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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2004 Google Inc. | 3 * Copyright 2004 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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2687 | 2687 |
2688 int median, std; | 2688 int median, std; |
2689 float dummy; | 2689 float dummy; |
2690 if (engine()->voe()->processing()->GetEcDelayMetrics( | 2690 if (engine()->voe()->processing()->GetEcDelayMetrics( |
2691 median, std, dummy) != -1) { | 2691 median, std, dummy) != -1) { |
2692 echo_delay_median_ms = median; | 2692 echo_delay_median_ms = median; |
2693 echo_delay_std_ms = std; | 2693 echo_delay_std_ms = std; |
2694 } | 2694 } |
2695 } | 2695 } |
2696 | 2696 |
| 2697 webrtc::CallStatistics cs; |
| 2698 unsigned int ssrc; |
| 2699 webrtc::CodecInst codec; |
| 2700 unsigned int level; |
| 2701 |
2697 for (const auto& ch : send_channels_) { | 2702 for (const auto& ch : send_channels_) { |
2698 const int channel = ch.second->channel(); | 2703 const int channel = ch.second->channel(); |
2699 | 2704 |
2700 // Fill in the sender info, based on what we know, and what the | 2705 // Fill in the sender info, based on what we know, and what the |
2701 // remote side told us it got from its RTCP report. | 2706 // remote side told us it got from its RTCP report. |
2702 VoiceSenderInfo sinfo; | 2707 VoiceSenderInfo sinfo; |
2703 | 2708 |
2704 webrtc::CallStatistics cs = {0}; | |
2705 unsigned int ssrc = 0; | |
2706 if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 || | 2709 if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 || |
2707 engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) { | 2710 engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) { |
2708 continue; | 2711 continue; |
2709 } | 2712 } |
2710 | 2713 |
2711 sinfo.add_ssrc(ssrc); | 2714 sinfo.add_ssrc(ssrc); |
2712 sinfo.codec_name = send_codec_.get() ? send_codec_->plname : ""; | 2715 sinfo.codec_name = send_codec_.get() ? send_codec_->plname : ""; |
2713 sinfo.bytes_sent = cs.bytesSent; | 2716 sinfo.bytes_sent = cs.bytesSent; |
2714 sinfo.packets_sent = cs.packetsSent; | 2717 sinfo.packets_sent = cs.packetsSent; |
2715 // RTT isn't known until a RTCP report is received. Until then, VoiceEngine | 2718 // RTT isn't known until a RTCP report is received. Until then, VoiceEngine |
2716 // returns 0 to indicate an error value. | 2719 // returns 0 to indicate an error value. |
2717 sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1; | 2720 sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1; |
2718 | 2721 |
2719 // Get data from the last remote RTCP report. Use default values if no data | 2722 // Get data from the last remote RTCP report. Use default values if no data |
2720 // available. | 2723 // available. |
2721 sinfo.fraction_lost = -1.0; | 2724 sinfo.fraction_lost = -1.0; |
2722 sinfo.jitter_ms = -1; | 2725 sinfo.jitter_ms = -1; |
2723 sinfo.packets_lost = -1; | 2726 sinfo.packets_lost = -1; |
2724 sinfo.ext_seqnum = -1; | 2727 sinfo.ext_seqnum = -1; |
2725 std::vector<webrtc::ReportBlock> receive_blocks; | 2728 std::vector<webrtc::ReportBlock> receive_blocks; |
2726 webrtc::CodecInst codec = {0}; | |
2727 if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks( | 2729 if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks( |
2728 channel, &receive_blocks) != -1 && | 2730 channel, &receive_blocks) != -1 && |
2729 engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) { | 2731 engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) { |
2730 for (const webrtc::ReportBlock& block : receive_blocks) { | 2732 for (const webrtc::ReportBlock& block : receive_blocks) { |
2731 // Lookup report for send ssrc only. | 2733 // Lookup report for send ssrc only. |
2732 if (block.source_SSRC == sinfo.ssrc()) { | 2734 if (block.source_SSRC == sinfo.ssrc()) { |
2733 // Convert Q8 to floating point. | 2735 // Convert Q8 to floating point. |
2734 sinfo.fraction_lost = static_cast<float>(block.fraction_lost) / 256; | 2736 sinfo.fraction_lost = static_cast<float>(block.fraction_lost) / 256; |
2735 // Convert samples to milliseconds. | 2737 // Convert samples to milliseconds. |
2736 if (codec.plfreq / 1000 > 0) { | 2738 if (codec.plfreq / 1000 > 0) { |
2737 sinfo.jitter_ms = block.interarrival_jitter / (codec.plfreq / 1000); | 2739 sinfo.jitter_ms = block.interarrival_jitter / (codec.plfreq / 1000); |
2738 } | 2740 } |
2739 sinfo.packets_lost = block.cumulative_num_packets_lost; | 2741 sinfo.packets_lost = block.cumulative_num_packets_lost; |
2740 sinfo.ext_seqnum = block.extended_highest_sequence_number; | 2742 sinfo.ext_seqnum = block.extended_highest_sequence_number; |
2741 break; | 2743 break; |
2742 } | 2744 } |
2743 } | 2745 } |
2744 } | 2746 } |
2745 | 2747 |
2746 // Local speech level. | 2748 // Local speech level. |
2747 unsigned int level = 0; | |
2748 sinfo.audio_level = (engine()->voe()->volume()-> | 2749 sinfo.audio_level = (engine()->voe()->volume()-> |
2749 GetSpeechInputLevelFullRange(level) != -1) ? level : -1; | 2750 GetSpeechInputLevelFullRange(level) != -1) ? level : -1; |
2750 | 2751 |
2751 // TODO(xians): We are injecting the same APM logging to all the send | 2752 // TODO(xians): We are injecting the same APM logging to all the send |
2752 // channels here because there is no good way to know which send channel | 2753 // channels here because there is no good way to know which send channel |
2753 // is using the APM. The correct fix is to allow the send channels to have | 2754 // is using the APM. The correct fix is to allow the send channels to have |
2754 // their own APM so that we can feed the correct APM logging to different | 2755 // their own APM so that we can feed the correct APM logging to different |
2755 // send channels. See issue crbug/264611 . | 2756 // send channels. See issue crbug/264611 . |
2756 sinfo.echo_return_loss = echo_return_loss; | 2757 sinfo.echo_return_loss = echo_return_loss; |
2757 sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement; | 2758 sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement; |
2758 sinfo.echo_delay_median_ms = echo_delay_median_ms; | 2759 sinfo.echo_delay_median_ms = echo_delay_median_ms; |
2759 sinfo.echo_delay_std_ms = echo_delay_std_ms; | 2760 sinfo.echo_delay_std_ms = echo_delay_std_ms; |
2760 // TODO(ajm): Re-enable this metric once we have a reliable implementation. | 2761 // TODO(ajm): Re-enable this metric once we have a reliable implementation. |
2761 sinfo.aec_quality_min = -1; | 2762 sinfo.aec_quality_min = -1; |
2762 sinfo.typing_noise_detected = typing_noise_detected_; | 2763 sinfo.typing_noise_detected = typing_noise_detected_; |
2763 | 2764 |
2764 info->senders.push_back(sinfo); | 2765 info->senders.push_back(sinfo); |
2765 } | 2766 } |
2766 | 2767 |
2767 // Get the SSRC and stats for each receiver. | 2768 // Get the SSRC and stats for each receiver. |
2768 info->receivers.clear(); | 2769 for (const auto& ch : receive_channels_) { |
2769 for (const auto& stream : receive_streams_) { | 2770 int ch_id = ch.second->channel(); |
2770 webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats(); | 2771 memset(&cs, 0, sizeof(cs)); |
2771 VoiceReceiverInfo rinfo; | 2772 if (engine()->voe()->rtp()->GetRemoteSSRC(ch_id, ssrc) != -1 && |
2772 rinfo.add_ssrc(stats.remote_ssrc); | 2773 engine()->voe()->rtp()->GetRTCPStatistics(ch_id, cs) != -1 && |
2773 rinfo.bytes_rcvd = stats.bytes_rcvd; | 2774 engine()->voe()->codec()->GetRecCodec(ch_id, codec) != -1) { |
2774 rinfo.packets_rcvd = stats.packets_rcvd; | 2775 VoiceReceiverInfo rinfo; |
2775 rinfo.packets_lost = stats.packets_lost; | 2776 rinfo.add_ssrc(ssrc); |
2776 rinfo.fraction_lost = stats.fraction_lost; | 2777 rinfo.bytes_rcvd = cs.bytesReceived; |
2777 rinfo.codec_name = stats.codec_name; | 2778 rinfo.packets_rcvd = cs.packetsReceived; |
2778 rinfo.ext_seqnum = stats.ext_seqnum; | 2779 // The next four fields are from the most recently sent RTCP report. |
2779 rinfo.jitter_ms = stats.jitter_ms; | 2780 // Convert Q8 to floating point. |
2780 rinfo.jitter_buffer_ms = stats.jitter_buffer_ms; | 2781 rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8); |
2781 rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms; | 2782 rinfo.packets_lost = cs.cumulativeLost; |
2782 rinfo.delay_estimate_ms = stats.delay_estimate_ms; | 2783 rinfo.ext_seqnum = cs.extendedMax; |
2783 rinfo.audio_level = stats.audio_level; | 2784 rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_; |
2784 rinfo.expand_rate = stats.expand_rate; | 2785 if (codec.pltype != -1) { |
2785 rinfo.speech_expand_rate = stats.speech_expand_rate; | 2786 rinfo.codec_name = codec.plname; |
2786 rinfo.secondary_decoded_rate = stats.secondary_decoded_rate; | 2787 } |
2787 rinfo.accelerate_rate = stats.accelerate_rate; | 2788 // Convert samples to milliseconds. |
2788 rinfo.preemptive_expand_rate = stats.preemptive_expand_rate; | 2789 if (codec.plfreq / 1000 > 0) { |
2789 rinfo.decoding_calls_to_silence_generator = | 2790 rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000); |
2790 stats.decoding_calls_to_silence_generator; | 2791 } |
2791 rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq; | 2792 |
2792 rinfo.decoding_normal = stats.decoding_normal; | 2793 // Get jitter buffer and total delay (alg + jitter + playout) stats. |
2793 rinfo.decoding_plc = stats.decoding_plc; | 2794 webrtc::NetworkStatistics ns; |
2794 rinfo.decoding_cng = stats.decoding_cng; | 2795 if (engine()->voe()->neteq() && |
2795 rinfo.decoding_plc_cng = stats.decoding_plc_cng; | 2796 engine()->voe()->neteq()->GetNetworkStatistics( |
2796 rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms; | 2797 ch_id, ns) != -1) { |
2797 info->receivers.push_back(rinfo); | 2798 rinfo.jitter_buffer_ms = ns.currentBufferSize; |
| 2799 rinfo.jitter_buffer_preferred_ms = ns.preferredBufferSize; |
| 2800 rinfo.expand_rate = |
| 2801 static_cast<float>(ns.currentExpandRate) / (1 << 14); |
| 2802 rinfo.speech_expand_rate = |
| 2803 static_cast<float>(ns.currentSpeechExpandRate) / (1 << 14); |
| 2804 rinfo.secondary_decoded_rate = |
| 2805 static_cast<float>(ns.currentSecondaryDecodedRate) / (1 << 14); |
| 2806 rinfo.accelerate_rate = |
| 2807 static_cast<float>(ns.currentAccelerateRate) / (1 << 14); |
| 2808 rinfo.preemptive_expand_rate = |
| 2809 static_cast<float>(ns.currentPreemptiveRate) / (1 << 14); |
| 2810 } |
| 2811 |
| 2812 webrtc::AudioDecodingCallStats ds; |
| 2813 if (engine()->voe()->neteq() && |
| 2814 engine()->voe()->neteq()->GetDecodingCallStatistics( |
| 2815 ch_id, &ds) != -1) { |
| 2816 rinfo.decoding_calls_to_silence_generator = |
| 2817 ds.calls_to_silence_generator; |
| 2818 rinfo.decoding_calls_to_neteq = ds.calls_to_neteq; |
| 2819 rinfo.decoding_normal = ds.decoded_normal; |
| 2820 rinfo.decoding_plc = ds.decoded_plc; |
| 2821 rinfo.decoding_cng = ds.decoded_cng; |
| 2822 rinfo.decoding_plc_cng = ds.decoded_plc_cng; |
| 2823 } |
| 2824 |
| 2825 if (engine()->voe()->sync()) { |
| 2826 int jitter_buffer_delay_ms = 0; |
| 2827 int playout_buffer_delay_ms = 0; |
| 2828 engine()->voe()->sync()->GetDelayEstimate( |
| 2829 ch_id, &jitter_buffer_delay_ms, &playout_buffer_delay_ms); |
| 2830 rinfo.delay_estimate_ms = jitter_buffer_delay_ms + |
| 2831 playout_buffer_delay_ms; |
| 2832 } |
| 2833 |
| 2834 // Get speech level. |
| 2835 rinfo.audio_level = (engine()->voe()->volume()-> |
| 2836 GetSpeechOutputLevelFullRange(ch_id, level) != -1) ? level : -1; |
| 2837 info->receivers.push_back(rinfo); |
| 2838 } |
2798 } | 2839 } |
2799 | 2840 |
2800 return true; | 2841 return true; |
2801 } | 2842 } |
2802 | 2843 |
2803 void WebRtcVoiceMediaChannel::OnError(int error) { | 2844 void WebRtcVoiceMediaChannel::OnError(int error) { |
2804 if (send_ == SEND_NOTHING) { | 2845 if (send_ == SEND_NOTHING) { |
2805 return; | 2846 return; |
2806 } | 2847 } |
2807 if (error == VE_TYPING_NOISE_WARNING) { | 2848 if (error == VE_TYPING_NOISE_WARNING) { |
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3011 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); | 3052 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); |
3012 return false; | 3053 return false; |
3013 } | 3054 } |
3014 } | 3055 } |
3015 return true; | 3056 return true; |
3016 } | 3057 } |
3017 | 3058 |
3018 } // namespace cricket | 3059 } // namespace cricket |
3019 | 3060 |
3020 #endif // HAVE_WEBRTC_VOICE | 3061 #endif // HAVE_WEBRTC_VOICE |
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