diff options
Diffstat (limited to 'modules/audio_coding/codecs/opus/opus_interface.c')
| -rw-r--r-- | modules/audio_coding/codecs/opus/opus_interface.c | 259 |
1 files changed, 31 insertions, 228 deletions
diff --git a/modules/audio_coding/codecs/opus/opus_interface.c b/modules/audio_coding/codecs/opus/opus_interface.c index 24fc4fc4..ea535ea9 100644 --- a/modules/audio_coding/codecs/opus/opus_interface.c +++ b/modules/audio_coding/codecs/opus/opus_interface.c @@ -15,9 +15,6 @@ #include "opus.h" -#include "webrtc/common_audio/signal_processing/resample_by_2_internal.h" -#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" - enum { /* Maximum supported frame size in WebRTC is 60 ms. */ kWebRtcOpusMaxEncodeFrameSizeMs = 60, @@ -31,17 +28,6 @@ enum { * milliseconds. */ kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs, - /* Maximum sample count per frame is 48 kHz * maximum frame size in - * milliseconds * maximum number of channels. */ - kWebRtcOpusMaxFrameSize = kWebRtcOpusMaxFrameSizePerChannel * 2, - - /* Maximum sample count per channel for output resampled to 32 kHz, - * 32 kHz * maximum frame size in milliseconds. */ - kWebRtcOpusMaxFrameSizePerChannel32kHz = 32 * kWebRtcOpusMaxDecodeFrameSizeMs, - - /* Number of samples in resampler state. */ - kWebRtcOpusStateSize = 7, - /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */ kWebRtcOpusDefaultFrameSize = 960, }; @@ -143,8 +129,6 @@ int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) { } struct WebRtcOpusDecInst { - int16_t state_48_32_left[8]; - int16_t state_48_32_right[8]; OpusDecoder* decoder_left; OpusDecoder* decoder_right; int prev_decoded_samples; @@ -205,8 +189,6 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst) { int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst) { int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE); if (error == OPUS_OK) { - memset(inst->state_48_32_left, 0, sizeof(inst->state_48_32_left)); - memset(inst->state_48_32_right, 0, sizeof(inst->state_48_32_right)); return 0; } return -1; @@ -215,7 +197,6 @@ int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst) { int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) { int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE); if (error == OPUS_OK) { - memset(inst->state_48_32_left, 0, sizeof(inst->state_48_32_left)); return 0; } return -1; @@ -224,7 +205,6 @@ int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) { int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst) { int error = opus_decoder_ctl(inst->decoder_right, OPUS_RESET_STATE); if (error == OPUS_OK) { - memset(inst->state_48_32_right, 0, sizeof(inst->state_48_32_right)); return 0; } return -1; @@ -267,124 +247,29 @@ static int DecodeFec(OpusDecoder* inst, const int16_t* encoded, return -1; } -/* Resample from 48 to 32 kHz. Length of state is assumed to be - * kWebRtcOpusStateSize (7). - */ -static int WebRtcOpus_Resample48to32(const int16_t* samples_in, int length, - int16_t* state, int16_t* samples_out) { - int i; - int blocks; - int16_t output_samples; - int32_t buffer32[kWebRtcOpusMaxFrameSizePerChannel + kWebRtcOpusStateSize]; - - /* Resample from 48 kHz to 32 kHz. */ - for (i = 0; i < kWebRtcOpusStateSize; i++) { - buffer32[i] = state[i]; - state[i] = samples_in[length - kWebRtcOpusStateSize + i]; - } - for (i = 0; i < length; i++) { - buffer32[kWebRtcOpusStateSize + i] = samples_in[i]; - } - /* Resampling 3 samples to 2. Function divides the input in |blocks| number - * of 3-sample groups, and output is |blocks| number of 2-sample groups. - * When this is removed, the compensation in WebRtcOpus_DurationEst should be - * removed too. */ - blocks = length / 3; - WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, blocks); - output_samples = (int16_t) (blocks * 2); - WebRtcSpl_VectorBitShiftW32ToW16(samples_out, output_samples, buffer32, 15); - - return output_samples; -} - -static int WebRtcOpus_DeInterleaveResample(OpusDecInst* inst, int16_t* input, - int sample_pairs, int16_t* output) { - int i; - int16_t buffer_left[kWebRtcOpusMaxFrameSizePerChannel]; - int16_t buffer_right[kWebRtcOpusMaxFrameSizePerChannel]; - int16_t buffer_out[kWebRtcOpusMaxFrameSizePerChannel32kHz]; - int resampled_samples; - - /* De-interleave the signal in left and right channel. */ - for (i = 0; i < sample_pairs; i++) { - /* Take every second sample, starting at the first sample. */ - buffer_left[i] = input[i * 2]; - buffer_right[i] = input[i * 2 + 1]; - } - - /* Resample from 48 kHz to 32 kHz for left channel. */ - resampled_samples = WebRtcOpus_Resample48to32( - buffer_left, sample_pairs, inst->state_48_32_left, buffer_out); - - /* Add samples interleaved to output vector. */ - for (i = 0; i < resampled_samples; i++) { - output[i * 2] = buffer_out[i]; - } - - /* Resample from 48 kHz to 32 kHz for right channel. */ - resampled_samples = WebRtcOpus_Resample48to32( - buffer_right, sample_pairs, inst->state_48_32_right, buffer_out); - - /* Add samples interleaved to output vector. */ - for (i = 0; i < resampled_samples; i++) { - output[i * 2 + 1] = buffer_out[i]; - } - - return resampled_samples; -} - int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { - /* |buffer| is big enough for 120 ms (the largest Opus packet size) of stereo - * audio at 48 kHz. */ - int16_t buffer[kWebRtcOpusMaxFrameSize]; int16_t* coded = (int16_t*)encoded; int decoded_samples; - int resampled_samples; - - /* If mono case, just do a regular call to the decoder. - * If stereo, we need to de-interleave the stereo output into blocks with - * left and right channel. Each block is resampled to 32 kHz, and then - * interleaved again. */ - /* Decode to a temporary buffer. */ decoded_samples = DecodeNative(inst->decoder_left, coded, encoded_bytes, kWebRtcOpusMaxFrameSizePerChannel, - buffer, audio_type); + decoded, audio_type); if (decoded_samples < 0) { return -1; } - if (inst->channels == 2) { - /* De-interleave and resample. */ - resampled_samples = WebRtcOpus_DeInterleaveResample(inst, - buffer, - decoded_samples, - decoded); - } else { - /* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is - * used for mono signals. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_left, - decoded); - } - /* Update decoded sample memory, to be used by the PLC in case of losses. */ inst->prev_decoded_samples = decoded_samples; - return resampled_samples; + return decoded_samples; } int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { - /* |buffer16| is big enough for 120 ms (the largestOpus packet size) of - * stereo audio at 48 kHz. */ - int16_t buffer16[kWebRtcOpusMaxFrameSize]; int decoded_samples; - int16_t output_samples; int i; /* If mono case, just do a regular call to the decoder. @@ -393,120 +278,82 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded, * This is to make stereo work with the current setup of NetEQ, which * requires two calls to the decoder to produce stereo. */ - /* Decode to a temporary buffer. */ decoded_samples = DecodeNative(inst->decoder_left, encoded, encoded_bytes, - kWebRtcOpusMaxFrameSizePerChannel, buffer16, + kWebRtcOpusMaxFrameSizePerChannel, decoded, audio_type); if (decoded_samples < 0) { return -1; } if (inst->channels == 2) { /* The parameter |decoded_samples| holds the number of samples pairs, in - * case of stereo. Number of samples in |buffer16| equals |decoded_samples| + * case of stereo. Number of samples in |decoded| equals |decoded_samples| * times 2. */ for (i = 0; i < decoded_samples; i++) { /* Take every second sample, starting at the first sample. This gives * the left channel. */ - buffer16[i] = buffer16[i * 2]; + decoded[i] = decoded[i * 2]; } } - /* Resample from 48 kHz to 32 kHz. */ - output_samples = WebRtcOpus_Resample48to32(buffer16, decoded_samples, - inst->state_48_32_left, decoded); - /* Update decoded sample memory, to be used by the PLC in case of losses. */ inst->prev_decoded_samples = decoded_samples; - return output_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { - /* |buffer16| is big enough for 120 ms (the largestOpus packet size) of - * stereo audio at 48 kHz. */ - int16_t buffer16[kWebRtcOpusMaxFrameSize]; int decoded_samples; - int16_t output_samples; int i; - /* Decode to a temporary buffer. */ decoded_samples = DecodeNative(inst->decoder_right, encoded, encoded_bytes, - kWebRtcOpusMaxFrameSizePerChannel, buffer16, + kWebRtcOpusMaxFrameSizePerChannel, decoded, audio_type); if (decoded_samples < 0) { return -1; } if (inst->channels == 2) { /* The parameter |decoded_samples| holds the number of samples pairs, in - * case of stereo. Number of samples in |buffer16| equals |decoded_samples| + * case of stereo. Number of samples in |decoded| equals |decoded_samples| * times 2. */ for (i = 0; i < decoded_samples; i++) { /* Take every second sample, starting at the second sample. This gives * the right channel. */ - buffer16[i] = buffer16[i * 2 + 1]; + decoded[i] = decoded[i * 2 + 1]; } } else { /* Decode slave should never be called for mono packets. */ return -1; } - /* Resample from 48 kHz to 32 kHz. */ - output_samples = WebRtcOpus_Resample48to32(buffer16, decoded_samples, - inst->state_48_32_right, decoded); - return output_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames) { - int16_t buffer[kWebRtcOpusMaxFrameSize]; int16_t audio_type = 0; int decoded_samples; - int resampled_samples; int plc_samples; - /* If mono case, just do a regular call to the plc function, before - * resampling. - * If stereo, we need to de-interleave the stereo output into blocks with - * left and right channel. Each block is resampled to 32 kHz, and then - * interleaved again. */ - - /* Decode to a temporary buffer. The number of samples we ask for is - * |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number - * of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */ + /* The number of samples we ask for is |number_of_lost_frames| times + * |prev_decoded_samples_|. Limit the number of samples to maximum + * |kWebRtcOpusMaxFrameSizePerChannel|. */ plc_samples = number_of_lost_frames * inst->prev_decoded_samples; plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ? plc_samples : kWebRtcOpusMaxFrameSizePerChannel; decoded_samples = DecodeNative(inst->decoder_left, NULL, 0, plc_samples, - buffer, &audio_type); + decoded, &audio_type); if (decoded_samples < 0) { return -1; } - if (inst->channels == 2) { - /* De-interleave and resample. */ - resampled_samples = WebRtcOpus_DeInterleaveResample(inst, - buffer, - decoded_samples, - decoded); - } else { - /* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is - * used for mono signals. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_left, - decoded); - } - - return resampled_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames) { - int16_t buffer[kWebRtcOpusMaxFrameSize]; int decoded_samples; - int resampled_samples; int16_t audio_type = 0; int plc_samples; int i; @@ -517,42 +364,35 @@ int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded, * output. This is to make stereo work with the current setup of NetEQ, which * requires two calls to the decoder to produce stereo. */ - /* Decode to a temporary buffer. The number of samples we ask for is - * |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number - * of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */ + /* The number of samples we ask for is |number_of_lost_frames| times + * |prev_decoded_samples_|. Limit the number of samples to maximum + * |kWebRtcOpusMaxFrameSizePerChannel|. */ plc_samples = number_of_lost_frames * inst->prev_decoded_samples; plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ? plc_samples : kWebRtcOpusMaxFrameSizePerChannel; decoded_samples = DecodeNative(inst->decoder_left, NULL, 0, plc_samples, - buffer, &audio_type); + decoded, &audio_type); if (decoded_samples < 0) { return -1; } if (inst->channels == 2) { /* The parameter |decoded_samples| holds the number of sample pairs, in - * case of stereo. The original number of samples in |buffer| equals + * case of stereo. The original number of samples in |decoded| equals * |decoded_samples| times 2. */ for (i = 0; i < decoded_samples; i++) { /* Take every second sample, starting at the first sample. This gives * the left channel. */ - buffer[i] = buffer[i * 2]; + decoded[i] = decoded[i * 2]; } } - /* Resample from 48 kHz to 32 kHz for left channel. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_left, - decoded); - return resampled_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames) { - int16_t buffer[kWebRtcOpusMaxFrameSize]; int decoded_samples; - int resampled_samples; int16_t audio_type = 0; int plc_samples; int i; @@ -563,44 +403,35 @@ int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded, return -1; } - /* Decode to a temporary buffer. The number of samples we ask for is - * |number_of_lost_frames| times |prev_decoded_samples_|. Limit the number - * of samples to maximum |kWebRtcOpusMaxFrameSizePerChannel|. */ + /* The number of samples we ask for is |number_of_lost_frames| times + * |prev_decoded_samples_|. Limit the number of samples to maximum + * |kWebRtcOpusMaxFrameSizePerChannel|. */ plc_samples = number_of_lost_frames * inst->prev_decoded_samples; plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ? plc_samples : kWebRtcOpusMaxFrameSizePerChannel; decoded_samples = DecodeNative(inst->decoder_right, NULL, 0, plc_samples, - buffer, &audio_type); + decoded, &audio_type); if (decoded_samples < 0) { return -1; } /* The parameter |decoded_samples| holds the number of sample pairs, - * The original number of samples in |buffer| equals |decoded_samples| + * The original number of samples in |decoded| equals |decoded_samples| * times 2. */ for (i = 0; i < decoded_samples; i++) { /* Take every second sample, starting at the second sample. This gives * the right channel. */ - buffer[i] = buffer[i * 2 + 1]; + decoded[i] = decoded[i * 2 + 1]; } - /* Resample from 48 kHz to 32 kHz for left channel. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_right, - decoded); - return resampled_samples; + return decoded_samples; } int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { - /* |buffer| is big enough for 120 ms (the largest Opus packet size) of stereo - * audio at 48 kHz. */ - int16_t buffer[kWebRtcOpusMaxFrameSize]; int16_t* coded = (int16_t*)encoded; int decoded_samples; - int resampled_samples; int fec_samples; if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) { @@ -609,33 +440,13 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, fec_samples = opus_packet_get_samples_per_frame(encoded, 48000); - /* Decode to a temporary buffer. */ decoded_samples = DecodeFec(inst->decoder_left, coded, encoded_bytes, - fec_samples, buffer, audio_type); + fec_samples, decoded, audio_type); if (decoded_samples < 0) { return -1; } - /* If mono case, just do a regular call to the decoder. - * If stereo, we need to de-interleave the stereo output into blocks with - * left and right channel. Each block is resampled to 32 kHz, and then - * interleaved again. */ - if (inst->channels == 2) { - /* De-interleave and resample. */ - resampled_samples = WebRtcOpus_DeInterleaveResample(inst, - buffer, - decoded_samples, - decoded); - } else { - /* Resample from 48 kHz to 32 kHz. Filter state memory for left channel is - * used for mono signals. */ - resampled_samples = WebRtcOpus_Resample48to32(buffer, - decoded_samples, - inst->state_48_32_left, - decoded); - } - - return resampled_samples; + return decoded_samples; } int WebRtcOpus_DurationEst(OpusDecInst* inst, @@ -652,10 +463,6 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst, /* Invalid payload duration. */ return 0; } - /* Compensate for the down-sampling from 48 kHz to 32 kHz. - * This should be removed when the resampling in WebRtcOpus_Decode is - * removed. */ - samples = samples * 2 / 3; return samples; } @@ -671,10 +478,6 @@ int WebRtcOpus_FecDurationEst(const uint8_t* payload, /* Invalid payload duration. */ return 0; } - /* Compensate for the down-sampling from 48 kHz to 32 kHz. - * This should be removed when the resampling in WebRtcOpus_Decode is - * removed. */ - samples = samples * 2 / 3; return samples; } |