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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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640 return AudioMixer::Source::AudioFrameInfo::kError; | 640 return AudioMixer::Source::AudioFrameInfo::kError; |
641 } | 641 } |
642 | 642 |
643 if (muted) { | 643 if (muted) { |
644 // TODO(henrik.lundin): We should be able to do better than this. But we | 644 // TODO(henrik.lundin): We should be able to do better than this. But we |
645 // will have to go through all the cases below where the audio samples may | 645 // will have to go through all the cases below where the audio samples may |
646 // be used, and handle the muted case in some way. | 646 // be used, and handle the muted case in some way. |
647 AudioFrameOperations::Mute(audio_frame); | 647 AudioFrameOperations::Mute(audio_frame); |
648 } | 648 } |
649 | 649 |
650 // Convert module ID to internal VoE channel ID | |
651 audio_frame->id_ = VoEChannelId(audio_frame->id_); | |
652 // Store speech type for dead-or-alive detection | 650 // Store speech type for dead-or-alive detection |
653 _outputSpeechType = audio_frame->speech_type_; | 651 _outputSpeechType = audio_frame->speech_type_; |
654 | 652 |
655 { | 653 { |
656 // Pass the audio buffers to an optional sink callback, before applying | 654 // Pass the audio buffers to an optional sink callback, before applying |
657 // scaling/panning, as that applies to the mix operation. | 655 // scaling/panning, as that applies to the mix operation. |
658 // External recipients of the audio (e.g. via AudioTrack), will do their | 656 // External recipients of the audio (e.g. via AudioTrack), will do their |
659 // own mixing/dynamic processing. | 657 // own mixing/dynamic processing. |
660 rtc::CritScope cs(&_callbackCritSect); | 658 rtc::CritScope cs(&_callbackCritSect); |
661 if (audio_sink_) { | 659 if (audio_sink_) { |
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790 seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), | 788 seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), |
791 rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), | 789 rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), |
792 retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), | 790 retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), |
793 kMaxRetransmissionWindowMs)), | 791 kMaxRetransmissionWindowMs)), |
794 decoder_factory_(config.acm_config.decoder_factory), | 792 decoder_factory_(config.acm_config.decoder_factory), |
795 use_twcc_plr_for_ana_( | 793 use_twcc_plr_for_ana_( |
796 webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") { | 794 webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") { |
797 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), | 795 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), |
798 "Channel::Channel() - ctor"); | 796 "Channel::Channel() - ctor"); |
799 AudioCodingModule::Config acm_config(config.acm_config); | 797 AudioCodingModule::Config acm_config(config.acm_config); |
800 acm_config.id = VoEModuleId(instanceId, channelId); | |
801 acm_config.neteq_config.enable_muted_state = true; | 798 acm_config.neteq_config.enable_muted_state = true; |
802 audio_coding_.reset(AudioCodingModule::Create(acm_config)); | 799 audio_coding_.reset(AudioCodingModule::Create(acm_config)); |
803 | 800 |
804 _outputAudioLevel.Clear(); | 801 _outputAudioLevel.Clear(); |
805 | 802 |
806 RtpRtcp::Configuration configuration; | 803 RtpRtcp::Configuration configuration; |
807 configuration.audio = true; | 804 configuration.audio = true; |
808 configuration.outgoing_transport = this; | 805 configuration.outgoing_transport = this; |
809 configuration.overhead_observer = this; | 806 configuration.overhead_observer = this; |
810 configuration.receive_statistics = rtp_receive_statistics_.get(); | 807 configuration.receive_statistics = rtp_receive_statistics_.get(); |
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1636 void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) { | 1633 void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) { |
1637 // Avoid posting any new tasks if sending was already stopped in StopSend(). | 1634 // Avoid posting any new tasks if sending was already stopped in StopSend(). |
1638 rtc::CritScope cs(&encoder_queue_lock_); | 1635 rtc::CritScope cs(&encoder_queue_lock_); |
1639 if (!encoder_queue_is_active_) { | 1636 if (!encoder_queue_is_active_) { |
1640 return; | 1637 return; |
1641 } | 1638 } |
1642 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); | 1639 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); |
1643 // TODO(henrika): try to avoid copying by moving ownership of audio frame | 1640 // TODO(henrika): try to avoid copying by moving ownership of audio frame |
1644 // either into pool of frames or into the task itself. | 1641 // either into pool of frames or into the task itself. |
1645 audio_frame->CopyFrom(audio_input); | 1642 audio_frame->CopyFrom(audio_input); |
1646 audio_frame->id_ = ChannelId(); | |
1647 // Profile time between when the audio frame is added to the task queue and | 1643 // Profile time between when the audio frame is added to the task queue and |
1648 // when the task is actually executed. | 1644 // when the task is actually executed. |
1649 audio_frame->UpdateProfileTimeStamp(); | 1645 audio_frame->UpdateProfileTimeStamp(); |
1650 encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( | 1646 encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( |
1651 new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); | 1647 new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); |
1652 } | 1648 } |
1653 | 1649 |
1654 void Channel::ProcessAndEncodeAudio(const int16_t* audio_data, | 1650 void Channel::ProcessAndEncodeAudio(const int16_t* audio_data, |
1655 int sample_rate, | 1651 int sample_rate, |
1656 size_t number_of_frames, | 1652 size_t number_of_frames, |
1657 size_t number_of_channels) { | 1653 size_t number_of_channels) { |
1658 // Avoid posting as new task if sending was already stopped in StopSend(). | 1654 // Avoid posting as new task if sending was already stopped in StopSend(). |
1659 rtc::CritScope cs(&encoder_queue_lock_); | 1655 rtc::CritScope cs(&encoder_queue_lock_); |
1660 if (!encoder_queue_is_active_) { | 1656 if (!encoder_queue_is_active_) { |
1661 return; | 1657 return; |
1662 } | 1658 } |
1663 CodecInst codec; | 1659 CodecInst codec; |
1664 const int result = GetSendCodec(codec); | 1660 const int result = GetSendCodec(codec); |
1665 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); | 1661 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); |
1666 audio_frame->id_ = ChannelId(); | |
1667 // TODO(ossu): Investigate how this could happen. b/62909493 | 1662 // TODO(ossu): Investigate how this could happen. b/62909493 |
1668 if (result == 0) { | 1663 if (result == 0) { |
1669 audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate); | 1664 audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate); |
1670 audio_frame->num_channels_ = std::min(number_of_channels, codec.channels); | 1665 audio_frame->num_channels_ = std::min(number_of_channels, codec.channels); |
1671 } else { | 1666 } else { |
1672 audio_frame->sample_rate_hz_ = sample_rate; | 1667 audio_frame->sample_rate_hz_ = sample_rate; |
1673 audio_frame->num_channels_ = number_of_channels; | 1668 audio_frame->num_channels_ = number_of_channels; |
1674 LOG(LS_WARNING) << "Unable to get send codec for channel " << ChannelId(); | 1669 LOG(LS_WARNING) << "Unable to get send codec for channel " << ChannelId(); |
1675 RTC_NOTREACHED(); | 1670 RTC_NOTREACHED(); |
1676 } | 1671 } |
1677 RemixAndResample(audio_data, number_of_frames, number_of_channels, | 1672 RemixAndResample(audio_data, number_of_frames, number_of_channels, |
1678 sample_rate, &input_resampler_, audio_frame.get()); | 1673 sample_rate, &input_resampler_, audio_frame.get()); |
1679 encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( | 1674 encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( |
1680 new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); | 1675 new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); |
1681 } | 1676 } |
1682 | 1677 |
1683 void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) { | 1678 void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) { |
1684 RTC_DCHECK_RUN_ON(encoder_queue_); | 1679 RTC_DCHECK_RUN_ON(encoder_queue_); |
1685 RTC_DCHECK_GT(audio_input->samples_per_channel_, 0); | 1680 RTC_DCHECK_GT(audio_input->samples_per_channel_, 0); |
1686 RTC_DCHECK_LE(audio_input->num_channels_, 2); | 1681 RTC_DCHECK_LE(audio_input->num_channels_, 2); |
1687 RTC_DCHECK_EQ(audio_input->id_, ChannelId()); | |
1688 | 1682 |
1689 // Measure time between when the audio frame is added to the task queue and | 1683 // Measure time between when the audio frame is added to the task queue and |
1690 // when the task is actually executed. Goal is to keep track of unwanted | 1684 // when the task is actually executed. Goal is to keep track of unwanted |
1691 // extra latency added by the task queue. | 1685 // extra latency added by the task queue. |
1692 RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs", | 1686 RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs", |
1693 audio_input->ElapsedProfileTimeMs()); | 1687 audio_input->ElapsedProfileTimeMs()); |
1694 | 1688 |
1695 bool is_muted = InputMute(); | 1689 bool is_muted = InputMute(); |
1696 AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted); | 1690 AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted); |
1697 | 1691 |
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1966 int64_t min_rtt = 0; | 1960 int64_t min_rtt = 0; |
1967 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 1961 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
1968 0) { | 1962 0) { |
1969 return 0; | 1963 return 0; |
1970 } | 1964 } |
1971 return rtt; | 1965 return rtt; |
1972 } | 1966 } |
1973 | 1967 |
1974 } // namespace voe | 1968 } // namespace voe |
1975 } // namespace webrtc | 1969 } // namespace webrtc |
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