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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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639 return AudioMixer::Source::AudioFrameInfo::kError; | 639 return AudioMixer::Source::AudioFrameInfo::kError; |
640 } | 640 } |
641 | 641 |
642 if (muted) { | 642 if (muted) { |
643 // TODO(henrik.lundin): We should be able to do better than this. But we | 643 // TODO(henrik.lundin): We should be able to do better than this. But we |
644 // will have to go through all the cases below where the audio samples may | 644 // will have to go through all the cases below where the audio samples may |
645 // be used, and handle the muted case in some way. | 645 // be used, and handle the muted case in some way. |
646 AudioFrameOperations::Mute(audio_frame); | 646 AudioFrameOperations::Mute(audio_frame); |
647 } | 647 } |
648 | 648 |
649 // Convert module ID to internal VoE channel ID | |
650 audio_frame->id_ = VoEChannelId(audio_frame->id_); | |
651 // Store speech type for dead-or-alive detection | 649 // Store speech type for dead-or-alive detection |
652 _outputSpeechType = audio_frame->speech_type_; | 650 _outputSpeechType = audio_frame->speech_type_; |
653 | 651 |
654 { | 652 { |
655 // Pass the audio buffers to an optional sink callback, before applying | 653 // Pass the audio buffers to an optional sink callback, before applying |
656 // scaling/panning, as that applies to the mix operation. | 654 // scaling/panning, as that applies to the mix operation. |
657 // External recipients of the audio (e.g. via AudioTrack), will do their | 655 // External recipients of the audio (e.g. via AudioTrack), will do their |
658 // own mixing/dynamic processing. | 656 // own mixing/dynamic processing. |
659 rtc::CritScope cs(&_callbackCritSect); | 657 rtc::CritScope cs(&_callbackCritSect); |
660 if (audio_sink_) { | 658 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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1708 void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) { | 1705 void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) { |
1709 // Avoid posting any new tasks if sending was already stopped in StopSend(). | 1706 // Avoid posting any new tasks if sending was already stopped in StopSend(). |
1710 rtc::CritScope cs(&encoder_queue_lock_); | 1707 rtc::CritScope cs(&encoder_queue_lock_); |
1711 if (!encoder_queue_is_active_) { | 1708 if (!encoder_queue_is_active_) { |
1712 return; | 1709 return; |
1713 } | 1710 } |
1714 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); | 1711 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); |
1715 // TODO(henrika): try to avoid copying by moving ownership of audio frame | 1712 // TODO(henrika): try to avoid copying by moving ownership of audio frame |
1716 // either into pool of frames or into the task itself. | 1713 // either into pool of frames or into the task itself. |
1717 audio_frame->CopyFrom(audio_input); | 1714 audio_frame->CopyFrom(audio_input); |
1718 audio_frame->id_ = ChannelId(); | |
1719 encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( | 1715 encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( |
1720 new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); | 1716 new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); |
1721 } | 1717 } |
1722 | 1718 |
1723 void Channel::ProcessAndEncodeAudio(const int16_t* audio_data, | 1719 void Channel::ProcessAndEncodeAudio(const int16_t* audio_data, |
1724 int sample_rate, | 1720 int sample_rate, |
1725 size_t number_of_frames, | 1721 size_t number_of_frames, |
1726 size_t number_of_channels) { | 1722 size_t number_of_channels) { |
1727 // Avoid posting as new task if sending was already stopped in StopSend(). | 1723 // Avoid posting as new task if sending was already stopped in StopSend(). |
1728 rtc::CritScope cs(&encoder_queue_lock_); | 1724 rtc::CritScope cs(&encoder_queue_lock_); |
1729 if (!encoder_queue_is_active_) { | 1725 if (!encoder_queue_is_active_) { |
1730 return; | 1726 return; |
1731 } | 1727 } |
1732 CodecInst codec; | 1728 CodecInst codec; |
1733 const int result = GetSendCodec(codec); | 1729 const int result = GetSendCodec(codec); |
1734 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); | 1730 std::unique_ptr<AudioFrame> audio_frame(new AudioFrame()); |
1735 audio_frame->id_ = ChannelId(); | |
1736 // TODO(ossu): Investigate how this could happen. b/62909493 | 1731 // TODO(ossu): Investigate how this could happen. b/62909493 |
1737 if (result == 0) { | 1732 if (result == 0) { |
1738 audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate); | 1733 audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate); |
1739 audio_frame->num_channels_ = std::min(number_of_channels, codec.channels); | 1734 audio_frame->num_channels_ = std::min(number_of_channels, codec.channels); |
1740 } else { | 1735 } else { |
1741 audio_frame->sample_rate_hz_ = sample_rate; | 1736 audio_frame->sample_rate_hz_ = sample_rate; |
1742 audio_frame->num_channels_ = number_of_channels; | 1737 audio_frame->num_channels_ = number_of_channels; |
1743 LOG(LS_WARNING) << "Unable to get send codec for channel " << ChannelId(); | 1738 LOG(LS_WARNING) << "Unable to get send codec for channel " << ChannelId(); |
1744 RTC_NOTREACHED(); | 1739 RTC_NOTREACHED(); |
1745 } | 1740 } |
1746 RemixAndResample(audio_data, number_of_frames, number_of_channels, | 1741 RemixAndResample(audio_data, number_of_frames, number_of_channels, |
1747 sample_rate, &input_resampler_, audio_frame.get()); | 1742 sample_rate, &input_resampler_, audio_frame.get()); |
1748 encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( | 1743 encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>( |
1749 new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); | 1744 new ProcessAndEncodeAudioTask(std::move(audio_frame), this))); |
1750 } | 1745 } |
1751 | 1746 |
1752 void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) { | 1747 void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) { |
1753 RTC_DCHECK_RUN_ON(encoder_queue_); | 1748 RTC_DCHECK_RUN_ON(encoder_queue_); |
1754 RTC_DCHECK_GT(audio_input->samples_per_channel_, 0); | 1749 RTC_DCHECK_GT(audio_input->samples_per_channel_, 0); |
1755 RTC_DCHECK_LE(audio_input->num_channels_, 2); | 1750 RTC_DCHECK_LE(audio_input->num_channels_, 2); |
1756 RTC_DCHECK_EQ(audio_input->id_, ChannelId()); | |
1757 | 1751 |
1758 bool is_muted = InputMute(); | 1752 bool is_muted = InputMute(); |
1759 AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted); | 1753 AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted); |
1760 | 1754 |
1761 if (_includeAudioLevelIndication) { | 1755 if (_includeAudioLevelIndication) { |
1762 size_t length = | 1756 size_t length = |
1763 audio_input->samples_per_channel_ * audio_input->num_channels_; | 1757 audio_input->samples_per_channel_ * audio_input->num_channels_; |
1764 RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes); | 1758 RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes); |
1765 if (is_muted && previous_frame_muted_) { | 1759 if (is_muted && previous_frame_muted_) { |
1766 rms_level_.AnalyzeMuted(length); | 1760 rms_level_.AnalyzeMuted(length); |
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2029 int64_t min_rtt = 0; | 2023 int64_t min_rtt = 0; |
2030 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 2024 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
2031 0) { | 2025 0) { |
2032 return 0; | 2026 return 0; |
2033 } | 2027 } |
2034 return rtt; | 2028 return rtt; |
2035 } | 2029 } |
2036 | 2030 |
2037 } // namespace voe | 2031 } // namespace voe |
2038 } // namespace webrtc | 2032 } // namespace webrtc |
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