 22354, + 14990, 16738, 21139, 23492, + 13735, 16971, 19026, 22158, + 14676, 17314, 20232, 22807, + 16196, 18146, 20459, 22339, + 14747, 17258, 19315, 22437, + 14973, 17778, 20692, 23367, + 15715, 17472, 20385, 22349, + 15702, 18228, 20829, 23410, + 14428, 16188, 20541, 23630, + 16824, 19394, 21365, 23246, + 13069, 16392, 18900, 21121, + 12047, 16640, 19463, 21689, + 14757, 17433, 19659, 23125, + 15185, 16930, 19900, 22540, + 16026, 17725, 19618, 22399, + 16086, 18643, 21179, 23472, + 15462, 17248, 19102, 21196, + 17368, 20016, 22396, 24096, + 12340, 14475, 19665, 23362, + 13636, 16229, 19462, 22728, + 14096, 16211, 19591, 21635, + 12152, 14867, 19943, 22301, + 14492, 17503, 21002, 22728, + 14834, 16788, 19447, 21411, + 14650, 16433, 19326, 22308, + 14624, 16328, 19659, 23204, + 13888, 16572, 20665, 22488, + 12977, 16102, 18841, 22246, + 15523, 18431, 21757, 23738, + 14095, 16349, 18837, 20947, + 13266, 17809, 21088, 22839, + 15427, 18190, 20270, 23143, + 11859, 16753, 20935, 22486, + 12310, 17667, 21736, 23319, + 14021, 15926, 18702, 22002, + 12286, 15299, 19178, 21126, + 15703, 17491, 21039, 23151, + 12272, 14018, 18213, 22570, + 14817, 16364, 18485, 22598, + 17109, 19683, 21851, 23677, + 12657, 14903, 19039, 22061, + 14713, 16487, 20527, 22814, + 14635, 16726, 18763, 21715, + 15878, 18550, 20718, 22906 +}; + +const int16_t WebRtcIlbcfix_kLsfDimCb[LSF_NSPLIT] = {3, 3, 4}; +const int16_t WebRtcIlbcfix_kLsfSizeCb[LSF_NSPLIT] = {64,128,128}; + +const int16_t WebRtcIlbcfix_kLsfMean[LPC_FILTERORDER] = { + 2308, 3652, 5434, 7885, + 10255, 12559, 15160, 17513, + 20328, 22752}; + +const int16_t WebRtcIlbcfix_kLspMean[LPC_FILTERORDER] = { + 31476, 29565, 25819, 18725, 10276, + 1236, -9049, -17600, -25884, -30618 +}; + +/* Q14 */ +const int16_t WebRtcIlbcfix_kLsfWeight20ms[4] = {12288, 8192, 4096, 0}; +const int16_t WebRtcIlbcfix_kLsfWeight30ms[6] = {8192, 16384, 10923, 5461, 0, 0}; + +/* + cos(x) in Q15 + WebRtcIlbcfix_kCos[i] = cos(pi*i/64.0) + used in WebRtcIlbcfix_Lsp2Lsf() +*/ + +const int16_t WebRtcIlbcfix_kCos[64] = { + 32767, 32729, 32610, 32413, 32138, 31786, 31357, 30853, + 30274, 29622, 28899, 28106, 27246, 26320, 25330, 24279, + 23170, 22006, 20788, 19520, 18205, 16846, 15447, 14010, + 12540, 11039, 9512, 7962, 6393, 4808, 3212, 1608, + 0, -1608, -3212, -4808, -6393, -7962, -9512, -11039, + -12540, -14010, -15447, -16846, -18205, -19520, -20788, -22006, + -23170, -24279, -25330, -26320, -27246, -28106, -28899, -29622, + -30274, -30853, -31357, -31786, -32138, -32413, -32610, -32729 +}; + +/* + Derivative in Q19, used to interpolate between the + WebRtcIlbcfix_kCos[] values to get a more exact y = cos(x) +*/ +const int16_t WebRtcIlbcfix_kCosDerivative[64] = { + -632, -1893, -3150, -4399, -5638, -6863, -8072, -9261, + -10428, -11570, -12684, -13767, -14817, -15832, -16808, -17744, + -18637, -19486, -20287, -21039, -21741, -22390, -22986, -23526, + -24009, -24435, -24801, -25108, -25354, -25540, -25664, -25726, + -25726, -25664, -25540, -25354, -25108, -24801, -24435, -24009, + -23526, -22986, -22390, -21741, -21039, -20287, -19486, -18637, + -17744, -16808, -15832, -14817, -13767, -12684, -11570, -10428, + -9261, -8072, -6863, -5638, -4399, -3150, -1893, -632}; + +/* + Table in Q15, used for a2lsf conversion + WebRtcIlbcfix_kCosGrid[i] = cos((2*pi*i)/(float)(2*COS_GRID_POINTS)); +*/ + +const int16_t WebRtcIlbcfix_kCosGrid[COS_GRID_POINTS + 1] = { + 32760, 32723, 32588, 32364, 32051, 31651, 31164, 30591, + 29935, 29196, 28377, 27481, 26509, 25465, 24351, 23170, + 21926, 20621, 19260, 17846, 16384, 14876, 13327, 11743, + 10125, 8480, 6812, 5126, 3425, 1714, 0, -1714, -3425, + -5126, -6812, -8480, -10125, -11743, -13327, -14876, + -16384, -17846, -19260, -20621, -21926, -23170, -24351, + -25465, -26509, -27481, -28377, -29196, -29935, -30591, + -31164, -31651, -32051, -32364, -32588, -32723, -32760 +}; + +/* + Derivative of y = acos(x) in Q12 + used in WebRtcIlbcfix_Lsp2Lsf() +*/ + +const int16_t WebRtcIlbcfix_kAcosDerivative[64] = { + -26887, -8812, -5323, -3813, -2979, -2444, -2081, -1811, + -1608, -1450, -1322, -1219, -1132, -1059, -998, -946, + -901, -861, -827, -797, -772, -750, -730, -713, + -699, -687, -677, -668, -662, -657, -654, -652, + -652, -654, -657, -662, -668, -677, -687, -699, + -713, -730, -750, -772, -797, -827, -861, -901, + -946, -998, -1059, -1132, -1219, -1322, -1450, -1608, + -1811, -2081, -2444, -2979, -3813, -5323, -8812, -26887 +}; + + +/* Tables for quantization of start state */ + +/* State quantization tables */ +const int16_t WebRtcIlbcfix_kStateSq3[8] = { /* Values in Q13 */ + -30473, -17838, -9257, -2537, + 3639, 10893, 19958, 32636 +}; + +/* This table defines the limits for the selection of the freqg + less or equal than value 0 => index = 0 + less or equal than value k => index = k +*/ +const int32_t WebRtcIlbcfix_kChooseFrgQuant[64] = { + 118, 163, 222, 305, 425, 604, + 851, 1174, 1617, 2222, 3080, 4191, + 5525, 7215, 9193, 11540, 14397, 17604, + 21204, 25209, 29863, 35720, 42531, 50375, + 59162, 68845, 80108, 93754, 110326, 129488, + 150654, 174328, 201962, 233195, 267843, 308239, + 354503, 405988, 464251, 531550, 608652, 697516, + 802526, 928793, 1080145, 1258120, 1481106, 1760881, + 2111111, 2546619, 3078825, 3748642, 4563142, 5573115, + 6887601, 8582108, 10797296, 14014513, 18625760, 25529599, + 37302935, 58819185, 109782723, WEBRTC_SPL_WORD32_MAX +}; + +const int16_t WebRtcIlbcfix_kScale[64] = { + /* Values in Q16 */ + 29485, 25003, 21345, 18316, 15578, 13128, 10973, 9310, 7955, + 6762, 5789, 4877, 4255, 3699, 3258, 2904, 2595, 2328, + 2123, 1932, 1785, 1631, 1493, 1370, 1260, 1167, 1083, + /* Values in Q21 */ + 32081, 29611, 27262, 25229, 23432, 21803, 20226, 18883, 17609, + 16408, 15311, 14327, 13390, 12513, 11693, 10919, 10163, 9435, + 8739, 8100, 7424, 6813, 6192, 5648, 5122, 4639, 4207, 3798, + 3404, 3048, 2706, 2348, 2036, 1713, 1393, 1087, 747 +}; + +/*frgq in fixpoint, but already computed like this: + for(i=0; i<64; i++){ + a = (pow(10,frgq[i])/4.5); + WebRtcIlbcfix_kFrgQuantMod[i] = round(a); + } + + Value 0 :36 in Q8 + 37:58 in Q5 + 59:63 in Q3 +*/ +const int16_t WebRtcIlbcfix_kFrgQuantMod[64] = { + /* First 37 values in Q8 */ + 569, 671, 786, 916, 1077, 1278, + 1529, 1802, 2109, 2481, 2898, 3440, + 3943, 4535, 5149, 5778, 6464, 7208, + 7904, 8682, 9397, 10285, 11240, 12246, + 13313, 14382, 15492, 16735, 18131, 19693, + 21280, 22912, 24624, 26544, 28432, 30488, + 32720, + /* 22 values in Q5 */ + 4383, 4684, 5012, 5363, 5739, 6146, + 6603, 7113, 7679, 8285, 9040, 9850, + 10838, 11882, 13103, 14467, 15950, 17669, + 19712, 22016, 24800, 28576, + /* 5 values in Q3 */ + 8240, 9792, 12040, 15440, 22472 +}; + +/* Constants for codebook search and creation */ + +/* Expansion filter to get additional cb section. + * Q12 and reversed compared to flp + */ +const int16_t WebRtcIlbcfix_kCbFiltersRev[CB_FILTERLEN]={ + -140, 446, -755, 3302, 2922, -590, 343, -138}; + +/* Weighting coefficients for short lags. + * [0.2 0.4 0.6 0.8] in Q15 */ +const int16_t WebRtcIlbcfix_kAlpha[4]={ + 6554, 13107, 19661, 26214}; + +/* Ranges for search and filters at different subframes */ + +const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={ + {58,58,58}, {108,44,44}, {108,108,108}, {108,108,108}, {108,108,108}}; + +const size_t WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147}; + +/* Gain Quantization for the codebook gains of the 3 stages */ + +/* Q14 (one extra value (max int16_t) to simplify for the search) */ +const int16_t WebRtcIlbcfix_kGainSq3[9]={ + -16384, -10813, -5407, 0, 4096, 8192, + 12288, 16384, 32767}; + +/* Q14 (one extra value (max int16_t) to simplify for the search) */ +const int16_t WebRtcIlbcfix_kGainSq4[17]={ + -17203, -14746, -12288, -9830, -7373, -4915, + -2458, 0, 2458, 4915, 7373, 9830, + 12288, 14746, 17203, 19661, 32767}; + +/* Q14 (one extra value (max int16_t) to simplify for the search) */ +const int16_t WebRtcIlbcfix_kGainSq5[33]={ + 614, 1229, 1843, 2458, 3072, 3686, + 4301, 4915, 5530, 6144, 6758, 7373, + 7987, 8602, 9216, 9830, 10445, 11059, + 11674, 12288, 12902, 13517, 14131, 14746, + 15360, 15974, 16589, 17203, 17818, 18432, + 19046, 19661, 32767}; + +/* Q14 gain_sq5Tbl squared in Q14 */ +const int16_t WebRtcIlbcfix_kGainSq5Sq[32] = { + 23, 92, 207, 368, 576, 829, + 1129, 1474, 1866, 2304, 2787, 3317, + 3893, 4516, 5184, 5897, 6658, 7464, + 8318, 9216, 10160, 11151, 12187, 13271, + 14400, 15574, 16796, 18062, 19377, 20736, + 22140, 23593 +}; + +const int16_t* const WebRtcIlbcfix_kGain[3] = +{WebRtcIlbcfix_kGainSq5, WebRtcIlbcfix_kGainSq4, WebRtcIlbcfix_kGainSq3}; + + +/* Tables for the Enhancer, using upsamling factor 4 (ENH_UPS0 = 4) */ + +const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]={ + {0, 0, 0, 4096, 0, 0, 0}, + {64, -315, 1181, 3531, -436, 77, -64}, + {97, -509, 2464, 2464, -509, 97, -97}, + {77, -436, 3531, 1181, -315, 64, -77} +}; + +const int16_t WebRtcIlbcfix_kEnhWt[3] = { + 4800, 16384, 27968 /* Q16 */ +}; + +const size_t WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = { + 160, 480, 800, 1120, 1440, 1760, 2080, 2400 /* Q(-2) */ +}; + +/* PLC table */ + +const int16_t WebRtcIlbcfix_kPlcPerSqr[6] = { /* Grid points for square of periodiciy in Q15 */ + 839, 1343, 2048, 2998, 4247, 5849 +}; + +const int16_t WebRtcIlbcfix_kPlcPitchFact[6] = { /* Value of y=(x^4-0.4)/(0.7-0.4) in grid points in Q15 */ + 0, 5462, 10922, 16384, 21846, 27306 +}; + +const int16_t WebRtcIlbcfix_kPlcPfSlope[6] = { /* Slope of y=(x^4-0.4)/(0.7-0.4) in Q11 */ + 26667, 18729, 13653, 10258, 7901, 6214 +}; diff --git a/webrtc/modules/audio_coding/codecs/ilbc/constants.h b/webrtc/modules/audio_coding/codecs/ilbc/constants.h new file mode 100644 index 0000000000..7c4ad4d928 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/constants.h @@ -0,0 +1,92 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + constants.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_ + +#include "defines.h" +#include "webrtc/typedefs.h" + +/* high pass filters */ + +extern const int16_t WebRtcIlbcfix_kHpInCoefs[]; +extern const int16_t WebRtcIlbcfix_kHpOutCoefs[]; + +/* Window for start state decision */ +extern const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[]; + +/* low pass filter used for downsampling */ +extern const int16_t WebRtcIlbcfix_kLpFiltCoefs[]; + +/* LPC analysis and quantization */ + +extern const int16_t WebRtcIlbcfix_kLpcWin[]; +extern const int16_t WebRtcIlbcfix_kLpcAsymWin[]; +extern const int32_t WebRtcIlbcfix_kLpcLagWin[]; +extern const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[]; +extern const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[]; +extern const int16_t WebRtcIlbcfix_kLsfDimCb[]; +extern const int16_t WebRtcIlbcfix_kLsfSizeCb[]; +extern const int16_t WebRtcIlbcfix_kLsfCb[]; +extern const int16_t WebRtcIlbcfix_kLsfWeight20ms[]; +extern const int16_t WebRtcIlbcfix_kLsfWeight30ms[]; +extern const int16_t WebRtcIlbcfix_kLsfMean[]; +extern const int16_t WebRtcIlbcfix_kLspMean[]; +extern const int16_t WebRtcIlbcfix_kCos[]; +extern const int16_t WebRtcIlbcfix_kCosDerivative[]; +extern const int16_t WebRtcIlbcfix_kCosGrid[]; +extern const int16_t WebRtcIlbcfix_kAcosDerivative[]; + +/* state quantization tables */ + +extern const int16_t WebRtcIlbcfix_kStateSq3[]; +extern const int32_t WebRtcIlbcfix_kChooseFrgQuant[]; +extern const int16_t WebRtcIlbcfix_kScale[]; +extern const int16_t WebRtcIlbcfix_kFrgQuantMod[]; + +/* Ranges for search and filters at different subframes */ + +extern const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]; +extern const size_t WebRtcIlbcfix_kFilterRange[]; + +/* gain quantization tables */ + +extern const int16_t WebRtcIlbcfix_kGainSq3[]; +extern const int16_t WebRtcIlbcfix_kGainSq4[]; +extern const int16_t WebRtcIlbcfix_kGainSq5[]; +extern const int16_t WebRtcIlbcfix_kGainSq5Sq[]; +extern const int16_t* const WebRtcIlbcfix_kGain[]; + +/* adaptive codebook definitions */ + +extern const int16_t WebRtcIlbcfix_kCbFiltersRev[]; +extern const int16_t WebRtcIlbcfix_kAlpha[]; + +/* enhancer definitions */ + +extern const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]; +extern const int16_t WebRtcIlbcfix_kEnhWt[]; +extern const size_t WebRtcIlbcfix_kEnhPlocs[]; + +/* PLC tables */ + +extern const int16_t WebRtcIlbcfix_kPlcPerSqr[]; +extern const int16_t WebRtcIlbcfix_kPlcPitchFact[]; +extern const int16_t WebRtcIlbcfix_kPlcPfSlope[]; + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c new file mode 100644 index 0000000000..8ae28ac3b9 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c @@ -0,0 +1,57 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CreateAugmentedVec.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * Recreate a specific codebook vector from the augmented part. + * + *----------------------------------------------------------------*/ + +void WebRtcIlbcfix_CreateAugmentedVec( + size_t index, /* (i) Index for the augmented vector to be created */ + int16_t *buffer, /* (i) Pointer to the end of the codebook memory that + is used for creation of the augmented codebook */ + int16_t *cbVec /* (o) The construced codebook vector */ + ) { + size_t ilow; + int16_t *ppo, *ppi; + int16_t cbVecTmp[4]; + + ilow = index-4; + + /* copy the first noninterpolated part */ + ppo = buffer-index; + WEBRTC_SPL_MEMCPY_W16(cbVec, ppo, index); + + /* interpolation */ + ppo = buffer - 4; + ppi = buffer - index - 4; + + /* perform cbVec[ilow+k] = ((ppi[k]*alphaTbl[k])>>15) + ((ppo[k]*alphaTbl[3-k])>>15); + for k = 0..3 + */ + WebRtcSpl_ElementwiseVectorMult(&cbVec[ilow], ppi, WebRtcIlbcfix_kAlpha, 4, 15); + WebRtcSpl_ReverseOrderMultArrayElements(cbVecTmp, ppo, &WebRtcIlbcfix_kAlpha[3], 4, 15); + WebRtcSpl_AddVectorsAndShift(&cbVec[ilow], &cbVec[ilow], cbVecTmp, 4, 0); + + /* copy the second noninterpolated part */ + ppo = buffer - index; + WEBRTC_SPL_MEMCPY_W16(cbVec+index,ppo,(SUBL-index)); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h new file mode 100644 index 0000000000..430dfe9b9d --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_CreateAugmentedVec.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Recreate a specific codebook vector from the augmented part. + * + *----------------------------------------------------------------*/ + +void WebRtcIlbcfix_CreateAugmentedVec( + size_t index, /* (i) Index for the augmented vector to be created */ + int16_t *buffer, /* (i) Pointer to the end of the codebook memory that + is used for creation of the augmented codebook */ + int16_t *cbVec /* (o) The construced codebook vector */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode.c b/webrtc/modules/audio_coding/codecs/ilbc/decode.c new file mode 100644 index 0000000000..4c8497a568 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/decode.c @@ -0,0 +1,244 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Decode.c + +******************************************************************/ + +#include "defines.h" +#include "simple_lsf_dequant.h" +#include "decoder_interpolate_lsf.h" +#include "index_conv_dec.h" +#include "do_plc.h" +#include "constants.h" +#include "enhancer_interface.h" +#include "xcorr_coef.h" +#include "lsf_check.h" +#include "decode_residual.h" +#include "unpack_bits.h" +#include "hp_output.h" +#ifndef WEBRTC_ARCH_BIG_ENDIAN +#include "swap_bytes.h" +#endif + +/*----------------------------------------------------------------* + * main decoder function + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_DecodeImpl( + int16_t *decblock, /* (o) decoded signal block */ + const uint16_t *bytes, /* (i) encoded signal bits */ + IlbcDecoder *iLBCdec_inst, /* (i/o) the decoder state + structure */ + int16_t mode /* (i) 0: bad packet, PLC, + 1: normal */ + ) { + size_t i; + int16_t order_plus_one; + + int16_t last_bit; + int16_t *data; + /* Stack based */ + int16_t decresidual[BLOCKL_MAX]; + int16_t PLCresidual[BLOCKL_MAX + LPC_FILTERORDER]; + int16_t syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)]; + int16_t PLClpc[LPC_FILTERORDER + 1]; +#ifndef WEBRTC_ARCH_BIG_ENDIAN + uint16_t swapped[NO_OF_WORDS_30MS]; +#endif + iLBC_bits *iLBCbits_inst = (iLBC_bits*)PLCresidual; + + /* Reuse some buffers that are non overlapping in order to save stack memory */ + data = &PLCresidual[LPC_FILTERORDER]; + + if (mode) { /* the data are good */ + + /* decode data */ + + /* Unpacketize bits into parameters */ + +#ifndef WEBRTC_ARCH_BIG_ENDIAN + WebRtcIlbcfix_SwapBytes(bytes, iLBCdec_inst->no_of_words, swapped); + last_bit = WebRtcIlbcfix_UnpackBits(swapped, iLBCbits_inst, iLBCdec_inst->mode); +#else + last_bit = WebRtcIlbcfix_UnpackBits(bytes, iLBCbits_inst, iLBCdec_inst->mode); +#endif + + /* Check for bit errors */ + if (iLBCbits_inst->startIdx<1) + mode = 0; + if ((iLBCdec_inst->mode==20) && (iLBCbits_inst->startIdx>3)) + mode = 0; + if ((iLBCdec_inst->mode==30) && (iLBCbits_inst->startIdx>5)) + mode = 0; + if (last_bit==1) + mode = 0; + + if (mode) { /* No bit errors was detected, continue decoding */ + /* Stack based */ + int16_t lsfdeq[LPC_FILTERORDER*LPC_N_MAX]; + int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX]; + + /* adjust index */ + WebRtcIlbcfix_IndexConvDec(iLBCbits_inst->cb_index); + + /* decode the lsf */ + WebRtcIlbcfix_SimpleLsfDeQ(lsfdeq, (int16_t*)(iLBCbits_inst->lsf), iLBCdec_inst->lpc_n); + WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCdec_inst->lpc_n); + WebRtcIlbcfix_DecoderInterpolateLsp(syntdenum, weightdenum, + lsfdeq, LPC_FILTERORDER, iLBCdec_inst); + + /* Decode the residual using the cb and gain indexes */ + WebRtcIlbcfix_DecodeResidual(iLBCdec_inst, iLBCbits_inst, decresidual, syntdenum); + + /* preparing the plc for a future loss! */ + WebRtcIlbcfix_DoThePlc( + PLCresidual, PLClpc, 0, decresidual, + syntdenum + (LPC_FILTERORDER + 1) * (iLBCdec_inst->nsub - 1), + iLBCdec_inst->last_lag, iLBCdec_inst); + + /* Use the output from doThePLC */ + WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl); + } + + } + + if (mode == 0) { + /* the data is bad (either a PLC call + * was made or a bit error was detected) + */ + + /* packet loss conceal */ + + WebRtcIlbcfix_DoThePlc(PLCresidual, PLClpc, 1, decresidual, syntdenum, + iLBCdec_inst->last_lag, iLBCdec_inst); + + WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl); + + order_plus_one = LPC_FILTERORDER + 1; + + for (i = 0; i < iLBCdec_inst->nsub; i++) { + WEBRTC_SPL_MEMCPY_W16(syntdenum+(i*order_plus_one), + PLClpc, order_plus_one); + } + } + + if ((*iLBCdec_inst).use_enhancer == 1) { /* Enhancer activated */ + + /* Update the filter and filter coefficients if there was a packet loss */ + if (iLBCdec_inst->prev_enh_pl==2) { + for (i=0;i<iLBCdec_inst->nsub;i++) { + WEBRTC_SPL_MEMCPY_W16(&(iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]), + syntdenum, (LPC_FILTERORDER+1)); + } + } + + /* post filtering */ + (*iLBCdec_inst).last_lag = + WebRtcIlbcfix_EnhancerInterface(data, decresidual, iLBCdec_inst); + + /* synthesis filtering */ + + /* Set up the filter state */ + WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER); + + if (iLBCdec_inst->mode==20) { + /* Enhancer has 40 samples delay */ + i=0; + WebRtcSpl_FilterARFastQ12( + data, data, + iLBCdec_inst->old_syntdenum + (i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1), + LPC_FILTERORDER+1, SUBL); + + for (i=1; i < iLBCdec_inst->nsub; i++) { + WebRtcSpl_FilterARFastQ12( + data+i*SUBL, data+i*SUBL, + syntdenum+(i-1)*(LPC_FILTERORDER+1), + LPC_FILTERORDER+1, SUBL); + } + + } else if (iLBCdec_inst->mode==30) { + /* Enhancer has 80 samples delay */ + for (i=0; i < 2; i++) { + WebRtcSpl_FilterARFastQ12( + data+i*SUBL, data+i*SUBL, + iLBCdec_inst->old_syntdenum + (i+4)*(LPC_FILTERORDER+1), + LPC_FILTERORDER+1, SUBL); + } + for (i=2; i < iLBCdec_inst->nsub; i++) { + WebRtcSpl_FilterARFastQ12( + data+i*SUBL, data+i*SUBL, + syntdenum+(i-2)*(LPC_FILTERORDER+1), + LPC_FILTERORDER+1, SUBL); + } + } + + /* Save the filter state */ + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER); + + } else { /* Enhancer not activated */ + size_t lag; + + /* Find last lag (since the enhancer is not called to give this info) */ + lag = 20; + if (iLBCdec_inst->mode==20) { + lag = WebRtcIlbcfix_XcorrCoef( + &decresidual[iLBCdec_inst->blockl-60], + &decresidual[iLBCdec_inst->blockl-60-lag], + 60, + 80, lag, -1); + } else { + lag = WebRtcIlbcfix_XcorrCoef( + &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL], + &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL-lag], + ENH_BLOCKL, + 100, lag, -1); + } + + /* Store lag (it is needed if next packet is lost) */ + (*iLBCdec_inst).last_lag = lag; + + /* copy data and run synthesis filter */ + WEBRTC_SPL_MEMCPY_W16(data, decresidual, iLBCdec_inst->blockl); + + /* Set up the filter state */ + WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER); + + for (i=0; i < iLBCdec_inst->nsub; i++) { + WebRtcSpl_FilterARFastQ12( + data+i*SUBL, data+i*SUBL, + syntdenum + i*(LPC_FILTERORDER+1), + LPC_FILTERORDER+1, SUBL); + } + + /* Save the filter state */ + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER); + } + + WEBRTC_SPL_MEMCPY_W16(decblock,data,iLBCdec_inst->blockl); + + /* High pass filter the signal (with upscaling a factor 2 and saturation) */ + WebRtcIlbcfix_HpOutput(decblock, (int16_t*)WebRtcIlbcfix_kHpOutCoefs, + iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx, + iLBCdec_inst->blockl); + + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->old_syntdenum, + syntdenum, iLBCdec_inst->nsub*(LPC_FILTERORDER+1)); + + iLBCdec_inst->prev_enh_pl=0; + + if (mode==0) { /* PLC was used */ + iLBCdec_inst->prev_enh_pl=1; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode.h b/webrtc/modules/audio_coding/codecs/ilbc/decode.h new file mode 100644 index 0000000000..0c4c2ef844 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/decode.h @@ -0,0 +1,37 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Decode.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * main decoder function + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_DecodeImpl( + int16_t *decblock, /* (o) decoded signal block */ + const uint16_t *bytes, /* (i) encoded signal bits */ + IlbcDecoder *iLBCdec_inst, /* (i/o) the decoder state + structure */ + int16_t mode /* (i) 0: bad packet, PLC, + 1: normal */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c new file mode 100644 index 0000000000..b8a067e0f3 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c @@ -0,0 +1,182 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_DecodeResidual.c + +******************************************************************/ + +#include <string.h> + +#include "defines.h" +#include "state_construct.h" +#include "cb_construct.h" +#include "index_conv_dec.h" +#include "do_plc.h" +#include "constants.h" +#include "enhancer_interface.h" +#include "xcorr_coef.h" +#include "lsf_check.h" + +/*----------------------------------------------------------------* + * frame residual decoder function (subrutine to iLBC_decode) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_DecodeResidual( + IlbcDecoder *iLBCdec_inst, + /* (i/o) the decoder state structure */ + iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used + for the decoding */ + int16_t *decresidual, /* (o) decoded residual frame */ + int16_t *syntdenum /* (i) the decoded synthesis filter + coefficients */ + ) { + size_t meml_gotten, diff, start_pos; + size_t subcount, subframe; + int16_t *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */ + int16_t *memVec = iLBCdec_inst->prevResidual; /* Memory for codebook and filter state (reuse memory in state) */ + int16_t *mem = &memVec[CB_HALFFILTERLEN]; /* Memory for codebook */ + + diff = STATE_LEN - iLBCdec_inst->state_short_len; + + if (iLBC_encbits->state_first == 1) { + start_pos = (iLBC_encbits->startIdx-1)*SUBL; + } else { + start_pos = (iLBC_encbits->startIdx-1)*SUBL + diff; + } + + /* decode scalar part of start state */ + + WebRtcIlbcfix_StateConstruct(iLBC_encbits->idxForMax, + iLBC_encbits->idxVec, &syntdenum[(iLBC_encbits->startIdx-1)*(LPC_FILTERORDER+1)], + &decresidual[start_pos], iLBCdec_inst->state_short_len + ); + + if (iLBC_encbits->state_first) { /* put adaptive part in the end */ + + /* setup memory */ + + WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCdec_inst->state_short_len); + WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos, + iLBCdec_inst->state_short_len); + + /* construct decoded vector */ + + WebRtcIlbcfix_CbConstruct( + &decresidual[start_pos+iLBCdec_inst->state_short_len], + iLBC_encbits->cb_index, iLBC_encbits->gain_index, + mem+CB_MEML-ST_MEM_L_TBL, + ST_MEM_L_TBL, diff); + + } + else {/* put adaptive part in the beginning */ + + /* setup memory */ + + meml_gotten = iLBCdec_inst->state_short_len; + WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1, + decresidual+start_pos, meml_gotten); + WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten); + + /* construct decoded vector */ + + WebRtcIlbcfix_CbConstruct( + reverseDecresidual, + iLBC_encbits->cb_index, iLBC_encbits->gain_index, + mem+CB_MEML-ST_MEM_L_TBL, + ST_MEM_L_TBL, diff + ); + + /* get decoded residual from reversed vector */ + + WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1], + reverseDecresidual, diff); + } + + /* counter for predicted subframes */ + + subcount=1; + + /* forward prediction of subframes */ + + if (iLBCdec_inst->nsub > iLBC_encbits->startIdx + 1) { + + /* setup memory */ + WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN); + WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN, + decresidual+(iLBC_encbits->startIdx-1)*SUBL, STATE_LEN); + + /* loop over subframes to encode */ + + size_t Nfor = iLBCdec_inst->nsub - iLBC_encbits->startIdx - 1; + for (subframe=0; subframe<Nfor; subframe++) { + + /* construct decoded vector */ + WebRtcIlbcfix_CbConstruct( + &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL], + iLBC_encbits->cb_index+subcount*CB_NSTAGES, + iLBC_encbits->gain_index+subcount*CB_NSTAGES, + mem, MEM_LF_TBL, SUBL + ); + + /* update memory */ + memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem)); + WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL, + &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL], SUBL); + + subcount++; + } + + } + + /* backward prediction of subframes */ + + if (iLBC_encbits->startIdx > 1) { + + /* setup memory */ + + meml_gotten = SUBL*(iLBCdec_inst->nsub+1-iLBC_encbits->startIdx); + if( meml_gotten > CB_MEML ) { + meml_gotten=CB_MEML; + } + + WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1, + decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten); + WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten); + + /* loop over subframes to decode */ + + size_t Nback = iLBC_encbits->startIdx - 1; + for (subframe=0; subframe<Nback; subframe++) { + + /* construct decoded vector */ + WebRtcIlbcfix_CbConstruct( + &reverseDecresidual[subframe*SUBL], + iLBC_encbits->cb_index+subcount*CB_NSTAGES, + iLBC_encbits->gain_index+subcount*CB_NSTAGES, + mem, MEM_LF_TBL, SUBL + ); + + /* update memory */ + memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem)); + WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL, + &reverseDecresidual[subframe*SUBL], SUBL); + + subcount++; + } + + /* get decoded residual from reversed vector */ + WebRtcSpl_MemCpyReversedOrder(decresidual+SUBL*Nback-1, + reverseDecresidual, SUBL*Nback); + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h new file mode 100644 index 0000000000..67f05a5768 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_DecodeResidual.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * frame residual decoder function (subrutine to iLBC_decode) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_DecodeResidual( + IlbcDecoder *iLBCdec_inst, + /* (i/o) the decoder state structure */ + iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used + for the decoding */ + int16_t *decresidual, /* (o) decoded residual frame */ + int16_t *syntdenum /* (i) the decoded synthesis filter + coefficients */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c new file mode 100644 index 0000000000..06ab2e7190 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c @@ -0,0 +1,83 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_DecoderInterpolateLsp.c + +******************************************************************/ + +#include "lsf_interpolate_to_poly_dec.h" +#include "bw_expand.h" +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * obtain synthesis and weighting filters form lsf coefficients + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_DecoderInterpolateLsp( + int16_t *syntdenum, /* (o) synthesis filter coefficients */ + int16_t *weightdenum, /* (o) weighting denumerator + coefficients */ + int16_t *lsfdeq, /* (i) dequantized lsf coefficients */ + int16_t length, /* (i) length of lsf coefficient vector */ + IlbcDecoder *iLBCdec_inst + /* (i) the decoder state structure */ + ){ + size_t i; + int pos, lp_length; + int16_t lp[LPC_FILTERORDER + 1], *lsfdeq2; + + lsfdeq2 = lsfdeq + length; + lp_length = length + 1; + + if (iLBCdec_inst->mode==30) { + /* subframe 1: Interpolation between old and first LSF */ + + WebRtcIlbcfix_LspInterpolate2PolyDec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq, + WebRtcIlbcfix_kLsfWeight30ms[0], length); + WEBRTC_SPL_MEMCPY_W16(syntdenum,lp,lp_length); + WebRtcIlbcfix_BwExpand(weightdenum, lp, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length); + + /* subframes 2 to 6: interpolation between first and last LSF */ + + pos = lp_length; + for (i = 1; i < 6; i++) { + WebRtcIlbcfix_LspInterpolate2PolyDec(lp, lsfdeq, lsfdeq2, + WebRtcIlbcfix_kLsfWeight30ms[i], length); + WEBRTC_SPL_MEMCPY_W16(syntdenum + pos,lp,lp_length); + WebRtcIlbcfix_BwExpand(weightdenum + pos, lp, + (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length); + pos += lp_length; + } + } else { /* iLBCdec_inst->mode=20 */ + /* subframes 1 to 4: interpolation between old and new LSF */ + pos = 0; + for (i = 0; i < iLBCdec_inst->nsub; i++) { + WebRtcIlbcfix_LspInterpolate2PolyDec(lp, iLBCdec_inst->lsfdeqold, lsfdeq, + WebRtcIlbcfix_kLsfWeight20ms[i], length); + WEBRTC_SPL_MEMCPY_W16(syntdenum+pos,lp,lp_length); + WebRtcIlbcfix_BwExpand(weightdenum+pos, lp, + (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length); + pos += lp_length; + } + } + + /* update memory */ + + if (iLBCdec_inst->mode==30) { + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq2, length); + } else { + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq, length); + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h new file mode 100644 index 0000000000..37ecf078ee --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_DecoderInterpolateLsp.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * obtain synthesis and weighting filters form lsf coefficients + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_DecoderInterpolateLsp( + int16_t *syntdenum, /* (o) synthesis filter coefficients */ + int16_t *weightdenum, /* (o) weighting denumerator + coefficients */ + int16_t *lsfdeq, /* (i) dequantized lsf coefficients */ + int16_t length, /* (i) length of lsf coefficient vector */ + IlbcDecoder *iLBCdec_inst + /* (i) the decoder state structure */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/defines.h b/webrtc/modules/audio_coding/codecs/ilbc/defines.h new file mode 100644 index 0000000000..2faaea15fc --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/defines.h @@ -0,0 +1,220 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + define.h + +******************************************************************/ +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_ + +#include <string.h> +#include "signal_processing_library.h" +#include "webrtc/typedefs.h" + +/* general codec settings */ + +#define FS 8000 +#define BLOCKL_20MS 160 +#define BLOCKL_30MS 240 +#define BLOCKL_MAX 240 +#define NSUB_20MS 4 +#define NSUB_30MS 6 +#define NSUB_MAX 6 +#define NASUB_20MS 2 +#define NASUB_30MS 4 +#define NASUB_MAX 4 +#define SUBL 40 +#define STATE_LEN 80 +#define STATE_SHORT_LEN_30MS 58 +#define STATE_SHORT_LEN_20MS 57 + +/* LPC settings */ + +#define LPC_FILTERORDER 10 +#define LPC_LOOKBACK 60 +#define LPC_N_20MS 1 +#define LPC_N_30MS 2 +#define LPC_N_MAX 2 +#define LPC_ASYMDIFF 20 +#define LSF_NSPLIT 3 +#define LSF_NUMBER_OF_STEPS 4 +#define LPC_HALFORDER 5 +#define COS_GRID_POINTS 60 + +/* cb settings */ + +#define CB_NSTAGES 3 +#define CB_EXPAND 2 +#define CB_MEML 147 +#define CB_FILTERLEN (2*4) +#define CB_HALFFILTERLEN 4 +#define CB_RESRANGE 34 +#define CB_MAXGAIN_FIXQ6 83 /* error = -0.24% */ +#define CB_MAXGAIN_FIXQ14 21299 + +/* enhancer */ + +#define ENH_BLOCKL 80 /* block length */ +#define ENH_BLOCKL_HALF (ENH_BLOCKL/2) +#define ENH_HL 3 /* 2*ENH_HL+1 is number blocks + in said second sequence */ +#define ENH_SLOP 2 /* max difference estimated and + correct pitch period */ +#define ENH_PLOCSL 8 /* pitch-estimates and + pitch-locations buffer length */ +#define ENH_OVERHANG 2 +#define ENH_UPS0 4 /* upsampling rate */ +#define ENH_FL0 3 /* 2*FLO+1 is the length of each filter */ +#define ENH_FLO_MULT2_PLUS1 7 +#define ENH_VECTL (ENH_BLOCKL+2*ENH_FL0) +#define ENH_CORRDIM (2*ENH_SLOP+1) +#define ENH_NBLOCKS (BLOCKL/ENH_BLOCKL) +#define ENH_NBLOCKS_EXTRA 5 +#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */ +#define ENH_BUFL (ENH_NBLOCKS_TOT)*ENH_BLOCKL +#define ENH_BUFL_FILTEROVERHEAD 3 +#define ENH_A0 819 /* Q14 */ +#define ENH_A0_MINUS_A0A0DIV4 848256041 /* Q34 */ +#define ENH_A0DIV2 26843546 /* Q30 */ + +/* PLC */ + +/* Down sampling */ + +#define FILTERORDER_DS_PLUS1 7 +#define DELAY_DS 3 +#define FACTOR_DS 2 + +/* bit stream defs */ + +#define NO_OF_BYTES_20MS 38 +#define NO_OF_BYTES_30MS 50 +#define NO_OF_WORDS_20MS 19 +#define NO_OF_WORDS_30MS 25 +#define STATE_BITS 3 +#define BYTE_LEN 8 +#define ULP_CLASSES 3 + +/* help parameters */ + +#define TWO_PI_FIX 25736 /* Q12 */ + +/* Constants for codebook search and creation */ + +#define ST_MEM_L_TBL 85 +#define MEM_LF_TBL 147 + + +/* Struct for the bits */ +typedef struct iLBC_bits_t_ { + int16_t lsf[LSF_NSPLIT*LPC_N_MAX]; + int16_t cb_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB index */ + int16_t gain_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB gain */ + size_t idxForMax; + int16_t state_first; + int16_t idxVec[STATE_SHORT_LEN_30MS]; + int16_t firstbits; + size_t startIdx; +} iLBC_bits; + +/* type definition encoder instance */ +typedef struct IlbcEncoder_ { + + /* flag for frame size mode */ + int16_t mode; + + /* basic parameters for different frame sizes */ + size_t blockl; + size_t nsub; + int16_t nasub; + size_t no_of_bytes, no_of_words; + int16_t lpc_n; + size_t state_short_len; + + /* analysis filter state */ + int16_t anaMem[LPC_FILTERORDER]; + + /* Fix-point old lsf parameters for interpolation */ + int16_t lsfold[LPC_FILTERORDER]; + int16_t lsfdeqold[LPC_FILTERORDER]; + + /* signal buffer for LP analysis */ + int16_t lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX]; + + /* state of input HP filter */ + int16_t hpimemx[2]; + int16_t hpimemy[4]; + +#ifdef SPLIT_10MS + int16_t weightdenumbuf[66]; + int16_t past_samples[160]; + uint16_t bytes[25]; + int16_t section; + int16_t Nfor_flag; + int16_t Nback_flag; + int16_t start_pos; + size_t diff; +#endif + +} IlbcEncoder; + +/* type definition decoder instance */ +typedef struct IlbcDecoder_ { + + /* flag for frame size mode */ + int16_t mode; + + /* basic parameters for different frame sizes */ + size_t blockl; + size_t nsub; + int16_t nasub; + size_t no_of_bytes, no_of_words; + int16_t lpc_n; + size_t state_short_len; + + /* synthesis filter state */ + int16_t syntMem[LPC_FILTERORDER]; + + /* old LSF for interpolation */ + int16_t lsfdeqold[LPC_FILTERORDER]; + + /* pitch lag estimated in enhancer and used in PLC */ + size_t last_lag; + + /* PLC state information */ + int consPLICount, prev_enh_pl; + int16_t perSquare; + + int16_t prevScale, prevPLI; + size_t prevLag; + int16_t prevLpc[LPC_FILTERORDER+1]; + int16_t prevResidual[NSUB_MAX*SUBL]; + int16_t seed; + + /* previous synthesis filter parameters */ + + int16_t old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX]; + + /* state of output HP filter */ + int16_t hpimemx[2]; + int16_t hpimemy[4]; + + /* enhancer state information */ + int use_enhancer; + int16_t enh_buf[ENH_BUFL+ENH_BUFL_FILTEROVERHEAD]; + size_t enh_period[ENH_NBLOCKS_TOT]; + +} IlbcDecoder; + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c new file mode 100644 index 0000000000..f74439ede6 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c @@ -0,0 +1,305 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_DoThePlc.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "comp_corr.h" +#include "bw_expand.h" + +/*----------------------------------------------------------------* + * Packet loss concealment routine. Conceals a residual signal + * and LP parameters. If no packet loss, update state. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_DoThePlc( + int16_t *PLCresidual, /* (o) concealed residual */ + int16_t *PLClpc, /* (o) concealed LP parameters */ + int16_t PLI, /* (i) packet loss indicator + 0 - no PL, 1 = PL */ + int16_t *decresidual, /* (i) decoded residual */ + int16_t *lpc, /* (i) decoded LPC (only used for no PL) */ + size_t inlag, /* (i) pitch lag */ + IlbcDecoder *iLBCdec_inst + /* (i/o) decoder instance */ + ){ + size_t i; + int32_t cross, ener, cross_comp, ener_comp = 0; + int32_t measure, maxMeasure, energy; + int16_t max, crossSquareMax, crossSquare; + size_t j, lag, randlag; + int16_t tmp1, tmp2; + int16_t shift1, shift2, shift3, shiftMax; + int16_t scale3; + size_t corrLen; + int32_t tmpW32, tmp2W32; + int16_t use_gain; + int16_t tot_gain; + int16_t max_perSquare; + int16_t scale1, scale2; + int16_t totscale; + int32_t nom; + int16_t denom; + int16_t pitchfact; + size_t use_lag; + int ind; + int16_t randvec[BLOCKL_MAX]; + + /* Packet Loss */ + if (PLI == 1) { + + (*iLBCdec_inst).consPLICount += 1; + + /* if previous frame not lost, + determine pitch pred. gain */ + + if (iLBCdec_inst->prevPLI != 1) { + + /* Maximum 60 samples are correlated, preserve as high accuracy + as possible without getting overflow */ + max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual, + iLBCdec_inst->blockl); + scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25; + if (scale3 < 0) { + scale3 = 0; + } + + /* Store scale for use when interpolating between the + * concealment and the received packet */ + iLBCdec_inst->prevScale = scale3; + + /* Search around the previous lag +/-3 to find the + best pitch period */ + lag = inlag - 3; + + /* Guard against getting outside the frame */ + corrLen = (size_t)WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3)); + + WebRtcIlbcfix_CompCorr( &cross, &ener, + iLBCdec_inst->prevResidual, lag, iLBCdec_inst->blockl, corrLen, scale3); + + /* Normalize and store cross^2 and the number of shifts */ + shiftMax = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross))-15; + crossSquareMax = (int16_t)(( + (int16_t)WEBRTC_SPL_SHIFT_W32(cross, -shiftMax) * + (int16_t)WEBRTC_SPL_SHIFT_W32(cross, -shiftMax)) >> 15); + + for (j=inlag-2;j<=inlag+3;j++) { + WebRtcIlbcfix_CompCorr( &cross_comp, &ener_comp, + iLBCdec_inst->prevResidual, j, iLBCdec_inst->blockl, corrLen, scale3); + + /* Use the criteria (corr*corr)/energy to compare if + this lag is better or not. To avoid the division, + do a cross multiplication */ + shift1 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross_comp))-15; + crossSquare = (int16_t)(( + (int16_t)WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1) * + (int16_t)WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1)) >> 15); + + shift2 = WebRtcSpl_GetSizeInBits(ener)-15; + measure = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, -shift2) * crossSquare; + + shift3 = WebRtcSpl_GetSizeInBits(ener_comp)-15; + maxMeasure = (int16_t)WEBRTC_SPL_SHIFT_W32(ener_comp, -shift3) * + crossSquareMax; + + /* Calculate shift value, so that the two measures can + be put in the same Q domain */ + if(((shiftMax<<1)+shift3) > ((shift1<<1)+shift2)) { + tmp1 = WEBRTC_SPL_MIN(31, (shiftMax<<1)+shift3-(shift1<<1)-shift2); + tmp2 = 0; + } else { + tmp1 = 0; + tmp2 = WEBRTC_SPL_MIN(31, (shift1<<1)+shift2-(shiftMax<<1)-shift3); + } + + if ((measure>>tmp1) > (maxMeasure>>tmp2)) { + /* New lag is better => record lag, measure and domain */ + lag = j; + crossSquareMax = crossSquare; + cross = cross_comp; + shiftMax = shift1; + ener = ener_comp; + } + } + + /* Calculate the periodicity for the lag with the maximum correlation. + + Definition of the periodicity: + abs(corr(vec1, vec2))/(sqrt(energy(vec1))*sqrt(energy(vec2))) + + Work in the Square domain to simplify the calculations + max_perSquare is less than 1 (in Q15) + */ + tmp2W32=WebRtcSpl_DotProductWithScale(&iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen], + &iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen], + corrLen, scale3); + + if ((tmp2W32>0)&&(ener_comp>0)) { + /* norm energies to int16_t, compute the product of the energies and + use the upper int16_t as the denominator */ + + scale1=(int16_t)WebRtcSpl_NormW32(tmp2W32)-16; + tmp1=(int16_t)WEBRTC_SPL_SHIFT_W32(tmp2W32, scale1); + + scale2=(int16_t)WebRtcSpl_NormW32(ener)-16; + tmp2=(int16_t)WEBRTC_SPL_SHIFT_W32(ener, scale2); + denom = (int16_t)((tmp1 * tmp2) >> 16); /* in Q(scale1+scale2-16) */ + + /* Square the cross correlation and norm it such that max_perSquare + will be in Q15 after the division */ + + totscale = scale1+scale2-1; + tmp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, (totscale>>1)); + tmp2 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, totscale-(totscale>>1)); + + nom = tmp1 * tmp2; + max_perSquare = (int16_t)WebRtcSpl_DivW32W16(nom, denom); + + } else { + max_perSquare = 0; + } + } + + /* previous frame lost, use recorded lag and gain */ + + else { + lag = iLBCdec_inst->prevLag; + max_perSquare = iLBCdec_inst->perSquare; + } + + /* Attenuate signal and scale down pitch pred gain if + several frames lost consecutively */ + + use_gain = 32767; /* 1.0 in Q15 */ + + if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320) { + use_gain = 29491; /* 0.9 in Q15 */ + } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>640) { + use_gain = 22938; /* 0.7 in Q15 */ + } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>960) { + use_gain = 16384; /* 0.5 in Q15 */ + } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>1280) { + use_gain = 0; /* 0.0 in Q15 */ + } + + /* Compute mixing factor of picth repeatition and noise: + for max_per>0.7 set periodicity to 1.0 + 0.4<max_per<0.7 set periodicity to (maxper-0.4)/0.7-0.4) + max_per<0.4 set periodicity to 0.0 + */ + + if (max_perSquare>7868) { /* periodicity > 0.7 (0.7^4=0.2401 in Q15) */ + pitchfact = 32767; + } else if (max_perSquare>839) { /* 0.4 < periodicity < 0.7 (0.4^4=0.0256 in Q15) */ + /* find best index and interpolate from that */ + ind = 5; + while ((max_perSquare<WebRtcIlbcfix_kPlcPerSqr[ind])&&(ind>0)) { + ind--; + } + /* pitch fact is approximated by first order */ + tmpW32 = (int32_t)WebRtcIlbcfix_kPlcPitchFact[ind] + + ((WebRtcIlbcfix_kPlcPfSlope[ind] * + (max_perSquare - WebRtcIlbcfix_kPlcPerSqr[ind])) >> 11); + + pitchfact = (int16_t)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */ + + } else { /* periodicity < 0.4 */ + pitchfact = 0; + } + + /* avoid repetition of same pitch cycle (buzzyness) */ + use_lag = lag; + if (lag<80) { + use_lag = 2*lag; + } + + /* compute concealed residual */ + energy = 0; + + for (i=0; i<iLBCdec_inst->blockl; i++) { + + /* noise component - 52 < randlagFIX < 117 */ + iLBCdec_inst->seed = (int16_t)(iLBCdec_inst->seed * 31821 + 13849); + randlag = 53 + (iLBCdec_inst->seed & 63); + if (randlag > i) { + randvec[i] = + iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - randlag]; + } else { + randvec[i] = iLBCdec_inst->prevResidual[i - randlag]; + } + + /* pitch repeatition component */ + if (use_lag > i) { + PLCresidual[i] = + iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - use_lag]; + } else { + PLCresidual[i] = PLCresidual[i - use_lag]; + } + + /* Attinuate total gain for each 10 ms */ + if (i<80) { + tot_gain=use_gain; + } else if (i<160) { + tot_gain = (int16_t)((31130 * use_gain) >> 15); /* 0.95*use_gain */ + } else { + tot_gain = (int16_t)((29491 * use_gain) >> 15); /* 0.9*use_gain */ + } + + + /* mix noise and pitch repeatition */ + PLCresidual[i] = (int16_t)((tot_gain * + ((pitchfact * PLCresidual[i] + (32767 - pitchfact) * randvec[i] + + 16384) >> 15)) >> 15); + + /* Shifting down the result one step extra to ensure that no overflow + will occur */ + energy += (PLCresidual[i] * PLCresidual[i]) >> + (iLBCdec_inst->prevScale + 1); + } + + /* less than 30 dB, use only noise */ + if (energy < (WEBRTC_SPL_SHIFT_W32(((int32_t)iLBCdec_inst->blockl*900),-(iLBCdec_inst->prevScale+1)))) { + energy = 0; + for (i=0; i<iLBCdec_inst->blockl; i++) { + PLCresidual[i] = randvec[i]; + } + } + + /* use the old LPC */ + WEBRTC_SPL_MEMCPY_W16(PLClpc, (*iLBCdec_inst).prevLpc, LPC_FILTERORDER+1); + + /* Update state in case there are multiple frame losses */ + iLBCdec_inst->prevLag = lag; + iLBCdec_inst->perSquare = max_perSquare; + } + + /* no packet loss, copy input */ + + else { + WEBRTC_SPL_MEMCPY_W16(PLCresidual, decresidual, iLBCdec_inst->blockl); + WEBRTC_SPL_MEMCPY_W16(PLClpc, lpc, (LPC_FILTERORDER+1)); + iLBCdec_inst->consPLICount = 0; + } + + /* update state */ + iLBCdec_inst->prevPLI = PLI; + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevLpc, PLClpc, (LPC_FILTERORDER+1)); + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevResidual, PLCresidual, iLBCdec_inst->blockl); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h new file mode 100644 index 0000000000..38b8fdb7c0 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h @@ -0,0 +1,41 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_DoThePlc.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Packet loss concealment routine. Conceals a residual signal + * and LP parameters. If no packet loss, update state. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_DoThePlc( + int16_t *PLCresidual, /* (o) concealed residual */ + int16_t *PLClpc, /* (o) concealed LP parameters */ + int16_t PLI, /* (i) packet loss indicator + 0 - no PL, 1 = PL */ + int16_t *decresidual, /* (i) decoded residual */ + int16_t *lpc, /* (i) decoded LPC (only used for no PL) */ + size_t inlag, /* (i) pitch lag */ + IlbcDecoder *iLBCdec_inst + /* (i/o) decoder instance */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/encode.c b/webrtc/modules/audio_coding/codecs/ilbc/encode.c new file mode 100644 index 0000000000..812ec8d6c7 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/encode.c @@ -0,0 +1,518 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Encode.c + +******************************************************************/ + +#include <string.h> + +#include "defines.h" +#include "lpc_encode.h" +#include "frame_classify.h" +#include "state_search.h" +#include "state_construct.h" +#include "constants.h" +#include "cb_search.h" +#include "cb_construct.h" +#include "index_conv_enc.h" +#include "pack_bits.h" +#include "hp_input.h" + +#ifdef SPLIT_10MS +#include "unpack_bits.h" +#include "index_conv_dec.h" +#endif +#ifndef WEBRTC_ARCH_BIG_ENDIAN +#include "swap_bytes.h" +#endif + +/*----------------------------------------------------------------* + * main encoder function + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_EncodeImpl( + uint16_t *bytes, /* (o) encoded data bits iLBC */ + const int16_t *block, /* (i) speech vector to encode */ + IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder + state */ + ){ + size_t n, meml_gotten, Nfor; + size_t diff, start_pos; + size_t index; + size_t subcount, subframe; + size_t start_count, end_count; + int16_t *residual; + int32_t en1, en2; + int16_t scale, max; + int16_t *syntdenum; + int16_t *decresidual; + int16_t *reverseResidual; + int16_t *reverseDecresidual; + /* Stack based */ + int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX]; + int16_t dataVec[BLOCKL_MAX + LPC_FILTERORDER]; + int16_t memVec[CB_MEML+CB_FILTERLEN]; + int16_t bitsMemory[sizeof(iLBC_bits)/sizeof(int16_t)]; + iLBC_bits *iLBCbits_inst = (iLBC_bits*)bitsMemory; + + +#ifdef SPLIT_10MS + int16_t *weightdenumbuf = iLBCenc_inst->weightdenumbuf; + int16_t last_bit; +#endif + + int16_t *data = &dataVec[LPC_FILTERORDER]; + int16_t *mem = &memVec[CB_HALFFILTERLEN]; + + /* Reuse som buffers to save stack memory */ + residual = &iLBCenc_inst->lpc_buffer[LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl]; + syntdenum = mem; /* syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX] and mem are used non overlapping in the code */ + decresidual = residual; /* Already encoded residual is overwritten by the decoded version */ + reverseResidual = data; /* data and reverseResidual are used non overlapping in the code */ + reverseDecresidual = reverseResidual; /* Already encoded residual is overwritten by the decoded version */ + +#ifdef SPLIT_10MS + + WebRtcSpl_MemSetW16 ( (int16_t *) iLBCbits_inst, 0, + sizeof(iLBC_bits) / sizeof(int16_t) ); + + start_pos = iLBCenc_inst->start_pos; + diff = iLBCenc_inst->diff; + + if (iLBCenc_inst->section != 0){ + WEBRTC_SPL_MEMCPY_W16 (weightdenum, weightdenumbuf, + SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM); + /* Un-Packetize the frame into parameters */ + last_bit = WebRtcIlbcfix_UnpackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode); + if (last_bit) + return; + /* adjust index */ + WebRtcIlbcfix_IndexConvDec (iLBCbits_inst->cb_index); + + if (iLBCenc_inst->section == 1){ + /* Save first 80 samples of a 160/240 sample frame for 20/30msec */ + WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples, block, 80); + } + else{ // iLBCenc_inst->section == 2 AND mode = 30ms + /* Save second 80 samples of a 240 sample frame for 30msec */ + WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples + 80, block, 80); + } + } + else{ // iLBCenc_inst->section == 0 + /* form a complete frame of 160/240 for 20msec/30msec mode */ + WEBRTC_SPL_MEMCPY_W16 (data + (iLBCenc_inst->mode * 8) - 80, block, 80); + WEBRTC_SPL_MEMCPY_W16 (data, iLBCenc_inst->past_samples, + (iLBCenc_inst->mode * 8) - 80); + iLBCenc_inst->Nfor_flag = 0; + iLBCenc_inst->Nback_flag = 0; +#else + /* copy input block to data*/ + WEBRTC_SPL_MEMCPY_W16(data,block,iLBCenc_inst->blockl); +#endif + + /* high pass filtering of input signal and scale down the residual (*0.5) */ + WebRtcIlbcfix_HpInput(data, (int16_t*)WebRtcIlbcfix_kHpInCoefs, + iLBCenc_inst->hpimemy, iLBCenc_inst->hpimemx, + iLBCenc_inst->blockl); + + /* LPC of hp filtered input data */ + WebRtcIlbcfix_LpcEncode(syntdenum, weightdenum, iLBCbits_inst->lsf, data, + iLBCenc_inst); + + /* Set up state */ + WEBRTC_SPL_MEMCPY_W16(dataVec, iLBCenc_inst->anaMem, LPC_FILTERORDER); + + /* inverse filter to get residual */ + for (n=0; n<iLBCenc_inst->nsub; n++ ) { + WebRtcSpl_FilterMAFastQ12( + &data[n*SUBL], &residual[n*SUBL], + &syntdenum[n*(LPC_FILTERORDER+1)], + LPC_FILTERORDER+1, SUBL); + } + + /* Copy the state for next frame */ + WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->anaMem, &data[iLBCenc_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER); + + /* find state location */ + + iLBCbits_inst->startIdx = WebRtcIlbcfix_FrameClassify(iLBCenc_inst,residual); + + /* check if state should be in first or last part of the + two subframes */ + + index = (iLBCbits_inst->startIdx-1)*SUBL; + max=WebRtcSpl_MaxAbsValueW16(&residual[index], 2*SUBL); + scale = WebRtcSpl_GetSizeInBits((uint32_t)(max * max)); + + /* Scale to maximum 25 bits so that the MAC won't cause overflow */ + scale = scale - 25; + if(scale < 0) { + scale = 0; + } + + diff = STATE_LEN - iLBCenc_inst->state_short_len; + en1=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index], + iLBCenc_inst->state_short_len, scale); + index += diff; + en2=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index], + iLBCenc_inst->state_short_len, scale); + if (en1 > en2) { + iLBCbits_inst->state_first = 1; + start_pos = (iLBCbits_inst->startIdx-1)*SUBL; + } else { + iLBCbits_inst->state_first = 0; + start_pos = (iLBCbits_inst->startIdx-1)*SUBL + diff; + } + + /* scalar quantization of state */ + + WebRtcIlbcfix_StateSearch(iLBCenc_inst, iLBCbits_inst, &residual[start_pos], + &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)], + &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)]); + + WebRtcIlbcfix_StateConstruct(iLBCbits_inst->idxForMax, iLBCbits_inst->idxVec, + &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)], + &decresidual[start_pos], iLBCenc_inst->state_short_len + ); + + /* predictive quantization in state */ + + if (iLBCbits_inst->state_first) { /* put adaptive part in the end */ + + /* setup memory */ + + WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len); + WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len, + decresidual+start_pos, iLBCenc_inst->state_short_len); + + /* encode subframes */ + + WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index, + &residual[start_pos+iLBCenc_inst->state_short_len], + mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff, + &weightdenum[iLBCbits_inst->startIdx*(LPC_FILTERORDER+1)], 0); + + /* construct decoded vector */ + + WebRtcIlbcfix_CbConstruct(&decresidual[start_pos+iLBCenc_inst->state_short_len], + iLBCbits_inst->cb_index, iLBCbits_inst->gain_index, + mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, + diff + ); + + } + else { /* put adaptive part in the beginning */ + + /* create reversed vectors for prediction */ + + WebRtcSpl_MemCpyReversedOrder(&reverseResidual[diff-1], + &residual[(iLBCbits_inst->startIdx+1)*SUBL-STATE_LEN], diff); + + /* setup memory */ + + meml_gotten = iLBCenc_inst->state_short_len; + WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten); + WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len); + + /* encode subframes */ + WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index, + reverseResidual, mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff, + &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)], + 0); + + /* construct decoded vector */ + + WebRtcIlbcfix_CbConstruct(reverseDecresidual, + iLBCbits_inst->cb_index, iLBCbits_inst->gain_index, + mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, + diff + ); + + /* get decoded residual from reversed vector */ + + WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1], reverseDecresidual, diff); + } + +#ifdef SPLIT_10MS + iLBCenc_inst->start_pos = start_pos; + iLBCenc_inst->diff = diff; + iLBCenc_inst->section++; + /* adjust index */ + WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index); + /* Packetize the parameters into the frame */ + WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode); + WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum, + SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM); + return; + } +#endif + + /* forward prediction of subframes */ + + Nfor = iLBCenc_inst->nsub-iLBCbits_inst->startIdx-1; + + /* counter for predicted subframes */ +#ifdef SPLIT_10MS + if (iLBCenc_inst->mode == 20) + { + subcount = 1; + } + if (iLBCenc_inst->mode == 30) + { + if (iLBCenc_inst->section == 1) + { + subcount = 1; + } + if (iLBCenc_inst->section == 2) + { + subcount = 3; + } + } +#else + subcount=1; +#endif + + if( Nfor > 0 ){ + + /* setup memory */ + + WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN); + WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN, + decresidual+(iLBCbits_inst->startIdx-1)*SUBL, STATE_LEN); + +#ifdef SPLIT_10MS + if (iLBCenc_inst->Nfor_flag > 0) + { + for (subframe = 0; subframe < WEBRTC_SPL_MIN (Nfor, 2); subframe++) + { + /* update memory */ + WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL)); + WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL, + &decresidual[(iLBCbits_inst->startIdx + 1 + + subframe) * SUBL], SUBL); + } + } + + iLBCenc_inst->Nfor_flag++; + + if (iLBCenc_inst->mode == 20) + { + start_count = 0; + end_count = Nfor; + } + if (iLBCenc_inst->mode == 30) + { + if (iLBCenc_inst->section == 1) + { + start_count = 0; + end_count = WEBRTC_SPL_MIN (Nfor, (size_t)2); + } + if (iLBCenc_inst->section == 2) + { + start_count = WEBRTC_SPL_MIN (Nfor, (size_t)2); + end_count = Nfor; + } + } +#else + start_count = 0; + end_count = Nfor; +#endif + + /* loop over subframes to encode */ + + for (subframe = start_count; subframe < end_count; subframe++){ + + /* encode subframe */ + + WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES, + iLBCbits_inst->gain_index+subcount*CB_NSTAGES, + &residual[(iLBCbits_inst->startIdx+1+subframe)*SUBL], + mem, MEM_LF_TBL, SUBL, + &weightdenum[(iLBCbits_inst->startIdx+1+subframe)*(LPC_FILTERORDER+1)], + subcount); + + /* construct decoded vector */ + + WebRtcIlbcfix_CbConstruct(&decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL], + iLBCbits_inst->cb_index+subcount*CB_NSTAGES, + iLBCbits_inst->gain_index+subcount*CB_NSTAGES, + mem, MEM_LF_TBL, + SUBL + ); + + /* update memory */ + + memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem)); + WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL, + &decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL], SUBL); + + subcount++; + } + } + +#ifdef SPLIT_10MS + if ((iLBCenc_inst->section == 1) && + (iLBCenc_inst->mode == 30) && (Nfor > 0) && (end_count == 2)) + { + iLBCenc_inst->section++; + /* adjust index */ + WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index); + /* Packetize the parameters into the frame */ + WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode); + WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum, + SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM); + return; + } +#endif + + /* backward prediction of subframes */ + + if (iLBCbits_inst->startIdx > 1) { + + /* create reverse order vectors + (The decresidual does not need to be copied since it is + contained in the same vector as the residual) + */ + + size_t Nback = iLBCbits_inst->startIdx - 1; + WebRtcSpl_MemCpyReversedOrder(&reverseResidual[Nback*SUBL-1], residual, Nback*SUBL); + + /* setup memory */ + + meml_gotten = SUBL*(iLBCenc_inst->nsub+1-iLBCbits_inst->startIdx); + if( meml_gotten > CB_MEML ) { + meml_gotten=CB_MEML; + } + + WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten); + WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten); + +#ifdef SPLIT_10MS + if (iLBCenc_inst->Nback_flag > 0) + { + for (subframe = 0; subframe < WEBRTC_SPL_MAX (2 - Nfor, 0); subframe++) + { + /* update memory */ + WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL)); + WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL, + &reverseDecresidual[subframe * SUBL], SUBL); + } + } + + iLBCenc_inst->Nback_flag++; + + + if (iLBCenc_inst->mode == 20) + { + start_count = 0; + end_count = Nback; + } + if (iLBCenc_inst->mode == 30) + { + if (iLBCenc_inst->section == 1) + { + start_count = 0; + end_count = (Nfor >= 2) ? 0 : (2 - NFor); + } + if (iLBCenc_inst->section == 2) + { + start_count = (Nfor >= 2) ? 0 : (2 - NFor); + end_count = Nback; + } + } +#else + start_count = 0; + end_count = Nback; +#endif + + /* loop over subframes to encode */ + + for (subframe = start_count; subframe < end_count; subframe++){ + + /* encode subframe */ + + WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES, + iLBCbits_inst->gain_index+subcount*CB_NSTAGES, &reverseResidual[subframe*SUBL], + mem, MEM_LF_TBL, SUBL, + &weightdenum[(iLBCbits_inst->startIdx-2-subframe)*(LPC_FILTERORDER+1)], + subcount); + + /* construct decoded vector */ + + WebRtcIlbcfix_CbConstruct(&reverseDecresidual[subframe*SUBL], + iLBCbits_inst->cb_index+subcount*CB_NSTAGES, + iLBCbits_inst->gain_index+subcount*CB_NSTAGES, + mem, MEM_LF_TBL, SUBL + ); + + /* update memory */ + memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem)); + WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL, + &reverseDecresidual[subframe*SUBL], SUBL); + + subcount++; + + } + + /* get decoded residual from reversed vector */ + + WebRtcSpl_MemCpyReversedOrder(&decresidual[SUBL*Nback-1], reverseDecresidual, SUBL*Nback); + } + /* end encoding part */ + + /* adjust index */ + + WebRtcIlbcfix_IndexConvEnc(iLBCbits_inst->cb_index); + + /* Packetize the parameters into the frame */ + +#ifdef SPLIT_10MS + if( (iLBCenc_inst->mode==30) && (iLBCenc_inst->section==1) ){ + WebRtcIlbcfix_PackBits(iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode); + } + else{ + WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode); + } +#else + WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode); +#endif + +#ifndef WEBRTC_ARCH_BIG_ENDIAN + /* Swap bytes for LITTLE ENDIAN since the packbits() + function assumes BIG_ENDIAN machine */ +#ifdef SPLIT_10MS + if (( (iLBCenc_inst->section == 1) && (iLBCenc_inst->mode == 20) ) || + ( (iLBCenc_inst->section == 2) && (iLBCenc_inst->mode == 30) )){ + WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words, bytes); + } +#else + WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words, bytes); +#endif +#endif + +#ifdef SPLIT_10MS + if (subcount == (iLBCenc_inst->nsub - 1)) + { + iLBCenc_inst->section = 0; + } + else + { + iLBCenc_inst->section++; + WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum, + SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM); + } +#endif + +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/encode.h b/webrtc/modules/audio_coding/codecs/ilbc/encode.h new file mode 100644 index 0000000000..fe6ae6250c --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/encode.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Encode.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * main encoder function + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_EncodeImpl( + uint16_t *bytes, /* (o) encoded data bits iLBC */ + const int16_t *block, /* (i) speech vector to encode */ + IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder + state */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c b/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c new file mode 100644 index 0000000000..b2bdcfffc3 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_EnergyInverse.c + +******************************************************************/ + +/* Inverses the in vector in into Q29 domain */ + +#include "energy_inverse.h" + +void WebRtcIlbcfix_EnergyInverse( + int16_t *energy, /* (i/o) Energy and inverse + energy (in Q29) */ + size_t noOfEnergies) /* (i) The length of the energy + vector */ +{ + int32_t Nom=(int32_t)0x1FFFFFFF; + int16_t *energyPtr; + size_t i; + + /* Set the minimum energy value to 16384 to avoid overflow */ + energyPtr=energy; + for (i=0; i<noOfEnergies; i++) { + (*energyPtr)=WEBRTC_SPL_MAX((*energyPtr),16384); + energyPtr++; + } + + /* Calculate inverse energy in Q29 */ + energyPtr=energy; + for (i=0; i<noOfEnergies; i++) { + (*energyPtr) = (int16_t)WebRtcSpl_DivW32W16(Nom, (*energyPtr)); + energyPtr++; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h b/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h new file mode 100644 index 0000000000..fe25094325 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h @@ -0,0 +1,32 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_EnergyInverse.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_ + +#include "defines.h" + +/* Inverses the in vector in into Q29 domain */ + +void WebRtcIlbcfix_EnergyInverse( + int16_t *energy, /* (i/o) Energy and inverse + energy (in Q29) */ + size_t noOfEnergies); /* (i) The length of the energy + vector */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c b/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c new file mode 100644 index 0000000000..929c630c68 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c @@ -0,0 +1,110 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_EnhUpsample.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * upsample finite array assuming zeros outside bounds + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_EnhUpsample( + int32_t *useq1, /* (o) upsampled output sequence */ + int16_t *seq1 /* (i) unupsampled sequence */ + ){ + int j; + int32_t *pu1, *pu11; + int16_t *ps, *w16tmp; + const int16_t *pp; + + /* filtering: filter overhangs left side of sequence */ + pu1=useq1; + for (j=0;j<ENH_UPS0; j++) { + pu11=pu1; + /* i = 2 */ + pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1; + ps=seq1+2; + *pu11 = (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + pu11+=ENH_UPS0; + /* i = 3 */ + pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1; + ps=seq1+3; + *pu11 = (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + pu11+=ENH_UPS0; + /* i = 4 */ + pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1; + ps=seq1+4; + *pu11 = (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + pu1++; + } + + /* filtering: simple convolution=inner products + (not needed since the sequence is so short) + */ + + /* filtering: filter overhangs right side of sequence */ + + /* Code with loops, which is equivivalent to the expanded version below + + filterlength = 5; + hf1 = 2; + for(j=0;j<ENH_UPS0; j++){ + pu = useq1 + (filterlength-hfl)*ENH_UPS0 + j; + for(i=1; i<=hfl; i++){ + *pu=0; + pp = polyp[j]+i; + ps = seq1+dim1-1; + for(k=0;k<filterlength-i;k++) { + *pu += (*ps--) * *pp++; + } + pu+=ENH_UPS0; + } + } + */ + pu1 = useq1 + 12; + w16tmp = seq1+4; + for (j=0;j<ENH_UPS0; j++) { + pu11 = pu1; + /* i = 1 */ + pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+2; + ps = w16tmp; + *pu11 = (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + pu11+=ENH_UPS0; + /* i = 2 */ + pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+3; + ps = w16tmp; + *pu11 = (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + *pu11 += (*ps--) * *pp++; + pu11+=ENH_UPS0; + + pu1++; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h b/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h new file mode 100644 index 0000000000..00bb28b0d8 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_EnhUpsample.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * upsample finite array assuming zeros outside bounds + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_EnhUpsample( + int32_t *useq1, /* (o) upsampled output sequence */ + int16_t *seq1 /* (i) unupsampled sequence */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c b/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c new file mode 100644 index 0000000000..521d00441c --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Enhancer.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "get_sync_seq.h" +#include "smooth.h" + +/*----------------------------------------------------------------* + * perform enhancement on idata+centerStartPos through + * idata+centerStartPos+ENH_BLOCKL-1 + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Enhancer( + int16_t *odata, /* (o) smoothed block, dimension blockl */ + int16_t *idata, /* (i) data buffer used for enhancing */ + size_t idatal, /* (i) dimension idata */ + size_t centerStartPos, /* (i) first sample current block within idata */ + size_t *period, /* (i) pitch period array (pitch bward-in time) */ + const size_t *plocs, /* (i) locations where period array values valid */ + size_t periodl /* (i) dimension of period and plocs */ + ){ + /* Stack based */ + int16_t surround[ENH_BLOCKL]; + + WebRtcSpl_MemSetW16(surround, 0, ENH_BLOCKL); + + /* get said second sequence of segments */ + + WebRtcIlbcfix_GetSyncSeq(idata, idatal, centerStartPos, period, plocs, + periodl, ENH_HL, surround); + + /* compute the smoothed output from said second sequence */ + + WebRtcIlbcfix_Smooth(odata, idata + centerStartPos, surround); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h b/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h new file mode 100644 index 0000000000..ed219fb1bb --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h @@ -0,0 +1,39 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Enhancer.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * perform enhancement on idata+centerStartPos through + * idata+centerStartPos+ENH_BLOCKL-1 + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Enhancer( + int16_t *odata, /* (o) smoothed block, dimension blockl */ + int16_t *idata, /* (i) data buffer used for enhancing */ + size_t idatal, /* (i) dimension idata */ + size_t centerStartPos, /* (i) first sample current block within idata */ + size_t *period, /* (i) pitch period array (pitch bward-in time) */ + const size_t *plocs, /* (i) locations where period array values valid */ + size_t periodl /* (i) dimension of period and plocs */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c new file mode 100644 index 0000000000..1c0fd42383 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c @@ -0,0 +1,377 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_EnhancerInterface.c + +******************************************************************/ + +#include <string.h> + +#include "defines.h" +#include "constants.h" +#include "xcorr_coef.h" +#include "enhancer.h" +#include "hp_output.h" + + + +/*----------------------------------------------------------------* + * interface for enhancer + *---------------------------------------------------------------*/ + +size_t WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */ + int16_t *out, /* (o) enhanced signal */ + int16_t *in, /* (i) unenhanced signal */ + IlbcDecoder *iLBCdec_inst /* (i) buffers etc */ + ){ + size_t iblock; + size_t lag=20, tlag=20; + size_t inLen=iLBCdec_inst->blockl+120; + int16_t scale, scale1; + size_t plc_blockl; + int16_t *enh_buf; + size_t *enh_period; + int32_t tmp1, tmp2, max; + size_t new_blocks; + int16_t *enh_bufPtr1; + size_t i; + size_t k; + int16_t EnChange; + int16_t SqrtEnChange; + int16_t inc; + int16_t win; + int16_t *tmpW16ptr; + size_t startPos; + int16_t *plc_pred; + int16_t *target, *regressor; + int16_t max16; + int shifts; + int32_t ener; + int16_t enerSh; + int16_t corrSh; + size_t ind; + int16_t sh; + size_t start, stop; + /* Stack based */ + int16_t totsh[3]; + int16_t downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */ + int32_t corr32[50]; + int32_t corrmax[3]; + int16_t corr16[3]; + int16_t en16[3]; + size_t lagmax[3]; + + plc_pred = downsampled; /* Reuse memory since plc_pred[ENH_BLOCKL] and + downsampled are non overlapping */ + enh_buf=iLBCdec_inst->enh_buf; + enh_period=iLBCdec_inst->enh_period; + + /* Copy in the new data into the enhancer buffer */ + memmove(enh_buf, &enh_buf[iLBCdec_inst->blockl], + (ENH_BUFL - iLBCdec_inst->blockl) * sizeof(*enh_buf)); + + WEBRTC_SPL_MEMCPY_W16(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in, + iLBCdec_inst->blockl); + + /* Set variables that are dependent on frame size */ + if (iLBCdec_inst->mode==30) { + plc_blockl=ENH_BLOCKL; + new_blocks=3; + startPos=320; /* Start position for enhancement + (640-new_blocks*ENH_BLOCKL-80) */ + } else { + plc_blockl=40; + new_blocks=2; + startPos=440; /* Start position for enhancement + (640-new_blocks*ENH_BLOCKL-40) */ + } + + /* Update the pitch prediction for each enhancer block, move the old ones */ + memmove(enh_period, &enh_period[new_blocks], + (ENH_NBLOCKS_TOT - new_blocks) * sizeof(*enh_period)); + + WebRtcSpl_DownsampleFast( + enh_buf+ENH_BUFL-inLen, /* Input samples */ + inLen + ENH_BUFL_FILTEROVERHEAD, + downsampled, + inLen / 2, + (int16_t*)WebRtcIlbcfix_kLpFiltCoefs, /* Coefficients in Q12 */ + FILTERORDER_DS_PLUS1, /* Length of filter (order-1) */ + FACTOR_DS, + DELAY_DS); + + /* Estimate the pitch in the down sampled domain. */ + for(iblock = 0; iblock<new_blocks; iblock++){ + + /* references */ + target = downsampled + 60 + iblock * ENH_BLOCKL_HALF; + regressor = target - 10; + + /* scaling */ + max16 = WebRtcSpl_MaxAbsValueW16(®ressor[-50], ENH_BLOCKL_HALF + 50 - 1); + shifts = WebRtcSpl_GetSizeInBits((uint32_t)(max16 * max16)) - 25; + shifts = WEBRTC_SPL_MAX(0, shifts); + + /* compute cross correlation */ + WebRtcSpl_CrossCorrelation(corr32, target, regressor, ENH_BLOCKL_HALF, 50, + shifts, -1); + + /* Find 3 highest correlations that should be compared for the + highest (corr*corr)/ener */ + + for (i=0;i<2;i++) { + lagmax[i] = WebRtcSpl_MaxIndexW32(corr32, 50); + corrmax[i] = corr32[lagmax[i]]; + start = WEBRTC_SPL_MAX(2, lagmax[i]) - 2; + stop = WEBRTC_SPL_MIN(47, lagmax[i]) + 2; + for (k = start; k <= stop; k++) { + corr32[k] = 0; + } + } + lagmax[2] = WebRtcSpl_MaxIndexW32(corr32, 50); + corrmax[2] = corr32[lagmax[2]]; + + /* Calculate normalized corr^2 and ener */ + for (i=0;i<3;i++) { + corrSh = 15-WebRtcSpl_GetSizeInBits(corrmax[i]); + ener = WebRtcSpl_DotProductWithScale(regressor - lagmax[i], + regressor - lagmax[i], + ENH_BLOCKL_HALF, shifts); + enerSh = 15-WebRtcSpl_GetSizeInBits(ener); + corr16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh); + corr16[i] = (int16_t)((corr16[i] * corr16[i]) >> 16); + en16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, enerSh); + totsh[i] = enerSh - (corrSh << 1); + } + + /* Compare lagmax[0..3] for the (corr^2)/ener criteria */ + ind = 0; + for (i=1; i<3; i++) { + if (totsh[ind] > totsh[i]) { + sh = WEBRTC_SPL_MIN(31, totsh[ind]-totsh[i]); + if (corr16[ind] * en16[i] < (corr16[i] * en16[ind]) >> sh) { + ind = i; + } + } else { + sh = WEBRTC_SPL_MIN(31, totsh[i]-totsh[ind]); + if ((corr16[ind] * en16[i]) >> sh < corr16[i] * en16[ind]) { + ind = i; + } + } + } + + lag = lagmax[ind] + 10; + + /* Store the estimated lag in the non-downsampled domain */ + enh_period[ENH_NBLOCKS_TOT - new_blocks + iblock] = lag * 8; + + /* Store the estimated lag for backward PLC */ + if (iLBCdec_inst->prev_enh_pl==1) { + if (!iblock) { + tlag = lag * 2; + } + } else { + if (iblock==1) { + tlag = lag * 2; + } + } + + lag *= 2; + } + + if ((iLBCdec_inst->prev_enh_pl==1)||(iLBCdec_inst->prev_enh_pl==2)) { + + /* Calculate the best lag of the new frame + This is used to interpolate backwards and mix with the PLC'd data + */ + + /* references */ + target=in; + regressor=in+tlag-1; + + /* scaling */ + max16 = WebRtcSpl_MaxAbsValueW16(regressor, plc_blockl + 3 - 1); + if (max16>5000) + shifts=2; + else + shifts=0; + + /* compute cross correlation */ + WebRtcSpl_CrossCorrelation(corr32, target, regressor, plc_blockl, 3, shifts, + 1); + + /* find lag */ + lag=WebRtcSpl_MaxIndexW32(corr32, 3); + lag+=tlag-1; + + /* Copy the backward PLC to plc_pred */ + + if (iLBCdec_inst->prev_enh_pl==1) { + if (lag>plc_blockl) { + WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-plc_blockl], plc_blockl); + } else { + WEBRTC_SPL_MEMCPY_W16(&plc_pred[plc_blockl-lag], in, lag); + WEBRTC_SPL_MEMCPY_W16( + plc_pred, &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl+lag], + (plc_blockl-lag)); + } + } else { + size_t pos; + + pos = plc_blockl; + + while (lag<pos) { + WEBRTC_SPL_MEMCPY_W16(&plc_pred[pos-lag], in, lag); + pos = pos - lag; + } + WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-pos], pos); + + } + + if (iLBCdec_inst->prev_enh_pl==1) { + /* limit energy change + if energy in backward PLC is more than 4 times higher than the forward + PLC, then reduce the energy in the backward PLC vector: + sample 1...len-16 set energy of the to 4 times forward PLC + sample len-15..len interpolate between 4 times fw PLC and bw PLC energy + + Note: Compared to floating point code there is a slight change, + the window is 16 samples long instead of 10 samples to simplify the + calculations + */ + + max=WebRtcSpl_MaxAbsValueW16( + &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], plc_blockl); + max16=WebRtcSpl_MaxAbsValueW16(plc_pred, plc_blockl); + max = WEBRTC_SPL_MAX(max, max16); + scale=22-(int16_t)WebRtcSpl_NormW32(max); + scale=WEBRTC_SPL_MAX(scale,0); + + tmp2 = WebRtcSpl_DotProductWithScale( + &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], + &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], + plc_blockl, scale); + tmp1 = WebRtcSpl_DotProductWithScale(plc_pred, plc_pred, + plc_blockl, scale); + + /* Check the energy difference */ + if ((tmp1>0)&&((tmp1>>2)>tmp2)) { + /* EnChange is now guaranteed to be <0.5 + Calculate EnChange=tmp2/tmp1 in Q16 + */ + + scale1=(int16_t)WebRtcSpl_NormW32(tmp1); + tmp1=WEBRTC_SPL_SHIFT_W32(tmp1, (scale1-16)); /* using 15 bits */ + + tmp2=WEBRTC_SPL_SHIFT_W32(tmp2, (scale1)); + EnChange = (int16_t)WebRtcSpl_DivW32W16(tmp2, + (int16_t)tmp1); + + /* Calculate the Sqrt of the energy in Q15 ((14+16)/2) */ + SqrtEnChange = (int16_t)WebRtcSpl_SqrtFloor(EnChange << 14); + + + /* Multiply first part of vector with 2*SqrtEnChange */ + WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange, plc_blockl-16, + 14); + + /* Calculate increase parameter for window part (16 last samples) */ + /* (1-2*SqrtEnChange)/16 in Q15 */ + inc = 2048 - (SqrtEnChange >> 3); + + win=0; + tmpW16ptr=&plc_pred[plc_blockl-16]; + + for (i=16;i>0;i--) { + *tmpW16ptr = (int16_t)( + (*tmpW16ptr * (SqrtEnChange + (win >> 1))) >> 14); + /* multiply by (2.0*SqrtEnChange+win) */ + + win += inc; + tmpW16ptr++; + } + } + + /* Make the linear interpolation between the forward PLC'd data + and the backward PLC'd data (from the new frame) + */ + + if (plc_blockl==40) { + inc=400; /* 1/41 in Q14 */ + } else { /* plc_blockl==80 */ + inc=202; /* 1/81 in Q14 */ + } + win=0; + enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl]; + for (i=0; i<plc_blockl; i++) { + win+=inc; + *enh_bufPtr1 = (int16_t)((*enh_bufPtr1 * win) >> 14); + *enh_bufPtr1 += (int16_t)( + ((16384 - win) * plc_pred[plc_blockl - 1 - i]) >> 14); + enh_bufPtr1--; + } + } else { + int16_t *synt = &downsampled[LPC_FILTERORDER]; + + enh_bufPtr1=&enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl]; + WEBRTC_SPL_MEMCPY_W16(enh_bufPtr1, plc_pred, plc_blockl); + + /* Clear fileter memory */ + WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER); + WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4); + WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2); + + /* Initialize filter memory by filtering through 2 lags */ + WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], iLBCdec_inst->syntMem, + LPC_FILTERORDER); + WebRtcSpl_FilterARFastQ12( + enh_bufPtr1, + synt, + &iLBCdec_inst->old_syntdenum[ + (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)], + LPC_FILTERORDER+1, lag); + + WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER], + LPC_FILTERORDER); + WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs, + iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx, + lag); + WebRtcSpl_FilterARFastQ12( + enh_bufPtr1, synt, + &iLBCdec_inst->old_syntdenum[ + (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)], + LPC_FILTERORDER+1, lag); + + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER], + LPC_FILTERORDER); + WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs, + iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx, + lag); + } + } + + + /* Perform enhancement block by block */ + + for (iblock = 0; iblock<new_blocks; iblock++) { + WebRtcIlbcfix_Enhancer(out + iblock * ENH_BLOCKL, + enh_buf, + ENH_BUFL, + iblock * ENH_BLOCKL + startPos, + enh_period, + WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT); + } + + return (lag); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h new file mode 100644 index 0000000000..61efd22604 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_EnhancerInterface.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * interface for enhancer + *---------------------------------------------------------------*/ + +size_t WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */ + int16_t *out, /* (o) enhanced signal */ + int16_t *in, /* (i) unenhanced signal */ + IlbcDecoder *iLBCdec_inst /* (i) buffers etc */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c new file mode 100644 index 0000000000..04d17a67ef --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c @@ -0,0 +1,48 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_FilteredCbVecs.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * Construct an additional codebook vector by filtering the + * initial codebook buffer. This vector is then used to expand + * the codebook with an additional section. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_FilteredCbVecs( + int16_t *cbvectors, /* (o) Codebook vector for the higher section */ + int16_t *CBmem, /* (i) Codebook memory that is filtered to create a + second CB section */ + size_t lMem, /* (i) Length of codebook memory */ + size_t samples /* (i) Number of samples to filter */ + ) { + + /* Set up the memory, start with zero state */ + WebRtcSpl_MemSetW16(CBmem+lMem, 0, CB_HALFFILTERLEN); + WebRtcSpl_MemSetW16(CBmem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN); + WebRtcSpl_MemSetW16(cbvectors, 0, lMem-samples); + + /* Filter to obtain the filtered CB memory */ + + WebRtcSpl_FilterMAFastQ12( + CBmem+CB_HALFFILTERLEN+lMem-samples, cbvectors+lMem-samples, + (int16_t*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h new file mode 100644 index 0000000000..d23b25c1ac --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_FilteredCbVecs.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Construct an additional codebook vector by filtering the + * initial codebook buffer. This vector is then used to expand + * the codebook with an additional section. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_FilteredCbVecs( + int16_t *cbvectors, /* (o) Codebook vector for the higher section */ + int16_t *CBmem, /* (i) Codebook memory that is filtered to create a + second CB section */ + size_t lMem, /* (i) Length of codebook memory */ + size_t samples /* (i) Number of samples to filter */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c new file mode 100644 index 0000000000..48332808e4 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c @@ -0,0 +1,88 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_FrameClassify.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * Classification of subframes to localize start state + *---------------------------------------------------------------*/ + +size_t WebRtcIlbcfix_FrameClassify( + /* (o) Index to the max-energy sub frame */ + IlbcEncoder *iLBCenc_inst, + /* (i/o) the encoder state structure */ + int16_t *residualFIX /* (i) lpc residual signal */ + ){ + int16_t max, scale; + int32_t ssqEn[NSUB_MAX-1]; + int16_t *ssqPtr; + int32_t *seqEnPtr; + int32_t maxW32; + int16_t scale1; + size_t pos; + size_t n; + + /* + Calculate the energy of each of the 80 sample blocks + in the draft the 4 first and last samples are windowed with 1/5...4/5 + and 4/5...1/5 respectively. To simplify for the fixpoint we have changed + this to 0 0 1 1 and 1 1 0 0 + */ + + max = WebRtcSpl_MaxAbsValueW16(residualFIX, iLBCenc_inst->blockl); + scale = WebRtcSpl_GetSizeInBits((uint32_t)(max * max)); + + /* Scale to maximum 24 bits so that it won't overflow for 76 samples */ + scale = scale-24; + scale1 = WEBRTC_SPL_MAX(0, scale); + + /* Calculate energies */ + ssqPtr=residualFIX + 2; + seqEnPtr=ssqEn; + for (n=(iLBCenc_inst->nsub-1); n>0; n--) { + (*seqEnPtr) = WebRtcSpl_DotProductWithScale(ssqPtr, ssqPtr, 76, scale1); + ssqPtr += 40; + seqEnPtr++; + } + + /* Scale to maximum 20 bits in order to allow for the 11 bit window */ + maxW32 = WebRtcSpl_MaxValueW32(ssqEn, iLBCenc_inst->nsub - 1); + scale = WebRtcSpl_GetSizeInBits(maxW32) - 20; + scale1 = WEBRTC_SPL_MAX(0, scale); + + /* Window each 80 block with the ssqEn_winTbl window to give higher probability for + the blocks in the middle + */ + seqEnPtr=ssqEn; + if (iLBCenc_inst->mode==20) { + ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin+1; + } else { + ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin; + } + for (n=(iLBCenc_inst->nsub-1); n>0; n--) { + (*seqEnPtr)=WEBRTC_SPL_MUL(((*seqEnPtr)>>scale1), (*ssqPtr)); + seqEnPtr++; + ssqPtr++; + } + + /* Extract the best choise of start state */ + pos = WebRtcSpl_MaxIndexW32(ssqEn, iLBCenc_inst->nsub - 1) + 1; + + return(pos); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h new file mode 100644 index 0000000000..99f7144782 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h @@ -0,0 +1,29 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_FrameClassify.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_ + +size_t WebRtcIlbcfix_FrameClassify( + /* (o) Index to the max-energy sub frame */ + IlbcEncoder *iLBCenc_inst, + /* (i/o) the encoder state structure */ + int16_t *residualFIX /* (i) lpc residual signal */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c b/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c new file mode 100644 index 0000000000..07e09bb887 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c @@ -0,0 +1,45 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GainDequant.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * decoder for quantized gains in the gain-shape coding of + * residual + *---------------------------------------------------------------*/ + +int16_t WebRtcIlbcfix_GainDequant( + /* (o) quantized gain value (Q14) */ + int16_t index, /* (i) quantization index */ + int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */ + int16_t stage /* (i) The stage of the search */ + ){ + int16_t scale; + const int16_t *gain; + + /* obtain correct scale factor */ + + scale=WEBRTC_SPL_ABS_W16(maxIn); + scale = WEBRTC_SPL_MAX(1638, scale); /* if lower than 0.1, set it to 0.1 */ + + /* select the quantization table and return the decoded value */ + gain = WebRtcIlbcfix_kGain[stage]; + + return (int16_t)((scale * gain[index] + 8192) >> 14); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h b/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h new file mode 100644 index 0000000000..6bda066a56 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GainDequant.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * decoder for quantized gains in the gain-shape coding of + * residual + *---------------------------------------------------------------*/ + +int16_t WebRtcIlbcfix_GainDequant( + /* (o) quantized gain value (Q14) */ + int16_t index, /* (i) quantization index */ + int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */ + int16_t stage /* (i) The stage of the search */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c b/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c new file mode 100644 index 0000000000..90b4114d3b --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c @@ -0,0 +1,103 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GainQuant.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * quantizer for the gain in the gain-shape coding of residual + *---------------------------------------------------------------*/ + +int16_t WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */ + int16_t gain, /* (i) gain value Q14 */ + int16_t maxIn, /* (i) maximum of gain value Q14 */ + int16_t stage, /* (i) The stage of the search */ + int16_t *index /* (o) quantization index */ + ) { + + int16_t scale, cblen; + int32_t gainW32, measure1, measure2; + const int16_t *cbPtr, *cb; + int loc, noMoves, noChecks, i; + + /* ensure a lower bound (0.1) on the scaling factor */ + + scale = WEBRTC_SPL_MAX(1638, maxIn); + + /* select the quantization table and calculate + the length of the table and the number of + steps in the binary search that are needed */ + cb = WebRtcIlbcfix_kGain[stage]; + cblen = 32>>stage; + noChecks = 4-stage; + + /* Multiply the gain with 2^14 to make the comparison + easier and with higher precision */ + gainW32 = gain << 14; + + /* Do a binary search, starting in the middle of the CB + loc - defines the current position in the table + noMoves - defines the number of steps to move in the CB in order + to get next CB location + */ + + loc = cblen>>1; + noMoves = loc; + cbPtr = cb + loc; /* Centre of CB */ + + for (i=noChecks;i>0;i--) { + noMoves>>=1; + measure1 = scale * *cbPtr; + + /* Move up if gain is larger, otherwise move down in table */ + measure1 = measure1 - gainW32; + + if (0>measure1) { + cbPtr+=noMoves; + loc+=noMoves; + } else { + cbPtr-=noMoves; + loc-=noMoves; + } + } + + /* Check which value is the closest one: loc-1, loc or loc+1 */ + + measure1 = scale * *cbPtr; + if (gainW32>measure1) { + /* Check against value above loc */ + measure2 = scale * cbPtr[1]; + if ((measure2-gainW32)<(gainW32-measure1)) { + loc+=1; + } + } else { + /* Check against value below loc */ + measure2 = scale * cbPtr[-1]; + if ((gainW32-measure2)<=(measure1-gainW32)) { + loc-=1; + } + } + + /* Guard against getting outside the table. The calculation above can give a location + which is one above the maximum value (in very rare cases) */ + loc=WEBRTC_SPL_MIN(loc, (cblen-1)); + *index=loc; + + /* Calculate and return the quantized gain value (in Q14) */ + return (int16_t)((scale * cb[loc] + 8192) >> 14); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.h b/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.h new file mode 100644 index 0000000000..395436410c --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GainQuant.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * quantizer for the gain in the gain-shape coding of residual + *---------------------------------------------------------------*/ + +int16_t WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */ + int16_t gain, /* (i) gain value Q14 */ + int16_t maxIn, /* (i) maximum of gain value Q14 */ + int16_t stage, /* (i) The stage of the search */ + int16_t *index /* (o) quantization index */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c new file mode 100644 index 0000000000..d7c2e75553 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c @@ -0,0 +1,111 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GetCbVec.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "create_augmented_vec.h" + +/*----------------------------------------------------------------* + * Construct codebook vector for given index. + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_GetCbVec( + int16_t *cbvec, /* (o) Constructed codebook vector */ + int16_t *mem, /* (i) Codebook buffer */ + size_t index, /* (i) Codebook index */ + size_t lMem, /* (i) Length of codebook buffer */ + size_t cbveclen /* (i) Codebook vector length */ + ){ + size_t k, base_size; + size_t lag; + /* Stack based */ + int16_t tempbuff2[SUBL+5]; + + /* Determine size of codebook sections */ + + base_size=lMem-cbveclen+1; + + if (cbveclen==SUBL) { + base_size += cbveclen / 2; + } + + /* No filter -> First codebook section */ + + if (index<lMem-cbveclen+1) { + + /* first non-interpolated vectors */ + + k=index+cbveclen; + /* get vector */ + WEBRTC_SPL_MEMCPY_W16(cbvec, mem+lMem-k, cbveclen); + + } else if (index < base_size) { + + /* Calculate lag */ + + k = (2 * (index - (lMem - cbveclen + 1))) + cbveclen; + + lag = k / 2; + + WebRtcIlbcfix_CreateAugmentedVec(lag, mem+lMem, cbvec); + + } + + /* Higher codebbok section based on filtering */ + + else { + + size_t memIndTest; + + /* first non-interpolated vectors */ + + if (index-base_size<lMem-cbveclen+1) { + + /* Set up filter memory, stuff zeros outside memory buffer */ + + memIndTest = lMem-(index-base_size+cbveclen); + + WebRtcSpl_MemSetW16(mem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN); + WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN); + + /* do filtering to get the codebook vector */ + + WebRtcSpl_FilterMAFastQ12( + &mem[memIndTest+4], cbvec, (int16_t*)WebRtcIlbcfix_kCbFiltersRev, + CB_FILTERLEN, cbveclen); + } + + /* interpolated vectors */ + + else { + /* Stuff zeros outside memory buffer */ + memIndTest = lMem-cbveclen-CB_FILTERLEN; + WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN); + + /* do filtering */ + WebRtcSpl_FilterMAFastQ12( + &mem[memIndTest+7], tempbuff2, (int16_t*)WebRtcIlbcfix_kCbFiltersRev, + CB_FILTERLEN, cbveclen+5); + + /* Calculate lag index */ + lag = (cbveclen<<1)-20+index-base_size-lMem-1; + + WebRtcIlbcfix_CreateAugmentedVec(lag, tempbuff2+SUBL+5, cbvec); + } + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h new file mode 100644 index 0000000000..07f67a2aa5 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h @@ -0,0 +1,30 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GetCbVec.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_ + +void WebRtcIlbcfix_GetCbVec( + int16_t *cbvec, /* (o) Constructed codebook vector */ + int16_t *mem, /* (i) Codebook buffer */ + size_t index, /* (i) Codebook index */ + size_t lMem, /* (i) Length of codebook buffer */ + size_t cbveclen /* (i) Codebook vector length */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c new file mode 100644 index 0000000000..62a686495b --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c @@ -0,0 +1,82 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GetLspPoly.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Construct the polynomials F1(z) and F2(z) from the LSP + * (Computations are done in Q24) + * + * The expansion is performed using the following recursion: + * + * f[0] = 1; + * tmp = -2.0 * lsp[0]; + * f[1] = tmp; + * for (i=2; i<=5; i++) { + * b = -2.0 * lsp[2*i-2]; + * f[i] = tmp*f[i-1] + 2.0*f[i-2]; + * for (j=i; j>=2; j--) { + * f[j] = f[j] + tmp*f[j-1] + f[j-2]; + * } + * f[i] = f[i] + tmp; + * } + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_GetLspPoly( + int16_t *lsp, /* (i) LSP in Q15 */ + int32_t *f) /* (o) polonymial in Q24 */ +{ + int32_t tmpW32; + int i, j; + int16_t high, low; + int16_t *lspPtr; + int32_t *fPtr; + + lspPtr = lsp; + fPtr = f; + /* f[0] = 1.0 (Q24) */ + (*fPtr) = (int32_t)16777216; + fPtr++; + + (*fPtr) = WEBRTC_SPL_MUL((*lspPtr), -1024); + fPtr++; + lspPtr+=2; + + for(i=2; i<=5; i++) + { + (*fPtr) = fPtr[-2]; + + for(j=i; j>1; j--) + { + /* Compute f[j] = f[j] + tmp*f[j-1] + f[j-2]; */ + high = (int16_t)(fPtr[-1] >> 16); + low = (int16_t)((fPtr[-1] - ((int32_t)high << 16)) >> 1); + + tmpW32 = ((high * *lspPtr) << 2) + (((low * *lspPtr) >> 15) << 2); + + (*fPtr) += fPtr[-2]; + (*fPtr) -= tmpW32; + fPtr--; + } + *fPtr -= *lspPtr << 10; + + fPtr+=i; + lspPtr+=2; + } + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h new file mode 100644 index 0000000000..46ade4870d --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h @@ -0,0 +1,47 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GetLspPoly.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Construct the polynomials F1(z) and F2(z) from the LSP + * (Computations are done in Q24) + * + * The expansion is performed using the following recursion: + * + * f[0] = 1; + * tmp = -2.0 * lsp[0]; + * f[1] = tmp; + * for (i=2; i<=5; i++) { + * b = -2.0 * lsp[2*i-2]; + * f[i] = tmp*f[i-1] + 2.0*f[i-2]; + * for (j=i; j>=2; j--) { + * f[j] = f[j] + tmp*f[j-1] + f[j-2]; + * } + * f[i] = f[i] + tmp; + * } + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_GetLspPoly( + int16_t *lsp, /* (i) LSP in Q15 */ + int32_t *f); /* (o) polonymial in Q24 */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c new file mode 100644 index 0000000000..a98a96cdf1 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c @@ -0,0 +1,109 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GetSyncSeq.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "refiner.h" +#include "nearest_neighbor.h" + +/*----------------------------------------------------------------* + * get the pitch-synchronous sample sequence + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_GetSyncSeq( + int16_t *idata, /* (i) original data */ + size_t idatal, /* (i) dimension of data */ + size_t centerStartPos, /* (i) where current block starts */ + size_t *period, /* (i) rough-pitch-period array (Q-2) */ + const size_t *plocs, /* (i) where periods of period array are taken (Q-2) */ + size_t periodl, /* (i) dimension period array */ + size_t hl, /* (i) 2*hl+1 is the number of sequences */ + int16_t *surround /* (i/o) The contribution from this sequence + summed with earlier contributions */ + ){ + size_t i, centerEndPos, q; + /* Stack based */ + size_t lagBlock[2 * ENH_HL + 1]; + size_t blockStartPos[2 * ENH_HL + 1]; /* The position to search around (Q2) */ + size_t plocs2[ENH_PLOCSL]; + + centerEndPos = centerStartPos + ENH_BLOCKL - 1; + + /* present (find predicted lag from this position) */ + + WebRtcIlbcfix_NearestNeighbor(lagBlock + hl, + plocs, + 2 * (centerStartPos + centerEndPos), + periodl); + + blockStartPos[hl] = 4 * centerStartPos; + + /* past (find predicted position and perform a refined + search to find the best sequence) */ + + for (q = hl; q > 0; q--) { + size_t qq = q - 1; + size_t period_q = period[lagBlock[q]]; + /* Stop if this sequence would be outside the buffer; that means all + further-past sequences would also be outside the buffer. */ + if (blockStartPos[q] < period_q + (4 * ENH_OVERHANG)) + break; + blockStartPos[qq] = blockStartPos[q] - period_q; + + size_t value = blockStartPos[qq] + 4 * ENH_BLOCKL_HALF; + value = (value > period_q) ? (value - period_q) : 0; + WebRtcIlbcfix_NearestNeighbor(lagBlock + qq, plocs, value, periodl); + + /* Find the best possible sequence in the 4 times upsampled + domain around blockStartPos+q */ + WebRtcIlbcfix_Refiner(blockStartPos + qq, idata, idatal, centerStartPos, + blockStartPos[qq], surround, + WebRtcIlbcfix_kEnhWt[qq]); + } + + /* future (find predicted position and perform a refined + search to find the best sequence) */ + + for (i = 0; i < periodl; i++) { + plocs2[i] = plocs[i] - period[i]; + } + + for (q = hl + 1; q <= (2 * hl); q++) { + + WebRtcIlbcfix_NearestNeighbor( + lagBlock + q, + plocs2, + blockStartPos[q - 1] + 4 * ENH_BLOCKL_HALF, + periodl); + + blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]]; + + if (blockStartPos[q] + 4 * (ENH_BLOCKL + ENH_OVERHANG) < 4 * idatal) { + + /* Find the best possible sequence in the 4 times upsampled + domain around blockStartPos+q */ + WebRtcIlbcfix_Refiner(blockStartPos + q, idata, idatal, centerStartPos, + blockStartPos[q], surround, + WebRtcIlbcfix_kEnhWt[2 * hl - q]); + + } else { + /* Don't add anything since this sequence would + be outside the buffer */ + } + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h new file mode 100644 index 0000000000..0e3b207753 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h @@ -0,0 +1,40 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_GetSyncSeq.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * get the pitch-synchronous sample sequence + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_GetSyncSeq( + int16_t *idata, /* (i) original data */ + size_t idatal, /* (i) dimension of data */ + size_t centerStartPos, /* (i) where current block starts */ + size_t *period, /* (i) rough-pitch-period array (Q-2) */ + const size_t *plocs, /* (i) where periods of period array are taken (Q-2) */ + size_t periodl, /* (i) dimension period array */ + size_t hl, /* (i) 2*hl+1 is the number of sequences */ + int16_t *surround /* (i/o) The contribution from this sequence + summed with earlier contributions */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c b/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c new file mode 100644 index 0000000000..5d8a86050e --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c @@ -0,0 +1,88 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_HpInput.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * high-pass filter of input with *0.5 and saturation + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_HpInput( + int16_t *signal, /* (i/o) signal vector */ + int16_t *ba, /* (i) B- and A-coefficients (2:nd order) + {b[0] b[1] b[2] -a[1] -a[2]} a[0] + is assumed to be 1.0 */ + int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1] + yhi[n-2] ylow[n-2] */ + int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */ + size_t len) /* (i) Number of samples to filter */ +{ + size_t i; + int32_t tmpW32; + int32_t tmpW32b; + + for (i=0; i<len; i++) { + + /* + y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2] + + (-a[1])*y[i-1] + (-a[2])*y[i-2]; + */ + + tmpW32 = y[1] * ba[3]; /* (-a[1])*y[i-1] (low part) */ + tmpW32 += y[3] * ba[4]; /* (-a[2])*y[i-2] (low part) */ + tmpW32 = (tmpW32>>15); + tmpW32 += y[0] * ba[3]; /* (-a[1])*y[i-1] (high part) */ + tmpW32 += y[2] * ba[4]; /* (-a[2])*y[i-2] (high part) */ + tmpW32 = (tmpW32<<1); + + tmpW32 += signal[i] * ba[0]; /* b[0]*x[0] */ + tmpW32 += x[0] * ba[1]; /* b[1]*x[i-1] */ + tmpW32 += x[1] * ba[2]; /* b[2]*x[i-2] */ + + /* Update state (input part) */ + x[1] = x[0]; + x[0] = signal[i]; + + /* Rounding in Q(12+1), i.e. add 2^12 */ + tmpW32b = tmpW32 + 4096; + + /* Saturate (to 2^28) so that the HP filtered signal does not overflow */ + tmpW32b = WEBRTC_SPL_SAT((int32_t)268435455, tmpW32b, (int32_t)-268435456); + + /* Convert back to Q0 and multiply with 0.5 */ + signal[i] = (int16_t)(tmpW32b >> 13); + + /* Update state (filtered part) */ + y[2] = y[0]; + y[3] = y[1]; + + /* upshift tmpW32 by 3 with saturation */ + if (tmpW32>268435455) { + tmpW32 = WEBRTC_SPL_WORD32_MAX; + } else if (tmpW32<-268435456) { + tmpW32 = WEBRTC_SPL_WORD32_MIN; + } else { + tmpW32 <<= 3; + } + + y[0] = (int16_t)(tmpW32 >> 16); + y[1] = (int16_t)((tmpW32 - (y[0] << 16)) >> 1); + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h b/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h new file mode 100644 index 0000000000..acdfa91262 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_HpInput.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_ + +#include "defines.h" + +void WebRtcIlbcfix_HpInput( + int16_t *signal, /* (i/o) signal vector */ + int16_t *ba, /* (i) B- and A-coefficients (2:nd order) + {b[0] b[1] b[2] -a[1] -a[2]} a[0] + is assumed to be 1.0 */ + int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1] + yhi[n-2] ylow[n-2] */ + int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */ + size_t len); /* (i) Number of samples to filter */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c b/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c new file mode 100644 index 0000000000..bd101bf30c --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c @@ -0,0 +1,89 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_HpOutput.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * high-pass filter of output and *2 with saturation + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_HpOutput( + int16_t *signal, /* (i/o) signal vector */ + int16_t *ba, /* (i) B- and A-coefficients (2:nd order) + {b[0] b[1] b[2] -a[1] -a[2]} a[0] + is assumed to be 1.0 */ + int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1] + yhi[n-2] ylow[n-2] */ + int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */ + size_t len) /* (i) Number of samples to filter */ +{ + size_t i; + int32_t tmpW32; + int32_t tmpW32b; + + for (i=0; i<len; i++) { + + /* + y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2] + + (-a[1])*y[i-1] + (-a[2])*y[i-2]; + */ + + tmpW32 = y[1] * ba[3]; /* (-a[1])*y[i-1] (low part) */ + tmpW32 += y[3] * ba[4]; /* (-a[2])*y[i-2] (low part) */ + tmpW32 = (tmpW32>>15); + tmpW32 += y[0] * ba[3]; /* (-a[1])*y[i-1] (high part) */ + tmpW32 += y[2] * ba[4]; /* (-a[2])*y[i-2] (high part) */ + tmpW32 = (tmpW32<<1); + + tmpW32 += signal[i] * ba[0]; /* b[0]*x[0] */ + tmpW32 += x[0] * ba[1]; /* b[1]*x[i-1] */ + tmpW32 += x[1] * ba[2]; /* b[2]*x[i-2] */ + + /* Update state (input part) */ + x[1] = x[0]; + x[0] = signal[i]; + + /* Rounding in Q(12-1), i.e. add 2^10 */ + tmpW32b = tmpW32 + 1024; + + /* Saturate (to 2^26) so that the HP filtered signal does not overflow */ + tmpW32b = WEBRTC_SPL_SAT((int32_t)67108863, tmpW32b, (int32_t)-67108864); + + /* Convert back to Q0 and multiply with 2 */ + signal[i] = (int16_t)(tmpW32b >> 11); + + /* Update state (filtered part) */ + y[2] = y[0]; + y[3] = y[1]; + + /* upshift tmpW32 by 3 with saturation */ + if (tmpW32>268435455) { + tmpW32 = WEBRTC_SPL_WORD32_MAX; + } else if (tmpW32<-268435456) { + tmpW32 = WEBRTC_SPL_WORD32_MIN; + } else { + tmpW32 <<= 3; + } + + y[0] = (int16_t)(tmpW32 >> 16); + y[1] = (int16_t)((tmpW32 - (y[0] << 16)) >> 1); + + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h b/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h new file mode 100644 index 0000000000..1840b68186 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_HpOutput.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_ + +#include "defines.h" + +void WebRtcIlbcfix_HpOutput( + int16_t *signal, /* (i/o) signal vector */ + int16_t *ba, /* (i) B- and A-coefficients (2:nd order) + {b[0] b[1] b[2] -a[1] -a[2]} a[0] + is assumed to be 1.0 */ + int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1] + yhi[n-2] ylow[n-2] */ + int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */ + size_t len); /* (i) Number of samples to filter */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c new file mode 100644 index 0000000000..6cd9a723fd --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c @@ -0,0 +1,282 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + iLBCInterface.c + +******************************************************************/ + +#include "ilbc.h" +#include "defines.h" +#include "init_encode.h" +#include "encode.h" +#include "init_decode.h" +#include "decode.h" +#include <stdlib.h> + +int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst, + int16_t* ILBCENC_inst_Addr, + int16_t* size) { + *iLBC_encinst=(IlbcEncoderInstance*)ILBCENC_inst_Addr; + *size=sizeof(IlbcEncoder)/sizeof(int16_t); + if (*iLBC_encinst!=NULL) { + return(0); + } else { + return(-1); + } +} + +int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance** iLBC_decinst, + int16_t* ILBCDEC_inst_Addr, + int16_t* size) { + *iLBC_decinst=(IlbcDecoderInstance*)ILBCDEC_inst_Addr; + *size=sizeof(IlbcDecoder)/sizeof(int16_t); + if (*iLBC_decinst!=NULL) { + return(0); + } else { + return(-1); + } +} + +int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst) { + *iLBC_encinst=(IlbcEncoderInstance*)malloc(sizeof(IlbcEncoder)); + if (*iLBC_encinst!=NULL) { + WebRtcSpl_Init(); + return(0); + } else { + return(-1); + } +} + +int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance **iLBC_decinst) { + *iLBC_decinst=(IlbcDecoderInstance*)malloc(sizeof(IlbcDecoder)); + if (*iLBC_decinst!=NULL) { + WebRtcSpl_Init(); + return(0); + } else { + return(-1); + } +} + +int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance *iLBC_encinst) { + free(iLBC_encinst); + return(0); +} + +int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance *iLBC_decinst) { + free(iLBC_decinst); + return(0); +} + +int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst, + int16_t mode) { + if ((mode==20)||(mode==30)) { + WebRtcIlbcfix_InitEncode((IlbcEncoder*) iLBCenc_inst, mode); + return(0); + } else { + return(-1); + } +} + +int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst, + const int16_t* speechIn, + size_t len, + uint8_t* encoded) { + size_t pos = 0; + size_t encpos = 0; + + if ((len != ((IlbcEncoder*)iLBCenc_inst)->blockl) && +#ifdef SPLIT_10MS + (len != 80) && +#endif + (len != 2*((IlbcEncoder*)iLBCenc_inst)->blockl) && + (len != 3*((IlbcEncoder*)iLBCenc_inst)->blockl)) + { + /* A maximum of 3 frames/packet is allowed */ + return(-1); + } else { + + /* call encoder */ + while (pos<len) { + WebRtcIlbcfix_EncodeImpl((uint16_t*)&encoded[2 * encpos], &speechIn[pos], + (IlbcEncoder*)iLBCenc_inst); +#ifdef SPLIT_10MS + pos += 80; + if(((IlbcEncoder*)iLBCenc_inst)->section == 0) +#else + pos += ((IlbcEncoder*)iLBCenc_inst)->blockl; +#endif + encpos += ((IlbcEncoder*)iLBCenc_inst)->no_of_words; + } + return (int)(encpos*2); + } +} + +int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst, + int16_t mode) { + if ((mode==20)||(mode==30)) { + WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, mode, 1); + return(0); + } else { + return(-1); + } +} +void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst) { + WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 20, 1); +} +void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst) { + WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 30, 1); +} + + +int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst, + const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType) +{ + size_t i=0; + /* Allow for automatic switching between the frame sizes + (although you do get some discontinuity) */ + if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)|| + (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)|| + (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) { + /* ok, do nothing */ + } else { + /* Test if the mode has changed */ + if (((IlbcDecoder*)iLBCdec_inst)->mode==20) { + if ((len==NO_OF_BYTES_30MS)|| + (len==2*NO_OF_BYTES_30MS)|| + (len==3*NO_OF_BYTES_30MS)) { + WebRtcIlbcfix_InitDecode( + ((IlbcDecoder*)iLBCdec_inst), 30, + ((IlbcDecoder*)iLBCdec_inst)->use_enhancer); + } else { + /* Unsupported frame length */ + return(-1); + } + } else { + if ((len==NO_OF_BYTES_20MS)|| + (len==2*NO_OF_BYTES_20MS)|| + (len==3*NO_OF_BYTES_20MS)) { + WebRtcIlbcfix_InitDecode( + ((IlbcDecoder*)iLBCdec_inst), 20, + ((IlbcDecoder*)iLBCdec_inst)->use_enhancer); + } else { + /* Unsupported frame length */ + return(-1); + } + } + } + + while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) { + WebRtcIlbcfix_DecodeImpl( + &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl], + (const uint16_t*)&encoded + [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words], + (IlbcDecoder*)iLBCdec_inst, 1); + i++; + } + /* iLBC does not support VAD/CNG yet */ + *speechType=1; + return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl); +} + +int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst, + const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType) +{ + size_t i=0; + if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)|| + (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)|| + (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) { + /* ok, do nothing */ + } else { + return(-1); + } + + while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) { + WebRtcIlbcfix_DecodeImpl( + &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl], + (const uint16_t*)&encoded + [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words], + (IlbcDecoder*)iLBCdec_inst, 1); + i++; + } + /* iLBC does not support VAD/CNG yet */ + *speechType=1; + return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl); +} + +int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst, + const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType) +{ + size_t i=0; + if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)|| + (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)|| + (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) { + /* ok, do nothing */ + } else { + return(-1); + } + + while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) { + WebRtcIlbcfix_DecodeImpl( + &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl], + (const uint16_t*)&encoded + [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words], + (IlbcDecoder*)iLBCdec_inst, 1); + i++; + } + /* iLBC does not support VAD/CNG yet */ + *speechType=1; + return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl); +} + +size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst, + int16_t* decoded, + size_t noOfLostFrames) { + size_t i; + uint16_t dummy; + + for (i=0;i<noOfLostFrames;i++) { + /* call decoder */ + WebRtcIlbcfix_DecodeImpl( + &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl], &dummy, + (IlbcDecoder*)iLBCdec_inst, 0); + } + return (noOfLostFrames*((IlbcDecoder*)iLBCdec_inst)->blockl); +} + +size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst, + int16_t* decoded, + size_t noOfLostFrames) { + /* Two input parameters not used, but needed for function pointers in NetEQ */ + (void)(decoded = NULL); + (void)(noOfLostFrames = 0); + + WebRtcSpl_MemSetW16(((IlbcDecoder*)iLBCdec_inst)->enh_buf, 0, ENH_BUFL); + ((IlbcDecoder*)iLBCdec_inst)->prev_enh_pl = 2; + + return (0); +} + +void WebRtcIlbcfix_version(char *version) +{ + strcpy((char*)version, "1.1.1"); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi new file mode 100644 index 0000000000..ac9f2e7b39 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi @@ -0,0 +1,191 @@ +# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +{ + 'targets': [ + { + 'target_name': 'ilbc', + 'type': 'static_library', + 'dependencies': [ + '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', + 'audio_encoder_interface', + ], + 'include_dirs': [ + 'include', + '<(webrtc_root)', + ], + 'direct_dependent_settings': { + 'include_dirs': [ + 'include', + '<(webrtc_root)', + ], + }, + 'sources': [ + 'include/audio_decoder_ilbc.h', + 'include/audio_encoder_ilbc.h', + 'include/ilbc.h', + 'abs_quant.c', + 'abs_quant_loop.c', + 'audio_decoder_ilbc.cc', + 'audio_encoder_ilbc.cc', + 'augmented_cb_corr.c', + 'bw_expand.c', + 'cb_construct.c', + 'cb_mem_energy.c', + 'cb_mem_energy_augmentation.c', + 'cb_mem_energy_calc.c', + 'cb_search.c', + 'cb_search_core.c', + 'cb_update_best_index.c', + 'chebyshev.c', + 'comp_corr.c', + 'constants.c', + 'create_augmented_vec.c', + 'decode.c', + 'decode_residual.c', + 'decoder_interpolate_lsf.c', + 'do_plc.c', + 'encode.c', + 'energy_inverse.c', + 'enh_upsample.c', + 'enhancer.c', + 'enhancer_interface.c', + 'filtered_cb_vecs.c', + 'frame_classify.c', + 'gain_dequant.c', + 'gain_quant.c', + 'get_cd_vec.c', + 'get_lsp_poly.c', + 'get_sync_seq.c', + 'hp_input.c', + 'hp_output.c', + 'ilbc.c', + 'index_conv_dec.c', + 'index_conv_enc.c', + 'init_decode.c', + 'init_encode.c', + 'interpolate.c', + 'interpolate_samples.c', + 'lpc_encode.c', + 'lsf_check.c', + 'lsf_interpolate_to_poly_dec.c', + 'lsf_interpolate_to_poly_enc.c', + 'lsf_to_lsp.c', + 'lsf_to_poly.c', + 'lsp_to_lsf.c', + 'my_corr.c', + 'nearest_neighbor.c', + 'pack_bits.c', + 'poly_to_lsf.c', + 'poly_to_lsp.c', + 'refiner.c', + 'simple_interpolate_lsf.c', + 'simple_lpc_analysis.c', + 'simple_lsf_dequant.c', + 'simple_lsf_quant.c', + 'smooth.c', + 'smooth_out_data.c', + 'sort_sq.c', + 'split_vq.c', + 'state_construct.c', + 'state_search.c', + 'swap_bytes.c', + 'unpack_bits.c', + 'vq3.c', + 'vq4.c', + 'window32_w32.c', + 'xcorr_coef.c', + 'abs_quant.h', + 'abs_quant_loop.h', + 'augmented_cb_corr.h', + 'bw_expand.h', + 'cb_construct.h', + 'cb_mem_energy.h', + 'cb_mem_energy_augmentation.h', + 'cb_mem_energy_calc.h', + 'cb_search.h', + 'cb_search_core.h', + 'cb_update_best_index.h', + 'chebyshev.h', + 'comp_corr.h', + 'constants.h', + 'create_augmented_vec.h', + 'decode.h', + 'decode_residual.h', + 'decoder_interpolate_lsf.h', + 'do_plc.h', + 'encode.h', + 'energy_inverse.h', + 'enh_upsample.h', + 'enhancer.h', + 'enhancer_interface.h', + 'filtered_cb_vecs.h', + 'frame_classify.h', + 'gain_dequant.h', + 'gain_quant.h', + 'get_cd_vec.h', + 'get_lsp_poly.h', + 'get_sync_seq.h', + 'hp_input.h', + 'hp_output.h', + 'defines.h', + 'index_conv_dec.h', + 'index_conv_enc.h', + 'init_decode.h', + 'init_encode.h', + 'interpolate.h', + 'interpolate_samples.h', + 'lpc_encode.h', + 'lsf_check.h', + 'lsf_interpolate_to_poly_dec.h', + 'lsf_interpolate_to_poly_enc.h', + 'lsf_to_lsp.h', + 'lsf_to_poly.h', + 'lsp_to_lsf.h', + 'my_corr.h', + 'nearest_neighbor.h', + 'pack_bits.h', + 'poly_to_lsf.h', + 'poly_to_lsp.h', + 'refiner.h', + 'simple_interpolate_lsf.h', + 'simple_lpc_analysis.h', + 'simple_lsf_dequant.h', + 'simple_lsf_quant.h', + 'smooth.h', + 'smooth_out_data.h', + 'sort_sq.h', + 'split_vq.h', + 'state_construct.h', + 'state_search.h', + 'swap_bytes.h', + 'unpack_bits.h', + 'vq3.h', + 'vq4.h', + 'window32_w32.h', + 'xcorr_coef.h', + ], # sources + }, # ilbc + ], # targets + 'conditions': [ + ['include_tests==1', { + 'targets': [ + { + 'target_name': 'ilbc_test', + 'type': 'executable', + 'dependencies': [ + 'ilbc', + ], + 'sources': [ + 'test/iLBC_test.c', + ], + }, # ilbc_test + ], # targets + }], # include_tests + ], # conditions +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h new file mode 100644 index 0000000000..fd52da7986 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/include/audio_decoder_ilbc.h @@ -0,0 +1,42 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_DECODER_ILBC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_DECODER_ILBC_H_ + +#include "webrtc/modules/audio_coding/codecs/audio_decoder.h" + +typedef struct iLBC_decinst_t_ IlbcDecoderInstance; + +namespace webrtc { + +class AudioDecoderIlbc final : public AudioDecoder { + public: + AudioDecoderIlbc(); + ~AudioDecoderIlbc() override; + bool HasDecodePlc() const override; + size_t DecodePlc(size_t num_frames, int16_t* decoded) override; + void Reset() override; + size_t Channels() const override; + + protected: + int DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) override; + + private: + IlbcDecoderInstance* dec_state_; + RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc); +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_DECODER_ILBC_H_ diff --git a/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h new file mode 100644 index 0000000000..2bb3101fd4 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h @@ -0,0 +1,63 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_ENCODER_ILBC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_ENCODER_ILBC_H_ + +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" +#include "webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h" + +namespace webrtc { + +struct CodecInst; + +class AudioEncoderIlbc final : public AudioEncoder { + public: + struct Config { + bool IsOk() const; + + int payload_type = 102; + int frame_size_ms = 30; // Valid values are 20, 30, 40, and 60 ms. + // Note that frame size 40 ms produces encodings with two 20 ms frames in + // them, and frame size 60 ms consists of two 30 ms frames. + }; + + explicit AudioEncoderIlbc(const Config& config); + explicit AudioEncoderIlbc(const CodecInst& codec_inst); + ~AudioEncoderIlbc() override; + + size_t MaxEncodedBytes() const override; + int SampleRateHz() const override; + int NumChannels() const override; + size_t Num10MsFramesInNextPacket() const override; + size_t Max10MsFramesInAPacket() const override; + int GetTargetBitrate() const override; + EncodedInfo EncodeInternal(uint32_t rtp_timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded) override; + void Reset() override; + + private: + size_t RequiredOutputSizeBytes() const; + + static const size_t kMaxSamplesPerPacket = 480; + const Config config_; + const size_t num_10ms_frames_per_packet_; + size_t num_10ms_frames_buffered_; + uint32_t first_timestamp_in_buffer_; + int16_t input_buffer_[kMaxSamplesPerPacket]; + IlbcEncoderInstance* encoder_; + RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbc); +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_AUDIO_ENCODER_ILBC_H_ diff --git a/webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h new file mode 100644 index 0000000000..3be9142c8c --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h @@ -0,0 +1,258 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/* + * ilbc.h + * + * This header file contains all of the API's for iLBC. + * + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_ILBC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INCLUDE_ILBC_H_ + +#include <stddef.h> + +/* + * Define the fixpoint numeric formats + */ + +#include "webrtc/typedefs.h" + +/* + * Solution to support multiple instances + * Customer has to cast instance to proper type + */ + +typedef struct iLBC_encinst_t_ IlbcEncoderInstance; + +typedef struct iLBC_decinst_t_ IlbcDecoderInstance; + +/* + * Comfort noise constants + */ + +#define ILBC_SPEECH 1 +#define ILBC_CNG 2 + +#ifdef __cplusplus +extern "C" { +#endif + + /**************************************************************************** + * WebRtcIlbcfix_XxxAssign(...) + * + * These functions assigns the encoder/decoder instance to the specified + * memory location + * + * Input: + * - XXX_xxxinst : Pointer to created instance that should be + * assigned + * - ILBCXXX_inst_Addr : Pointer to the desired memory space + * - size : The size that this structure occupies (in Word16) + * + * Return value : 0 - Ok + * -1 - Error + */ + + int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance **iLBC_encinst, + int16_t *ILBCENC_inst_Addr, + int16_t *size); + int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance **iLBC_decinst, + int16_t *ILBCDEC_inst_Addr, + int16_t *size); + + + /**************************************************************************** + * WebRtcIlbcfix_XxxAssign(...) + * + * These functions create a instance to the specified structure + * + * Input: + * - XXX_inst : Pointer to created instance that should be created + * + * Return value : 0 - Ok + * -1 - Error + */ + + int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst); + int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance **iLBC_decinst); + + /**************************************************************************** + * WebRtcIlbcfix_XxxFree(...) + * + * These functions frees the dynamic memory of a specified instance + * + * Input: + * - XXX_inst : Pointer to created instance that should be freed + * + * Return value : 0 - Ok + * -1 - Error + */ + + int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance *iLBC_encinst); + int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance *iLBC_decinst); + + + /**************************************************************************** + * WebRtcIlbcfix_EncoderInit(...) + * + * This function initializes a iLBC instance + * + * Input: + * - iLBCenc_inst : iLBC instance, i.e. the user that should receive + * be initialized + * - frameLen : The frame length of the codec 20/30 (ms) + * + * Return value : 0 - Ok + * -1 - Error + */ + + int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance *iLBCenc_inst, + int16_t frameLen); + + /**************************************************************************** + * WebRtcIlbcfix_Encode(...) + * + * This function encodes one iLBC frame. Input speech length has be a + * multiple of the frame length. + * + * Input: + * - iLBCenc_inst : iLBC instance, i.e. the user that should encode + * a package + * - speechIn : Input speech vector + * - len : Samples in speechIn (160, 240, 320 or 480) + * + * Output: + * - encoded : The encoded data vector + * + * Return value : >0 - Length (in bytes) of coded data + * -1 - Error + */ + + int WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst, + const int16_t *speechIn, + size_t len, + uint8_t* encoded); + + /**************************************************************************** + * WebRtcIlbcfix_DecoderInit(...) + * + * This function initializes a iLBC instance with either 20 or 30 ms frames + * Alternatively the WebRtcIlbcfix_DecoderInit_XXms can be used. Then it's + * not needed to specify the frame length with a variable. + * + * Input: + * - IlbcDecoderInstance : iLBC decoder instance + * - frameLen : The frame length of the codec 20/30 (ms) + * + * Return value : 0 - Ok + * -1 - Error + */ + + int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance *iLBCdec_inst, + int16_t frameLen); + void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst); + void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst); + + /**************************************************************************** + * WebRtcIlbcfix_Decode(...) + * + * This function decodes a packet with iLBC frame(s). Output speech length + * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet). + * + * Input: + * - iLBCdec_inst : iLBC instance, i.e. the user that should decode + * a packet + * - encoded : Encoded iLBC frame(s) + * - len : Bytes in encoded vector + * + * Output: + * - decoded : The decoded vector + * - speechType : 1 normal, 2 CNG + * + * Return value : >0 - Samples in decoded vector + * -1 - Error + */ + + int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst, + const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType); + int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst, + const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType); + int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst, + const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType); + + /**************************************************************************** + * WebRtcIlbcfix_DecodePlc(...) + * + * This function conducts PLC for iLBC frame(s). Output speech length + * will be a multiple of 160 or 240 samples. + * + * Input: + * - iLBCdec_inst : iLBC instance, i.e. the user that should perform + * a PLC + * - noOfLostFrames : Number of PLC frames to produce + * + * Output: + * - decoded : The "decoded" vector + * + * Return value : Samples in decoded PLC vector + */ + + size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance *iLBCdec_inst, + int16_t *decoded, + size_t noOfLostFrames); + + /**************************************************************************** + * WebRtcIlbcfix_NetEqPlc(...) + * + * This function updates the decoder when a packet loss has occured, but it + * does not produce any PLC data. Function can be used if another PLC method + * is used (i.e NetEq). + * + * Input: + * - iLBCdec_inst : iLBC instance that should be updated + * - noOfLostFrames : Number of lost frames + * + * Output: + * - decoded : The "decoded" vector (nothing in this case) + * + * Return value : Samples in decoded PLC vector + */ + + size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance *iLBCdec_inst, + int16_t *decoded, + size_t noOfLostFrames); + + /**************************************************************************** + * WebRtcIlbcfix_version(...) + * + * This function returns the version number of iLBC + * + * Output: + * - version : Version number of iLBC (maximum 20 char) + */ + + void WebRtcIlbcfix_version(char *version); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c b/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c new file mode 100644 index 0000000000..033a0a3584 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_IndexConvDec.c + +******************************************************************/ + +#include "defines.h" + +void WebRtcIlbcfix_IndexConvDec( + int16_t *index /* (i/o) Codebook indexes */ + ){ + int k; + + for (k=4;k<6;k++) { + /* Readjust the second and third codebook index for the first 40 sample + so that they look the same as the first (in terms of lag) + */ + if ((index[k]>=44)&&(index[k]<108)) { + index[k]+=64; + } else if ((index[k]>=108)&&(index[k]<128)) { + index[k]+=128; + } else { + /* ERROR */ + } + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h new file mode 100644 index 0000000000..354c5b8a58 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h @@ -0,0 +1,28 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_IndexConvDec.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_ + +#include "defines.h" + +void WebRtcIlbcfix_IndexConvDec( + int16_t *index /* (i/o) Codebook indexes */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c b/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c new file mode 100644 index 0000000000..0e7701b4f6 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c @@ -0,0 +1,42 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + IiLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_IndexConvEnc.c + +******************************************************************/ + +#include "defines.h" +/*----------------------------------------------------------------* + * Convert the codebook indexes to make the search easier + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_IndexConvEnc( + int16_t *index /* (i/o) Codebook indexes */ + ){ + int k; + + for (k=4;k<6;k++) { + /* Readjust the second and third codebook index so that it is + packetized into 7 bits (before it was put in lag-wise the same + way as for the first codebook which uses 8 bits) + */ + if ((index[k]>=108)&&(index[k]<172)) { + index[k]-=64; + } else if (index[k]>=236) { + index[k]-=128; + } else { + /* ERROR */ + } + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h new file mode 100644 index 0000000000..d686331f26 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h @@ -0,0 +1,32 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_IndexConvEnc.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Convert the codebook indexes to make the search easier + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_IndexConvEnc( + int16_t *index /* (i/o) Codebook indexes */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c new file mode 100644 index 0000000000..1f92480d9f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c @@ -0,0 +1,96 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_InitDecode.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * Initiation of decoder instance. + *---------------------------------------------------------------*/ + +int WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */ + IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */ + int16_t mode, /* (i) frame size mode */ + int use_enhancer) { /* (i) 1: use enhancer, 0: no enhancer */ + int i; + + iLBCdec_inst->mode = mode; + + /* Set all the variables that are dependent on the frame size mode */ + if (mode==30) { + iLBCdec_inst->blockl = BLOCKL_30MS; + iLBCdec_inst->nsub = NSUB_30MS; + iLBCdec_inst->nasub = NASUB_30MS; + iLBCdec_inst->lpc_n = LPC_N_30MS; + iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS; + iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS; + iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS; + } + else if (mode==20) { + iLBCdec_inst->blockl = BLOCKL_20MS; + iLBCdec_inst->nsub = NSUB_20MS; + iLBCdec_inst->nasub = NASUB_20MS; + iLBCdec_inst->lpc_n = LPC_N_20MS; + iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS; + iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS; + iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS; + } + else { + return(-1); + } + + /* Reset all the previous LSF to mean LSF */ + WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER); + + /* Clear the synthesis filter memory */ + WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER); + + /* Set the old synthesis filter to {1.0 0.0 ... 0.0} */ + WebRtcSpl_MemSetW16(iLBCdec_inst->old_syntdenum, 0, ((LPC_FILTERORDER + 1)*NSUB_MAX)); + for (i=0; i<NSUB_MAX; i++) { + iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)] = 4096; + } + + /* Clear the variables that are used for the PLC */ + iLBCdec_inst->last_lag = 20; + iLBCdec_inst->consPLICount = 0; + iLBCdec_inst->prevPLI = 0; + iLBCdec_inst->perSquare = 0; + iLBCdec_inst->prevLag = 120; + iLBCdec_inst->prevLpc[0] = 4096; + WebRtcSpl_MemSetW16(iLBCdec_inst->prevLpc+1, 0, LPC_FILTERORDER); + WebRtcSpl_MemSetW16(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX); + + /* Initialize the seed for the random number generator */ + iLBCdec_inst->seed = 777; + + /* Set the filter state of the HP filter to 0 */ + WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2); + WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4); + + /* Set the variables that are used in the ehnahcer */ + iLBCdec_inst->use_enhancer = use_enhancer; + WebRtcSpl_MemSetW16(iLBCdec_inst->enh_buf, 0, (ENH_BUFL+ENH_BUFL_FILTEROVERHEAD)); + for (i=0;i<ENH_NBLOCKS_TOT;i++) { + iLBCdec_inst->enh_period[i]=160; /* Q(-4) */ + } + + iLBCdec_inst->prev_enh_pl = 0; + + return (int)(iLBCdec_inst->blockl); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h new file mode 100644 index 0000000000..cdd2192079 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_InitDecode.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Initiation of decoder instance. + *---------------------------------------------------------------*/ + +int WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */ + IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */ + int16_t mode, /* (i) frame size mode */ + int use_enhancer /* (i) 1 to use enhancer + 0 to run without enhancer */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c new file mode 100644 index 0000000000..f559d8441f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c @@ -0,0 +1,71 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_InitEncode.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * Initiation of encoder instance. + *---------------------------------------------------------------*/ + +int WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */ + IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */ + int16_t mode) { /* (i) frame size mode */ + iLBCenc_inst->mode = mode; + + /* Set all the variables that are dependent on the frame size mode */ + if (mode==30) { + iLBCenc_inst->blockl = BLOCKL_30MS; + iLBCenc_inst->nsub = NSUB_30MS; + iLBCenc_inst->nasub = NASUB_30MS; + iLBCenc_inst->lpc_n = LPC_N_30MS; + iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS; + iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS; + iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS; + } + else if (mode==20) { + iLBCenc_inst->blockl = BLOCKL_20MS; + iLBCenc_inst->nsub = NSUB_20MS; + iLBCenc_inst->nasub = NASUB_20MS; + iLBCenc_inst->lpc_n = LPC_N_20MS; + iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS; + iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS; + iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS; + } + else { + return(-1); + } + + /* Clear the buffers and set the previous LSF and LSP to the mean value */ + WebRtcSpl_MemSetW16(iLBCenc_inst->anaMem, 0, LPC_FILTERORDER); + WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER); + WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER); + WebRtcSpl_MemSetW16(iLBCenc_inst->lpc_buffer, 0, LPC_LOOKBACK + BLOCKL_MAX); + + /* Set the filter state of the HP filter to 0 */ + WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemx, 0, 2); + WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemy, 0, 4); + +#ifdef SPLIT_10MS + /*Zeroing the past samples for 10msec Split*/ + WebRtcSpl_MemSetW16(iLBCenc_inst->past_samples,0,160); + iLBCenc_inst->section = 0; +#endif + + return (int)(iLBCenc_inst->no_of_bytes); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h new file mode 100644 index 0000000000..7154661fbd --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_InitEncode.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Initiation of encoder instance. + *---------------------------------------------------------------*/ + +int WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */ + IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */ + int16_t mode /* (i) frame size mode */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c b/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c new file mode 100644 index 0000000000..d0869a5dc2 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Interpolate.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * interpolation between vectors + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Interpolate( + int16_t *out, /* (o) output vector */ + int16_t *in1, /* (i) first input vector */ + int16_t *in2, /* (i) second input vector */ + int16_t coef, /* (i) weight coefficient in Q14 */ + int16_t length) /* (i) number of sample is vectors */ +{ + int i; + int16_t invcoef; + + /* + Performs the operation out[i] = in[i]*coef + (1-coef)*in2[i] (with rounding) + */ + + invcoef = 16384 - coef; /* 16384 = 1.0 (Q14)*/ + for (i = 0; i < length; i++) { + out[i] = (int16_t)((coef * in1[i] + invcoef * in2[i] + 8192) >> 14); + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interpolate.h b/webrtc/modules/audio_coding/codecs/ilbc/interpolate.h new file mode 100644 index 0000000000..04832320a3 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/interpolate.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Interpolate.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * interpolation between vectors + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Interpolate( + int16_t *out, /* (o) output vector */ + int16_t *in1, /* (i) first input vector */ + int16_t *in2, /* (i) second input vector */ + int16_t coef, /* (i) weight coefficient in Q14 */ + int16_t length); /* (i) number of sample is vectors */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c new file mode 100644 index 0000000000..376dbbb668 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_InterpolateSamples.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +void WebRtcIlbcfix_InterpolateSamples( + int16_t *interpSamples, /* (o) The interpolated samples */ + int16_t *CBmem, /* (i) The CB memory */ + size_t lMem /* (i) Length of the CB memory */ + ) { + int16_t *ppi, *ppo, i, j, temp1, temp2; + int16_t *tmpPtr; + + /* Calculate the 20 vectors of interpolated samples (4 samples each) + that are used in the codebooks for lag 20 to 39 */ + tmpPtr = interpSamples; + for (j=0; j<20; j++) { + temp1 = 0; + temp2 = 3; + ppo = CBmem+lMem-4; + ppi = CBmem+lMem-j-24; + for (i=0; i<4; i++) { + + *tmpPtr++ = (int16_t)((WebRtcIlbcfix_kAlpha[temp2] * *ppo) >> 15) + + (int16_t)((WebRtcIlbcfix_kAlpha[temp1] * *ppi) >> 15); + + ppo++; + ppi++; + temp1++; + temp2--; + } + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h new file mode 100644 index 0000000000..7549d2c216 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_InterpolateSamples.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Construct the interpolated samples for the Augmented CB + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_InterpolateSamples( + int16_t *interpSamples, /* (o) The interpolated samples */ + int16_t *CBmem, /* (i) The CB memory */ + size_t lMem /* (i) Length of the CB memory */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c b/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c new file mode 100644 index 0000000000..258b6b1552 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c @@ -0,0 +1,60 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_LpcEncode.c + +******************************************************************/ + +#include "defines.h" +#include "simple_lpc_analysis.h" +#include "simple_interpolate_lsf.h" +#include "simple_lsf_quant.h" +#include "lsf_check.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * lpc encoder + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_LpcEncode( + int16_t *syntdenum, /* (i/o) synthesis filter coefficients + before/after encoding */ + int16_t *weightdenum, /* (i/o) weighting denumerator coefficients + before/after encoding */ + int16_t *lsf_index, /* (o) lsf quantization index */ + int16_t *data, /* (i) Speech to do LPC analysis on */ + IlbcEncoder *iLBCenc_inst + /* (i/o) the encoder state structure */ + ) { + /* Stack based */ + int16_t lsf[LPC_FILTERORDER * LPC_N_MAX]; + int16_t lsfdeq[LPC_FILTERORDER * LPC_N_MAX]; + + /* Calculate LSF's from the input speech */ + WebRtcIlbcfix_SimpleLpcAnalysis(lsf, data, iLBCenc_inst); + + /* Quantize the LSF's */ + WebRtcIlbcfix_SimpleLsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n); + + /* Stableize the LSF's if needed */ + WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n); + + /* Calculate the synthesis and weighting filter coefficients from + the optimal LSF and the dequantized LSF */ + WebRtcIlbcfix_SimpleInterpolateLsf(syntdenum, weightdenum, + lsf, lsfdeq, iLBCenc_inst->lsfold, + iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h b/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h new file mode 100644 index 0000000000..9f6f50416f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h @@ -0,0 +1,39 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_LpcEncode.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * lpc encoder + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_LpcEncode( + int16_t *syntdenum, /* (i/o) synthesis filter coefficients + before/after encoding */ + int16_t *weightdenum, /* (i/o) weighting denumerator coefficients + before/after encoding */ + int16_t *lsf_index, /* (o) lsf quantization index */ + int16_t *data, /* (i) Speech to do LPC analysis on */ + IlbcEncoder *iLBCenc_inst + /* (i/o) the encoder state structure */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c b/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c new file mode 100644 index 0000000000..0e3bd8485a --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c @@ -0,0 +1,71 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_LsfCheck.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * check for stability of lsf coefficients + *---------------------------------------------------------------*/ + +int WebRtcIlbcfix_LsfCheck( + int16_t *lsf, /* LSF parameters */ + int dim, /* dimension of LSF */ + int NoAn) /* No of analysis per frame */ +{ + int k,n,m, Nit=2, change=0,pos; + const int16_t eps=319; /* 0.039 in Q13 (50 Hz)*/ + const int16_t eps2=160; /* eps/2.0 in Q13;*/ + const int16_t maxlsf=25723; /* 3.14; (4000 Hz)*/ + const int16_t minlsf=82; /* 0.01; (0 Hz)*/ + + /* LSF separation check*/ + for (n=0;n<Nit;n++) { /* Run through a 2 times */ + for (m=0;m<NoAn;m++) { /* Number of analyses per frame */ + for (k=0;k<(dim-1);k++) { + pos=m*dim+k; + + /* Seperate coefficients with a safety margin of 50 Hz */ + if ((lsf[pos+1]-lsf[pos])<eps) { + + if (lsf[pos+1]<lsf[pos]) { + lsf[pos+1]= lsf[pos]+eps2; + lsf[pos]= lsf[pos+1]-eps2; + } else { + lsf[pos]-=eps2; + lsf[pos+1]+=eps2; + } + change=1; + } + + /* Limit minimum and maximum LSF */ + if (lsf[pos]<minlsf) { + lsf[pos]=minlsf; + change=1; + } + + if (lsf[pos]>maxlsf) { + lsf[pos]=maxlsf; + change=1; + } + } + } + } + + return change; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.h b/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.h new file mode 100644 index 0000000000..2f4ac8cc89 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_LsfCheck.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * check for stability of lsf coefficients + *---------------------------------------------------------------*/ + +int WebRtcIlbcfix_LsfCheck( + int16_t *lsf, /* LSF parameters */ + int dim, /* dimension of LSF */ + int NoAn); /* No of analysis per frame */ + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c b/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c new file mode 100644 index 0000000000..66bbde8e7f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c @@ -0,0 +1,42 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_LspInterpolate2PolyDec.c + +******************************************************************/ + +#include "interpolate.h" +#include "lsf_to_poly.h" +#include "defines.h" + +/*----------------------------------------------------------------* + * interpolation of lsf coefficients for the decoder + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_LspInterpolate2PolyDec( + int16_t *a, /* (o) lpc coefficients Q12 */ + int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */ + int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */ + int16_t coef, /* (i) weighting coefficient to use between + lsf1 and lsf2 Q14 */ + int16_t length /* (i) length of coefficient vectors */ + ){ + int16_t lsftmp[LPC_FILTERORDER]; + + /* interpolate LSF */ + WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length); + + /* Compute the filter coefficients from the LSF */ + WebRtcIlbcfix_Lsf2Poly(a, lsftmp); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h new file mode 100644 index 0000000000..3540c1c3b5 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h @@ -0,0 +1,37 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_LspInterpolate2PolyDec.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * interpolation of lsf coefficients for the decoder + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_LspInterpolate2PolyDec( + int16_t *a, /* (o) lpc coefficients Q12 */ + int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */ + int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */ + int16_t coef, /* (i) weighting coefficient to use between + lsf1 and lsf2 Q14 */ + int16_t length /* (i) length of coefficient vectors */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c b/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c new file mode 100644 index 0000000000..cf67ecc3be --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_LsfInterpolate2PloyEnc.c + +******************************************************************/ + +#include "defines.h" +#include "interpolate.h" +#include "lsf_to_poly.h" + +/*----------------------------------------------------------------* + * lsf interpolator and conversion from lsf to a coefficients + * (subrutine to SimpleInterpolateLSF) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_LsfInterpolate2PloyEnc( + int16_t *a, /* (o) lpc coefficients Q12 */ + int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */ + int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */ + int16_t coef, /* (i) weighting coefficient to use between + lsf1 and lsf2 Q14 */ + int16_t length /* (i) length of coefficient vectors */ + ) { + /* Stack based */ + int16_t lsftmp[LPC_FILTERORDER]; + + /* interpolate LSF */ + WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length); + + /* Compute the filter coefficients from the LSF */ + WebRtcIlbcfix_Lsf2Poly(a, lsftmp); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h new file mode 100644 index 0000000000..799c1003a4 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_LsfInterpolate2PloyEnc.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * lsf interpolator and conversion from lsf to a coefficients + * (subrutine to SimpleInterpolateLSF) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_LsfInterpolate2PloyEnc( + int16_t *a, /* (o) lpc coefficients Q12 */ + int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */ + int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */ + int16_t coef, /* (i) weighting coefficient to use between + lsf1 and lsf2 Q14 */ + int16_t length /* (i) length of coefficient vectors */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c b/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c new file mode 100644 index 0000000000..cfab013ab7 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c @@ -0,0 +1,61 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Lsf2Lsp.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * conversion from lsf to lsp coefficients + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Lsf2Lsp( + int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */ + int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */ + int16_t m /* (i) number of coefficients */ + ) { + int16_t i, k; + int16_t diff; /* difference, which is used for the + linear approximation (Q8) */ + int16_t freq; /* normalized frequency in Q15 (0..1) */ + int32_t tmpW32; + + for(i=0; i<m; i++) + { + freq = (int16_t)((lsf[i] * 20861) >> 15); + /* 20861: 1.0/(2.0*PI) in Q17 */ + /* + Upper 8 bits give the index k and + Lower 8 bits give the difference, which needs + to be approximated linearly + */ + k = freq >> 8; + diff = (freq&0x00ff); + + /* Guard against getting outside table */ + + if (k>63) { + k = 63; + } + + /* Calculate linear approximation */ + tmpW32 = WebRtcIlbcfix_kCosDerivative[k] * diff; + lsp[i] = WebRtcIlbcfix_kCos[k] + (int16_t)(tmpW32 >> 12); + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h new file mode 100644 index 0000000000..b2104d7d21 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Lsf2Lsp.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * conversion from lsf to lsp coefficients + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Lsf2Lsp( + int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */ + int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */ + int16_t m /* (i) number of coefficients */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c b/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c new file mode 100644 index 0000000000..c3a34cab00 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c @@ -0,0 +1,86 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Lsf2Poly.c + +******************************************************************/ + +#include "defines.h" +#include "lsf_to_lsp.h" +#include "get_lsp_poly.h" +#include "constants.h" + +void WebRtcIlbcfix_Lsf2Poly( + int16_t *a, /* (o) predictor coefficients (order = 10) in Q12 */ + int16_t *lsf /* (i) line spectral frequencies in Q13 */ + ) { + int32_t f[2][6]; /* f[0][] and f[1][] corresponds to + F1(z) and F2(z) respectivly */ + int32_t *f1ptr, *f2ptr; + int16_t *a1ptr, *a2ptr; + int32_t tmpW32; + int16_t lsp[10]; + int i; + + /* Convert lsf to lsp */ + WebRtcIlbcfix_Lsf2Lsp(lsf, lsp, LPC_FILTERORDER); + + /* Get F1(z) and F2(z) from the lsp */ + f1ptr=f[0]; + f2ptr=f[1]; + WebRtcIlbcfix_GetLspPoly(&lsp[0],f1ptr); + WebRtcIlbcfix_GetLspPoly(&lsp[1],f2ptr); + + /* for i = 5 down to 1 + Compute f1[i] += f1[i-1]; + and f2[i] += f2[i-1]; + */ + f1ptr=&f[0][5]; + f2ptr=&f[1][5]; + for (i=5; i>0; i--) + { + (*f1ptr) += (*(f1ptr-1)); + (*f2ptr) -= (*(f2ptr-1)); + f1ptr--; + f2ptr--; + } + + /* Get the A(z) coefficients + a[0] = 1.0 + for i = 1 to 5 + a[i] = (f1[i] + f2[i] + round)>>13; + for i = 1 to 5 + a[11-i] = (f1[i] - f2[i] + round)>>13; + */ + a[0]=4096; + a1ptr=&a[1]; + a2ptr=&a[10]; + f1ptr=&f[0][1]; + f2ptr=&f[1][1]; + for (i=5; i>0; i--) + { + tmpW32 = (*f1ptr) + (*f2ptr); + *a1ptr = (int16_t)((tmpW32 + 4096) >> 13); + + tmpW32 = (*f1ptr) - (*f2ptr); + *a2ptr = (int16_t)((tmpW32 + 4096) >> 13); + + a1ptr++; + a2ptr--; + f1ptr++; + f2ptr++; + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h new file mode 100644 index 0000000000..d85f510b27 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Lsf2Poly.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Convert from LSF coefficients to A coefficients + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Lsf2Poly( + int16_t *a, /* (o) predictor coefficients (order = 10) in Q12 */ + int16_t *lsf /* (i) line spectral frequencies in Q13 */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c b/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c new file mode 100644 index 0000000000..be95de7a2f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c @@ -0,0 +1,84 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Lsp2Lsf.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * conversion from LSP coefficients to LSF coefficients + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Lsp2Lsf( + int16_t *lsp, /* (i) lsp vector -1...+1 in Q15 */ + int16_t *lsf, /* (o) Lsf vector 0...Pi in Q13 + (ordered, so that lsf[i]<lsf[i+1]) */ + int16_t m /* (i) Number of coefficients */ + ) +{ + int16_t i, k; + int16_t diff; /* diff between table value and desired value (Q15) */ + int16_t freq; /* lsf/(2*pi) (Q16) */ + int16_t *lspPtr, *lsfPtr, *cosTblPtr; + int16_t tmp; + + /* set the index to maximum index value in WebRtcIlbcfix_kCos */ + k = 63; + + /* + Start with the highest LSP and then work the way down + For each LSP the lsf is calculated by first order approximation + of the acos(x) function + */ + lspPtr = &lsp[9]; + lsfPtr = &lsf[9]; + cosTblPtr=(int16_t*)&WebRtcIlbcfix_kCos[k]; + for(i=m-1; i>=0; i--) + { + /* + locate value in the table, which is just above lsp[i], + basically an approximation to acos(x) + */ + while( (((int32_t)(*cosTblPtr)-(*lspPtr)) < 0)&&(k>0) ) + { + k-=1; + cosTblPtr--; + } + + /* Calculate diff, which is used in the linear approximation of acos(x) */ + diff = (*lspPtr)-(*cosTblPtr); + + /* + The linear approximation of acos(lsp[i]) : + acos(lsp[i])= k*512 + (WebRtcIlbcfix_kAcosDerivative[ind]*offset >> 11) + */ + + /* tmp (linear offset) in Q16 */ + tmp = (int16_t)((WebRtcIlbcfix_kAcosDerivative[k] * diff) >> 11); + + /* freq in Q16 */ + freq = (k << 9) + tmp; + + /* lsf = freq*2*pi */ + (*lsfPtr) = (int16_t)(((int32_t)freq*25736)>>15); + + lsfPtr--; + lspPtr--; + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h new file mode 100644 index 0000000000..a2bcaff349 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Lsp2Lsf.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * conversion from LSP coefficients to LSF coefficients + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Lsp2Lsf( + int16_t *lsp, /* (i) lsp vector -1...+1 in Q15 */ + int16_t *lsf, /* (o) Lsf vector 0...Pi in Q13 + (ordered, so that lsf[i]<lsf[i+1]) */ + int16_t m /* (i) Number of coefficients */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c new file mode 100644 index 0000000000..bd6ff561c2 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c @@ -0,0 +1,53 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_MyCorr.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * compute cross correlation between sequences + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_MyCorr( + int32_t* corr, /* (o) correlation of seq1 and seq2 */ + const int16_t* seq1, /* (i) first sequence */ + size_t dim1, /* (i) dimension first seq1 */ + const int16_t* seq2, /* (i) second sequence */ + size_t dim2 /* (i) dimension seq2 */ + ){ + int16_t max; + size_t loops; + int scale; + + /* Calculate correlation between the two sequences. Scale the + result of the multiplcication to maximum 26 bits in order + to avoid overflow */ + max=WebRtcSpl_MaxAbsValueW16(seq1, dim1); + scale=WebRtcSpl_GetSizeInBits(max); + + scale = 2 * scale - 26; + if (scale<0) { + scale=0; + } + + loops=dim1-dim2+1; + + /* Calculate the cross correlations */ + WebRtcSpl_CrossCorrelation(corr, seq2, seq1, dim2, loops, scale, 1); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h new file mode 100644 index 0000000000..214946410e --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_MyCorr.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * compute cross correlation between sequences + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_MyCorr( + int32_t* corr, /* (o) correlation of seq1 and seq2 */ + const int16_t* seq1, /* (i) first sequence */ + size_t dim1, /* (i) dimension first seq1 */ + const int16_t* seq2, /* (i) second sequence */ + size_t dim2 /* (i) dimension seq2 */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c new file mode 100644 index 0000000000..2b58abc4f9 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_NearestNeighbor.c + +******************************************************************/ + +#include "defines.h" + +void WebRtcIlbcfix_NearestNeighbor(size_t* index, + const size_t* array, + size_t value, + size_t arlength) { + size_t i; + size_t min_diff = (size_t)-1; + for (i = 0; i < arlength; i++) { + const size_t diff = + (array[i] < value) ? (value - array[i]) : (array[i] - value); + if (diff < min_diff) { + *index = i; + min_diff = diff; + } + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h new file mode 100644 index 0000000000..7d7fb6fccc --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_NearestNeighbor.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Find index in array such that the array element with said + * index is the element of said array closest to "value" + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_NearestNeighbor( + size_t* index, /* (o) index of array element closest to value */ + const size_t* array, /* (i) data array (Q2) */ + size_t value, /* (i) value (Q2) */ + size_t arlength /* (i) dimension of data array (==ENH_NBLOCKS_TOT) */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c b/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c new file mode 100644 index 0000000000..9475ecba01 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c @@ -0,0 +1,251 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_PackBits.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * unpacking of bits from bitstream, i.e., vector of bytes + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_PackBits( + uint16_t *bitstream, /* (o) The packetized bitstream */ + iLBC_bits *enc_bits, /* (i) Encoded bits */ + int16_t mode /* (i) Codec mode (20 or 30) */ + ){ + uint16_t *bitstreamPtr; + int i, k; + int16_t *tmpPtr; + + bitstreamPtr=bitstream; + + /* Class 1 bits of ULP */ + /* First int16_t */ + (*bitstreamPtr) = ((uint16_t)enc_bits->lsf[0])<<10; /* Bit 0..5 */ + (*bitstreamPtr) |= (enc_bits->lsf[1])<<3; /* Bit 6..12 */ + (*bitstreamPtr) |= (enc_bits->lsf[2]&0x70)>>4; /* Bit 13..15 */ + bitstreamPtr++; + /* Second int16_t */ + (*bitstreamPtr) = ((uint16_t)enc_bits->lsf[2]&0xF)<<12; /* Bit 0..3 */ + + if (mode==20) { + (*bitstreamPtr) |= (enc_bits->startIdx)<<10; /* Bit 4..5 */ + (*bitstreamPtr) |= (enc_bits->state_first)<<9; /* Bit 6 */ + (*bitstreamPtr) |= (enc_bits->idxForMax)<<3; /* Bit 7..12 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[0])&0x70)>>4; /* Bit 13..15 */ + bitstreamPtr++; + /* Third int16_t */ + (*bitstreamPtr) = ((enc_bits->cb_index[0])&0xE)<<12; /* Bit 0..2 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x18)<<8; /* Bit 3..4 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x8)<<7; /* Bit 5 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0xFE)<<2; /* Bit 6..12 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x10)>>2; /* Bit 13 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x8)>>2; /* Bit 14 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x10)>>4; /* Bit 15 */ + } else { /* mode==30 */ + (*bitstreamPtr) |= (enc_bits->lsf[3])<<6; /* Bit 4..9 */ + (*bitstreamPtr) |= (enc_bits->lsf[4]&0x7E)>>1; /* Bit 10..15 */ + bitstreamPtr++; + /* Third int16_t */ + (*bitstreamPtr) = ((uint16_t)enc_bits->lsf[4]&0x1)<<15; /* Bit 0 */ + (*bitstreamPtr) |= (enc_bits->lsf[5])<<8; /* Bit 1..7 */ + (*bitstreamPtr) |= (enc_bits->startIdx)<<5; /* Bit 8..10 */ + (*bitstreamPtr) |= (enc_bits->state_first)<<4; /* Bit 11 */ + (*bitstreamPtr) |= ((enc_bits->idxForMax)&0x3C)>>2; /* Bit 12..15 */ + bitstreamPtr++; + /* 4:th int16_t */ + (*bitstreamPtr) = ((uint16_t)enc_bits->idxForMax&0x3)<<14; /* Bit 0..1 */ + (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x78)<<7; /* Bit 2..5 */ + (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x10)<<5; /* Bit 6 */ + (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x8)<<5; /* Bit 7 */ + (*bitstreamPtr) |= (enc_bits->cb_index[3]&0xFC); /* Bit 8..13 */ + (*bitstreamPtr) |= (enc_bits->gain_index[3]&0x10)>>3; /* Bit 14 */ + (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x8)>>3; /* Bit 15 */ + } + /* Class 2 bits of ULP */ + /* 4:th to 6:th int16_t for 20 ms case + 5:th to 7:th int16_t for 30 ms case */ + bitstreamPtr++; + tmpPtr=enc_bits->idxVec; + for (k=0; k<3; k++) { + (*bitstreamPtr) = 0; + for (i=15; i>=0; i--) { + (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i; + /* Bit 15-i */ + tmpPtr++; + } + bitstreamPtr++; + } + + if (mode==20) { + /* 7:th int16_t */ + (*bitstreamPtr) = 0; + for (i=15; i>6; i--) { + (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i; + /* Bit 15-i */ + tmpPtr++; + } + (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4)<<4; /* Bit 9 */ + (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<2; /* Bit 10..11 */ + (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x4)<<1; /* Bit 12 */ + (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x8)>>1; /* Bit 13 */ + (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)>>2; /* Bit 14..15 */ + + } else { /* mode==30 */ + /* 8:th int16_t */ + (*bitstreamPtr) = 0; + for (i=15; i>5; i--) { + (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i; + /* Bit 15-i */ + tmpPtr++; + } + (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x6)<<3; /* Bit 10..11 */ + (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x8); /* Bit 12 */ + (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4); /* Bit 13 */ + (*bitstreamPtr) |= (enc_bits->cb_index[3]&0x2); /* Bit 14 */ + (*bitstreamPtr) |= (enc_bits->cb_index[6]&0x80)>>7; /* Bit 15 */ + bitstreamPtr++; + /* 9:th int16_t */ + (*bitstreamPtr) = ((uint16_t)enc_bits->cb_index[6]&0x7E)<<9;/* Bit 0..5 */ + (*bitstreamPtr) |= (enc_bits->cb_index[9]&0xFE)<<2; /* Bit 6..12 */ + (*bitstreamPtr) |= (enc_bits->cb_index[12]&0xE0)>>5; /* Bit 13..15 */ + bitstreamPtr++; + /* 10:th int16_t */ + (*bitstreamPtr) = ((uint16_t)enc_bits->cb_index[12]&0x1E)<<11;/* Bit 0..3 */ + (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<8; /* Bit 4..5 */ + (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x6)<<7; /* Bit 6..7 */ + (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x18)<<3; /* Bit 8..9 */ + (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)<<2; /* Bit 10..11 */ + (*bitstreamPtr) |= (enc_bits->gain_index[9]&0x10)>>1; /* Bit 12 */ + (*bitstreamPtr) |= (enc_bits->gain_index[10]&0x8)>>1; /* Bit 13 */ + (*bitstreamPtr) |= (enc_bits->gain_index[12]&0x10)>>3; /* Bit 14 */ + (*bitstreamPtr) |= (enc_bits->gain_index[13]&0x8)>>3; /* Bit 15 */ + } + bitstreamPtr++; + /* Class 3 bits of ULP */ + /* 8:th to 14:th int16_t for 20 ms case + 11:th to 17:th int16_t for 30 ms case */ + tmpPtr=enc_bits->idxVec; + for (k=0; k<7; k++) { + (*bitstreamPtr) = 0; + for (i=14; i>=0; i-=2) { + (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x3))<<i; /* Bit 15-i..14-i*/ + tmpPtr++; + } + bitstreamPtr++; + } + + if (mode==20) { + /* 15:th int16_t */ + (*bitstreamPtr) = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */ + (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<13; /* Bit 2 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<6; /* Bit 3..9 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x7E)>>1; /* Bit 10..15 */ + bitstreamPtr++; + /* 16:th int16_t */ + (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[2])&0x1))<<15; + /* Bit 0 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<12; /* Bit 1..3 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<10; /* Bit 4..5 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[2]))<<7; /* Bit 6..8 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<6; /* Bit 9 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x7E)>>1; /* Bit 10..15 */ + bitstreamPtr++; + /* 17:th int16_t */ + (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[4])&0x1))<<15; + /* Bit 0 */ + (*bitstreamPtr) |= (enc_bits->cb_index[5])<<8; /* Bit 1..7 */ + (*bitstreamPtr) |= (enc_bits->cb_index[6]); /* Bit 8..15 */ + bitstreamPtr++; + /* 18:th int16_t */ + (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[7]))<<8; /* Bit 0..7 */ + (*bitstreamPtr) |= (enc_bits->cb_index[8]); /* Bit 8..15 */ + bitstreamPtr++; + /* 19:th int16_t */ + (*bitstreamPtr) = ((uint16_t)((enc_bits->gain_index[3])&0x3))<<14; + /* Bit 0..1 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x3)<<12; /* Bit 2..3 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[5]))<<9; /* Bit 4..6 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<6; /* Bit 7..9 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<4; /* Bit 10..11 */ + (*bitstreamPtr) |= (enc_bits->gain_index[8])<<1; /* Bit 12..14 */ + } else { /* mode==30 */ + /* 18:th int16_t */ + (*bitstreamPtr) = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */ + (*bitstreamPtr) |= (((enc_bits->idxVec[57])&0x3))<<12; /* Bit 2..3 */ + (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<11; /* Bit 4 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<4; /* Bit 5..11 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x78)>>3; /* Bit 12..15 */ + bitstreamPtr++; + /* 19:th int16_t */ + (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[2])&0x7)<<13; + /* Bit 0..2 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<10; /* Bit 3..5 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<8; /* Bit 6..7 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[2])&0x7)<<5; /* Bit 8..10 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<4; /* Bit 11 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x78)>>3; /* Bit 12..15 */ + bitstreamPtr++; + /* 20:th int16_t */ + (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[4])&0x7)<<13; + /* Bit 0..2 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[5]))<<6; /* Bit 3..9 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[6])&0x1)<<5; /* Bit 10 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[7])&0xF8)>>3; /* Bit 11..15 */ + bitstreamPtr++; + /* 21:st int16_t */ + (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[7])&0x7)<<13; + /* Bit 0..2 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[8]))<<5; /* Bit 3..10 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[9])&0x1)<<4; /* Bit 11 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[10])&0xF0)>>4; /* Bit 12..15 */ + bitstreamPtr++; + /* 22:nd int16_t */ + (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[10])&0xF)<<12; + /* Bit 0..3 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[11]))<<4; /* Bit 4..11 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[12])&0x1)<<3; /* Bit 12 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[13])&0xE0)>>5; /* Bit 13..15 */ + bitstreamPtr++; + /* 23:rd int16_t */ + (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[13])&0x1F)<<11; + /* Bit 0..4 */ + (*bitstreamPtr) |= ((enc_bits->cb_index[14]))<<3; /* Bit 5..12 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x3)<<1; /* Bit 13..14 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x1); /* Bit 15 */ + bitstreamPtr++; + /* 24:rd int16_t */ + (*bitstreamPtr) = ((uint16_t)(enc_bits->gain_index[5]))<<13; + /* Bit 0..2 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<10; /* Bit 3..5 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<8; /* Bit 6..7 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[8]))<<5; /* Bit 8..10 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[9])&0xF)<<1; /* Bit 11..14 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[10])&0x4)>>2; /* Bit 15 */ + bitstreamPtr++; + /* 25:rd int16_t */ + (*bitstreamPtr) = ((uint16_t)(enc_bits->gain_index[10])&0x3)<<14; + /* Bit 0..1 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[11]))<<11; /* Bit 2..4 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[12])&0xF)<<7; /* Bit 5..8 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[13])&0x7)<<4; /* Bit 9..11 */ + (*bitstreamPtr) |= ((enc_bits->gain_index[14]))<<1; /* Bit 12..14 */ + } + /* Last bit is automatically zero */ + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.h b/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.h new file mode 100644 index 0000000000..603ddd4c83 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_PackBits.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * unpacking of bits from bitstream, i.e., vector of bytes + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_PackBits( + uint16_t *bitstream, /* (o) The packetized bitstream */ + iLBC_bits *enc_bits, /* (i) Encoded bits */ + int16_t mode /* (i) Codec mode (20 or 30) */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c b/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c new file mode 100644 index 0000000000..df8a78b1ed --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c @@ -0,0 +1,31 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Poly2Lsf.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "poly_to_lsp.h" +#include "lsp_to_lsf.h" + +void WebRtcIlbcfix_Poly2Lsf( + int16_t *lsf, /* (o) lsf coefficients (Q13) */ + int16_t *a /* (i) A coefficients (Q12) */ + ) { + int16_t lsp[10]; + WebRtcIlbcfix_Poly2Lsp(a, lsp, (int16_t*)WebRtcIlbcfix_kLspMean); + WebRtcIlbcfix_Lsp2Lsf(lsp, lsf, 10); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h new file mode 100644 index 0000000000..5a7f7bbbbe --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Poly2Lsf.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * conversion from lpc coefficients to lsf coefficients + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Poly2Lsf( + int16_t *lsf, /* (o) lsf coefficients (Q13) */ + int16_t *a /* (i) A coefficients (Q12) */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c b/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c new file mode 100644 index 0000000000..ca376b34de --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c @@ -0,0 +1,158 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Poly2Lsp.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "chebyshev.h" + +/*----------------------------------------------------------------* + * conversion from lpc coefficients to lsp coefficients + * function is only for 10:th order LPC + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Poly2Lsp( + int16_t *a, /* (o) A coefficients in Q12 */ + int16_t *lsp, /* (i) LSP coefficients in Q15 */ + int16_t *old_lsp /* (i) old LSP coefficients that are used if the new + coefficients turn out to be unstable */ + ) { + int16_t f[2][6]; /* f[0][] represents f1 and f[1][] represents f2 */ + int16_t *a_i_ptr, *a_10mi_ptr; + int16_t *f1ptr, *f2ptr; + int32_t tmpW32; + int16_t x, y, xlow, ylow, xmid, ymid, xhigh, yhigh, xint; + int16_t shifts, sign; + int i, j; + int foundFreqs; + int fi_select; + + /* + Calculate the two polynomials f1(z) and f2(z) + (the sum and the diff polynomial) + f1[0] = f2[0] = 1.0; + f1[i+1] = a[i+1] + a[10-i] - f1[i]; + f2[i+1] = a[i+1] - a[10-i] - f1[i]; + */ + + a_i_ptr = a + 1; + a_10mi_ptr = a + 10; + f1ptr = f[0]; + f2ptr = f[1]; + (*f1ptr) = 1024; /* 1.0 in Q10 */ + (*f2ptr) = 1024; /* 1.0 in Q10 */ + for (i = 0; i < 5; i++) { + *(f1ptr + 1) = + (int16_t)((((int32_t)(*a_i_ptr) + *a_10mi_ptr) >> 2) - *f1ptr); + *(f2ptr + 1) = + (int16_t)((((int32_t)(*a_i_ptr) - *a_10mi_ptr) >> 2) + *f2ptr); + a_i_ptr++; + a_10mi_ptr--; + f1ptr++; + f2ptr++; + } + + /* + find the LSPs using the Chebychev pol. evaluation + */ + + fi_select = 0; /* selector between f1 and f2, start with f1 */ + + foundFreqs = 0; + + xlow = WebRtcIlbcfix_kCosGrid[0]; + ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]); + + /* + Iterate until all the 10 LSP's have been found or + all the grid points have been tried. If the 10 LSP's can + not be found, set the LSP vector to previous LSP + */ + + for (j = 1; j < COS_GRID_POINTS && foundFreqs < 10; j++) { + xhigh = xlow; + yhigh = ylow; + xlow = WebRtcIlbcfix_kCosGrid[j]; + ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]); + + if (ylow * yhigh <= 0) { + /* Run 4 times to reduce the interval */ + for (i = 0; i < 4; i++) { + /* xmid =(xlow + xhigh)/2 */ + xmid = (xlow >> 1) + (xhigh >> 1); + ymid = WebRtcIlbcfix_Chebyshev(xmid, f[fi_select]); + + if (ylow * ymid <= 0) { + yhigh = ymid; + xhigh = xmid; + } else { + ylow = ymid; + xlow = xmid; + } + } + + /* + Calculater xint by linear interpolation: + xint = xlow - ylow*(xhigh-xlow)/(yhigh-ylow); + */ + + x = xhigh - xlow; + y = yhigh - ylow; + + if (y == 0) { + xint = xlow; + } else { + sign = y; + y = WEBRTC_SPL_ABS_W16(y); + shifts = (int16_t)WebRtcSpl_NormW32(y)-16; + y <<= shifts; + y = (int16_t)WebRtcSpl_DivW32W16(536838144, y); /* 1/(yhigh-ylow) */ + + tmpW32 = (x * y) >> (19 - shifts); + + /* y=(xhigh-xlow)/(yhigh-ylow) */ + y = (int16_t)(tmpW32&0xFFFF); + + if (sign < 0) { + y = -y; + } + /* tmpW32 = ylow*(xhigh-xlow)/(yhigh-ylow) */ + tmpW32 = (ylow * y) >> 10; + xint = xlow-(int16_t)(tmpW32&0xFFFF); + } + + /* Store the calculated lsp */ + lsp[foundFreqs] = (int16_t)xint; + foundFreqs++; + + /* if needed, set xlow and ylow for next recursion */ + if (foundFreqs<10) { + xlow = xint; + /* Swap between f1 and f2 (f[0][] and f[1][]) */ + fi_select = ((fi_select+1)&0x1); + + ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]); + } + } + } + + /* Check if M roots found, if not then use the old LSP */ + if (foundFreqs < 10) { + WEBRTC_SPL_MEMCPY_W16(lsp, old_lsp, 10); + } + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h new file mode 100644 index 0000000000..ed20fd970b --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Poly2Lsp.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * conversion from lpc coefficients to lsp coefficients + * function is only for 10:th order LPC + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Poly2Lsp( + int16_t *a, /* (o) A coefficients in Q12 */ + int16_t *lsp, /* (i) LSP coefficients in Q15 */ + int16_t *old_lsp /* (i) old LSP coefficients that are used if the new + coefficients turn out to be unstable */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/refiner.c b/webrtc/modules/audio_coding/codecs/ilbc/refiner.c new file mode 100644 index 0000000000..3c1265e5d5 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/refiner.c @@ -0,0 +1,140 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Refiner.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "enh_upsample.h" +#include "my_corr.h" + +/*----------------------------------------------------------------* + * find segment starting near idata+estSegPos that has highest + * correlation with idata+centerStartPos through + * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a + * resolution of ENH_UPSO times the original of the original + * sampling rate + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Refiner( + size_t *updStartPos, /* (o) updated start point (Q-2) */ + int16_t *idata, /* (i) original data buffer */ + size_t idatal, /* (i) dimension of idata */ + size_t centerStartPos, /* (i) beginning center segment */ + size_t estSegPos, /* (i) estimated beginning other segment (Q-2) */ + int16_t *surround, /* (i/o) The contribution from this sequence + summed with earlier contributions */ + int16_t gain /* (i) Gain to use for this sequence */ + ){ + size_t estSegPosRounded, searchSegStartPos, searchSegEndPos, corrdim; + size_t tloc, tloc2, i; + + int32_t maxtemp, scalefact; + int16_t *filtStatePtr, *polyPtr; + /* Stack based */ + int16_t filt[7]; + int32_t corrVecUps[ENH_CORRDIM*ENH_UPS0]; + int32_t corrVecTemp[ENH_CORRDIM]; + int16_t vect[ENH_VECTL]; + int16_t corrVec[ENH_CORRDIM]; + + /* defining array bounds */ + + estSegPosRounded = (estSegPos - 2) >> 2; + + searchSegStartPos = + (estSegPosRounded < ENH_SLOP) ? 0 : (estSegPosRounded - ENH_SLOP); + + searchSegEndPos = estSegPosRounded + ENH_SLOP; + if ((searchSegEndPos + ENH_BLOCKL) >= idatal) { + searchSegEndPos = idatal - ENH_BLOCKL - 1; + } + + corrdim = searchSegEndPos + 1 - searchSegStartPos; + + /* compute upsampled correlation and find + location of max */ + + WebRtcIlbcfix_MyCorr(corrVecTemp, idata + searchSegStartPos, + corrdim + ENH_BLOCKL - 1, idata + centerStartPos, + ENH_BLOCKL); + + /* Calculate the rescaling factor for the correlation in order to + put the correlation in a int16_t vector instead */ + maxtemp = WebRtcSpl_MaxAbsValueW32(corrVecTemp, corrdim); + + scalefact = WebRtcSpl_GetSizeInBits(maxtemp) - 15; + + if (scalefact > 0) { + for (i = 0; i < corrdim; i++) { + corrVec[i] = (int16_t)(corrVecTemp[i] >> scalefact); + } + } else { + for (i = 0; i < corrdim; i++) { + corrVec[i] = (int16_t)corrVecTemp[i]; + } + } + /* In order to guarantee that all values are initialized */ + for (i = corrdim; i < ENH_CORRDIM; i++) { + corrVec[i] = 0; + } + + /* Upsample the correlation */ + WebRtcIlbcfix_EnhUpsample(corrVecUps, corrVec); + + /* Find maximum */ + tloc = WebRtcSpl_MaxIndexW32(corrVecUps, ENH_UPS0 * corrdim); + + /* make vector can be upsampled without ever running outside + bounds */ + *updStartPos = searchSegStartPos * 4 + tloc + 4; + + tloc2 = (tloc + 3) >> 2; + + /* initialize the vector to be filtered, stuff with zeros + when data is outside idata buffer */ + if (ENH_FL0 > (searchSegStartPos + tloc2)) { + const size_t st = ENH_FL0 - searchSegStartPos - tloc2; + WebRtcSpl_MemSetW16(vect, 0, st); + WEBRTC_SPL_MEMCPY_W16(&vect[st], idata, ENH_VECTL - st); + } else { + const size_t st = searchSegStartPos + tloc2 - ENH_FL0; + if ((st + ENH_VECTL) > idatal) { + const size_t en = st + ENH_VECTL - idatal; + WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL - en); + WebRtcSpl_MemSetW16(&vect[ENH_VECTL - en], 0, en); + } else { + WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL); + } + } + + /* compute the segment (this is actually a convolution) */ + filtStatePtr = filt + 6; + polyPtr = (int16_t*)WebRtcIlbcfix_kEnhPolyPhaser[tloc2 * ENH_UPS0 - tloc]; + for (i = 0; i < 7; i++) { + *filtStatePtr-- = *polyPtr++; + } + + WebRtcSpl_FilterMAFastQ12(&vect[6], vect, filt, ENH_FLO_MULT2_PLUS1, + ENH_BLOCKL); + + /* Add the contribution from this vector (scaled with gain) to the total + surround vector */ + WebRtcSpl_AddAffineVectorToVector(surround, vect, gain, 32768, 16, + ENH_BLOCKL); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/refiner.h b/webrtc/modules/audio_coding/codecs/ilbc/refiner.h new file mode 100644 index 0000000000..f8a2abc2d6 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/refiner.h @@ -0,0 +1,43 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Refiner.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * find segment starting near idata+estSegPos that has highest + * correlation with idata+centerStartPos through + * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a + * resolution of ENH_UPSO times the original of the original + * sampling rate + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Refiner( + size_t *updStartPos, /* (o) updated start point (Q-2) */ + int16_t *idata, /* (i) original data buffer */ + size_t idatal, /* (i) dimension of idata */ + size_t centerStartPos, /* (i) beginning center segment */ + size_t estSegPos, /* (i) estimated beginning other segment (Q-2) */ + int16_t *surround, /* (i/o) The contribution from this sequence + summed with earlier contributions */ + int16_t gain /* (i) Gain to use for this sequence */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c new file mode 100644 index 0000000000..e63dda8c8f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c @@ -0,0 +1,131 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SimpleInterpolateLsf.c + +******************************************************************/ + +#include "defines.h" +#include "lsf_interpolate_to_poly_enc.h" +#include "bw_expand.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * lsf interpolator (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SimpleInterpolateLsf( + int16_t *syntdenum, /* (o) the synthesis filter denominator + resulting from the quantized + interpolated lsf Q12 */ + int16_t *weightdenum, /* (o) the weighting filter denominator + resulting from the unquantized + interpolated lsf Q12 */ + int16_t *lsf, /* (i) the unquantized lsf coefficients Q13 */ + int16_t *lsfdeq, /* (i) the dequantized lsf coefficients Q13 */ + int16_t *lsfold, /* (i) the unquantized lsf coefficients of + the previous signal frame Q13 */ + int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the + previous signal frame Q13 */ + int16_t length, /* (i) should equate FILTERORDER */ + IlbcEncoder *iLBCenc_inst + /* (i/o) the encoder state structure */ + ) { + size_t i; + int pos, lp_length; + + int16_t *lsf2, *lsfdeq2; + /* Stack based */ + int16_t lp[LPC_FILTERORDER + 1]; + + lsf2 = lsf + length; + lsfdeq2 = lsfdeq + length; + lp_length = length + 1; + + if (iLBCenc_inst->mode==30) { + /* subframe 1: Interpolation between old and first set of + lsf coefficients */ + + /* Calculate Analysis/Syntehsis filter from quantized LSF */ + WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq, + WebRtcIlbcfix_kLsfWeight30ms[0], + length); + WEBRTC_SPL_MEMCPY_W16(syntdenum, lp, lp_length); + + /* Calculate Weighting filter from quantized LSF */ + WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf, + WebRtcIlbcfix_kLsfWeight30ms[0], + length); + WebRtcIlbcfix_BwExpand(weightdenum, lp, + (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum, + (int16_t)lp_length); + + /* subframe 2 to 6: Interpolation between first and second + set of lsf coefficients */ + + pos = lp_length; + for (i = 1; i < iLBCenc_inst->nsub; i++) { + + /* Calculate Analysis/Syntehsis filter from quantized LSF */ + WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeq, lsfdeq2, + WebRtcIlbcfix_kLsfWeight30ms[i], + length); + WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length); + + /* Calculate Weighting filter from quantized LSF */ + WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsf, lsf2, + WebRtcIlbcfix_kLsfWeight30ms[i], + length); + WebRtcIlbcfix_BwExpand(weightdenum + pos, lp, + (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum, + (int16_t)lp_length); + + pos += lp_length; + } + + /* update memory */ + + WEBRTC_SPL_MEMCPY_W16(lsfold, lsf2, length); + WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq2, length); + + } else { /* iLBCenc_inst->mode==20 */ + pos = 0; + for (i = 0; i < iLBCenc_inst->nsub; i++) { + + /* Calculate Analysis/Syntehsis filter from quantized LSF */ + WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq, + WebRtcIlbcfix_kLsfWeight20ms[i], + length); + WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length); + + /* Calculate Weighting filter from quantized LSF */ + WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf, + WebRtcIlbcfix_kLsfWeight20ms[i], + length); + WebRtcIlbcfix_BwExpand(weightdenum+pos, lp, + (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum, + (int16_t)lp_length); + + pos += lp_length; + } + + /* update memory */ + + WEBRTC_SPL_MEMCPY_W16(lsfold, lsf, length); + WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq, length); + + } + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h new file mode 100644 index 0000000000..6abcc00b9d --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SimpleInterpolateLsf.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * lsf interpolator (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SimpleInterpolateLsf( + int16_t *syntdenum, /* (o) the synthesis filter denominator + resulting from the quantized + interpolated lsf Q12 */ + int16_t *weightdenum, /* (o) the weighting filter denominator + resulting from the unquantized + interpolated lsf Q12 */ + int16_t *lsf, /* (i) the unquantized lsf coefficients Q13 */ + int16_t *lsfdeq, /* (i) the dequantized lsf coefficients Q13 */ + int16_t *lsfold, /* (i) the unquantized lsf coefficients of + the previous signal frame Q13 */ + int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the + previous signal frame Q13 */ + int16_t length, /* (i) should equate FILTERORDER */ + IlbcEncoder *iLBCenc_inst + /* (i/o) the encoder state structure */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c new file mode 100644 index 0000000000..72d80e0430 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c @@ -0,0 +1,94 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SimpleLpcAnalysis.c + +******************************************************************/ + +#include "defines.h" +#include "window32_w32.h" +#include "bw_expand.h" +#include "poly_to_lsf.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * lpc analysis (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SimpleLpcAnalysis( + int16_t *lsf, /* (o) lsf coefficients */ + int16_t *data, /* (i) new block of speech */ + IlbcEncoder *iLBCenc_inst + /* (i/o) the encoder state structure */ + ) { + int k; + int scale; + size_t is; + int16_t stability; + /* Stack based */ + int16_t A[LPC_FILTERORDER + 1]; + int32_t R[LPC_FILTERORDER + 1]; + int16_t windowedData[BLOCKL_MAX]; + int16_t rc[LPC_FILTERORDER]; + + is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl; + WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer+is,data,iLBCenc_inst->blockl); + + /* No lookahead, last window is asymmetric */ + + for (k = 0; k < iLBCenc_inst->lpc_n; k++) { + + is = LPC_LOOKBACK; + + if (k < (iLBCenc_inst->lpc_n - 1)) { + + /* Hanning table WebRtcIlbcfix_kLpcWin[] is in Q15-domain so the output is right-shifted 15 */ + WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer, WebRtcIlbcfix_kLpcWin, BLOCKL_MAX, 15); + } else { + + /* Hanning table WebRtcIlbcfix_kLpcAsymWin[] is in Q15-domain so the output is right-shifted 15 */ + WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer+is, WebRtcIlbcfix_kLpcAsymWin, BLOCKL_MAX, 15); + } + + /* Compute autocorrelation */ + WebRtcSpl_AutoCorrelation(windowedData, BLOCKL_MAX, LPC_FILTERORDER, R, &scale); + + /* Window autocorrelation vector */ + WebRtcIlbcfix_Window32W32(R, R, WebRtcIlbcfix_kLpcLagWin, LPC_FILTERORDER + 1 ); + + /* Calculate the A coefficients from the Autocorrelation using Levinson Durbin algorithm */ + stability=WebRtcSpl_LevinsonDurbin(R, A, rc, LPC_FILTERORDER); + + /* + Set the filter to {1.0, 0.0, 0.0,...} if filter from Levinson Durbin algorithm is unstable + This should basically never happen... + */ + if (stability!=1) { + A[0]=4096; + WebRtcSpl_MemSetW16(&A[1], 0, LPC_FILTERORDER); + } + + /* Bandwidth expand the filter coefficients */ + WebRtcIlbcfix_BwExpand(A, A, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1); + + /* Convert from A to LSF representation */ + WebRtcIlbcfix_Poly2Lsf(lsf + k*LPC_FILTERORDER, A); + } + + is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl; + WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer, + iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is, is); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h new file mode 100644 index 0000000000..7808da140d --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SimpleLpcAnalysis.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * lpc analysis (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SimpleLpcAnalysis( + int16_t *lsf, /* (o) lsf coefficients */ + int16_t *data, /* (i) new block of speech */ + IlbcEncoder *iLBCenc_inst + /* (i/o) the encoder state structure */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c b/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c new file mode 100644 index 0000000000..9fa0d61ded --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c @@ -0,0 +1,60 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SimpleLsfDeQ.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * obtain dequantized lsf coefficients from quantization index + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SimpleLsfDeQ( + int16_t *lsfdeq, /* (o) dequantized lsf coefficients */ + int16_t *index, /* (i) quantization index */ + int16_t lpc_n /* (i) number of LPCs */ + ){ + int i, j, pos, cb_pos; + + /* decode first LSF */ + + pos = 0; + cb_pos = 0; + for (i = 0; i < LSF_NSPLIT; i++) { + for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) { + lsfdeq[pos + j] = WebRtcIlbcfix_kLsfCb[cb_pos + j + index[i] * + WebRtcIlbcfix_kLsfDimCb[i]]; + } + pos += WebRtcIlbcfix_kLsfDimCb[i]; + cb_pos += WebRtcIlbcfix_kLsfSizeCb[i] * WebRtcIlbcfix_kLsfDimCb[i]; + } + + if (lpc_n>1) { + /* decode last LSF */ + pos = 0; + cb_pos = 0; + for (i = 0; i < LSF_NSPLIT; i++) { + for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) { + lsfdeq[LPC_FILTERORDER + pos + j] = WebRtcIlbcfix_kLsfCb[ + cb_pos + index[LSF_NSPLIT + i] * WebRtcIlbcfix_kLsfDimCb[i] + j]; + } + pos += WebRtcIlbcfix_kLsfDimCb[i]; + cb_pos += WebRtcIlbcfix_kLsfSizeCb[i] * WebRtcIlbcfix_kLsfDimCb[i]; + } + } + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h new file mode 100644 index 0000000000..353edb2225 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SimpleLsfDeQ.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * obtain dequantized lsf coefficients from quantization index + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SimpleLsfDeQ( + int16_t *lsfdeq, /* (o) dequantized lsf coefficients */ + int16_t *index, /* (i) quantization index */ + int16_t lpc_n /* (i) number of LPCs */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c b/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c new file mode 100644 index 0000000000..8daba50150 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c @@ -0,0 +1,47 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SimpleLsfQ.c + +******************************************************************/ + +#include "defines.h" +#include "split_vq.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * lsf quantizer (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SimpleLsfQ( + int16_t *lsfdeq, /* (o) dequantized lsf coefficients + (dimension FILTERORDER) Q13 */ + int16_t *index, /* (o) quantization index */ + int16_t *lsf, /* (i) the lsf coefficient vector to be + quantized (dimension FILTERORDER) Q13 */ + int16_t lpc_n /* (i) number of lsf sets to quantize */ + ){ + + /* Quantize first LSF with memoryless split VQ */ + WebRtcIlbcfix_SplitVq( lsfdeq, index, lsf, + (int16_t*)WebRtcIlbcfix_kLsfCb, (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb); + + if (lpc_n==2) { + /* Quantize second LSF with memoryless split VQ */ + WebRtcIlbcfix_SplitVq( lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT, + lsf + LPC_FILTERORDER, (int16_t*)WebRtcIlbcfix_kLsfCb, + (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb); + } + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h new file mode 100644 index 0000000000..94f804bfdd --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h @@ -0,0 +1,37 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SimpleLsfQ.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * lsf quantizer (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SimpleLsfQ( + int16_t *lsfdeq, /* (o) dequantized lsf coefficients + (dimension FILTERORDER) Q13 */ + int16_t *index, /* (o) quantization index */ + int16_t *lsf, /* (i) the lsf coefficient vector to be + quantized (dimension FILTERORDER) Q13 */ + int16_t lpc_n /* (i) number of lsf sets to quantize */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/smooth.c b/webrtc/modules/audio_coding/codecs/ilbc/smooth.c new file mode 100644 index 0000000000..f58e72c178 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/smooth.c @@ -0,0 +1,207 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Smooth.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "smooth_out_data.h" + +/*----------------------------------------------------------------* + * find the smoothed output data + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Smooth( + int16_t *odata, /* (o) smoothed output */ + int16_t *current, /* (i) the un enhanced residual for + this block */ + int16_t *surround /* (i) The approximation from the + surrounding sequences */ + ) { + int16_t maxtot, scale, scale1, scale2; + int16_t A, B, C, denomW16; + int32_t B_W32, denom, num; + int32_t errs; + int32_t w00,w10,w11, endiff, crit; + int32_t w00prim, w10prim, w11_div_w00; + int16_t w11prim; + int16_t bitsw00, bitsw10, bitsw11; + int32_t w11w00, w10w10, w00w00; + int16_t max1, max2; + + /* compute some inner products (ensure no overflow by first calculating proper scale factor) */ + + w00 = w10 = w11 = 0; + + max1=WebRtcSpl_MaxAbsValueW16(current, ENH_BLOCKL); + max2=WebRtcSpl_MaxAbsValueW16(surround, ENH_BLOCKL); + maxtot=WEBRTC_SPL_MAX(max1, max2); + + scale=WebRtcSpl_GetSizeInBits(maxtot); + scale = (int16_t)(2 * scale) - 26; + scale=WEBRTC_SPL_MAX(0, scale); + + w00=WebRtcSpl_DotProductWithScale(current,current,ENH_BLOCKL,scale); + w11=WebRtcSpl_DotProductWithScale(surround,surround,ENH_BLOCKL,scale); + w10=WebRtcSpl_DotProductWithScale(surround,current,ENH_BLOCKL,scale); + + if (w00<0) w00 = WEBRTC_SPL_WORD32_MAX; + if (w11<0) w11 = WEBRTC_SPL_WORD32_MAX; + + /* Rescale w00 and w11 to w00prim and w11prim, so that w00prim/w11prim + is in Q16 */ + + bitsw00 = WebRtcSpl_GetSizeInBits(w00); + bitsw11 = WebRtcSpl_GetSizeInBits(w11); + bitsw10 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(w10)); + scale1 = 31 - bitsw00; + scale2 = 15 - bitsw11; + + if (scale2>(scale1-16)) { + scale2 = scale1 - 16; + } else { + scale1 = scale2 + 16; + } + + w00prim = w00 << scale1; + w11prim = (int16_t) WEBRTC_SPL_SHIFT_W32(w11, scale2); + + /* Perform C = sqrt(w11/w00) (C is in Q11 since (16+6)/2=11) */ + if (w11prim>64) { + endiff = WebRtcSpl_DivW32W16(w00prim, w11prim) << 6; + C = (int16_t)WebRtcSpl_SqrtFloor(endiff); /* C is in Q11 */ + } else { + C = 1; + } + + /* first try enhancement without power-constraint */ + + errs = WebRtcIlbcfix_Smooth_odata(odata, current, surround, C); + + + + /* if constraint violated by first try, add constraint */ + + if ( (6-scale+scale1) > 31) { + crit=0; + } else { + /* crit = 0.05 * w00 (Result in Q-6) */ + crit = WEBRTC_SPL_SHIFT_W32( + WEBRTC_SPL_MUL(ENH_A0, w00prim >> 14), + -(6-scale+scale1)); + } + + if (errs > crit) { + + if( w00 < 1) { + w00=1; + } + + /* Calculate w11*w00, w10*w10 and w00*w00 in the same Q domain */ + + scale1 = bitsw00-15; + scale2 = bitsw11-15; + + if (scale2>scale1) { + scale = scale2; + } else { + scale = scale1; + } + + w11w00 = (int16_t)WEBRTC_SPL_SHIFT_W32(w11, -scale) * + (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale); + + w10w10 = (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale) * + (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale); + + w00w00 = (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale) * + (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale); + + /* Calculate (w11*w00-w10*w10)/(w00*w00) in Q16 */ + if (w00w00>65536) { + endiff = (w11w00-w10w10); + endiff = WEBRTC_SPL_MAX(0, endiff); + /* denom is in Q16 */ + denom = WebRtcSpl_DivW32W16(endiff, (int16_t)(w00w00 >> 16)); + } else { + denom = 65536; + } + + if( denom > 7){ /* eliminates numerical problems + for if smooth */ + + scale=WebRtcSpl_GetSizeInBits(denom)-15; + + if (scale>0) { + /* denomW16 is in Q(16+scale) */ + denomW16 = (int16_t)(denom >> scale); + + /* num in Q(34-scale) */ + num = ENH_A0_MINUS_A0A0DIV4 >> scale; + } else { + /* denomW16 is in Q16 */ + denomW16=(int16_t)denom; + + /* num in Q34 */ + num=ENH_A0_MINUS_A0A0DIV4; + } + + /* A sqrt( (ENH_A0-(ENH_A0^2)/4)*(w00*w00)/(w11*w00 + w10*w10) ) in Q9 */ + A = (int16_t)WebRtcSpl_SqrtFloor(WebRtcSpl_DivW32W16(num, denomW16)); + + /* B_W32 is in Q30 ( B = 1 - ENH_A0/2 - A * w10/w00 ) */ + scale1 = 31-bitsw10; + scale2 = 21-scale1; + w10prim = w10 << scale1; + w00prim = WEBRTC_SPL_SHIFT_W32(w00, -scale2); + scale = bitsw00-scale2-15; + + if (scale>0) { + w10prim >>= scale; + w00prim >>= scale; + } + + if ((w00prim>0)&&(w10prim>0)) { + w11_div_w00=WebRtcSpl_DivW32W16(w10prim, (int16_t)w00prim); + + if (WebRtcSpl_GetSizeInBits(w11_div_w00)+WebRtcSpl_GetSizeInBits(A)>31) { + B_W32 = 0; + } else { + B_W32 = (int32_t)1073741824 - (int32_t)ENH_A0DIV2 - + WEBRTC_SPL_MUL(A, w11_div_w00); + } + B = (int16_t)(B_W32 >> 16); /* B in Q14. */ + } else { + /* No smoothing */ + A = 0; + B = 16384; /* 1 in Q14 */ + } + } + else{ /* essentially no difference between cycles; + smoothing not needed */ + + A = 0; + B = 16384; /* 1 in Q14 */ + } + + /* create smoothed sequence */ + + WebRtcSpl_ScaleAndAddVectors(surround, A, 9, + current, B, 14, + odata, ENH_BLOCKL); + } + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/smooth.h b/webrtc/modules/audio_coding/codecs/ilbc/smooth.h new file mode 100644 index 0000000000..add0c7ba47 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/smooth.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Smooth.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * find the smoothed output data + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Smooth( + int16_t *odata, /* (o) smoothed output */ + int16_t *current, /* (i) the un enhanced residual for + this block */ + int16_t *surround /* (i) The approximation from the + surrounding sequences */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c b/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c new file mode 100644 index 0000000000..9c41e0422b --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c @@ -0,0 +1,44 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Smooth_odata.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +int32_t WebRtcIlbcfix_Smooth_odata( + int16_t *odata, + int16_t *psseq, + int16_t *surround, + int16_t C) +{ + int i; + + int16_t err; + int32_t errs; + + for(i=0;i<80;i++) { + odata[i]= (int16_t)((C * surround[i] + 1024) >> 11); + } + + errs=0; + for(i=0;i<80;i++) { + err = (psseq[i] - odata[i]) >> 3; + errs += err * err; /* errs in Q-6 */ + } + + return errs; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h new file mode 100644 index 0000000000..83244392bb --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Smooth_odata.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * help function to WebRtcIlbcfix_Smooth() + *---------------------------------------------------------------*/ + +int32_t WebRtcIlbcfix_Smooth_odata( + int16_t *odata, + int16_t *psseq, + int16_t *surround, + int16_t C); + + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c b/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c new file mode 100644 index 0000000000..c51bf6dc63 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SortSq.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * scalar quantization + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SortSq( + int16_t *xq, /* (o) the quantized value */ + int16_t *index, /* (o) the quantization index */ + int16_t x, /* (i) the value to quantize */ + const int16_t *cb, /* (i) the quantization codebook */ + int16_t cb_size /* (i) the size of the quantization codebook */ + ){ + int i; + + if (x <= cb[0]) { + *index = 0; + *xq = cb[0]; + } else { + i = 0; + while ((x > cb[i]) && (i < (cb_size-1))) { + i++; + } + + if (x > (((int32_t)cb[i] + cb[i - 1] + 1) >> 1)) { + *index = i; + *xq = cb[i]; + } else { + *index = i - 1; + *xq = cb[i - 1]; + } + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.h b/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.h new file mode 100644 index 0000000000..eaf175be1d --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SortSq.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * scalar quantization + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SortSq( + int16_t *xq, /* (o) the quantized value */ + int16_t *index, /* (o) the quantization index */ + int16_t x, /* (i) the value to quantize */ + const int16_t *cb, /* (i) the quantization codebook */ + int16_t cb_size /* (i) the size of the quantization codebook */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c b/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c new file mode 100644 index 0000000000..39b6e1b56a --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c @@ -0,0 +1,61 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SplitVq.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "vq3.h" +#include "vq4.h" + +/*----------------------------------------------------------------* + * split vector quantization + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SplitVq( + int16_t *qX, /* (o) the quantized vector in Q13 */ + int16_t *index, /* (o) a vector of indexes for all vector + codebooks in the split */ + int16_t *X, /* (i) the vector to quantize */ + int16_t *CB, /* (i) the quantizer codebook in Q13 */ + int16_t *dim, /* (i) the dimension of X and qX */ + int16_t *cbsize /* (i) the number of vectors in the codebook */ + ) { + + int16_t *qXPtr, *indexPtr, *CBPtr, *XPtr; + + /* Quantize X with the 3 vectror quantization tables */ + + qXPtr=qX; + indexPtr=index; + CBPtr=CB; + XPtr=X; + WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[0]); + + qXPtr+=3; + indexPtr+=1; + CBPtr+=(dim[0]*cbsize[0]); + XPtr+=3; + WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[1]); + + qXPtr+=3; + indexPtr+=1; + CBPtr+=(dim[1]*cbsize[1]); + XPtr+=3; + WebRtcIlbcfix_Vq4(qXPtr, indexPtr, CBPtr, XPtr, cbsize[2]); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/split_vq.h b/webrtc/modules/audio_coding/codecs/ilbc/split_vq.h new file mode 100644 index 0000000000..2ca98cb5b1 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/split_vq.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SplitVq.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * split vector quantization + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SplitVq( + int16_t *qX, /* (o) the quantized vector in Q13 */ + int16_t *index, /* (o) a vector of indexes for all vector + codebooks in the split */ + int16_t *X, /* (i) the vector to quantize */ + int16_t *CB, /* (i) the quantizer codebook in Q13 */ + int16_t *dim, /* (i) the dimension of X and qX */ + int16_t *cbsize /* (i) the number of vectors in the codebook */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c b/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c new file mode 100644 index 0000000000..29fe91b87e --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c @@ -0,0 +1,114 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_StateConstruct.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * decoding of the start state + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_StateConstruct( + size_t idxForMax, /* (i) 6-bit index for the quantization of + max amplitude */ + int16_t *idxVec, /* (i) vector of quantization indexes */ + int16_t *syntDenum, /* (i) synthesis filter denumerator */ + int16_t *Out_fix, /* (o) the decoded state vector */ + size_t len /* (i) length of a state vector */ + ) { + size_t k; + int16_t maxVal; + int16_t *tmp1, *tmp2, *tmp3; + /* Stack based */ + int16_t numerator[1+LPC_FILTERORDER]; + int16_t sampleValVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER]; + int16_t sampleMaVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER]; + int16_t *sampleVal = &sampleValVec[LPC_FILTERORDER]; + int16_t *sampleMa = &sampleMaVec[LPC_FILTERORDER]; + int16_t *sampleAr = &sampleValVec[LPC_FILTERORDER]; + + /* initialization of coefficients */ + + for (k=0; k<LPC_FILTERORDER+1; k++){ + numerator[k] = syntDenum[LPC_FILTERORDER-k]; + } + + /* decoding of the maximum value */ + + maxVal = WebRtcIlbcfix_kFrgQuantMod[idxForMax]; + + /* decoding of the sample values */ + tmp1 = sampleVal; + tmp2 = &idxVec[len-1]; + + if (idxForMax<37) { + for(k=0; k<len; k++){ + /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 2097152 (= 0.5 << 22) + maxVal is in Q8 and result is in Q(-1) */ + *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 2097152) >> + 22); + tmp1++; + tmp2--; + } + } else if (idxForMax<59) { + for(k=0; k<len; k++){ + /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 262144 (= 0.5 << 19) + maxVal is in Q5 and result is in Q(-1) */ + *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 262144) >> + 19); + tmp1++; + tmp2--; + } + } else { + for(k=0; k<len; k++){ + /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 65536 (= 0.5 << 17) + maxVal is in Q3 and result is in Q(-1) */ + *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 65536) >> + 17); + tmp1++; + tmp2--; + } + } + + /* Set the rest of the data to zero */ + WebRtcSpl_MemSetW16(&sampleVal[len], 0, len); + + /* circular convolution with all-pass filter */ + + /* Set the state to zero */ + WebRtcSpl_MemSetW16(sampleValVec, 0, (LPC_FILTERORDER)); + + /* Run MA filter + AR filter */ + WebRtcSpl_FilterMAFastQ12( + sampleVal, sampleMa, + numerator, LPC_FILTERORDER+1, len + LPC_FILTERORDER); + WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER)); + WebRtcSpl_FilterARFastQ12( + sampleMa, sampleAr, + syntDenum, LPC_FILTERORDER+1, 2 * len); + + tmp1 = &sampleAr[len-1]; + tmp2 = &sampleAr[2*len-1]; + tmp3 = Out_fix; + for(k=0;k<len;k++){ + (*tmp3) = (*tmp1) + (*tmp2); + tmp1--; + tmp2--; + tmp3++; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h b/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h new file mode 100644 index 0000000000..26319193b8 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_StateConstruct.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_ + +/*----------------------------------------------------------------* + * Generate the start state from the quantized indexes + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_StateConstruct( + size_t idxForMax, /* (i) 6-bit index for the quantization of + max amplitude */ + int16_t *idxVec, /* (i) vector of quantization indexes */ + int16_t *syntDenum, /* (i) synthesis filter denumerator */ + int16_t *Out_fix, /* (o) the decoded state vector */ + size_t len /* (i) length of a state vector */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_search.c b/webrtc/modules/audio_coding/codecs/ilbc/state_search.c new file mode 100644 index 0000000000..295c543d84 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/state_search.c @@ -0,0 +1,119 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_StateSearch.c + +******************************************************************/ + +#include "defines.h" +#include "constants.h" +#include "abs_quant.h" + +/*----------------------------------------------------------------* + * encoding of start state + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_StateSearch( + IlbcEncoder *iLBCenc_inst, + /* (i) Encoder instance */ + iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax + and idxVec, input state_first) */ + int16_t *residual, /* (i) target residual vector */ + int16_t *syntDenum, /* (i) lpc synthesis filter */ + int16_t *weightDenum /* (i) weighting filter denuminator */ + ) { + size_t k, index; + int16_t maxVal; + int16_t scale, shift; + int32_t maxValsq; + int16_t scaleRes; + int16_t max; + int i; + /* Stack based */ + int16_t numerator[1+LPC_FILTERORDER]; + int16_t residualLongVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER]; + int16_t sampleMa[2*STATE_SHORT_LEN_30MS]; + int16_t *residualLong = &residualLongVec[LPC_FILTERORDER]; + int16_t *sampleAr = residualLong; + + /* Scale to maximum 12 bits to avoid saturation in circular convolution filter */ + max = WebRtcSpl_MaxAbsValueW16(residual, iLBCenc_inst->state_short_len); + scaleRes = WebRtcSpl_GetSizeInBits(max)-12; + scaleRes = WEBRTC_SPL_MAX(0, scaleRes); + /* Set up the filter coefficients for the circular convolution */ + for (i=0; i<LPC_FILTERORDER+1; i++) { + numerator[i] = (syntDenum[LPC_FILTERORDER-i]>>scaleRes); + } + + /* Copy the residual to a temporary buffer that we can filter + * and set the remaining samples to zero. + */ + WEBRTC_SPL_MEMCPY_W16(residualLong, residual, iLBCenc_inst->state_short_len); + WebRtcSpl_MemSetW16(residualLong + iLBCenc_inst->state_short_len, 0, iLBCenc_inst->state_short_len); + + /* Run the Zero-Pole filter (Ciurcular convolution) */ + WebRtcSpl_MemSetW16(residualLongVec, 0, LPC_FILTERORDER); + WebRtcSpl_FilterMAFastQ12(residualLong, sampleMa, numerator, + LPC_FILTERORDER + 1, + iLBCenc_inst->state_short_len + LPC_FILTERORDER); + WebRtcSpl_MemSetW16(&sampleMa[iLBCenc_inst->state_short_len + LPC_FILTERORDER], 0, iLBCenc_inst->state_short_len - LPC_FILTERORDER); + + WebRtcSpl_FilterARFastQ12( + sampleMa, sampleAr, + syntDenum, LPC_FILTERORDER+1, 2 * iLBCenc_inst->state_short_len); + + for(k=0;k<iLBCenc_inst->state_short_len;k++){ + sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len]; + } + + /* Find maximum absolute value in the vector */ + maxVal=WebRtcSpl_MaxAbsValueW16(sampleAr, iLBCenc_inst->state_short_len); + + /* Find the best index */ + + if ((((int32_t)maxVal)<<scaleRes)<23170) { + maxValsq=((int32_t)maxVal*maxVal)<<(2+2*scaleRes); + } else { + maxValsq=(int32_t)WEBRTC_SPL_WORD32_MAX; + } + + index=0; + for (i=0;i<63;i++) { + + if (maxValsq>=WebRtcIlbcfix_kChooseFrgQuant[i]) { + index=i+1; + } else { + i=63; + } + } + iLBC_encbits->idxForMax=index; + + /* Rescale the vector before quantization */ + scale=WebRtcIlbcfix_kScale[index]; + + if (index<27) { /* scale table is in Q16, fout[] is in Q(-1) and we want the result to be in Q11 */ + shift=4; + } else { /* scale table is in Q21, fout[] is in Q(-1) and we want the result to be in Q11 */ + shift=9; + } + + /* Set up vectors for AbsQuant and rescale it with the scale factor */ + WebRtcSpl_ScaleVectorWithSat(sampleAr, sampleAr, scale, + iLBCenc_inst->state_short_len, (int16_t)(shift-scaleRes)); + + /* Quantize the values in fout[] */ + WebRtcIlbcfix_AbsQuant(iLBCenc_inst, iLBC_encbits, sampleAr, weightDenum); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_search.h b/webrtc/modules/audio_coding/codecs/ilbc/state_search.h new file mode 100644 index 0000000000..800beac09b --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/state_search.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_StateSearch.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * encoding of start state + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_StateSearch( + IlbcEncoder *iLBCenc_inst, + /* (i) Encoder instance */ + iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax + and idxVec, input state_first) */ + int16_t *residual, /* (i) target residual vector */ + int16_t *syntDenum, /* (i) lpc synthesis filter */ + int16_t *weightDenum /* (i) weighting filter denuminator */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c b/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c new file mode 100644 index 0000000000..b795e56ac4 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SwapBytes.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Swap bytes (to simplify operations on Little Endian machines) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SwapBytes( + const uint16_t* input, /* (i) the sequence to swap */ + size_t wordLength, /* (i) number or uint16_t to swap */ + uint16_t* output /* (o) the swapped sequence */ + ) { + size_t k; + for (k = wordLength; k > 0; k--) { + *output++ = (*input >> 8)|(*input << 8); + input++; + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h b/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h new file mode 100644 index 0000000000..a4484d621e --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_SwapBytes.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * Swap bytes (to simplify operations on Little Endian machines) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_SwapBytes( + const uint16_t* input, /* (i) the sequence to swap */ + size_t wordLength, /* (i) number or uint16_t to swap */ + uint16_t* output /* (o) the swapped sequence */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c new file mode 100644 index 0000000000..1199c816d8 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c @@ -0,0 +1,240 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + iLBC_test.c + +******************************************************************/ + +#include <stdlib.h> +#include <stdio.h> +#include <string.h> +#include "ilbc.h" + +/*---------------------------------------------------------------* + * Main program to test iLBC encoding and decoding + * + * Usage: + * exefile_name.exe <infile> <bytefile> <outfile> <channel> + * + * <infile> : Input file, speech for encoder (16-bit pcm file) + * <bytefile> : Bit stream output from the encoder + * <outfile> : Output file, decoded speech (16-bit pcm file) + * <channel> : Bit error file, optional (16-bit) + * 1 - Packet received correctly + * 0 - Packet Lost + * + *--------------------------------------------------------------*/ + +#define BLOCKL_MAX 240 +#define ILBCNOOFWORDS_MAX 25 + + +int main(int argc, char* argv[]) +{ + + FILE *ifileid,*efileid,*ofileid, *cfileid; + int16_t data[BLOCKL_MAX]; + uint8_t encoded_data[2 * ILBCNOOFWORDS_MAX]; + int16_t decoded_data[BLOCKL_MAX]; + int len_int, mode; + short pli; + int blockcount = 0; + int packetlosscount = 0; + size_t frameLen, len, len_i16s; + int16_t speechType; + IlbcEncoderInstance *Enc_Inst; + IlbcDecoderInstance *Dec_Inst; + +#ifdef __ILBC_WITH_40BITACC + /* Doublecheck that long long exists */ + if (sizeof(long)>=sizeof(long long)) { + fprintf(stderr, "40-bit simulation is not be supported on this platform\n"); + exit(0); + } +#endif + + /* get arguments and open files */ + + if ((argc!=5) && (argc!=6)) { + fprintf(stderr, + "\n*-----------------------------------------------*\n"); + fprintf(stderr, + " %s <20,30> input encoded decoded (channel)\n\n", + argv[0]); + fprintf(stderr, + " mode : Frame size for the encoding/decoding\n"); + fprintf(stderr, + " 20 - 20 ms\n"); + fprintf(stderr, + " 30 - 30 ms\n"); + fprintf(stderr, + " input : Speech for encoder (16-bit pcm file)\n"); + fprintf(stderr, + " encoded : Encoded bit stream\n"); + fprintf(stderr, + " decoded : Decoded speech (16-bit pcm file)\n"); + fprintf(stderr, + " channel : Packet loss pattern, optional (16-bit)\n"); + fprintf(stderr, + " 1 - Packet received correctly\n"); + fprintf(stderr, + " 0 - Packet Lost\n"); + fprintf(stderr, + "*-----------------------------------------------*\n\n"); + exit(1); + } + mode=atoi(argv[1]); + if (mode != 20 && mode != 30) { + fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", + argv[1]); + exit(2); + } + if ( (ifileid=fopen(argv[2],"rb")) == NULL) { + fprintf(stderr,"Cannot open input file %s\n", argv[2]); + exit(2);} + if ( (efileid=fopen(argv[3],"wb")) == NULL) { + fprintf(stderr, "Cannot open encoded file file %s\n", + argv[3]); exit(1);} + if ( (ofileid=fopen(argv[4],"wb")) == NULL) { + fprintf(stderr, "Cannot open decoded file %s\n", + argv[4]); exit(1);} + if (argc==6) { + if( (cfileid=fopen(argv[5],"rb")) == NULL) { + fprintf(stderr, "Cannot open channel file %s\n", + argv[5]); + exit(1); + } + } else { + cfileid=NULL; + } + + /* print info */ + + fprintf(stderr, "\n"); + fprintf(stderr, + "*---------------------------------------------------*\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "* iLBC test program *\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "*---------------------------------------------------*\n"); + fprintf(stderr,"\nMode : %2d ms\n", mode); + fprintf(stderr,"Input file : %s\n", argv[2]); + fprintf(stderr,"Encoded file : %s\n", argv[3]); + fprintf(stderr,"Output file : %s\n", argv[4]); + if (argc==6) { + fprintf(stderr,"Channel file : %s\n", argv[5]); + } + fprintf(stderr,"\n"); + + /* Create structs */ + WebRtcIlbcfix_EncoderCreate(&Enc_Inst); + WebRtcIlbcfix_DecoderCreate(&Dec_Inst); + + + /* Initialization */ + + WebRtcIlbcfix_EncoderInit(Enc_Inst, mode); + WebRtcIlbcfix_DecoderInit(Dec_Inst, mode); + frameLen = (size_t)(mode*8); + + /* loop over input blocks */ + + while (fread(data,sizeof(int16_t),frameLen,ifileid) == frameLen) { + + blockcount++; + + /* encoding */ + + fprintf(stderr, "--- Encoding block %i --- ",blockcount); + len_int = WebRtcIlbcfix_Encode(Enc_Inst, data, frameLen, encoded_data); + if (len_int < 0) { + fprintf(stderr, "Error encoding\n"); + exit(0); + } + len = (size_t)len_int; + fprintf(stderr, "\r"); + + /* write byte file */ + + len_i16s = (len + 1) / sizeof(int16_t); + if (fwrite(encoded_data, sizeof(int16_t), len_i16s, efileid) != len_i16s) { + return -1; + } + + /* get channel data if provided */ + if (argc==6) { + if (fread(&pli, sizeof(int16_t), 1, cfileid)) { + if ((pli!=0)&&(pli!=1)) { + fprintf(stderr, "Error in channel file\n"); + exit(0); + } + if (pli==0) { + /* Packet loss -> remove info from frame */ + memset(encoded_data, 0, + sizeof(int16_t)*ILBCNOOFWORDS_MAX); + packetlosscount++; + } + } else { + fprintf(stderr, "Error. Channel file too short\n"); + exit(0); + } + } else { + pli=1; + } + + /* decoding */ + + fprintf(stderr, "--- Decoding block %i --- ",blockcount); + if (pli==1) { + len_int=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, + len, decoded_data,&speechType); + if (len_int < 0) { + fprintf(stderr, "Error decoding\n"); + exit(0); + } + len = (size_t)len_int; + } else { + len=WebRtcIlbcfix_DecodePlc(Dec_Inst, decoded_data, 1); + } + fprintf(stderr, "\r"); + + /* write output file */ + + if (fwrite(decoded_data, sizeof(int16_t), len, ofileid) != len) { + return -1; + } + } + + /* close files */ + + fclose(ifileid); fclose(efileid); fclose(ofileid); + if (argc==6) { + fclose(cfileid); + } + + /* Free structs */ + WebRtcIlbcfix_EncoderFree(Enc_Inst); + WebRtcIlbcfix_DecoderFree(Dec_Inst); + + + printf("\nDone with simulation\n\n"); + + return(0); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c new file mode 100644 index 0000000000..f14192c2ae --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c @@ -0,0 +1,215 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + +iLBC Speech Coder ANSI-C Source Code + +iLBC_test.c + +******************************************************************/ + +#include <math.h> +#include <stdlib.h> +#include <stdio.h> +#include <string.h> +#include <time.h> +#include "ilbc.h" + +//#define JUNK_DATA +#ifdef JUNK_DATA +#define SEED_FILE "randseed.txt" +#endif + + +/*----------------------------------------------------------------* +* Main program to test iLBC encoding and decoding +* +* Usage: +* exefile_name.exe <infile> <bytefile> <outfile> +* +*---------------------------------------------------------------*/ + +int main(int argc, char* argv[]) +{ + FILE *ifileid,*efileid,*ofileid, *chfileid; + short encoded_data[55], data[240], speechType; + int len_int, mode; + short pli; + size_t len, readlen; + int blockcount = 0; + + IlbcEncoderInstance *Enc_Inst; + IlbcDecoderInstance *Dec_Inst; +#ifdef JUNK_DATA + size_t i; + FILE *seedfile; + unsigned int random_seed = (unsigned int) time(NULL);//1196764538 +#endif + + /* Create structs */ + WebRtcIlbcfix_EncoderCreate(&Enc_Inst); + WebRtcIlbcfix_DecoderCreate(&Dec_Inst); + + /* get arguments and open files */ + + if (argc != 6 ) { + fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n", + argv[0]); + fprintf(stderr, "Example:\n"); + fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]); + exit(1); + } + mode=atoi(argv[1]); + if (mode != 20 && mode != 30) { + fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]); + exit(2); + } + if ( (ifileid=fopen(argv[2],"rb")) == NULL) { + fprintf(stderr,"Cannot open input file %s\n", argv[2]); + exit(2);} + if ( (efileid=fopen(argv[3],"wb")) == NULL) { + fprintf(stderr, "Cannot open channelfile file %s\n", + argv[3]); exit(3);} + if( (ofileid=fopen(argv[4],"wb")) == NULL) { + fprintf(stderr, "Cannot open output file %s\n", + argv[4]); exit(3);} + if ( (chfileid=fopen(argv[5],"rb")) == NULL) { + fprintf(stderr,"Cannot open channel file file %s\n", argv[5]); + exit(2); + } + /* print info */ + fprintf(stderr, "\n"); + fprintf(stderr, + "*---------------------------------------------------*\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "* iLBCtest *\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "*---------------------------------------------------*\n"); +#ifdef SPLIT_10MS + fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode); +#else + fprintf(stderr,"\nMode : %2d ms\n", mode); +#endif + fprintf(stderr,"\nInput file : %s\n", argv[2]); + fprintf(stderr,"Coded file : %s\n", argv[3]); + fprintf(stderr,"Output file : %s\n\n", argv[4]); + fprintf(stderr,"Channel file : %s\n\n", argv[5]); + +#ifdef JUNK_DATA + srand(random_seed); + + if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) { + fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE); + } + else { + fprintf(seedfile, "%u\n", random_seed); + fclose(seedfile); + } +#endif + + /* Initialization */ + WebRtcIlbcfix_EncoderInit(Enc_Inst, mode); + WebRtcIlbcfix_DecoderInit(Dec_Inst, mode); + + /* loop over input blocks */ +#ifdef SPLIT_10MS + readlen = 80; +#else + readlen = (size_t)(mode << 3); +#endif + while(fread(data, sizeof(short), readlen, ifileid) == readlen) { + blockcount++; + + /* encoding */ + fprintf(stderr, "--- Encoding block %i --- ",blockcount); + len_int=WebRtcIlbcfix_Encode(Enc_Inst, data, readlen, encoded_data); + if (len_int < 0) { + fprintf(stderr, "Error encoding\n"); + exit(0); + } + len = (size_t)len_int; + fprintf(stderr, "\r"); + +#ifdef JUNK_DATA + for ( i = 0; i < len; i++) { + encoded_data[i] = (short) (encoded_data[i] + (short) rand()); + } +#endif + /* write byte file */ + if(len != 0){ //len may be 0 in 10ms split case + fwrite(encoded_data,1,len,efileid); + + /* get channel data if provided */ + if (argc==6) { + if (fread(&pli, sizeof(int16_t), 1, chfileid)) { + if ((pli!=0)&&(pli!=1)) { + fprintf(stderr, "Error in channel file\n"); + exit(0); + } + if (pli==0) { + /* Packet loss -> remove info from frame */ + memset(encoded_data, 0, sizeof(int16_t)*25); + } + } else { + fprintf(stderr, "Error. Channel file too short\n"); + exit(0); + } + } else { + pli=1; + } + + /* decoding */ + fprintf(stderr, "--- Decoding block %i --- ",blockcount); + if (pli==1) { + len_int = WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data, + &speechType); + if (len_int < 0) { + fprintf(stderr, "Error decoding\n"); + exit(0); + } + len = (size_t)len_int; + } else { + len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1); + } + fprintf(stderr, "\r"); + + /* write output file */ + fwrite(data,sizeof(short),len,ofileid); + } + } + +#ifdef JUNK_DATA + if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) { + fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE); + } + else { + fprintf(seedfile, "ok\n\n"); + fclose(seedfile); + } +#endif + + /* free structs */ + WebRtcIlbcfix_EncoderFree(Enc_Inst); + WebRtcIlbcfix_DecoderFree(Dec_Inst); + + /* close files */ + fclose(ifileid); + fclose(efileid); + fclose(ofileid); + + return 0; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c new file mode 100644 index 0000000000..303ede3e63 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c @@ -0,0 +1,343 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + iLBC_test.c + +******************************************************************/ + +#include <math.h> +#include <stdlib.h> +#include <stdio.h> +#include <string.h> + +#include "defines.h" +#include "nit_encode.h" +#include "encode.h" +#include "init_decode.h" +#include "decode.h" +#include "constants.h" +#include "ilbc.h" + +#define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2 + +/* Runtime statistics */ +#include <time.h> +/* #define CLOCKS_PER_SEC 1000 */ + +/*----------------------------------------------------------------* + * Encoder interface function + *---------------------------------------------------------------*/ + +short encode( /* (o) Number of bytes encoded */ + IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */ + int16_t *encoded_data, /* (o) The encoded bytes */ + int16_t *data /* (i) The signal block to encode */ + ){ + + /* do the actual encoding */ + WebRtcIlbcfix_Encode((uint16_t *)encoded_data, data, iLBCenc_inst); + + return (iLBCenc_inst->no_of_bytes); +} + +/*----------------------------------------------------------------* + * Decoder interface function + *---------------------------------------------------------------*/ + +short decode( /* (o) Number of decoded samples */ + IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */ + short *decoded_data, /* (o) Decoded signal block */ + short *encoded_data, /* (i) Encoded bytes */ + short mode /* (i) 0=PL, 1=Normal */ + ){ + + /* check if mode is valid */ + + if (mode<0 || mode>1) { + printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);} + + /* do actual decoding of block */ + + WebRtcIlbcfix_Decode(decoded_data, (uint16_t *)encoded_data, + iLBCdec_inst, mode); + + return (iLBCdec_inst->blockl); +} + +/*----------------------------------------------------------------* + * Main program to test iLBC encoding and decoding + * + * Usage: + * exefile_name.exe <infile> <bytefile> <outfile> <channelfile> + * + *---------------------------------------------------------------*/ + +#define MAXFRAMES 10000 +#define MAXFILELEN (BLOCKL_MAX*MAXFRAMES) + +int main(int argc, char* argv[]) +{ + + /* Runtime statistics */ + + float starttime1, starttime2; + float runtime1, runtime2; + float outtime; + + FILE *ifileid,*efileid,*ofileid, *chfileid; + short *inputdata, *encodeddata, *decodeddata; + short *channeldata; + int blockcount = 0, noOfBlocks=0, i, noOfLostBlocks=0; + short mode; + IlbcEncoder Enc_Inst; + IlbcDecoder Dec_Inst; + + short frameLen; + short count; +#ifdef SPLIT_10MS + short size; +#endif + + inputdata=(short*) malloc(MAXFILELEN*sizeof(short)); + if (inputdata==NULL) { + fprintf(stderr,"Could not allocate memory for vector\n"); + exit(0); + } + encodeddata=(short*) malloc(ILBCNOOFWORDS_MAX*MAXFRAMES*sizeof(short)); + if (encodeddata==NULL) { + fprintf(stderr,"Could not allocate memory for vector\n"); + free(inputdata); + exit(0); + } + decodeddata=(short*) malloc(MAXFILELEN*sizeof(short)); + if (decodeddata==NULL) { + fprintf(stderr,"Could not allocate memory for vector\n"); + free(inputdata); + free(encodeddata); + exit(0); + } + channeldata=(short*) malloc(MAXFRAMES*sizeof(short)); + if (channeldata==NULL) { + fprintf(stderr,"Could not allocate memory for vector\n"); + free(inputdata); + free(encodeddata); + free(decodeddata); + exit(0); + } + + /* get arguments and open files */ + + if (argc != 6 ) { + fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n", + argv[0]); + fprintf(stderr, "Example:\n"); + fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]); + exit(1); + } + mode=atoi(argv[1]); + if (mode != 20 && mode != 30) { + fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]); + exit(2); + } + if ( (ifileid=fopen(argv[2],"rb")) == NULL) { + fprintf(stderr,"Cannot open input file %s\n", argv[2]); + exit(2);} + if ( (efileid=fopen(argv[3],"wb")) == NULL) { + fprintf(stderr, "Cannot open channelfile file %s\n", + argv[3]); exit(3);} + if( (ofileid=fopen(argv[4],"wb")) == NULL) { + fprintf(stderr, "Cannot open output file %s\n", + argv[4]); exit(3);} + if ( (chfileid=fopen(argv[5],"rb")) == NULL) { + fprintf(stderr,"Cannot open channel file file %s\n", argv[5]); + exit(2);} + + + /* print info */ +#ifndef PRINT_MIPS + fprintf(stderr, "\n"); + fprintf(stderr, + "*---------------------------------------------------*\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "* iLBCtest *\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "* *\n"); + fprintf(stderr, + "*---------------------------------------------------*\n"); +#ifdef SPLIT_10MS + fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode); +#else + fprintf(stderr,"\nMode : %2d ms\n", mode); +#endif + fprintf(stderr,"\nInput file : %s\n", argv[2]); + fprintf(stderr,"Coded file : %s\n", argv[3]); + fprintf(stderr,"Output file : %s\n\n", argv[4]); + fprintf(stderr,"Channel file : %s\n\n", argv[5]); +#endif + + /* Initialization */ + + WebRtcIlbcfix_EncoderInit(&Enc_Inst, mode); + WebRtcIlbcfix_DecoderInit(&Dec_Inst, mode, 1); + + /* extract the input file and channel file */ + +#ifdef SPLIT_10MS + frameLen = (mode==20)? 80:160; + fread(Enc_Inst.past_samples, sizeof(short), frameLen, ifileid); + Enc_Inst.section = 0; + + while( fread(&inputdata[noOfBlocks*80], sizeof(short), + 80, ifileid) == 80 ) { + noOfBlocks++; + } + + noOfBlocks += frameLen/80; + frameLen = 80; +#else + frameLen = Enc_Inst.blockl; + + while( fread(&inputdata[noOfBlocks*Enc_Inst.blockl],sizeof(short), + Enc_Inst.blockl,ifileid)==(uint16_t)Enc_Inst.blockl){ + noOfBlocks++; + } +#endif + + + while ((fread(&channeldata[blockcount],sizeof(short), 1,chfileid)==1) + && ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) )) { + blockcount++; + } + + if ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) ) { + fprintf(stderr,"Channel file %s is too short\n", argv[4]); + free(inputdata); + free(encodeddata); + free(decodeddata); + free(channeldata); + exit(0); + } + + count=0; + + /* Runtime statistics */ + + starttime1 = clock()/(float)CLOCKS_PER_SEC; + + /* Encoding loop */ +#ifdef PRINT_MIPS + printf("-1 -1\n"); +#endif + +#ifdef SPLIT_10MS + /* "Enc_Inst.section != 0" is to make sure we run through full + lengths of all vectors for 10ms split mode. + */ + // while( (count < noOfBlocks) || (Enc_Inst.section != 0) ) { + while( count < blockcount * (Enc_Inst.blockl/frameLen) ) { + + encode(&Enc_Inst, &encodeddata[Enc_Inst.no_of_words * + (count/(Enc_Inst.nsub/2))], + &inputdata[frameLen * count] ); +#else + while (count < noOfBlocks) { + encode( &Enc_Inst, &encodeddata[Enc_Inst.no_of_words * count], + &inputdata[frameLen * count] ); +#endif + +#ifdef PRINT_MIPS + printf("-1 -1\n"); +#endif + + count++; + } + + count=0; + + /* Runtime statistics */ + + starttime2=clock()/(float)CLOCKS_PER_SEC; + runtime1 = (float)(starttime2-starttime1); + + /* Decoding loop */ + + while (count < blockcount) { + if (channeldata[count]==1) { + /* Normal decoding */ + decode(&Dec_Inst, &decodeddata[count * Dec_Inst.blockl], + &encodeddata[Dec_Inst.no_of_words * count], 1); + } else if (channeldata[count]==0) { + /* PLC */ + short emptydata[ILBCNOOFWORDS_MAX]; + memset(emptydata, 0, Dec_Inst.no_of_words*sizeof(short)); + decode(&Dec_Inst, &decodeddata[count*Dec_Inst.blockl], + emptydata, 0); + noOfLostBlocks++; + } else { + printf("Error in channel file (values have to be either 1 or 0)\n"); + exit(0); + } +#ifdef PRINT_MIPS + printf("-1 -1\n"); +#endif + + count++; + } + + /* Runtime statistics */ + + runtime2 = (float)(clock()/(float)CLOCKS_PER_SEC-starttime2); + + outtime = (float)((float)blockcount* + (float)mode/1000.0); + +#ifndef PRINT_MIPS + printf("\nLength of speech file: %.1f s\n", outtime); + printf("Lost frames : %.1f%%\n\n", 100*(float)noOfLostBlocks/(float)blockcount); + + printf("Time to run iLBC_encode+iLBC_decode:"); + printf(" %.1f s (%.1f%% of realtime)\n", runtime1+runtime2, + (100*(runtime1+runtime2)/outtime)); + + printf("Time in iLBC_encode :"); + printf(" %.1f s (%.1f%% of total runtime)\n", + runtime1, 100.0*runtime1/(runtime1+runtime2)); + + printf("Time in iLBC_decode :"); + printf(" %.1f s (%.1f%% of total runtime)\n\n", + runtime2, 100.0*runtime2/(runtime1+runtime2)); +#endif + + /* Write data to files */ + for (i=0; i<blockcount; i++) { + fwrite(&encodeddata[i*Enc_Inst.no_of_words], sizeof(short), + Enc_Inst.no_of_words, efileid); + } + for (i=0;i<blockcount;i++) { + fwrite(&decodeddata[i*Enc_Inst.blockl],sizeof(short),Enc_Inst.blockl,ofileid); + } + + /* return memory and close files */ + + free(inputdata); + free(encodeddata); + free(decodeddata); + free(channeldata); + fclose(ifileid); fclose(efileid); fclose(ofileid); + return(0); + } diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt new file mode 100644 index 0000000000..db0e9a0979 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt @@ -0,0 +1,73 @@ +#!/bin/bash +(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character + +# +# This script can be used to verify the bit exactness of iLBC fixed-point version 1.0.6 +# + +INP=../../../../../../../resources/audio_coding +EXEP=../../../../../../../out/Release +OUTP=./GeneratedFiles +mkdir ./GeneratedFiles + +$EXEP/iLBCtest 20 $INP/F00.INP $OUTP/F00.BIT20 $OUTP/F00.OUT20 $INP/clean.chn +$EXEP/iLBCtest 20 $INP/F01.INP $OUTP/F01.BIT20 $OUTP/F01.OUT20 $INP/clean.chn +$EXEP/iLBCtest 20 $INP/F02.INP $OUTP/F02.BIT20 $OUTP/F02.OUT20 $INP/clean.chn +$EXEP/iLBCtest 20 $INP/F03.INP $OUTP/F03.BIT20 $OUTP/F03.OUT20 $INP/clean.chn +$EXEP/iLBCtest 20 $INP/F04.INP $OUTP/F04.BIT20 $OUTP/F04.OUT20 $INP/clean.chn +$EXEP/iLBCtest 20 $INP/F05.INP $OUTP/F05.BIT20 $OUTP/F05.OUT20 $INP/clean.chn +$EXEP/iLBCtest 20 $INP/F06.INP $OUTP/F06.BIT20 $OUTP/F06.OUT20 $INP/clean.chn + +$EXEP/iLBCtest 30 $INP/F00.INP $OUTP/F00.BIT30 $OUTP/F00.OUT30 $INP/clean.chn +$EXEP/iLBCtest 30 $INP/F01.INP $OUTP/F01.BIT30 $OUTP/F01.OUT30 $INP/clean.chn +$EXEP/iLBCtest 30 $INP/F02.INP $OUTP/F02.BIT30 $OUTP/F02.OUT30 $INP/clean.chn +$EXEP/iLBCtest 30 $INP/F03.INP $OUTP/F03.BIT30 $OUTP/F03.OUT30 $INP/clean.chn +$EXEP/iLBCtest 30 $INP/F04.INP $OUTP/F04.BIT30 $OUTP/F04.OUT30 $INP/clean.chn +$EXEP/iLBCtest 30 $INP/F05.INP $OUTP/F05.BIT30 $OUTP/F05.OUT30 $INP/clean.chn +$EXEP/iLBCtest 30 $INP/F06.INP $OUTP/F06.BIT30 $OUTP/F06.OUT30 $INP/clean.chn + +$EXEP/iLBCtest 20 $INP/F00.INP $OUTP/F00.BIT20 $OUTP/F00_tlm10.OUT20 $INP/tlm10.chn +$EXEP/iLBCtest 20 $INP/F01.INP $OUTP/F01.BIT20 $OUTP/F01_tlm10.OUT20 $INP/tlm10.chn +$EXEP/iLBCtest 20 $INP/F02.INP $OUTP/F02.BIT20 $OUTP/F02_tlm10.OUT20 $INP/tlm10.chn +$EXEP/iLBCtest 30 $INP/F00.INP $OUTP/F00.BIT30 $OUTP/F00_tlm10.OUT30 $INP/tlm10.chn +$EXEP/iLBCtest 30 $INP/F01.INP $OUTP/F01.BIT30 $OUTP/F01_tlm10.OUT30 $INP/tlm10.chn +$EXEP/iLBCtest 30 $INP/F02.INP $OUTP/F02.BIT30 $OUTP/F02_tlm10.OUT30 $INP/tlm10.chn + + +diff $OUTP/F00.BIT20 $INP/F00.BIT20 +diff $OUTP/F01.BIT20 $INP/F01.BIT20 +diff $OUTP/F02.BIT20 $INP/F02.BIT20 +diff $OUTP/F03.BIT20 $INP/F03.BIT20 +diff $OUTP/F04.BIT20 $INP/F04.BIT20 +diff $OUTP/F05.BIT20 $INP/F05.BIT20 +diff $OUTP/F06.BIT20 $INP/F06.BIT20 +diff $OUTP/F00.OUT20 $INP/F00.OUT20 +diff $OUTP/F01.OUT20 $INP/F01.OUT20 +diff $OUTP/F02.OUT20 $INP/F02.OUT20 +diff $OUTP/F03.OUT20 $INP/F03.OUT20 +diff $OUTP/F04.OUT20 $INP/F04.OUT20 +diff $OUTP/F05.OUT20 $INP/F05.OUT20 +diff $OUTP/F06.OUT20 $INP/F06.OUT20 + +diff $OUTP/F00.BIT30 $INP/F00.BIT30 +diff $OUTP/F01.BIT30 $INP/F01.BIT30 +diff $OUTP/F02.BIT30 $INP/F02.BIT30 +diff $OUTP/F03.BIT30 $INP/F03.BIT30 +diff $OUTP/F04.BIT30 $INP/F04.BIT30 +diff $OUTP/F05.BIT30 $INP/F05.BIT30 +diff $OUTP/F06.BIT30 $INP/F06.BIT30 +diff $OUTP/F00.OUT30 $INP/F00.OUT30 +diff $OUTP/F01.OUT30 $INP/F01.OUT30 +diff $OUTP/F02.OUT30 $INP/F02.OUT30 +diff $OUTP/F03.OUT30 $INP/F03.OUT30 +diff $OUTP/F04.OUT30 $INP/F04.OUT30 +diff $OUTP/F05.OUT30 $INP/F05.OUT30 +diff $OUTP/F06.OUT30 $INP/F06.OUT30 + +diff $OUTP/F00_tlm10.OUT20 $INP/F00_tlm10.OUT20 +diff $OUTP/F01_tlm10.OUT20 $INP/F01_tlm10.OUT20 +diff $OUTP/F02_tlm10.OUT20 $INP/F02_tlm10.OUT20 +diff $OUTP/F00_tlm10.OUT30 $INP/F00_tlm10.OUT30 +diff $OUTP/F01_tlm10.OUT30 $INP/F01_tlm10.OUT30 +diff $OUTP/F02_tlm10.OUT30 $INP/F02_tlm10.OUT30 + diff --git a/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c b/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c new file mode 100644 index 0000000000..fae06f6339 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c @@ -0,0 +1,239 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_UnpackBits.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * unpacking of bits from bitstream, i.e., vector of bytes + *---------------------------------------------------------------*/ + +int16_t WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */ + const uint16_t *bitstream, /* (i) The packatized bitstream */ + iLBC_bits *enc_bits, /* (o) Paramerers from bitstream */ + int16_t mode /* (i) Codec mode (20 or 30) */ + ) { + const uint16_t *bitstreamPtr; + int i, k; + int16_t *tmpPtr; + + bitstreamPtr=bitstream; + + /* First int16_t */ + enc_bits->lsf[0] = (*bitstreamPtr)>>10; /* Bit 0..5 */ + enc_bits->lsf[1] = ((*bitstreamPtr)>>3)&0x7F; /* Bit 6..12 */ + enc_bits->lsf[2] = ((*bitstreamPtr)&0x7)<<4; /* Bit 13..15 */ + bitstreamPtr++; + /* Second int16_t */ + enc_bits->lsf[2] |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3 */ + + if (mode==20) { + enc_bits->startIdx = ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5 */ + enc_bits->state_first = ((*bitstreamPtr)>>9)&0x1; /* Bit 6 */ + enc_bits->idxForMax = ((*bitstreamPtr)>>3)&0x3F; /* Bit 7..12 */ + enc_bits->cb_index[0] = ((*bitstreamPtr)&0x7)<<4; /* Bit 13..15 */ + bitstreamPtr++; + /* Third int16_t */ + enc_bits->cb_index[0] |= ((*bitstreamPtr)>>12)&0xE; /* Bit 0..2 */ + enc_bits->gain_index[0] = ((*bitstreamPtr)>>8)&0x18; /* Bit 3..4 */ + enc_bits->gain_index[1] = ((*bitstreamPtr)>>7)&0x8; /* Bit 5 */ + enc_bits->cb_index[3] = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */ + enc_bits->gain_index[3] = ((*bitstreamPtr)<<2)&0x10; /* Bit 13 */ + enc_bits->gain_index[4] = ((*bitstreamPtr)<<2)&0x8; /* Bit 14 */ + enc_bits->gain_index[6] = ((*bitstreamPtr)<<4)&0x10; /* Bit 15 */ + } else { /* mode==30 */ + enc_bits->lsf[3] = ((*bitstreamPtr)>>6)&0x3F; /* Bit 4..9 */ + enc_bits->lsf[4] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */ + bitstreamPtr++; + /* Third int16_t */ + enc_bits->lsf[4] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */ + enc_bits->lsf[5] = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7 */ + enc_bits->startIdx = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */ + enc_bits->state_first = ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */ + enc_bits->idxForMax = ((*bitstreamPtr)<<2)&0x3C; /* Bit 12..15 */ + bitstreamPtr++; + /* 4:th int16_t */ + enc_bits->idxForMax |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */ + enc_bits->cb_index[0] = ((*bitstreamPtr)>>7)&0x78; /* Bit 2..5 */ + enc_bits->gain_index[0] = ((*bitstreamPtr)>>5)&0x10; /* Bit 6 */ + enc_bits->gain_index[1] = ((*bitstreamPtr)>>5)&0x8; /* Bit 7 */ + enc_bits->cb_index[3] = ((*bitstreamPtr))&0xFC; /* Bit 8..13 */ + enc_bits->gain_index[3] = ((*bitstreamPtr)<<3)&0x10; /* Bit 14 */ + enc_bits->gain_index[4] = ((*bitstreamPtr)<<3)&0x8; /* Bit 15 */ + } + /* Class 2 bits of ULP */ + /* 4:th to 6:th int16_t for 20 ms case + 5:th to 7:th int16_t for 30 ms case */ + bitstreamPtr++; + tmpPtr=enc_bits->idxVec; + for (k=0; k<3; k++) { + for (i=15; i>=0; i--) { + (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4; + /* Bit 15-i */ + tmpPtr++; + } + bitstreamPtr++; + } + + if (mode==20) { + /* 7:th int16_t */ + for (i=15; i>6; i--) { + (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4; + /* Bit 15-i */ + tmpPtr++; + } + enc_bits->gain_index[1] |= ((*bitstreamPtr)>>4)&0x4; /* Bit 9 */ + enc_bits->gain_index[3] |= ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */ + enc_bits->gain_index[4] |= ((*bitstreamPtr)>>1)&0x4; /* Bit 12 */ + enc_bits->gain_index[6] |= ((*bitstreamPtr)<<1)&0x8; /* Bit 13 */ + enc_bits->gain_index[7] = ((*bitstreamPtr)<<2)&0xC; /* Bit 14..15 */ + + } else { /* mode==30 */ + /* 8:th int16_t */ + for (i=15; i>5; i--) { + (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4; + /* Bit 15-i */ + tmpPtr++; + } + enc_bits->cb_index[0] |= ((*bitstreamPtr)>>3)&0x6; /* Bit 10..11 */ + enc_bits->gain_index[0] |= ((*bitstreamPtr))&0x8; /* Bit 12 */ + enc_bits->gain_index[1] |= ((*bitstreamPtr))&0x4; /* Bit 13 */ + enc_bits->cb_index[3] |= ((*bitstreamPtr))&0x2; /* Bit 14 */ + enc_bits->cb_index[6] = ((*bitstreamPtr)<<7)&0x80; /* Bit 15 */ + bitstreamPtr++; + /* 9:th int16_t */ + enc_bits->cb_index[6] |= ((*bitstreamPtr)>>9)&0x7E; /* Bit 0..5 */ + enc_bits->cb_index[9] = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */ + enc_bits->cb_index[12] = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */ + bitstreamPtr++; + /* 10:th int16_t */ + enc_bits->cb_index[12] |= ((*bitstreamPtr)>>11)&0x1E;/* Bit 0..3 */ + enc_bits->gain_index[3] |= ((*bitstreamPtr)>>8)&0xC; /* Bit 4..5 */ + enc_bits->gain_index[4] |= ((*bitstreamPtr)>>7)&0x6; /* Bit 6..7 */ + enc_bits->gain_index[6] = ((*bitstreamPtr)>>3)&0x18; /* Bit 8..9 */ + enc_bits->gain_index[7] = ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */ + enc_bits->gain_index[9] = ((*bitstreamPtr)<<1)&0x10; /* Bit 12 */ + enc_bits->gain_index[10] = ((*bitstreamPtr)<<1)&0x8; /* Bit 13 */ + enc_bits->gain_index[12] = ((*bitstreamPtr)<<3)&0x10; /* Bit 14 */ + enc_bits->gain_index[13] = ((*bitstreamPtr)<<3)&0x8; /* Bit 15 */ + } + bitstreamPtr++; + /* Class 3 bits of ULP */ + /* 8:th to 14:th int16_t for 20 ms case + 11:th to 17:th int16_t for 30 ms case */ + tmpPtr=enc_bits->idxVec; + for (k=0; k<7; k++) { + for (i=14; i>=0; i-=2) { + (*tmpPtr) |= ((*bitstreamPtr)>>i)&0x3; /* Bit 15-i..14-i*/ + tmpPtr++; + } + bitstreamPtr++; + } + + if (mode==20) { + /* 15:th int16_t */ + enc_bits->idxVec[56] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */ + enc_bits->cb_index[0] |= ((*bitstreamPtr)>>13)&0x1; /* Bit 2 */ + enc_bits->cb_index[1] = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9 */ + enc_bits->cb_index[2] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */ + bitstreamPtr++; + /* 16:th int16_t */ + enc_bits->cb_index[2] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */ + enc_bits->gain_index[0] |= ((*bitstreamPtr)>>12)&0x7; /* Bit 1..3 */ + enc_bits->gain_index[1] |= ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5 */ + enc_bits->gain_index[2] = ((*bitstreamPtr)>>7)&0x7; /* Bit 6..8 */ + enc_bits->cb_index[3] |= ((*bitstreamPtr)>>6)&0x1; /* Bit 9 */ + enc_bits->cb_index[4] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */ + bitstreamPtr++; + /* 17:th int16_t */ + enc_bits->cb_index[4] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */ + enc_bits->cb_index[5] = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7 */ + enc_bits->cb_index[6] = ((*bitstreamPtr))&0xFF; /* Bit 8..15 */ + bitstreamPtr++; + /* 18:th int16_t */ + enc_bits->cb_index[7] = (*bitstreamPtr)>>8; /* Bit 0..7 */ + enc_bits->cb_index[8] = (*bitstreamPtr)&0xFF; /* Bit 8..15 */ + bitstreamPtr++; + /* 19:th int16_t */ + enc_bits->gain_index[3] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */ + enc_bits->gain_index[4] |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3 */ + enc_bits->gain_index[5] = ((*bitstreamPtr)>>9)&0x7; /* Bit 4..6 */ + enc_bits->gain_index[6] |= ((*bitstreamPtr)>>6)&0x7; /* Bit 7..9 */ + enc_bits->gain_index[7] |= ((*bitstreamPtr)>>4)&0x3; /* Bit 10..11 */ + enc_bits->gain_index[8] = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */ + } else { /* mode==30 */ + /* 18:th int16_t */ + enc_bits->idxVec[56] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */ + enc_bits->idxVec[57] |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3 */ + enc_bits->cb_index[0] |= ((*bitstreamPtr)>>11)&1; /* Bit 4 */ + enc_bits->cb_index[1] = ((*bitstreamPtr)>>4)&0x7F; /* Bit 5..11 */ + enc_bits->cb_index[2] = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */ + bitstreamPtr++; + /* 19:th int16_t */ + enc_bits->cb_index[2] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */ + enc_bits->gain_index[0] |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5 */ + enc_bits->gain_index[1] |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7 */ + enc_bits->gain_index[2] = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */ + enc_bits->cb_index[3] |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */ + enc_bits->cb_index[4] = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */ + bitstreamPtr++; + /* 20:th int16_t */ + enc_bits->cb_index[4] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */ + enc_bits->cb_index[5] = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9 */ + enc_bits->cb_index[6] |= ((*bitstreamPtr)>>5)&0x1; /* Bit 10 */ + enc_bits->cb_index[7] = ((*bitstreamPtr)<<3)&0xF8; /* Bit 11..15 */ + bitstreamPtr++; + /* 21:st int16_t */ + enc_bits->cb_index[7] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */ + enc_bits->cb_index[8] = ((*bitstreamPtr)>>5)&0xFF; /* Bit 3..10 */ + enc_bits->cb_index[9] |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */ + enc_bits->cb_index[10] = ((*bitstreamPtr)<<4)&0xF0; /* Bit 12..15 */ + bitstreamPtr++; + /* 22:nd int16_t */ + enc_bits->cb_index[10] |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3 */ + enc_bits->cb_index[11] = ((*bitstreamPtr)>>4)&0xFF; /* Bit 4..11 */ + enc_bits->cb_index[12] |= ((*bitstreamPtr)>>3)&0x1; /* Bit 12 */ + enc_bits->cb_index[13] = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */ + bitstreamPtr++; + /* 23:rd int16_t */ + enc_bits->cb_index[13] |= ((*bitstreamPtr)>>11)&0x1F;/* Bit 0..4 */ + enc_bits->cb_index[14] = ((*bitstreamPtr)>>3)&0xFF; /* Bit 5..12 */ + enc_bits->gain_index[3] |= ((*bitstreamPtr)>>1)&0x3; /* Bit 13..14 */ + enc_bits->gain_index[4] |= ((*bitstreamPtr)&0x1); /* Bit 15 */ + bitstreamPtr++; + /* 24:rd int16_t */ + enc_bits->gain_index[5] = ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */ + enc_bits->gain_index[6] |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5 */ + enc_bits->gain_index[7] |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7 */ + enc_bits->gain_index[8] = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */ + enc_bits->gain_index[9] |= ((*bitstreamPtr)>>1)&0xF; /* Bit 11..14 */ + enc_bits->gain_index[10] |= ((*bitstreamPtr)<<2)&0x4; /* Bit 15 */ + bitstreamPtr++; + /* 25:rd int16_t */ + enc_bits->gain_index[10] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */ + enc_bits->gain_index[11] = ((*bitstreamPtr)>>11)&0x7; /* Bit 2..4 */ + enc_bits->gain_index[12] |= ((*bitstreamPtr)>>7)&0xF; /* Bit 5..8 */ + enc_bits->gain_index[13] |= ((*bitstreamPtr)>>4)&0x7; /* Bit 9..11 */ + enc_bits->gain_index[14] = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */ + } + /* Last bit should be zero, otherwise it's an "empty" frame */ + if (((*bitstreamPtr)&0x1) == 1) { + return(1); + } else { + return(0); + } +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h b/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h new file mode 100644 index 0000000000..9586a120d0 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_UnpackBits.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * unpacking of bits from bitstream, i.e., vector of bytes + *---------------------------------------------------------------*/ + +int16_t WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */ + const uint16_t *bitstream, /* (i) The packatized bitstream */ + iLBC_bits *enc_bits, /* (o) Paramerers from bitstream */ + int16_t mode /* (i) Codec mode (20 or 30) */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/vq3.c b/webrtc/modules/audio_coding/codecs/ilbc/vq3.c new file mode 100644 index 0000000000..ee3b26e039 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/vq3.c @@ -0,0 +1,63 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Vq3.c + +******************************************************************/ + +#include "vq3.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * vector quantization + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Vq3( + int16_t *Xq, /* quantized vector (Q13) */ + int16_t *index, + int16_t *CB, /* codebook in Q13 */ + int16_t *X, /* vector to quantize (Q13) */ + int16_t n_cb + ){ + int16_t i, j; + int16_t pos, minindex=0; + int16_t tmp; + int32_t dist, mindist; + + pos = 0; + mindist = WEBRTC_SPL_WORD32_MAX; /* start value */ + + /* Find the codebook with the lowest square distance */ + for (j = 0; j < n_cb; j++) { + tmp = X[0] - CB[pos]; + dist = tmp * tmp; + for (i = 1; i < 3; i++) { + tmp = X[i] - CB[pos + i]; + dist += tmp * tmp; + } + + if (dist < mindist) { + mindist = dist; + minindex = j; + } + pos += 3; + } + + /* Store the quantized codebook and the index */ + for (i = 0; i < 3; i++) { + Xq[i] = CB[minindex*3 + i]; + } + *index = minindex; + +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/vq3.h b/webrtc/modules/audio_coding/codecs/ilbc/vq3.h new file mode 100644 index 0000000000..b146ea3d85 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/vq3.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Vq3.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_ + +#include "webrtc/typedefs.h" + +/*----------------------------------------------------------------* + * Vector quantization of order 3 (based on MSE) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Vq3( + int16_t *Xq, /* (o) the quantized vector (Q13) */ + int16_t *index, /* (o) the quantization index */ + int16_t *CB, /* (i) the vector quantization codebook (Q13) */ + int16_t *X, /* (i) the vector to quantize (Q13) */ + int16_t n_cb /* (i) the number of vectors in the codebook */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/vq4.c b/webrtc/modules/audio_coding/codecs/ilbc/vq4.c new file mode 100644 index 0000000000..bc20952132 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/vq4.c @@ -0,0 +1,62 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Vq4.c + +******************************************************************/ + +#include "vq4.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * vector quantization + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Vq4( + int16_t *Xq, /* quantized vector (Q13) */ + int16_t *index, + int16_t *CB, /* codebook in Q13 */ + int16_t *X, /* vector to quantize (Q13) */ + int16_t n_cb + ){ + int16_t i, j; + int16_t pos, minindex=0; + int16_t tmp; + int32_t dist, mindist; + + pos = 0; + mindist = WEBRTC_SPL_WORD32_MAX; /* start value */ + + /* Find the codebook with the lowest square distance */ + for (j = 0; j < n_cb; j++) { + tmp = X[0] - CB[pos]; + dist = tmp * tmp; + for (i = 1; i < 4; i++) { + tmp = X[i] - CB[pos + i]; + dist += tmp * tmp; + } + + if (dist < mindist) { + mindist = dist; + minindex = j; + } + pos += 4; + } + + /* Store the quantized codebook and the index */ + for (i = 0; i < 4; i++) { + Xq[i] = CB[minindex*4 + i]; + } + *index = minindex; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/vq4.h b/webrtc/modules/audio_coding/codecs/ilbc/vq4.h new file mode 100644 index 0000000000..9263759367 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/vq4.h @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Vq4.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_ + +#include "webrtc/typedefs.h" + +/*----------------------------------------------------------------* + * Vector quantization of order 4 (based on MSE) + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Vq4( + int16_t *Xq, /* (o) the quantized vector (Q13) */ + int16_t *index, /* (o) the quantization index */ + int16_t *CB, /* (i) the vector quantization codebook (Q13) */ + int16_t *X, /* (i) the vector to quantize (Q13) */ + int16_t n_cb /* (i) the number of vectors in the codebook */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c b/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c new file mode 100644 index 0000000000..dc12a5a7c4 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c @@ -0,0 +1,62 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Window32W32.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * window multiplication + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Window32W32( + int32_t *z, /* Output */ + int32_t *x, /* Input (same domain as Output)*/ + const int32_t *y, /* Q31 Window */ + size_t N /* length to process */ + ) { + size_t i; + int16_t x_low, x_hi, y_low, y_hi; + int16_t left_shifts; + int32_t temp; + + left_shifts = (int16_t)WebRtcSpl_NormW32(x[0]); + WebRtcSpl_VectorBitShiftW32(x, N, x, (int16_t)(-left_shifts)); + + + /* The double precision numbers use a special representation: + * w32 = hi<<16 + lo<<1 + */ + for (i = 0; i < N; i++) { + /* Extract higher bytes */ + x_hi = (int16_t)(x[i] >> 16); + y_hi = (int16_t)(y[i] >> 16); + + /* Extract lower bytes, defined as (w32 - hi<<16)>>1 */ + x_low = (int16_t)((x[i] - (x_hi << 16)) >> 1); + + y_low = (int16_t)((y[i] - (y_hi << 16)) >> 1); + + /* Calculate z by a 32 bit multiplication using both low and high from x and y */ + temp = ((x_hi * y_hi) << 1) + ((x_hi * y_low) >> 14); + + z[i] = temp + ((x_low * y_hi) >> 14); + } + + WebRtcSpl_VectorBitShiftW32(z, N, z, left_shifts); + + return; +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h b/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h new file mode 100644 index 0000000000..27ed1b6a33 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h @@ -0,0 +1,35 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_Window32W32.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * window multiplication + *---------------------------------------------------------------*/ + +void WebRtcIlbcfix_Window32W32( + int32_t *z, /* Output */ + int32_t *x, /* Input (same domain as Output)*/ + const int32_t *y, /* Q31 Window */ + size_t N /* length to process */ + ); + +#endif diff --git a/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c new file mode 100644 index 0000000000..0d898c54a4 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c @@ -0,0 +1,140 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_XcorrCoef.c + +******************************************************************/ + +#include "defines.h" + +/*----------------------------------------------------------------* + * cross correlation which finds the optimal lag for the + * crossCorr*crossCorr/(energy) criteria + *---------------------------------------------------------------*/ + +size_t WebRtcIlbcfix_XcorrCoef( + int16_t *target, /* (i) first array */ + int16_t *regressor, /* (i) second array */ + size_t subl, /* (i) dimension arrays */ + size_t searchLen, /* (i) the search lenght */ + size_t offset, /* (i) samples offset between arrays */ + int16_t step /* (i) +1 or -1 */ + ){ + size_t k; + size_t maxlag; + int16_t pos; + int16_t max; + int16_t crossCorrScale, Energyscale; + int16_t crossCorrSqMod, crossCorrSqMod_Max; + int32_t crossCorr, Energy; + int16_t crossCorrmod, EnergyMod, EnergyMod_Max; + int16_t *tp, *rp; + int16_t *rp_beg, *rp_end; + int16_t totscale, totscale_max; + int16_t scalediff; + int32_t newCrit, maxCrit; + int shifts; + + /* Initializations, to make sure that the first one is selected */ + crossCorrSqMod_Max=0; + EnergyMod_Max=WEBRTC_SPL_WORD16_MAX; + totscale_max=-500; + maxlag=0; + pos=0; + + /* Find scale value and start position */ + if (step==1) { + max=WebRtcSpl_MaxAbsValueW16(regressor, subl + searchLen - 1); + rp_beg = regressor; + rp_end = regressor + subl; + } else { /* step==-1 */ + max = WebRtcSpl_MaxAbsValueW16(regressor - searchLen, subl + searchLen - 1); + rp_beg = regressor - 1; + rp_end = regressor + subl - 1; + } + + /* Introduce a scale factor on the Energy in int32_t in + order to make sure that the calculation does not + overflow */ + + if (max>5000) { + shifts=2; + } else { + shifts=0; + } + + /* Calculate the first energy, then do a +/- to get the other energies */ + Energy=WebRtcSpl_DotProductWithScale(regressor, regressor, subl, shifts); + + for (k=0;k<searchLen;k++) { + tp = target; + rp = ®ressor[pos]; + + crossCorr=WebRtcSpl_DotProductWithScale(tp, rp, subl, shifts); + + if ((Energy>0)&&(crossCorr>0)) { + + /* Put cross correlation and energy on 16 bit word */ + crossCorrScale=(int16_t)WebRtcSpl_NormW32(crossCorr)-16; + crossCorrmod=(int16_t)WEBRTC_SPL_SHIFT_W32(crossCorr, crossCorrScale); + Energyscale=(int16_t)WebRtcSpl_NormW32(Energy)-16; + EnergyMod=(int16_t)WEBRTC_SPL_SHIFT_W32(Energy, Energyscale); + + /* Square cross correlation and store upper int16_t */ + crossCorrSqMod = (int16_t)((crossCorrmod * crossCorrmod) >> 16); + + /* Calculate the total number of (dynamic) right shifts that have + been performed on (crossCorr*crossCorr)/energy + */ + totscale=Energyscale-(crossCorrScale<<1); + + /* Calculate the shift difference in order to be able to compare the two + (crossCorr*crossCorr)/energy in the same domain + */ + scalediff=totscale-totscale_max; + scalediff=WEBRTC_SPL_MIN(scalediff,31); + scalediff=WEBRTC_SPL_MAX(scalediff,-31); + + /* Compute the cross multiplication between the old best criteria + and the new one to be able to compare them without using a + division */ + + if (scalediff<0) { + newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max)>>(-scalediff); + maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod); + } else { + newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max); + maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod)>>scalediff; + } + + /* Store the new lag value if the new criteria is larger + than previous largest criteria */ + + if (newCrit > maxCrit) { + crossCorrSqMod_Max = crossCorrSqMod; + EnergyMod_Max = EnergyMod; + totscale_max = totscale; + maxlag = k; + } + } + pos+=step; + + /* Do a +/- to get the next energy */ + Energy += step * ((*rp_end * *rp_end - *rp_beg * *rp_beg) >> shifts); + rp_beg+=step; + rp_end+=step; + } + + return(maxlag+offset); +} diff --git a/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h new file mode 100644 index 0000000000..9b81c0fe97 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + WebRtcIlbcfix_XcorrCoef.h + +******************************************************************/ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_ + +#include "defines.h" + +/*----------------------------------------------------------------* + * cross correlation which finds the optimal lag for the + * crossCorr*crossCorr/(energy) criteria + *---------------------------------------------------------------*/ + +size_t WebRtcIlbcfix_XcorrCoef( + int16_t *target, /* (i) first array */ + int16_t *regressor, /* (i) second array */ + size_t subl, /* (i) dimension arrays */ + size_t searchLen, /* (i) the search lenght */ + size_t offset, /* (i) samples offset between arrays */ + int16_t step /* (i) +1 or -1 */ + ); + +#endif |