| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| audio_multi_vector.h | 109 virtual size_t Channels() const { return num_channels_; }
127 size_t num_channels_; member in class:webrtc::AudioMultiVector
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| audio_multi_vector_unittest.cc | 34 : num_channels_(GetParam()), // Get the test parameter.
35 interleaved_length_(num_channels_ * array_length()) {
36 array_interleaved_ = new int16_t[num_channels_ * array_length()];
53 for (size_t j = 1; j <= num_channels_; ++j) {
64 const size_t num_channels_; member in class:webrtc::AudioMultiVectorTest
73 AudioMultiVector vec1(num_channels_);
75 EXPECT_EQ(num_channels_, vec1.Channels());
79 AudioMultiVector vec2(num_channels_, initial_size);
81 EXPECT_EQ(num_channels_, vec2.Channels());
87 AudioMultiVector vec(num_channels_, array_length())
[all...] |
| merge.h | 38 num_channels_(num_channels),
43 expanded_(num_channels_) {
44 assert(num_channels_ > 0);
55 // must have |num_channels_| elements.
64 const size_t num_channels_; member in class:webrtc::Merge
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| background_noise.h | 128 size_t num_channels_; member in class:webrtc::BackgroundNoise
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| time_stretch.h | 42 num_channels_(static_cast<int>(num_channels)),
50 assert(num_channels_ > 0);
51 assert(static_cast<int>(master_channel_) < num_channels_);
89 const int num_channels_; member in class:webrtc::TimeStretch
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| neteq_stereo_unittest.cc | 51 : num_channels_(GetParam().num_channels),
69 input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
71 num_channels_];
72 output_multi_channel_ = new int16_t[kMaxBlockSize * num_channels_];
94 if (num_channels_ == 2) {
96 } else if (num_channels_ == 5) {
104 if (num_channels_ == 2) {
112 if (num_channels_ == 2) {
120 if (num_channels_ == 2) {
152 num_channels_,
241 const int num_channels_; member in class:webrtc::NetEqStereoTest
[all...] |
| /external/chromium_org/media/audio/sounds/ |
| wav_audio_handler.h | 50 uint16 num_channels_; member in class:media::WavAudioHandler
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| /external/chromium_org/media/cast/test/utility/ |
| audio_utility.h | 39 const int num_channels_; member in class:media::cast::TestAudioBusFactory
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| /external/chromium_org/media/base/android/ |
| audio_decoder_job.h | 58 int num_channels_; member in class:media::AudioDecoderJob
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| /external/chromium_org/third_party/webrtc/common_audio/resampler/include/ |
| push_resampler.h | 43 int num_channels_; member in class:webrtc::PushResampler
|
| /frameworks/ex/variablespeed/jni/ |
| ring_buffer.h | 104 int num_channels_; member in class:video_editing::RingBuffer
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| sola_time_scaler.h | 48 num_channels_ = num_channels;
65 int num_channels_; member in class:video_editing::SolaAnalyzer
125 int num_channels() const { return num_channels_; }
133 int num_channels_; // channel valence of audio stream member in class:video_editing::SolaTimeScaler
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| sola_time_scaler.cc | 37 num_frames *= num_channels_;
51 return num_frames * num_channels_;
60 num_channels_ = 0;
88 num_channels_ = num_channels;
162 (sample_rate_ * duration), num_channels_, 1); local
167 (sample_rate_ * ratio_ * duration), num_channels_, 2); local
176 analyzer_->Init(sample_rate_, num_channels_);
285 output_pointer + ((merge_offset + i) * num_channels_),
290 if (score == (num_overlap_frames_ * num_channels_)) {
296 output_pointer + ((merge_offset - i) * num_channels_),
[all...] |
| /external/webrtc/src/modules/audio_processing/ |
| audio_buffer.h | 61 int num_channels_; member in class:webrtc::AudioBuffer
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_generic_codec.h | 918 uint16_t num_channels_; member in class:webrtc::acm2::ACMGenericCodec
[all...] |
| /external/chromium_org/media/cast/audio_sender/ |
| audio_encoder.cc | 53 num_channels_(num_channels),
62 if (num_channels_ <= 0 || samples_per_frame_ <= 0 ||
64 samples_per_frame_ * num_channels_ > kMaxSamplesTimesChannelsPerFrame) {
111 DCHECK_EQ(audio_bus->channels(), num_channels_);
155 const int num_channels_; member in class:media::cast::AudioEncoder::ImplBase
237 float* dest = buffer_.get() + buffer_fill_offset * num_channels_ + ch;
238 for (; src < src_end; ++src, dest += num_channels_)
307 buffer_.get() + buffer_fill_offset * num_channels_);
312 out->resize(num_channels_ * samples_per_frame_ * sizeof(int16));
314 const int16* const src_end = src + num_channels_ * samples_per_frame_
[all...] |
| /external/chromium_org/media/cast/receiver/ |
| audio_decoder.cc | 31 num_channels_(num_channels),
34 if (num_channels_ <= 0 || sampling_rate <= 0 || sampling_rate % 100 != 0)
81 const int num_channels_; member in class:media::cast::AudioDecoder::ImplBase
138 audio_bus = AudioBus::Create(num_channels_, num_samples_decoded).Pass();
140 for (int ch = 0; ch < num_channels_; ++ch) {
142 const float* const src_end = src + num_samples_decoded * num_channels_;
144 for (; src < src_end; src += num_channels_, ++dest)
182 const int num_samples = len / sizeof(int16) / num_channels_;
189 const int num_elements = num_samples * num_channels_;
193 audio_bus = AudioBus::Create(num_channels_, num_samples).Pass()
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| audio_processing_impl.h | 67 num_channels_(num_channels) {}
72 num_channels_ = num_channels;
75 int num_channels() const { return num_channels_; }
78 int num_channels_; member in class:webrtc::AudioFormat
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| /external/chromium_org/third_party/webrtc/modules/interface/ |
| module_common_types.h | 642 * samples_per_channel_ * num_channels_
702 int num_channels_; member in class:webrtc::AudioFrame
730 num_channels_ = 0;
749 num_channels_ = num_channels;
772 num_channels_ = src.num_channels_;
776 const int length = samples_per_channel_ * num_channels_;
782 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
786 assert((num_channels_ > 0) && (num_channels_ < 3))
[all...] |
