| /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);
|
| 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;
|
| /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)
|
| fakemediaprocessor.h | 36 class AudioFrame;
50 AudioFrame* frame) {
|
| 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,
|
| 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,
|
| /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;
|
| /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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