| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| H A D | neteq_unittest.cc | 329 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
473 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
511 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
541 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
572 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
613 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
639 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
654 neteq_->GetAudio(
660 // If |pull_once| is true, GetAudio will be called once half-way through
669 neteq_->GetAudio(
[all...] |
| H A D | neteq_external_decoder_unittest.cc | 144 neteq_->GetAudio(kMaxBlockSize,
153 neteq_external_->GetAudio(kMaxBlockSize,
284 neteq_external_->GetAudio(kMaxBlockSize,
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| H A D | neteq_stereo_unittest.cc | 218 neteq_mono_->GetAudio(kMaxBlockSize, output_,
225 neteq_->GetAudio(kMaxBlockSize * num_channels_,
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| H A D | neteq_impl.h | 107 virtual int GetAudio(size_t max_length, int16_t* output_audio,
162 // Enables post-decode VAD. When enabled, GetAudio() will return
323 // GetAudio().
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| H A D | neteq_impl_unittest.cc | 468 neteq_->GetAudio(
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| H A D | neteq_impl.cc | 158 int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio, function in class:webrtc::NetEqImpl
162 LOG(LS_VERBOSE) << "GetAudio";
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| H A D | acm_receiver.h | 88 int GetAudio(int desired_freq_hz, AudioFrame* audio_frame);
238 // Gets the RTP timestamp of the last sample delivered by GetAudio().
311 // Get statistics of calls to GetAudio().
335 // Used in GetAudio, declared as member to avoid allocating every 10ms.
336 // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
353 // |late_packets_sync_stream_| is only used in GetAudio(). Both of these
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| H A D | acm_receiver_unittest.cc | 238 EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
274 ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
284 ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
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| H A D | acm_receiver_unittest_oldapi.cc | 242 EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
278 ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
288 ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
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| H A D | acm_receiver.cc | 343 int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { function in class:webrtc::AcmReceiver
384 if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
388 LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed.";
423 LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
443 LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
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| H A D | audio_coding_module_impl.cc | 1768 // GetAudio always returns 10 ms, at the requested sample rate.
1769 if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
| H A D | neteq.h | 156 virtual int GetAudio(size_t max_length, int16_t* output_audio,
226 // Enables post-decode VAD. When enabled, GetAudio() will return
233 // Gets the RTP timestamp for the last sample delivered by GetAudio().
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| H A D | neteq_performance_test.cc | 112 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
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| H A D | neteq_quality_test.cc | 292 int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
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| H A D | neteq_rtpplay.cc | 269 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
272 std::cerr << "GetAudio returned error code " <<
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