Implement AudioSendStream::GetStats().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1414743004
Cr-Commit-Position: refs/heads/master@{#10424}
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index fd93535..1d12fbf 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1321,7 +1321,11 @@
: channel_(ch),
voe_audio_transport_(voe_audio_transport),
call_(call) {
+ RTC_DCHECK_GE(ch, 0);
+ // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
+ // RTC_DCHECK(voe_audio_transport);
RTC_DCHECK(call);
+ audio_capture_thread_checker_.DetachFromThread();
webrtc::AudioSendStream::Config config(nullptr);
config.voe_channel_id = channel_;
config.rtp.ssrc = ssrc;
@@ -1329,6 +1333,7 @@
RTC_DCHECK(stream_);
}
~WebRtcAudioSendStream() override {
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
Stop();
call_->DestroyAudioSendStream(stream_);
}
@@ -1338,7 +1343,7 @@
// 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(signal_thread_checker_.CalledOnValidThread());
RTC_DCHECK(renderer);
if (renderer_) {
RTC_DCHECK(renderer_ == renderer);
@@ -1348,11 +1353,16 @@
renderer_ = renderer;
}
+ webrtc::AudioSendStream::Stats GetStats() const {
+ RTC_DCHECK(signal_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(signal_thread_checker_.CalledOnValidThread());
if (renderer_) {
renderer_->SetSink(nullptr);
renderer_ = nullptr;
@@ -1366,6 +1376,7 @@
int sample_rate,
int number_of_channels,
size_t number_of_frames) override {
+ RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
RTC_DCHECK(voe_audio_transport_);
voe_audio_transport_->OnData(channel_,
audio_data,
@@ -1378,16 +1389,21 @@
// 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(signal_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(signal_thread_checker_.CalledOnValidThread());
+ return channel_;
+ }
private:
+ rtc::ThreadChecker signal_thread_checker_;
+ rtc::ThreadChecker audio_capture_thread_checker_;
const int channel_ = -1;
webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
webrtc::Call* call_ = nullptr;
@@ -1398,9 +1414,6 @@
// goes away.
AudioRenderer* renderer_ = nullptr;
- // Protects |renderer_| in Start(), Stop() and OnClose().
- rtc::CriticalSection lock_;
-
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
};
@@ -1433,7 +1446,6 @@
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 +2630,36 @@
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;
