/dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/ |
H A D | PCMFile.cc | 126 audio_frame.data_[k] = 0; in Read10MsData() 135 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 136 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 137 audio_frame.num_channels_ = channels; in Read10MsData() 138 audio_frame.timestamp_ = timestamp_; in Read10MsData() 143 void PCMFile::Write10MsData(AudioFrame& audio_frame) { in Write10MsData() argument 144 if (audio_frame.num_channels_ == 1) { in Write10MsData() 146 if (fwrite(audio_frame.data_, sizeof(uint16_t), in Write10MsData() 155 stereo_audio[k << 1] = audio_frame.data_[k]; in Write10MsData() 166 if (fwrite(audio_frame.data_, sizeof(int16_t), in Write10MsData() [all …]
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/dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/webrtc/modules/audio_coding/main/test/ |
H A D | PCMFile.cc | 126 audio_frame.data_[k] = 0; in Read10MsData() 135 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 136 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 137 audio_frame.num_channels_ = channels; in Read10MsData() 138 audio_frame.timestamp_ = timestamp_; in Read10MsData() 143 void PCMFile::Write10MsData(AudioFrame& audio_frame) { in Write10MsData() argument 144 if (audio_frame.num_channels_ == 1) { in Write10MsData() 146 if (fwrite(audio_frame.data_, sizeof(uint16_t), in Write10MsData() 154 stereo_audio[k << 1] = audio_frame.data_[k]; in Write10MsData() 165 if (fwrite(audio_frame.data_, sizeof(int16_t), in Write10MsData() [all …]
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/test/ |
H A D | PCMFile.cc | 132 audio_frame.data_[k] = 0; in Read10MsData() 141 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 142 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 143 audio_frame.num_channels_ = channels; in Read10MsData() 144 audio_frame.timestamp_ = timestamp_; in Read10MsData() 152 void PCMFile::Write10MsData(AudioFrame& audio_frame) { in Write10MsData() argument 153 if (audio_frame.num_channels_ == 1) { in Write10MsData() 155 if (fwrite(audio_frame.data_, sizeof(uint16_t), in Write10MsData() 163 stereo_audio[k << 1] = audio_frame.data_[k]; in Write10MsData() 174 if (fwrite(audio_frame.data_, sizeof(int16_t), in Write10MsData() [all …]
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/modules/audio_coding/test/ |
H A D | PCMFile.cc | 125 int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) { in Read10MsData() argument 135 int16_t* frame_data = audio_frame.mutable_data(); in Read10MsData() 146 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 147 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 148 audio_frame.num_channels_ = channels; in Read10MsData() 149 audio_frame.timestamp_ = timestamp_; in Read10MsData() 158 if (audio_frame.num_channels_ == 1) { in Write10MsData() 160 if (fwrite(audio_frame.data(), sizeof(uint16_t), in Write10MsData() 166 const int16_t* frame_data = audio_frame.data(); in Write10MsData() 180 if (fwrite(audio_frame.data(), sizeof(int16_t), in Write10MsData() [all …]
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/dports/net-im/tg_owt/tg_owt-d578c76/src/modules/audio_coding/test/ |
H A D | PCMFile.cc | 125 int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) { in Read10MsData() argument 135 int16_t* frame_data = audio_frame.mutable_data(); in Read10MsData() 146 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 147 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 148 audio_frame.num_channels_ = channels; in Read10MsData() 149 audio_frame.timestamp_ = timestamp_; in Read10MsData() 158 if (audio_frame.num_channels_ == 1) { in Write10MsData() 160 if (fwrite(audio_frame.data(), sizeof(uint16_t), in Write10MsData() 166 const int16_t* frame_data = audio_frame.data(); in Write10MsData() 180 if (fwrite(audio_frame.data(), sizeof(int16_t), in Write10MsData() [all …]
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/dports/www/firefox/firefox-99.0/third_party/libwebrtc/modules/audio_coding/test/ |
H A D | PCMFile.cc | 125 int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) { in Read10MsData() argument 135 int16_t* frame_data = audio_frame.mutable_data(); in Read10MsData() 146 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 147 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 148 audio_frame.num_channels_ = channels; in Read10MsData() 149 audio_frame.timestamp_ = timestamp_; in Read10MsData() 158 if (audio_frame.num_channels_ == 1) { in Write10MsData() 160 if (fwrite(audio_frame.data(), sizeof(uint16_t), in Write10MsData() 166 const int16_t* frame_data = audio_frame.data(); in Write10MsData() 180 if (fwrite(audio_frame.data(), sizeof(int16_t), in Write10MsData() [all …]
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/test/ |
H A D | PCMFile.cc | 122 int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) { in Read10MsData() argument 132 int16_t* frame_data = audio_frame.mutable_data(); in Read10MsData() 143 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 144 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 145 audio_frame.num_channels_ = channels; in Read10MsData() 146 audio_frame.timestamp_ = timestamp_; in Read10MsData() 155 if (audio_frame.num_channels_ == 1) { in Write10MsData() 157 if (fwrite(audio_frame.data(), sizeof(uint16_t), in Write10MsData() 163 const int16_t* frame_data = audio_frame.data(); in Write10MsData() 177 if (fwrite(audio_frame.data(), sizeof(int16_t), in Write10MsData() [all …]
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/modules/audio_coding/test/ |
H A D | PCMFile.cc | 125 int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) { in Read10MsData() argument 135 int16_t* frame_data = audio_frame.mutable_data(); in Read10MsData() 146 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 147 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 148 audio_frame.num_channels_ = channels; in Read10MsData() 149 audio_frame.timestamp_ = timestamp_; in Read10MsData() 158 if (audio_frame.num_channels_ == 1) { in Write10MsData() 160 if (fwrite(audio_frame.data(), sizeof(uint16_t), in Write10MsData() 166 const int16_t* frame_data = audio_frame.data(); in Write10MsData() 180 if (fwrite(audio_frame.data(), sizeof(int16_t), in Write10MsData() [all …]
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/test/ |
H A D | PCMFile.cc | 122 int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) { in Read10MsData() argument 132 int16_t* frame_data = audio_frame.mutable_data(); in Read10MsData() 143 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 144 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 145 audio_frame.num_channels_ = channels; in Read10MsData() 146 audio_frame.timestamp_ = timestamp_; in Read10MsData() 155 if (audio_frame.num_channels_ == 1) { in Write10MsData() 157 if (fwrite(audio_frame.data(), sizeof(uint16_t), in Write10MsData() 163 const int16_t* frame_data = audio_frame.data(); in Write10MsData() 177 if (fwrite(audio_frame.data(), sizeof(int16_t), in Write10MsData() [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/test/ |
H A D | PCMFile.cc | 122 int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) { in Read10MsData() argument 132 int16_t* frame_data = audio_frame.mutable_data(); in Read10MsData() 143 audio_frame.samples_per_channel_ = samples_10ms_; in Read10MsData() 144 audio_frame.sample_rate_hz_ = frequency_; in Read10MsData() 145 audio_frame.num_channels_ = channels; in Read10MsData() 146 audio_frame.timestamp_ = timestamp_; in Read10MsData() 155 if (audio_frame.num_channels_ == 1) { in Write10MsData() 157 if (fwrite(audio_frame.data(), sizeof(uint16_t), in Write10MsData() 163 const int16_t* frame_data = audio_frame.data(); in Write10MsData() 177 if (fwrite(audio_frame.data(), sizeof(int16_t), in Write10MsData() [all …]
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/dports/net-im/tg_owt/tg_owt-d578c76/src/audio/ |
H A D | audio_transport_impl.cc | 36 AudioFrame* audio_frame) { in InitializeCaptureFrame() argument 37 RTC_DCHECK(audio_frame); in InitializeCaptureFrame() 40 audio_frame->sample_rate_hz_ = native_rate_hz; in InitializeCaptureFrame() 52 AudioFrame* audio_frame) { in ProcessCaptureFrame() argument 53 RTC_DCHECK(audio_frame); in ProcessCaptureFrame() 146 audio_frame.get()); in RecordedDataIsAvailable() 167 RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0); in RecordedDataIsAvailable() 171 SendProcessedData(std::move(audio_frame)); in RecordedDataIsAvailable() 177 std::unique_ptr<AudioFrame> audio_frame) { in SendProcessedData() argument 178 RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0); in SendProcessedData() [all …]
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/audio/ |
H A D | audio_transport_impl.cc | 36 AudioFrame* audio_frame) { in InitializeCaptureFrame() argument 37 RTC_DCHECK(audio_frame); in InitializeCaptureFrame() 40 audio_frame->sample_rate_hz_ = native_rate_hz; in InitializeCaptureFrame() 52 AudioFrame* audio_frame) { in ProcessCaptureFrame() argument 53 RTC_DCHECK(audio_frame); in ProcessCaptureFrame() 146 audio_frame.get()); in RecordedDataIsAvailable() 167 RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0); in RecordedDataIsAvailable() 171 SendProcessedData(std::move(audio_frame)); in RecordedDataIsAvailable() 177 std::unique_ptr<AudioFrame> audio_frame) { in SendProcessedData() argument 178 RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0); in SendProcessedData() [all …]
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/modules/audio_mixer/ |
H A D | audio_frame_manipulator.cc | 19 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { in AudioMixerCalculateEnergy() argument 20 if (audio_frame.muted()) { in AudioMixerCalculateEnergy() 25 const int16_t* frame_data = audio_frame.data(); in AudioMixerCalculateEnergy() 27 position < audio_frame.samples_per_channel_ * audio_frame.num_channels_; in AudioMixerCalculateEnergy() 35 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { in Ramp() argument 36 RTC_DCHECK(audio_frame); in Ramp() 39 if (start_gain == target_gain || audio_frame->muted()) { in Ramp() 43 size_t samples = audio_frame->samples_per_channel_; in Ramp() 47 int16_t* frame_data = audio_frame->mutable_data(); in Ramp() 51 for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { in Ramp() [all …]
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H A D | audio_mixer_impl.cc | 31 AudioFrame* audio_frame, in SourceFrame() 33 : source_status(source_status), audio_frame(audio_frame), muted(muted) { in SourceFrame() 35 RTC_DCHECK(audio_frame); in SourceFrame() 37 energy = AudioMixerCalculateEnergy(*audio_frame); in SourceFrame() 42 AudioFrame* audio_frame, in SourceFrame() 46 audio_frame(audio_frame), in SourceFrame() 50 RTC_DCHECK(audio_frame); in SourceFrame() 54 AudioFrame* audio_frame = nullptr; member 65 const auto a_activity = a.audio_frame->vad_activity_; in ShouldMixBefore() 80 source_frame.audio_frame); in RampAndUpdateGain() [all …]
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/dports/net-im/tg_owt/tg_owt-d578c76/src/modules/audio_mixer/ |
H A D | audio_frame_manipulator.cc | 19 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { in AudioMixerCalculateEnergy() argument 20 if (audio_frame.muted()) { in AudioMixerCalculateEnergy() 25 const int16_t* frame_data = audio_frame.data(); in AudioMixerCalculateEnergy() 27 position < audio_frame.samples_per_channel_ * audio_frame.num_channels_; in AudioMixerCalculateEnergy() 35 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { in Ramp() argument 36 RTC_DCHECK(audio_frame); in Ramp() 39 if (start_gain == target_gain || audio_frame->muted()) { in Ramp() 43 size_t samples = audio_frame->samples_per_channel_; in Ramp() 47 int16_t* frame_data = audio_frame->mutable_data(); in Ramp() 51 for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { in Ramp() [all …]
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/dports/www/firefox/firefox-99.0/third_party/libwebrtc/modules/audio_mixer/ |
H A D | audio_frame_manipulator.cc | 19 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { in AudioMixerCalculateEnergy() argument 20 if (audio_frame.muted()) { in AudioMixerCalculateEnergy() 25 const int16_t* frame_data = audio_frame.data(); in AudioMixerCalculateEnergy() 27 position < audio_frame.samples_per_channel_ * audio_frame.num_channels_; in AudioMixerCalculateEnergy() 35 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { in Ramp() argument 36 RTC_DCHECK(audio_frame); in Ramp() 39 if (start_gain == target_gain || audio_frame->muted()) { in Ramp() 43 size_t samples = audio_frame->samples_per_channel_; in Ramp() 47 int16_t* frame_data = audio_frame->mutable_data(); in Ramp() 51 for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { in Ramp() [all …]
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H A D | audio_mixer_impl.cc | 31 AudioFrame* audio_frame, in SourceFrame() 33 : source_status(source_status), audio_frame(audio_frame), muted(muted) { in SourceFrame() 35 RTC_DCHECK(audio_frame); in SourceFrame() 37 energy = AudioMixerCalculateEnergy(*audio_frame); in SourceFrame() 42 AudioFrame* audio_frame, in SourceFrame() 46 audio_frame(audio_frame), in SourceFrame() 50 RTC_DCHECK(audio_frame); in SourceFrame() 54 AudioFrame* audio_frame = nullptr; member 65 const auto a_activity = a.audio_frame->vad_activity_; in ShouldMixBefore() 80 source_frame.audio_frame); in RampAndUpdateGain() [all …]
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/modules/audio_mixer/ |
H A D | audio_frame_manipulator.cc | 19 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { in AudioMixerCalculateEnergy() argument 20 if (audio_frame.muted()) { in AudioMixerCalculateEnergy() 25 const int16_t* frame_data = audio_frame.data(); in AudioMixerCalculateEnergy() 27 position < audio_frame.samples_per_channel_ * audio_frame.num_channels_; in AudioMixerCalculateEnergy() 35 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { in Ramp() argument 36 RTC_DCHECK(audio_frame); in Ramp() 39 if (start_gain == target_gain || audio_frame->muted()) { in Ramp() 43 size_t samples = audio_frame->samples_per_channel_; in Ramp() 47 int16_t* frame_data = audio_frame->mutable_data(); in Ramp() 51 for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { in Ramp() [all …]
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/audio_mixer/ |
H A D | audio_frame_manipulator.cc | 18 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { in AudioMixerCalculateEnergy() argument 19 if (audio_frame.muted()) { in AudioMixerCalculateEnergy() 24 const int16_t* frame_data = audio_frame.data(); in AudioMixerCalculateEnergy() 25 for (size_t position = 0; position < audio_frame.samples_per_channel_; in AudioMixerCalculateEnergy() 33 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { in Ramp() argument 34 RTC_DCHECK(audio_frame); in Ramp() 37 if (start_gain == target_gain || audio_frame->muted()) { in Ramp() 41 size_t samples = audio_frame->samples_per_channel_; in Ramp() 45 int16_t* frame_data = audio_frame->mutable_data(); in Ramp() 49 for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { in Ramp() [all …]
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/audio_mixer/ |
H A D | audio_frame_manipulator.cc | 18 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { in AudioMixerCalculateEnergy() argument 19 if (audio_frame.muted()) { in AudioMixerCalculateEnergy() 24 const int16_t* frame_data = audio_frame.data(); in AudioMixerCalculateEnergy() 25 for (size_t position = 0; position < audio_frame.samples_per_channel_; in AudioMixerCalculateEnergy() 33 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { in Ramp() argument 34 RTC_DCHECK(audio_frame); in Ramp() 37 if (start_gain == target_gain || audio_frame->muted()) { in Ramp() 41 size_t samples = audio_frame->samples_per_channel_; in Ramp() 45 int16_t* frame_data = audio_frame->mutable_data(); in Ramp() 49 for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { in Ramp() [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_mixer/ |
H A D | audio_frame_manipulator.cc | 18 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { in AudioMixerCalculateEnergy() argument 19 if (audio_frame.muted()) { in AudioMixerCalculateEnergy() 24 const int16_t* frame_data = audio_frame.data(); in AudioMixerCalculateEnergy() 25 for (size_t position = 0; position < audio_frame.samples_per_channel_; in AudioMixerCalculateEnergy() 33 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { in Ramp() argument 34 RTC_DCHECK(audio_frame); in Ramp() 37 if (start_gain == target_gain || audio_frame->muted()) { in Ramp() 41 size_t samples = audio_frame->samples_per_channel_; in Ramp() 45 int16_t* frame_data = audio_frame->mutable_data(); in Ramp() 49 for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { in Ramp() [all …]
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/audio_mixer/ |
H A D | audio_frame_manipulator.cc | 18 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { in AudioMixerCalculateEnergy() argument 20 for (size_t position = 0; position < audio_frame.samples_per_channel_; in AudioMixerCalculateEnergy() 23 energy += audio_frame.data_[position] * audio_frame.data_[position]; in AudioMixerCalculateEnergy() 28 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { in Ramp() argument 29 RTC_DCHECK(audio_frame); in Ramp() 36 size_t samples = audio_frame->samples_per_channel_; in Ramp() 43 for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { in Ramp() 44 audio_frame->data_[audio_frame->num_channels_ * i + ch] *= gain; in Ramp()
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/audio/ |
H A D | audio_transport_impl.cc | 35 AudioFrame* audio_frame) { in InitializeCaptureFrame() argument 36 RTC_DCHECK(audio_frame); in InitializeCaptureFrame() 39 audio_frame->sample_rate_hz_ = native_rate_hz; in InitializeCaptureFrame() 51 AudioFrame* audio_frame) { in ProcessCaptureFrame() argument 53 RTC_DCHECK(audio_frame); in ProcessCaptureFrame() 56 int error = ProcessAudioFrame(audio_processing, audio_frame); in ProcessCaptureFrame() 60 AudioFrameOperations::SwapStereoChannels(audio_frame); in ProcessCaptureFrame() 126 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); in RecordedDataIsAvailable() local 134 audio_frame.get()); in RecordedDataIsAvailable() 153 RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0); in RecordedDataIsAvailable() [all …]
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/dports/www/firefox/firefox-99.0/third_party/libwebrtc/audio/ |
H A D | audio_transport_impl.cc | 35 AudioFrame* audio_frame) { in InitializeCaptureFrame() argument 36 RTC_DCHECK(audio_frame); in InitializeCaptureFrame() 39 audio_frame->sample_rate_hz_ = native_rate_hz; in InitializeCaptureFrame() 40 if (audio_frame->sample_rate_hz_ >= min_processing_rate_hz) { in InitializeCaptureFrame() 51 AudioFrame* audio_frame) { in ProcessCaptureFrame() argument 52 RTC_DCHECK(audio_frame); in ProcessCaptureFrame() 62 AudioFrameOperations::SwapStereoChannels(audio_frame); in ProcessCaptureFrame() 127 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); in RecordedDataIsAvailable() local 135 audio_frame.get()); in RecordedDataIsAvailable() 155 RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0); in RecordedDataIsAvailable() [all …]
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/dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_receiver.cc | 57 audio_frame->speech_type_ = AudioFrame::kCNG; in SetAudioFrameActivityAndType() 63 audio_frame->speech_type_ = AudioFrame::kPLC; in SetAudioFrameActivityAndType() 68 audio_frame->speech_type_ = AudioFrame::kPLCCNG; in SetAudioFrameActivityAndType() 83 audio_frame->speech_type_ = AudioFrame::kCNG; in SetAudioFrameActivityAndType() 87 audio_frame->speech_type_ = AudioFrame::kPLC; in SetAudioFrameActivityAndType() 91 audio_frame->speech_type_ = AudioFrame::kPLCCNG; in SetAudioFrameActivityAndType() 264 audio_frame->data_); in GetAudio() 274 memcpy(audio_frame->data_, in GetAudio() 283 audio_frame->num_channels_ = num_channels; in GetAudio() 298 audio_frame->timestamp_ = playout_timestamp - in GetAudio() [all …]
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