/external/webrtc/webrtc/modules/audio_coding/test/ |
SpatialAudio.cc | 153 AudioFrame audioFrame; 161 _inFile.Read10MsData(audioFrame); 162 for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) { 163 audioFrame.data_[n] = (int16_t) floor( 164 audioFrame.data_[n] * leftPanning + 0.5); 166 CHECK_ERROR(_acmLeft->Add10MsData(audioFrame)); 168 for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) { 169 audioFrame.data_[n] = (int16_t) floor( 170 audioFrame.data_[n] * rightToLeftRatio + 0.5) [all...] |
EncodeDecodeTest.h | 65 AudioFrame _audioFrame;
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PCMFile.cc | 116 int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) { 146 void PCMFile::Write10MsData(AudioFrame& audio_frame) {
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target_delay_unittest.cc | 151 AudioFrame frame;
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APITest.cc | 325 AudioFrame audioFrame; 326 if (_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame) < 0) { 337 _outFileA.Write10MsData(audioFrame); 345 AudioFrame audioFrame; 346 if (_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame) < 0) { 358 _outFileB.Write10MsData(audioFrame); 366 AudioFrame audioFrame; [all...] |
TestVADDTX.cc | 105 AudioFrame audio_frame;
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delay_test.cc | 178 AudioFrame audio_frame;
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insert_packet_with_timing.cc | 265 AudioFrame frame_;
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opus_test.cc | 210 AudioFrame audio_frame;
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/external/webrtc/webrtc/modules/audio_processing/test/ |
test_utils.h | 91 void SetFrameSampleRate(AudioFrame* frame, 97 AudioFrame* frame,
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audio_processing_unittest.cc | 82 void ConvertToFloat(const AudioFrame& frame, ChannelBuffer<float>* cb) { 129 void SetFrameTo(AudioFrame* frame, int16_t value) { 136 void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) { 144 void ScaleFrame(AudioFrame* frame, float scale) { 151 bool FrameDataAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) { 196 int16_t MaxAudioFrame(const AudioFrame& frame) { 366 bool ReadFrame(FILE* file, AudioFrame* frame); 367 bool ReadFrame(FILE* file, AudioFrame* frame, ChannelBuffer<float>* cb); 368 void ReadFrameWithRewind(FILE* file, AudioFrame* frame) [all...] |
test_utils.cc | 112 void SetFrameSampleRate(AudioFrame* frame,
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/external/webrtc/webrtc/modules/audio_processing/include/ |
audio_processing.h | 32 class AudioFrame; 317 virtual int ProcessStream(AudioFrame* frame) = 0; 365 virtual int AnalyzeReverseStream(AudioFrame* frame) = 0; 369 virtual int ProcessReverseStream(AudioFrame* frame) = 0; [all...] |
mock_audio_processing.h | 214 int(AudioFrame* frame)); 229 int(AudioFrame* frame)); 230 MOCK_METHOD1(ProcessReverseStream, int(AudioFrame* frame));
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/external/webrtc/webrtc/modules/audio_processing/ |
audio_processing_impl.h | 67 int ProcessStream(AudioFrame* frame) override; 88 int AnalyzeReverseStream(AudioFrame* frame) override; 89 int ProcessReverseStream(AudioFrame* frame) override;
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
acm_receiver_unittest_oldapi.cc | 99 AudioFrame frame; 279 AudioFrame frame; 306 AudioFrame frame; 312 EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_); 322 EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
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audio_coding_module_unittest_oldapi.cc | 173 static_assert(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples, 208 AudioFrame audio_frame; 233 AudioFrame input_frame_; 298 AudioFrame audio_frame; 310 AudioFrame audio_frame; 812 AudioFrame audio_frame; [all...] |
/external/webrtc/webrtc/voice_engine/ |
channel.h | 308 int UpdateRxVadDetection(AudioFrame& audioFrame); 401 int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override; 447 uint32_t Demultiplex(const AudioFrame& audioFrame); 483 int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); 517 AudioFrame _audioFrame; 585 AudioFrame::SpeechType _outputSpeechType [all...] |
output_mixer.cc | 28 const AudioFrame& generalAudioFrame, 29 const AudioFrame** uniqueAudioFrames, 467 AudioFrame* frame) { 525 AudioFrame frame;
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voe_base_impl.h | 143 AudioFrame audioFrame_;
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transmit_mixer.h | 200 AudioFrame _audioFrame;
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voe_external_media_impl.cc | 126 AudioFrame* frame) {
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/external/webrtc/webrtc/tools/agc/ |
activity_metric.cc | 58 static void DitherSilence(AudioFrame* frame) { 96 int AddAudio(const AudioFrame& frame, double p_video, 226 webrtc::AudioFrame frame;
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/external/webrtc/webrtc/modules/audio_processing/agc/ |
agc_unittest.cc | 58 AudioFrame frame;
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/external/webrtc/webrtc/modules/utility/source/ |
file_player_impl.cc | 109 AudioFrame unresampledAudioFrame;
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