| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| common.h | 41 ChannelBuffer(int samples_per_channel, int num_channels)
42 : data_(new T[samples_per_channel * num_channels]),
44 samples_per_channel_(samples_per_channel),
49 ChannelBuffer(const T* data, int samples_per_channel, int num_channels)
50 : data_(new T[samples_per_channel * num_channels]),
52 samples_per_channel_(samples_per_channel),
58 ChannelBuffer(const T* const* channels, int samples_per_channel,
60 : data_(new T[samples_per_channel * num_channels]),
62 samples_per_channel_(samples_per_channel),
92 int samples_per_channel() const { return samples_per_channel_; function in class:webrtc::ChannelBuffer
[all...] |
| audio_buffer.h | 54 int samples_per_channel() const;
96 int samples_per_channel,
98 void CopyTo(int samples_per_channel,
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| audio_processing_impl.cc | 174 render_audio_.reset(new AudioBuffer(rev_in_format_.samples_per_channel(),
176 rev_proc_format_.samples_per_channel(),
178 rev_proc_format_.samples_per_channel()));
179 capture_audio_.reset(new AudioBuffer(fwd_in_format_.samples_per_channel(),
181 fwd_proc_format_.samples_per_channel(),
183 fwd_out_format_.samples_per_channel()));
345 int samples_per_channel,
362 if (samples_per_channel != fwd_in_format_.samples_per_channel()) {
371 sizeof(float) * fwd_in_format_.samples_per_channel();
[all...] |
| audio_buffer.cc | 55 int samples_per_channel) {
56 for (int i = 0; i < samples_per_channel; ++i) {
62 int samples_per_channel) {
63 for (int i = 0; i < samples_per_channel; ++i) {
78 IFChannelBuffer(int samples_per_channel, int num_channels)
80 ibuf_(samples_per_channel, num_channels),
82 fbuf_(samples_per_channel, num_channels) {}
200 int samples_per_channel,
202 assert(samples_per_channel == input_samples_per_channel_);
238 void AudioBuffer::CopyTo(int samples_per_channel,
393 int AudioBuffer::samples_per_channel() const { function in class:webrtc::AudioBuffer
[all...] |
| audio_processing_impl.h | 56 int samples_per_channel() const { return samples_per_channel_; } function in class:webrtc::AudioRate
107 int samples_per_channel,
115 int samples_per_channel,
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| level_estimator_impl.cc | 34 rms_level->Process(audio->data(i), audio->samples_per_channel());
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| /external/chromium_org/third_party/webrtc/modules/utility/interface/ |
| audio_frame_operations.h | 29 static void MonoToStereo(const int16_t* src_audio, int samples_per_channel,
38 static void StereoToMono(const int16_t* src_audio, int samples_per_channel,
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| /external/chromium_org/third_party/webrtc/common_audio/include/ |
| audio_util.h | 61 // |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
64 void Deinterleave(const T* interleaved, int samples_per_channel,
69 for (int j = 0; j < samples_per_channel; ++j) {
78 // (|samples_per_channel| * |num_channels|).
80 void Interleave(const T* const* deinterleaved, int samples_per_channel,
85 for (int j = 0; j < samples_per_channel; ++j) {
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| /external/chromium_org/third_party/webrtc/voice_engine/ |
| utility.h | 45 int samples_per_channel,
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| transmit_mixer_unittest.cc | 23 int16_t audio[], int samples_per_channel,
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| utility.cc | 75 int samples_per_channel,
83 assert(samples_per_channel <= kMaxMonoDataSizeSamples);
95 AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
111 const int in_length = samples_per_channel * num_channels;
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| /external/chromium_org/third_party/webrtc/voice_engine/include/mock/ |
| fake_voe_external_media.h | 53 int samples_per_channel, int sample_rate_hz,
55 const int length = samples_per_channel * num_channels;
66 it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
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| /external/chromium_org/third_party/webrtc/modules/utility/source/ |
| audio_frame_operations.cc | 17 int samples_per_channel,
19 for (int i = 0; i < samples_per_channel; i++) {
44 int samples_per_channel,
46 for (int i = 0; i < samples_per_channel; i++) {
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| /external/webrtc/src/modules/audio_processing/ |
| audio_buffer.h | 25 AudioBuffer(int max_num_channels, int samples_per_channel);
29 int samples_per_channel() const;
|
| audio_buffer.cc | 25 int16_t* out, int samples_per_channel) {
27 for (int i = 0; i < samples_per_channel; i++) {
65 int samples_per_channel)
71 samples_per_channel_(samples_per_channel),
72 samples_per_split_channel_(samples_per_channel),
181 int AudioBuffer::samples_per_channel() const { function in class:webrtc::AudioBuffer
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| level_estimator_impl.cc | 100 level->ProcessMuted(audio->samples_per_channel());
110 level->Process(mixed_data, audio->samples_per_channel());
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_performance_test.cc | 111 int samples_per_channel; local
112 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
117 assert(samples_per_channel == kSampRateHz * 10 / 1000);
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_external_decoder_unittest.cc | 141 int samples_per_channel; local
146 &samples_per_channel,
150 EXPECT_EQ(output_size_samples_, samples_per_channel);
155 &samples_per_channel,
159 EXPECT_EQ(output_size_samples_, samples_per_channel);
280 int samples_per_channel; variable
286 &samples_per_channel,
290 EXPECT_EQ(output_size_samples_, samples_per_channel);
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| neteq_unittest.cc | 826 int samples_per_channel; local
829 &samples_per_channel, &num_channels, &type));
859 int samples_per_channel; local
861 &samples_per_channel,
913 int samples_per_channel = 0; local
1141 int samples_per_channel; local
1219 int samples_per_channel; local
1288 int samples_per_channel; local
[all...] |
| neteq_stereo_unittest.cc | 215 int samples_per_channel; local
219 &samples_per_channel, &num_channels,
222 EXPECT_EQ(output_size_samples_, samples_per_channel);
227 &samples_per_channel, &num_channels,
230 EXPECT_EQ(output_size_samples_, samples_per_channel);
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| neteq_impl_unittest.cc | 463 int samples_per_channel; local
469 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
470 ASSERT_EQ(kMaxOutputSize, samples_per_channel);
482 EXPECT_EQ(rtp_header.header.timestamp + output[samples_per_channel - 1],
494 EXPECT_EQ(kPayloadLengthSamples - output[samples_per_channel - 1],
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| /external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
| test_utils.h | 63 size_t samples_per_channel,
67 size_t length = num_channels * samples_per_channel;
69 Interleave(data, samples_per_channel, num_channels, buffer.get());
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| process_test.cc | 165 int samples_per_channel = sample_rate_hz / 100; local
203 samples_per_channel = sample_rate_hz / 100;
612 samples_per_channel = msg.sample_rate() / 100;
617 near_frame.samples_per_channel_ = samples_per_channel;
622 primary_cb.reset(new ChannelBuffer<float>(samples_per_channel,
699 ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
789 const size_t samples_per_channel = output_sample_rate / 100; local
795 apm->num_output_channels() * samples_per_channel,
803 samples_per_channel,
845 far_frame.samples_per_channel_ = samples_per_channel;
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_receiver.cc | 346 int samples_per_channel; local
386 &samples_per_channel,
406 current_sample_rate_hz_ = samples_per_channel * 100;
415 samples_per_channel =
422 if (samples_per_channel < 0) {
428 memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
435 samples_per_channel =
442 if (samples_per_channel < 0) {
446 memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
452 audio_frame->samples_per_channel_ = samples_per_channel;
[all...] |
| /external/webrtc/src/modules/audio_processing/test/ |
| process_test.cc | 162 int samples_per_channel = sample_rate_hz / 100; local
199 samples_per_channel = sample_rate_hz / 100;
548 samples_per_channel = msg.sample_rate() / 100;
550 far_frame._payloadDataLengthInSamples = samples_per_channel;
553 near_frame._payloadDataLengthInSamples = samples_per_channel;
571 ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
603 ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
662 size_t size = samples_per_channel * near_frame._audioChannel;
704 far_frame._payloadDataLengthInSamples = samples_per_channel;
707 near_frame._payloadDataLengthInSamples = samples_per_channel;
[all...] |
