| /external/chromium_org/third_party/WebKit/Source/core/platform/audio/ |
| AudioChannel.cpp | 33 #include "core/platform/audio/AudioChannel.h"
44 void AudioChannel::resizeSmaller(size_t newLength)
51 void AudioChannel::scale(float scale)
59 void AudioChannel::copyFrom(const AudioChannel* sourceChannel)
73 void AudioChannel::copyFromRange(const AudioChannel* sourceChannel, unsigned startFrame, unsigned endFrame)
103 void AudioChannel::sumFrom(const AudioChannel* sourceChannel)
119 float AudioChannel::maxAbsValue() cons
[all...] |
| AudioChannel.h | 37 // An AudioChannel represents a buffer of non-interleaved floating-point audio samples.
39 class AudioChannel {
40 WTF_MAKE_NONCOPYABLE(AudioChannel);
45 AudioChannel(float* storage, size_t length)
53 explicit AudioChannel(size_t length)
62 AudioChannel()
118 void copyFrom(const AudioChannel* sourceChannel);
121 void copyFromRange(const AudioChannel* sourceChannel, unsigned startFrame, unsigned endFrame);
124 void sumFrom(const AudioChannel* sourceChannel);
|
| Reverb.cpp | 123 AudioChannel* channel = impulseResponseBuffer->channel(i);
154 AudioChannel* destinationChannelL = destinationBus->channel(0);
155 const AudioChannel* sourceChannelL = sourceBus->channel(0);
164 const AudioChannel* sourceChannelR = sourceBus->channel(1);
165 AudioChannel* destinationChannelR = destinationBus->channel(1);
171 AudioChannel* destinationChannel = destinationBus->channel(i);
179 AudioChannel* destinationChannelR = destinationBus->channel(1);
190 const AudioChannel* sourceChannelR = sourceBus->channel(1);
191 AudioChannel* destinationChannelR = destinationBus->channel(1);
193 AudioChannel* tempChannelL = m_tempBuffer->channel(0)
[all...] |
| HRTFKernel.h | 42 class AudioChannel;
52 // Note: this is destructive on the passed in AudioChannel.
54 static PassRefPtr<HRTFKernel> create(AudioChannel* channel, size_t fftSize, float sampleRate)
76 PassOwnPtr<AudioChannel> createImpulseResponse();
79 // Note: this is destructive on the passed in AudioChannel.
80 HRTFKernel(AudioChannel*, size_t fftSize, float sampleRate);
|
| HRTFKernel.cpp | 36 #include "core/platform/audio/AudioChannel.h"
44 // Takes the input AudioChannel as an input impulse response and calculates the average group delay.
47 // the length of the passed in AudioChannel must be a power of 2.
48 static float extractAverageGroupDelay(AudioChannel* channel, size_t analysisFFTSize)
71 HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate)
100 PassOwnPtr<AudioChannel> HRTFKernel::createImpulseResponse()
102 OwnPtr<AudioChannel> channel = adoptPtr(new AudioChannel(fftSize()));
|
| ReverbConvolver.h | 45 class AudioChannel;
53 ReverbConvolver(AudioChannel* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads);
56 void process(const AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess);
|
| AudioBus.h | 32 #include "core/platform/audio/AudioChannel.h"
77 AudioChannel* channel(unsigned channel) { return m_channels[channel].get(); }
78 const AudioChannel* channel(unsigned channel) const { return const_cast<AudioBus*>(this)->m_channels[channel].get(); }
79 AudioChannel* channelByType(unsigned type);
80 const AudioChannel* channelByType(unsigned type) const;
161 Vector<OwnPtr<AudioChannel> > m_channels;
|
| AudioBus.cpp | 69 PassOwnPtr<AudioChannel> channel = allocate ? adoptPtr(new AudioChannel(length)) : adoptPtr(new AudioChannel(0, length));
100 AudioChannel* AudioBus::channelByType(unsigned channelType)
154 const AudioChannel* AudioBus::channelByType(unsigned type) const
198 const AudioChannel* channel = this->channel(i);
279 const AudioChannel* sourceChannel = sourceBus.channel(0);
321 const AudioChannel* sourceChannel = sourceBus.channel(0);
|
| ReverbConvolver.cpp | 62 ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads)
181 void ReverbConvolver::process(const AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess)
|
| HRTFElevation.cpp | 173 AudioChannel* leftEarImpulseResponse = response->channel(AudioBus::ChannelLeft);
174 AudioChannel* rightEarImpulseResponse = response->channel(AudioBus::ChannelRight);
193 AudioChannel* leftEarImpulseResponse = impulseResponse->channelByType(AudioBus::ChannelLeft);
194 AudioChannel* rightEarImpulseResponse = impulseResponse->channelByType(AudioBus::ChannelRight);
|
| HRTFPanner.cpp | 157 const AudioChannel* inputChannelL = inputBus->channelByType(AudioBus::ChannelLeft);
158 const AudioChannel* inputChannelR = numInputChannels > 1 ? inputBus->channelByType(AudioBus::ChannelRight) : 0;
|
| /external/webrtc/src/modules/audio_processing/ |
| audio_buffer.h | 20 struct AudioChannel;
73 scoped_array<AudioChannel> channels_;
75 scoped_array<AudioChannel> mixed_channels_;
77 scoped_array<AudioChannel> mixed_low_pass_channels_;
78 scoped_array<AudioChannel> low_pass_reference_channels_;
|
| audio_buffer.cc | 36 struct AudioChannel {
37 AudioChannel() {
83 channels_.reset(new AudioChannel[max_num_channels_]);
84 mixed_channels_.reset(new AudioChannel[max_num_channels_]);
85 mixed_low_pass_channels_.reset(new AudioChannel[max_num_channels_]);
87 low_pass_reference_channels_.reset(new AudioChannel[max_num_channels_]);
|
| /frameworks/base/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/functional/videoeditor/ |
| MediaPropertiesTest.java | 75 int audioSamplingFrequency, int audioChannel, int audioBitrate,
99 assertEquals("Audio Channels " + mvi.getAudioChannels(), audioChannel,
105 int audioSamplingFrequency, int audioChannel, int audioBitrate,
115 assertEquals("Audio Channels " + aT.getAudioChannels(), audioChannel,
146 final int audioChannel = 2;
158 audioSamplingFrequency, audioChannel, audioBitrate, mvi);
178 final int audioChannel = 1;
190 audioSamplingFrequency, audioChannel, audioBitrate, mvi);
209 final int audioChannel = 2;
221 audioSamplingFrequency, audioChannel, audioBitrate, mvi)
[all...] |
| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| ChannelMergerNode.cpp | 86 AudioChannel* inputChannel = input->bus()->channel(j);
87 AudioChannel* outputChannel = output->bus()->channel(outputChannelIndex);
|
| /external/webrtc/src/modules/interface/ |
| module_common_types.h | 741 const WebRtc_UWord8 audioChannel = 1,
800 const WebRtc_UWord8 audioChannel,
810 _audioChannel = audioChannel;
814 (audioChannel > 2) || (audioChannel < 1))
|
| /external/smack/asmack-master/jingle/ |
| 60-remove-jingle_mediaimpl.patch | [all...] |
| /frameworks/base/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/functional/mediarecorder/ |
| MediaRecorderTest.java | 144 Log.v(TAG, "audioChannel : " + audioChannels);
|
| /frameworks/base/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/stress/ |
| VideoEditorStressTest.java | 676 final int audioChannel = 2;
726 audioChannel, mediaItem3.getAudioChannels());
[all...] |
| /external/chromium_org/third_party/WebKit/Source/core/ |
| webcore_platform.target.darwin-arm.mk | 76 third_party/WebKit/Source/core/platform/audio/AudioChannel.cpp \
[all...] |
| webcore_platform.target.darwin-mips.mk | 76 third_party/WebKit/Source/core/platform/audio/AudioChannel.cpp \
[all...] |
| webcore_platform.target.darwin-x86.mk | 76 third_party/WebKit/Source/core/platform/audio/AudioChannel.cpp \
[all...] |
| webcore_platform.target.linux-arm.mk | 76 third_party/WebKit/Source/core/platform/audio/AudioChannel.cpp \
[all...] |
| webcore_platform.target.linux-mips.mk | 76 third_party/WebKit/Source/core/platform/audio/AudioChannel.cpp \
[all...] |
| webcore_platform.target.linux-x86.mk | 76 third_party/WebKit/Source/core/platform/audio/AudioChannel.cpp \
[all...] |
