diff options
Diffstat (limited to 'modules/audio_coding/main/acm2/acm_receiver.cc')
| -rw-r--r-- | modules/audio_coding/main/acm2/acm_receiver.cc | 109 |
1 files changed, 56 insertions, 53 deletions
diff --git a/modules/audio_coding/main/acm2/acm_receiver.cc b/modules/audio_coding/main/acm2/acm_receiver.cc index 0c7e6c72..07447541 100644 --- a/modules/audio_coding/main/acm2/acm_receiver.cc +++ b/modules/audio_coding/main/acm2/acm_receiver.cc @@ -122,11 +122,14 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config) last_audio_decoder_(-1), // Invalid value. previous_audio_activity_(AudioFrame::kVadPassive), current_sample_rate_hz_(config.neteq_config.sample_rate_hz), + audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), + last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]), nack_(), nack_enabled_(false), neteq_(NetEq::Create(config.neteq_config)), vad_enabled_(true), clock_(config.clock), + resampled_last_output_frame_(true), av_sync_(false), initial_delay_manager_(), missing_packets_sync_stream_(), @@ -143,6 +146,9 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config) neteq_->EnableVad(); else neteq_->DisableVad(); + + memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples); + memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples); } AcmReceiver::~AcmReceiver() { @@ -342,7 +348,6 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header, int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { enum NetEqOutputType type; - int16_t* ptr_audio_buffer = audio_frame->data_; int samples_per_channel; int num_channels; bool return_silence = false; @@ -359,18 +364,6 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { initial_delay_manager_->LatePackets(timestamp_now, late_packets_sync_stream_.get()); } - - if (!return_silence) { - // This is our initial guess regarding whether a resampling will be - // required. It is based on previous sample rate of netEq. Most often, - // this is a correct guess, however, in case that incoming payload changes - // the resampling might might be needed. By doing so, we avoid an - // unnecessary memcpy(). - if (desired_freq_hz != -1 && - current_sample_rate_hz_ != desired_freq_hz) { - ptr_audio_buffer = audio_buffer_; - } - } } // If |late_packets_sync_stream_| is allocated then we have been in AV-sync @@ -381,17 +374,19 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { return 0; } + // Accessing members, take the lock. + CriticalSectionScoped lock(crit_sect_.get()); + + // Always write the output to |audio_buffer_| first. if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples, - ptr_audio_buffer, + audio_buffer_.get(), &samples_per_channel, - &num_channels, &type) != NetEq::kOK) { + &num_channels, + &type) != NetEq::kOK) { LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed."; return -1; } - // Accessing members, take the lock. - CriticalSectionScoped lock(crit_sect_.get()); - // Update NACK. int decoded_sequence_num = 0; uint32_t decoded_timestamp = 0; @@ -409,45 +404,53 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { bool need_resampling = (desired_freq_hz != -1) && (current_sample_rate_hz_ != desired_freq_hz); - if (ptr_audio_buffer == audio_buffer_) { - // Data is written to local buffer. - if (need_resampling) { - samples_per_channel = - resampler_.Resample10Msec(audio_buffer_, - current_sample_rate_hz_, - desired_freq_hz, - num_channels, - AudioFrame::kMaxDataSizeSamples, - audio_frame->data_); - if (samples_per_channel < 0) { - LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed."; - return -1; - } - } else { - // We might end up here ONLY if codec is changed. - memcpy(audio_frame->data_, audio_buffer_, samples_per_channel * - num_channels * sizeof(int16_t)); + if (need_resampling && !resampled_last_output_frame_) { + // Prime the resampler with the last frame. + int16_t temp_output[AudioFrame::kMaxDataSizeSamples]; + samples_per_channel = + resampler_.Resample10Msec(last_audio_buffer_.get(), + current_sample_rate_hz_, + desired_freq_hz, + num_channels, + AudioFrame::kMaxDataSizeSamples, + temp_output); + if (samples_per_channel < 0) { + LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") + << "Resampling last_audio_buffer_ failed."; + return -1; } - } else { - // Data is written into |audio_frame|. - if (need_resampling) { - // We might end up here ONLY if codec is changed. - samples_per_channel = - resampler_.Resample10Msec(audio_frame->data_, - current_sample_rate_hz_, - desired_freq_hz, - num_channels, - AudioFrame::kMaxDataSizeSamples, - audio_buffer_); - if (samples_per_channel < 0) { - LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed."; - return -1; - } - memcpy(audio_frame->data_, audio_buffer_, samples_per_channel * - num_channels * sizeof(int16_t)); + } + + // The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either + // through resampling, or through straight memcpy. + // TODO(henrik.lundin) Glitches in the output may appear if the output rate + // from NetEq changes. See WebRTC issue 3923. + if (need_resampling) { + samples_per_channel = + resampler_.Resample10Msec(audio_buffer_.get(), + current_sample_rate_hz_, + desired_freq_hz, + num_channels, + AudioFrame::kMaxDataSizeSamples, + audio_frame->data_); + if (samples_per_channel < 0) { + LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") + << "Resampling audio_buffer_ failed."; + return -1; } + resampled_last_output_frame_ = true; + } else { + resampled_last_output_frame_ = false; + // We might end up here ONLY if codec is changed. + memcpy(audio_frame->data_, + audio_buffer_.get(), + samples_per_channel * num_channels * sizeof(int16_t)); } + // Swap buffers, so that the current audio is stored in |last_audio_buffer_| + // for next time. + audio_buffer_.swap(last_audio_buffer_); + audio_frame->num_channels_ = num_channels; audio_frame->samples_per_channel_ = samples_per_channel; audio_frame->sample_rate_hz_ = samples_per_channel * 100; |