Implement AudioSendStream::GetStats().

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index fd93535bea0672c35b64e13e80d1d794475a0dd2..b4311229396ba5d5691a292adf2a1227143a0ef9 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1321,6 +1321,9 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
       : channel_(ch),
         voe_audio_transport_(voe_audio_transport),
         call_(call) {
+    RTC_DCHECK(ch >= 0);

[kwiberg-webrtc, 2015/10/26 13:19:53]

RTC_DCHECK_GE

[the sun, 2015/10/26 14:44:02]

Done.

+    // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
+    // RTC_DCHECK(voe_audio_transport);
     RTC_DCHECK(call);
     webrtc::AudioSendStream::Config config(nullptr);
     config.voe_channel_id = channel_;
@@ -1329,6 +1332,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     RTC_DCHECK(stream_);
   }
   ~WebRtcAudioSendStream() override {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     Stop();
     call_->DestroyAudioSendStream(stream_);
   }
@@ -1338,7 +1342,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // This method is called on the libjingle worker thread.
   // TODO(xians): Make sure Start() is called only once.
   void Start(AudioRenderer* renderer) {
-    rtc::CritScope lock(&lock_);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     RTC_DCHECK(renderer);
     if (renderer_) {
       RTC_DCHECK(renderer_ == renderer);
@@ -1348,11 +1352,16 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     renderer_ = renderer;
   }
 
+  webrtc::AudioSendStream::Stats GetStats() const {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    return stream_->GetStats();
+  }
+
   // Stops rendering by setting the sink of the renderer to nullptr. No data
   // callback will be received after this method.
   // This method is called on the libjingle worker thread.
   void Stop() {
-    rtc::CritScope lock(&lock_);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (renderer_) {
       renderer_->SetSink(nullptr);
       renderer_ = nullptr;
@@ -1366,6 +1375,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
               int sample_rate,
               int number_of_channels,
               size_t number_of_frames) override {
+    RTC_DCHECK(!thread_checker_.CalledOnValidThread());

[kwiberg-webrtc, 2015/10/26 13:19:53]

How much work would it be to have separate thread checkers for the audio and
worker threads? This check will let all other threads through too... and
malfunction if this is the first CalledOnValidThread call.

[the sun, 2015/10/26 14:44:02]

Done.

     RTC_DCHECK(voe_audio_transport_);
     voe_audio_transport_->OnData(channel_,
                                  audio_data,
@@ -1378,16 +1388,20 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // Callback from the |renderer_| when it is going away. In case Start() has
   // never been called, this callback won't be triggered.
   void OnClose() override {
-    rtc::CritScope lock(&lock_);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     // Set |renderer_| to nullptr to make sure no more callback will get into
     // the renderer.
     renderer_ = nullptr;
   }
 
   // Accessor to the VoE channel ID.
-  int channel() const { return channel_; }
+  int channel() const {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    return channel_;
+  }
 
  private:
+  rtc::ThreadChecker thread_checker_;
   const int channel_ = -1;
   webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
   webrtc::Call* call_ = nullptr;
@@ -1398,9 +1412,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // goes away.
   AudioRenderer* renderer_ = nullptr;
 
-  // Protects |renderer_| in Start(), Stop() and OnClose().
-  rtc::CriticalSection lock_;
-
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
 };
 
@@ -1433,7 +1444,6 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
       desired_send_(SEND_NOTHING),
       send_(SEND_NOTHING),
       call_(call) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
   RTC_DCHECK(nullptr != call);
   engine->RegisterChannel(this);
@@ -2618,109 +2628,36 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
 
 bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(info);
 
-  bool echo_metrics_on = false;
-  // These can take on valid negative values, so use the lowest possible level
-  // as default rather than -1.
-  int echo_return_loss = -100;
-  int echo_return_loss_enhancement = -100;
-  // These can also be negative, but in practice -1 is only used to signal
-  // insufficient data, since the resolution is limited to multiples of 4 ms.
-  int echo_delay_median_ms = -1;
-  int echo_delay_std_ms = -1;
-  if (engine()->voe()->processing()->GetEcMetricsStatus(
-          echo_metrics_on) != -1 && echo_metrics_on) {
-    // TODO(ajm): we may want to use VoECallReport::GetEchoMetricsSummary
-    // here, but it appears to be unsuitable currently. Revisit after this is
-    // investigated: http://b/issue?id=5666755
-    int erl, erle, rerl, anlp;
-    if (engine()->voe()->processing()->GetEchoMetrics(
-            erl, erle, rerl, anlp) != -1) {
-      echo_return_loss = erl;
-      echo_return_loss_enhancement = erle;
-    }
-
-    int median, std;
-    float dummy;
-    if (engine()->voe()->processing()->GetEcDelayMetrics(
-        median, std, dummy) != -1) {
-      echo_delay_median_ms = median;
-      echo_delay_std_ms = std;
-    }
-  }
-
-  for (const auto& ch : send_streams_) {
-    const int channel = ch.second->channel();
-
-    // Fill in the sender info, based on what we know, and what the
-    // remote side told us it got from its RTCP report.
+  // Get SSRC and stats for each sender.
+  RTC_DCHECK(info->senders.size() == 0);
+  for (const auto& stream : send_streams_) {
+    webrtc::AudioSendStream::Stats stats = stream.second->GetStats();
     VoiceSenderInfo sinfo;
-
-    webrtc::CallStatistics cs = {0};
-    unsigned int ssrc = 0;
-    if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 ||
-        engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) {
-      continue;
-    }
-
-    sinfo.add_ssrc(ssrc);
-    sinfo.codec_name = send_codec_.get() ? send_codec_->plname : "";
-    sinfo.bytes_sent = cs.bytesSent;
-    sinfo.packets_sent = cs.packetsSent;
-    // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
-    // returns 0 to indicate an error value.
-    sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1;
-
-    // Get data from the last remote RTCP report. Use default values if no data
-    // available.
-    sinfo.fraction_lost = -1.0;
-    sinfo.jitter_ms = -1;
-    sinfo.packets_lost = -1;
-    sinfo.ext_seqnum = -1;
-    std::vector<webrtc::ReportBlock> receive_blocks;
-    webrtc::CodecInst codec = {0};
-    if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks(
-            channel, &receive_blocks) != -1 &&
-        engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) {
-      for (const webrtc::ReportBlock& block : receive_blocks) {
-        // Lookup report for send ssrc only.
-        if (block.source_SSRC == sinfo.ssrc()) {
-          // Convert Q8 to floating point.
-          sinfo.fraction_lost = static_cast<float>(block.fraction_lost) / 256;
-          // Convert samples to milliseconds.
-          if (codec.plfreq / 1000 > 0) {
-            sinfo.jitter_ms = block.interarrival_jitter / (codec.plfreq / 1000);
-          }
-          sinfo.packets_lost = block.cumulative_num_packets_lost;
-          sinfo.ext_seqnum = block.extended_highest_sequence_number;
-          break;
-        }
-      }
-    }
-
-    // Local speech level.
-    unsigned int level = 0;
-    sinfo.audio_level = (engine()->voe()->volume()->
-        GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
-
-    // TODO(xians): We are injecting the same APM logging to all the send
-    // channels here because there is no good way to know which send channel
-    // is using the APM. The correct fix is to allow the send channels to have
-    // their own APM so that we can feed the correct APM logging to different
-    // send channels. See issue crbug/264611 .
-    sinfo.echo_return_loss = echo_return_loss;
-    sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement;
-    sinfo.echo_delay_median_ms = echo_delay_median_ms;
-    sinfo.echo_delay_std_ms = echo_delay_std_ms;
-    // TODO(ajm): Re-enable this metric once we have a reliable implementation.
-    sinfo.aec_quality_min = -1;
+    sinfo.add_ssrc(stats.local_ssrc);
+    sinfo.bytes_sent = stats.bytes_sent;
+    sinfo.packets_sent = stats.packets_sent;
+    sinfo.packets_lost = stats.packets_lost;
+    sinfo.fraction_lost = stats.fraction_lost;
+    sinfo.codec_name = stats.codec_name;
+    sinfo.ext_seqnum = stats.ext_seqnum;
+    sinfo.jitter_ms = stats.jitter_ms;
+    sinfo.rtt_ms = stats.rtt_ms;
+    sinfo.audio_level = stats.audio_level;
+    sinfo.aec_quality_min = stats.aec_quality_min;
+    sinfo.echo_delay_median_ms = stats.echo_delay_median_ms;
+    sinfo.echo_delay_std_ms = stats.echo_delay_std_ms;
+    sinfo.echo_return_loss = stats.echo_return_loss;
+    sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement;
     sinfo.typing_noise_detected = typing_noise_detected_;
-
+    // TODO(solenberg): Move to AudioSendStream.
+    //  sinfo.typing_noise_detected = stats.typing_noise_detected;
     info->senders.push_back(sinfo);
   }
 
-  // Get the SSRC and stats for each receiver.
-  info->receivers.clear();
+  // Get SSRC and stats for each receiver.
+  RTC_DCHECK(info->receivers.size() == 0);
   for (const auto& stream : receive_streams_) {
     webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
     VoiceReceiverInfo rinfo;