/external/chromium_org/third_party/webrtc/modules/audio_conference_mixer/source/ |
audio_frame_manipulator.h | 15 class AudioFrame; 17 // Updates the audioFrame's energy (based on its samples). 18 void CalculateEnergy(AudioFrame& audioFrame); 20 // Apply linear step function that ramps in/out the audio samples in audioFrame 21 void RampIn(AudioFrame& audioFrame); 22 void RampOut(AudioFrame& audioFrame);
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/external/chromium_org/third_party/webrtc/modules/utility/interface/ |
audio_frame_operations.h | 18 class AudioFrame; 33 static int MonoToStereo(AudioFrame* frame); 42 static int StereoToMono(AudioFrame* frame); 46 static void SwapStereoChannels(AudioFrame* frame); 49 static void Mute(AudioFrame& frame); 51 static int Scale(float left, float right, AudioFrame& frame); 53 static int ScaleWithSat(float scale, AudioFrame& frame);
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
call_statistics.cc | 19 void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type) { 22 case AudioFrame::kNormalSpeech: { 26 case AudioFrame::kPLC: { 30 case AudioFrame::kCNG: { 34 case AudioFrame::kPLCCNG: { 38 case AudioFrame::kUndefined: {
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call_statistics.h | 41 void DecodedByNetEq(AudioFrame::SpeechType speech_type);
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call_statistics_unittest.cc | 36 call_stats.DecodedByNetEq(AudioFrame::kNormalSpeech); 37 call_stats.DecodedByNetEq(AudioFrame::kPLC); 38 call_stats.DecodedByNetEq(AudioFrame::kPLCCNG); 39 call_stats.DecodedByNetEq(AudioFrame::kCNG);
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/external/chromium_org/third_party/webrtc/voice_engine/ |
utility.h | 23 class AudioFrame; 31 void RemixAndResample(const AudioFrame& src_frame, 33 AudioFrame* dst_frame); 52 AudioFrame* dst_af);
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level_indicator.h | 19 class AudioFrame; 38 void ComputeLevel(const AudioFrame& audioFrame);
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voe_external_media_impl.h | 34 AudioFrame* frame);
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utility_unittest.cc | 43 AudioFrame src_frame_; 44 AudioFrame dst_frame_; 45 AudioFrame golden_frame_; 51 void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) { 62 void SetMonoFrame(AudioFrame* frame, float data) { 68 void SetStereoFrame(AudioFrame* frame, float left, float right, 81 void SetStereoFrame(AudioFrame* frame, float left, float right) { 85 void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) { 94 float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame [all...] |
output_mixer.h | 68 AudioFrame* audioFrame); 92 const AudioFrame& generalAudioFrame, 93 const AudioFrame** uniqueAudioFrames, 131 AudioFrame _audioFrame;
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/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
audioframe.h | 33 class AudioFrame { 35 AudioFrame() 41 AudioFrame(int16* audio, size_t audio_length, int sample_freq, bool stereo)
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fakemediaprocessor.h | 36 class AudioFrame; 50 AudioFrame* frame) {
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voiceprocessor.h | 33 #include "talk/media/base/audioframe.h" 52 AudioFrame* frame) = 0;
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
coder.h | 20 class AudioFrame; 36 int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz, 39 int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz); 41 int32_t Encode(const AudioFrame& audio, int8_t* encodedData,
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audio_frame_operations.cc | 25 int AudioFrameOperations::MonoToStereo(AudioFrame* frame) { 29 if ((frame->samples_per_channel_ * 2) >= AudioFrame::kMaxDataSizeSamples) { 34 int16_t data_copy[AudioFrame::kMaxDataSizeSamples]; 51 int AudioFrameOperations::StereoToMono(AudioFrame* frame) { 62 void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) { 72 void AudioFrameOperations::Mute(AudioFrame& frame) { 77 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) { 91 int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
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audio_frame_operations_unittest.cc | 27 AudioFrame frame_; 30 void SetFrameData(AudioFrame* frame, int16_t left, int16_t right) { 37 void SetFrameData(AudioFrame* frame, int16_t data) { 43 void VerifyFramesAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) { 57 frame_.samples_per_channel_ = AudioFrame::kMaxDataSizeSamples; 65 AudioFrame temp_frame; 69 AudioFrame stereo_frame; 90 AudioFrame temp_frame; 94 AudioFrame mono_frame [all...] |
coder.cc | 54 int32_t AudioCoder::Decode(AudioFrame& decodedAudio, 74 int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio, 80 int32_t AudioCoder::Encode(const AudioFrame& audio, 86 AudioFrame audioFrame; 87 audioFrame.CopyFrom(audio); 88 audioFrame.timestamp_ = _encodeTimestamp; 89 _encodeTimestamp += audioFrame.samples_per_channel_; 94 if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1 [all...] |
/external/webrtc/src/modules/audio_processing/ |
audio_buffer.h | 44 void set_activity(AudioFrame::VADActivity activity); 45 AudioFrame::VADActivity activity() const; 49 void DeinterleaveFrom(AudioFrame* audioFrame); 50 void InterleaveTo(AudioFrame* audioFrame) const; 53 void InterleaveTo(AudioFrame* frame, bool data_changed) const; 69 AudioFrame::VADActivity activity_;
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/external/chromium_org/third_party/webrtc/modules/audio_conference_mixer/interface/ |
audio_conference_mixer_defines.h | 24 // The implementation of this function should update audioFrame with new 28 virtual int32_t GetAudioFrame(const int32_t id, AudioFrame& audioFrame) = 0; 33 // This function specifies the sampling frequency needed for the AudioFrame 79 // Note that uniqueAudioFrames is an array of AudioFrame pointers with the 82 const AudioFrame& generalAudioFrame, 83 const AudioFrame** uniqueAudioFrames,
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/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
audio_buffer.h | 79 void set_activity(AudioFrame::VADActivity activity); 80 AudioFrame::VADActivity activity() const; 83 void DeinterleaveFrom(AudioFrame* audioFrame); 84 void InterleaveTo(AudioFrame* audioFrame) const; 87 void InterleaveTo(AudioFrame* frame, bool data_changed) const; 114 AudioFrame::VADActivity activity_;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
audio_sink.h | 34 bool WriteAudioFrame(const AudioFrame& audio_frame) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
PCMFile.h | 37 int32_t Read10MsData(AudioFrame& audio_frame); 40 void Write10MsData(AudioFrame& audio_frame);
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/external/chromium_org/third_party/webrtc/voice_engine/include/ |
voe_external_media.h | 18 class AudioFrame; 69 AudioFrame* frame) = 0;
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/external/webrtc/src/modules/interface/ |
module_common_types.h | 687 * AudioFrame class 689 * The AudioFrame class holds up to 60 ms wideband 710 class AudioFrame 730 AudioFrame(); 731 virtual ~AudioFrame(); 745 AudioFrame& Append(const AudioFrame& rhs); 749 AudioFrame& operator=(const AudioFrame& rhs); 750 AudioFrame& operator>>=(const WebRtc_Word32 rhs) [all...] |
/external/chromium_org/content/renderer/media/ |
media_stream_audio_processor.h | 32 class AudioFrame; 146 int ProcessData(webrtc::AudioFrame* audio_frame, 162 // AudioFrame used to hold the output of |capture_converter_|. 163 webrtc::AudioFrame capture_frame_; 169 // AudioFrame used to hold the output of |render_converter_|. 170 webrtc::AudioFrame render_frame_;
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