| /aosp_15_r20/external/webrtc/modules/audio_device/ |
| H A D | audio_device_buffer.cc | 233 size_t samples_per_channel) { in SetRecordedBuffer()
238 size_t samples_per_channel, in SetRecordedBuffer()
300 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) { in RequestPlayoutData()
503 size_t samples_per_channel) { in UpdateRecStats()
513 size_t samples_per_channel) { in UpdatePlayStats()
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| /aosp_15_r20/external/webrtc/modules/audio_mixer/ |
| H A D | frame_combiner.cc | 48 const size_t samples_per_channel = static_cast<size_t>( in SetAudioFrameFields() local
95 size_t samples_per_channel, in MixToFloatFrame()
131 const size_t samples_per_channel = mixing_buffer_view.samples_per_channel(); in InterleaveToAudioFrame() local
170 const size_t samples_per_channel = static_cast<size_t>( in Combine() local
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| H A D | audio_frame_manipulator_unittest.cc | 20 void FillFrameWithConstants(size_t samples_per_channel, in FillFrameWithConstants()
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| /aosp_15_r20/external/webrtc/modules/audio_processing/agc2/ |
| H A D | limiter.cc | 49 int samples_per_channel, in ComputePerSampleSubframeFactors()
78 const int samples_per_channel = signal.samples_per_channel(); in ScaleSamples() local
119 const int samples_per_channel = signal.samples_per_channel(); in Process() local
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| H A D | input_volume_controller.cc | 106 size_t samples_per_channel) { in ComputeClippedRatio()
428 size_t samples_per_channel = audio_buffer.num_frames(); in AnalyzePreProcess() local
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| H A D | clipping_predictor.cc | 114 const int samples_per_channel = frame.samples_per_channel(); in Analyze() local
253 const int samples_per_channel = frame.samples_per_channel(); in Analyze() local
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| H A D | vector_float_frame.cc | 28 int samples_per_channel, in VectorFloatFrame()
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| H A D | gain_applier.cc | 97 void GainApplier::Initialize(int samples_per_channel) { in Initialize()
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| H A D | noise_level_estimator_unittest.cc | 35 const int samples_per_channel = in RunEstimator() local
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| /aosp_15_r20/external/webrtc/modules/audio_processing/aec_dump/ |
| H A D | aec_dump_impl.cc | 114 int samples_per_channel) { in AddCaptureStreamInput()
120 int samples_per_channel) { in AddCaptureStreamOutput()
134 int samples_per_channel) { in WriteRenderStreamMessage()
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| H A D | capture_stream_info.cc | 37 int samples_per_channel) { in AddInput()
45 int samples_per_channel) { in AddOutput()
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| /aosp_15_r20/external/webrtc/modules/audio_processing/test/ |
| H A D | bitexactness_tools.cc | 57 void ReadFloatSamplesFromStereoFile(size_t samples_per_channel, in ReadFloatSamplesFromStereoFile()
76 size_t samples_per_channel, in VerifyDeinterleavedArray()
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| H A D | simulator_buffers.cc | 60 int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); in CreateConfigAndBuffer() local
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| /aosp_15_r20/external/webrtc/modules/audio_coding/codecs/g722/ |
| H A D | audio_encoder_g722.cc | 38 const size_t samples_per_channel = in AudioEncoderG722Impl() local
109 const size_t samples_per_channel = SamplesPerChannel(); in EncodeImpl() local
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| /aosp_15_r20/external/webrtc/common_audio/include/ |
| H A D | audio_util.h | 115 size_t samples_per_channel, in Deinterleave()
133 size_t samples_per_channel, in Interleave()
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| /aosp_15_r20/external/webrtc/audio/utility/ |
| H A D | audio_frame_operations.cc | 92 size_t samples_per_channel, in QuadToStereo()
122 size_t samples_per_channel, in DownmixChannels()
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| /aosp_15_r20/external/webrtc/modules/audio_processing/ |
| H A D | audio_processing_unittest.cc | 93 size_t samples_per_channel) { in MixStereoToMono()
100 size_t samples_per_channel) { in MixStereoToMono()
105 void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) { in CopyLeftToRightChannel()
111 void VerifyChannelsAreEqual(const int16_t* stereo, size_t samples_per_channel) { in VerifyChannelsAreEqual()
894 const size_t samples_per_channel = frame_.samples_per_channel; in TEST_F() local
963 const size_t samples_per_channel = frame_.samples_per_channel; in TEST_F() local
997 const size_t samples_per_channel = frame_.samples_per_channel; in TEST_F() local
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| H A D | audio_buffer.cc | 269 float* y) { in CopyFrom()
329 int16_t* y) { in CopyTo()
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| H A D | audio_processing_impl_locking_unittest.cc | 450 size_t samples_per_channel, in PopulateAudioFrame()
463 size_t samples_per_channel, in PopulateAudioFrame()
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| /aosp_15_r20/external/webrtc/modules/audio_processing/agc/ |
| H A D | agc_manager_direct.cc | 116 size_t samples_per_channel) { in ComputeClippedRatio()
543 size_t samples_per_channel = audio_buffer.num_frames(); in AnalyzePreProcess() local
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| /aosp_15_r20/external/webrtc/audio/ |
| H A D | audio_state_unittest.cc | 145 const int samples_per_channel = sample_rate_hz / 100; in Create10msTestData() local
159 const size_t samples_per_channel = audio_frame->samples_per_channel_; in ComputeChannelLevels() local
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| H A D | remix_resample.cc | 34 size_t samples_per_channel, in RemixAndResample()
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| /aosp_15_r20/external/webrtc/api/call/ |
| H A D | audio_sink.h | 37 size_t samples_per_channel; // Number of frames in the buffer. member
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| /aosp_15_r20/external/webrtc/modules/audio_processing/include/ |
| H A D | audio_frame_view.h | 45 int samples_per_channel() const { return channel_size_; } in samples_per_channel() function
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| /aosp_15_r20/external/webrtc/api/audio/ |
| H A D | audio_frame.cc | 49 size_t samples_per_channel, in UpdateFrame()
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