| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_unittest.cc | 330 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
473 ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
491 ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
508 ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
595 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
633 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
663 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
694 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
735 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
761 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len
[all...] |
| neteq_external_decoder_unittest.cc | 164 neteq_->GetAudio(kMaxBlockSize, output_,
171 neteq_external_->GetAudio(kMaxBlockSize, output_external_,
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| neteq_stereo_unittest.cc | 218 neteq_mono_->GetAudio(kMaxBlockSize, output_,
225 neteq_->GetAudio(kMaxBlockSize * num_channels_,
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| 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
328 // GetAudio().
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| neteq_impl_unittest.cc | 468 neteq_->GetAudio(
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| neteq_impl.cc | 156 int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
160 LOG(LS_VERBOSE) << "GetAudio";
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_receiver.h | 89 int GetAudio(int desired_freq_hz, AudioFrame* audio_frame);
246 // Gets the RTP timestamp of the last sample delivered by GetAudio().
325 // Get statistics of calls to GetAudio().
349 // Used in GetAudio, declared as member to avoid allocating every 10ms.
350 // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
368 // |late_packets_sync_stream_| is only used in GetAudio(). Both of these
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| acm_receiver_unittest.cc | 242 EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
293 ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
303 ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
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| acm_receiver.cc | 354 int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
398 if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
402 LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed.";
438 LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
458 LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
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| audio_coding_module_impl.cc | [all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
| neteq.h | 154 virtual int GetAudio(size_t max_length, int16_t* output_audio,
224 // Enables post-decode VAD. When enabled, GetAudio() will return
231 // 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/ |
| neteq_performance_test.cc | 112 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
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| neteq_quality_test.cc | 83 int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
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| neteq_rtpplay.cc | 276 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
279 std::cerr << "GetAudio returned error code " <<
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