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Diffstat (limited to 'src/modules/audio_processing/main/source/audio_buffer.cc')
| -rw-r--r-- | src/modules/audio_processing/main/source/audio_buffer.cc | 278 |
1 files changed, 278 insertions, 0 deletions
diff --git a/src/modules/audio_processing/main/source/audio_buffer.cc b/src/modules/audio_processing/main/source/audio_buffer.cc new file mode 100644 index 0000000000..6b20fcecee --- /dev/null +++ b/src/modules/audio_processing/main/source/audio_buffer.cc @@ -0,0 +1,278 @@ +/* + * 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. + */ + +#include "audio_buffer.h" + +#include "module_common_types.h" + +namespace webrtc { +namespace { + +enum { + kSamplesPer8kHzChannel = 80, + kSamplesPer16kHzChannel = 160, + kSamplesPer32kHzChannel = 320 +}; + +void StereoToMono(const WebRtc_Word16* left, const WebRtc_Word16* right, + WebRtc_Word16* out, int samples_per_channel) { + WebRtc_Word32 data_int32 = 0; + for (int i = 0; i < samples_per_channel; i++) { + data_int32 = (left[i] + right[i]) >> 1; + if (data_int32 > 32767) { + data_int32 = 32767; + } else if (data_int32 < -32768) { + data_int32 = -32768; + } + + out[i] = static_cast<WebRtc_Word16>(data_int32); + } +} +} // namespace + +struct AudioChannel { + AudioChannel() { + memset(data, 0, sizeof(data)); + } + + WebRtc_Word16 data[kSamplesPer32kHzChannel]; +}; + +struct SplitAudioChannel { + SplitAudioChannel() { + memset(low_pass_data, 0, sizeof(low_pass_data)); + memset(high_pass_data, 0, sizeof(high_pass_data)); + memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1)); + memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2)); + memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1)); + memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2)); + } + + WebRtc_Word16 low_pass_data[kSamplesPer16kHzChannel]; + WebRtc_Word16 high_pass_data[kSamplesPer16kHzChannel]; + + WebRtc_Word32 analysis_filter_state1[6]; + WebRtc_Word32 analysis_filter_state2[6]; + WebRtc_Word32 synthesis_filter_state1[6]; + WebRtc_Word32 synthesis_filter_state2[6]; +}; + +// TODO(am): check range of input parameters? +AudioBuffer::AudioBuffer(WebRtc_Word32 max_num_channels, + WebRtc_Word32 samples_per_channel) + : max_num_channels_(max_num_channels), + num_channels_(0), + num_mixed_channels_(0), + num_mixed_low_pass_channels_(0), + samples_per_channel_(samples_per_channel), + samples_per_split_channel_(samples_per_channel), + reference_copied_(false), + data_(NULL), + channels_(NULL), + split_channels_(NULL), + mixed_low_pass_channels_(NULL), + low_pass_reference_channels_(NULL) { + if (max_num_channels_ > 1) { + channels_ = new AudioChannel[max_num_channels_]; + mixed_low_pass_channels_ = new AudioChannel[max_num_channels_]; + } + low_pass_reference_channels_ = new AudioChannel[max_num_channels_]; + + if (samples_per_channel_ == kSamplesPer32kHzChannel) { + split_channels_ = new SplitAudioChannel[max_num_channels_]; + samples_per_split_channel_ = kSamplesPer16kHzChannel; + } +} + +AudioBuffer::~AudioBuffer() { + if (channels_ != NULL) { + delete [] channels_; + } + + if (mixed_low_pass_channels_ != NULL) { + delete [] mixed_low_pass_channels_; + } + + if (low_pass_reference_channels_ != NULL) { + delete [] low_pass_reference_channels_; + } + + if (split_channels_ != NULL) { + delete [] split_channels_; + } +} + +WebRtc_Word16* AudioBuffer::data(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_channels_); + if (data_ != NULL) { + return data_; + } + + return channels_[channel].data; +} + +WebRtc_Word16* AudioBuffer::low_pass_split_data(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_channels_); + if (split_channels_ == NULL) { + return data(channel); + } + + return split_channels_[channel].low_pass_data; +} + +WebRtc_Word16* AudioBuffer::high_pass_split_data(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_channels_); + if (split_channels_ == NULL) { + return NULL; + } + + return split_channels_[channel].high_pass_data; +} + +WebRtc_Word16* AudioBuffer::mixed_low_pass_data(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_mixed_low_pass_channels_); + + return mixed_low_pass_channels_[channel].data; +} + +WebRtc_Word16* AudioBuffer::low_pass_reference(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_channels_); + if (!reference_copied_) { + return NULL; + } + + return low_pass_reference_channels_[channel].data; +} + +WebRtc_Word32* AudioBuffer::analysis_filter_state1(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_channels_); + return split_channels_[channel].analysis_filter_state1; +} + +WebRtc_Word32* AudioBuffer::analysis_filter_state2(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_channels_); + return split_channels_[channel].analysis_filter_state2; +} + +WebRtc_Word32* AudioBuffer::synthesis_filter_state1(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_channels_); + return split_channels_[channel].synthesis_filter_state1; +} + +WebRtc_Word32* AudioBuffer::synthesis_filter_state2(WebRtc_Word32 channel) const { + assert(channel >= 0 && channel < num_channels_); + return split_channels_[channel].synthesis_filter_state2; +} + +WebRtc_Word32 AudioBuffer::num_channels() const { + return num_channels_; +} + +WebRtc_Word32 AudioBuffer::samples_per_channel() const { + return samples_per_channel_; +} + +WebRtc_Word32 AudioBuffer::samples_per_split_channel() const { + return samples_per_split_channel_; +} + +// TODO(ajm): Do deinterleaving and mixing in one step? +void AudioBuffer::DeinterleaveFrom(AudioFrame* audioFrame) { + assert(audioFrame->_audioChannel <= max_num_channels_); + assert(audioFrame->_payloadDataLengthInSamples == samples_per_channel_); + + num_channels_ = audioFrame->_audioChannel; + num_mixed_channels_ = 0; + num_mixed_low_pass_channels_ = 0; + reference_copied_ = false; + + if (num_channels_ == 1) { + // We can get away with a pointer assignment in this case. + data_ = audioFrame->_payloadData; + return; + } + + for (int i = 0; i < num_channels_; i++) { + WebRtc_Word16* deinterleaved = channels_[i].data; + WebRtc_Word16* interleaved = audioFrame->_payloadData; + WebRtc_Word32 interleaved_idx = i; + for (int j = 0; j < samples_per_channel_; j++) { + deinterleaved[j] = interleaved[interleaved_idx]; + interleaved_idx += num_channels_; + } + } +} + +void AudioBuffer::InterleaveTo(AudioFrame* audioFrame) const { + assert(audioFrame->_audioChannel == num_channels_); + assert(audioFrame->_payloadDataLengthInSamples == samples_per_channel_); + + if (num_channels_ == 1) { + if (num_mixed_channels_ == 1) { + memcpy(audioFrame->_payloadData, + channels_[0].data, + sizeof(WebRtc_Word16) * samples_per_channel_); + } else { + // These should point to the same buffer in this case. + assert(data_ == audioFrame->_payloadData); + } + + return; + } + + for (int i = 0; i < num_channels_; i++) { + WebRtc_Word16* deinterleaved = channels_[i].data; + WebRtc_Word16* interleaved = audioFrame->_payloadData; + WebRtc_Word32 interleaved_idx = i; + for (int j = 0; j < samples_per_channel_; j++) { + interleaved[interleaved_idx] = deinterleaved[j]; + interleaved_idx += num_channels_; + } + } +} + +// TODO(ajm): would be good to support the no-mix case with pointer assignment. +// TODO(ajm): handle mixing to multiple channels? +void AudioBuffer::Mix(WebRtc_Word32 num_mixed_channels) { + // We currently only support the stereo to mono case. + assert(num_channels_ == 2); + assert(num_mixed_channels == 1); + + StereoToMono(channels_[0].data, + channels_[1].data, + channels_[0].data, + samples_per_channel_); + + num_channels_ = num_mixed_channels; + num_mixed_channels_ = num_mixed_channels; +} + +void AudioBuffer::CopyAndMixLowPass(WebRtc_Word32 num_mixed_channels) { + // We currently only support the stereo to mono case. + assert(num_channels_ == 2); + assert(num_mixed_channels == 1); + + StereoToMono(low_pass_split_data(0), + low_pass_split_data(1), + mixed_low_pass_channels_[0].data, + samples_per_split_channel_); + + num_mixed_low_pass_channels_ = num_mixed_channels; +} + +void AudioBuffer::CopyLowPassToReference() { + reference_copied_ = true; + for (int i = 0; i < num_channels_; i++) { + memcpy(low_pass_reference_channels_[i].data, + low_pass_split_data(i), + sizeof(WebRtc_Word16) * samples_per_split_channel_); + } +} +} // namespace webrtc |