| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_external_decoder_test.cc | 53 neteq_->GetAudio(max_length,
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| neteq_performance_test.cc | 114 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
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| neteq_quality_test.cc | 399 int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
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| neteq_rtpplay.cc | 614 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
617 std::cerr << "GetAudio returned error code " <<
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| neteq_unittest.cc | 429 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
613 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
658 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
689 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
730 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
758 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
773 neteq_->GetAudio(
779 // If |pull_once| is true, GetAudio will be called once half-way through
788 neteq_->GetAudio(
823 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len
[all...] |
| neteq_impl_unittest.cc | 473 neteq_->GetAudio(
552 neteq_->GetAudio(
583 neteq_->GetAudio(
628 neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
651 neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
756 neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
766 neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
785 neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
[all...] |
| neteq_stereo_unittest.cc | 219 neteq_mono_->GetAudio(kMaxBlockSize, output_,
226 neteq_->GetAudio(kMaxBlockSize * num_channels_,
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| neteq_external_decoder_unittest.cc | 194 neteq_internal_->GetAudio(kMaxBlockSize,
313 // timestamp happens, NetEq will do Expand during one GetAudio call. In the
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| neteq_impl.h | 107 int GetAudio(size_t max_length,
160 // Enables post-decode VAD. When enabled, GetAudio() will return
331 // GetAudio().
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| neteq_impl.cc | 153 int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
156 TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
667 // Update the codec on the next GetAudio call.
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_receiver.h | 89 int GetAudio(int desired_freq_hz, AudioFrame* audio_frame);
219 // Gets the RTP timestamp of the last sample delivered by GetAudio().
274 // Get statistics of calls to GetAudio().
289 // Used in GetAudio, declared as member to avoid allocating every 10ms.
290 // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
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| acm_receiver_unittest_oldapi.cc | 286 EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
310 ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
320 ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
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| acm_receiver.cc | 213 int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
222 if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
227 LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
244 LOG(LERROR) << "AcmReceiver::GetAudio - "
260 LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
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| audio_coding_module_impl.cc | 684 // GetAudio always returns 10 ms, at the requested sample rate.
685 if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
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| /external/webrtc/webrtc/modules/audio_coding/neteq/include/ |
| neteq.h | 173 virtual int GetAudio(size_t max_length, int16_t* output_audio,
250 // Enables post-decode VAD. When enabled, GetAudio() will return
257 // Gets the RTP timestamp for the last sample delivered by GetAudio().
261 // Returns the sample rate in Hz of the audio produced in the last GetAudio
262 // call. If GetAudio has not been called yet, the configured sample rate
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