Initial WebRtcVideoEngine2::GetStats().
Also forward-declaring and moving WebRtcVideoRenderer out of header.
BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6729 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/media/webrtc/webrtcvideoengine2.cc b/media/webrtc/webrtcvideoengine2.cc
index 03c5e14..1ce1b99 100644
--- a/media/webrtc/webrtcvideoengine2.cc
+++ b/media/webrtc/webrtcvideoengine2.cc
@@ -638,8 +638,6 @@
const webrtc::I420VideoFrame* const frame_;
};
-// WebRtcVideoChannel2
-
WebRtcVideoChannel2::WebRtcVideoChannel2(
WebRtcVideoEngine2* engine,
VoiceMediaChannel* voice_channel,
@@ -1048,10 +1046,36 @@
bool WebRtcVideoChannel2::GetStats(const StatsOptions& options,
VideoMediaInfo* info) {
- // TODO(pbos): Implement.
+ info->Clear();
+ FillSenderStats(info);
+ FillReceiverStats(info);
+ FillBandwidthEstimationStats(info);
return true;
}
+void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info) {
+ for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
+ send_streams_.begin();
+ it != send_streams_.end();
+ ++it) {
+ video_media_info->senders.push_back(it->second->GetVideoSenderInfo());
+ }
+}
+
+void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info) {
+ for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
+ receive_streams_.begin();
+ it != receive_streams_.end();
+ ++it) {
+ video_media_info->receivers.push_back(it->second->GetVideoReceiverInfo());
+ }
+}
+
+void WebRtcVideoChannel2::FillBandwidthEstimationStats(
+ VideoMediaInfo* video_media_info) {
+ // TODO(pbos): Implement.
+}
+
bool WebRtcVideoChannel2::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
<< (capturer != NULL ? "(capturer)" : "NULL");
@@ -1549,6 +1573,60 @@
sending_ = false;
}
+VideoSenderInfo
+WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
+ VideoSenderInfo info;
+ talk_base::CritScope cs(&lock_);
+ for (size_t i = 0; i < parameters_.config.rtp.ssrcs.size(); ++i) {
+ info.add_ssrc(parameters_.config.rtp.ssrcs[i]);
+ }
+
+ webrtc::VideoSendStream::Stats stats = stream_->GetStats();
+ info.framerate_input = stats.input_frame_rate;
+ info.framerate_sent = stats.encode_frame_rate;
+
+ for (std::map<uint32_t, webrtc::StreamStats>::iterator it =
+ stats.substreams.begin();
+ it != stats.substreams.end();
+ ++it) {
+ // TODO(pbos): Wire up additional stats, such as padding bytes.
+ webrtc::StreamStats stream_stats = it->second;
+ info.bytes_sent += stream_stats.rtp_stats.bytes +
+ stream_stats.rtp_stats.header_bytes +
+ stream_stats.rtp_stats.padding_bytes;
+ info.packets_sent += stream_stats.rtp_stats.packets;
+ info.packets_lost += stream_stats.rtcp_stats.cumulative_lost;
+ }
+
+ if (!stats.substreams.empty()) {
+ // TODO(pbos): Report fraction lost per SSRC.
+ webrtc::StreamStats first_stream_stats = stats.substreams.begin()->second;
+ info.fraction_lost =
+ static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
+ (1 << 8);
+ }
+
+ if (capturer_ != NULL && !capturer_->IsMuted()) {
+ VideoFormat last_captured_frame_format;
+ capturer_->GetStats(&info.adapt_frame_drops,
+ &info.effects_frame_drops,
+ &info.capturer_frame_time,
+ &last_captured_frame_format);
+ info.input_frame_width = last_captured_frame_format.width;
+ info.input_frame_height = last_captured_frame_format.height;
+ info.send_frame_width =
+ static_cast<int>(parameters_.video_streams.front().width);
+ info.send_frame_height =
+ static_cast<int>(parameters_.video_streams.front().height);
+ }
+
+ // TODO(pbos): Support or remove the following stats.
+ info.packets_cached = -1;
+ info.rtt_ms = -1;
+
+ return info;
+}
+
void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
if (stream_ != NULL) {
call_->DestroyVideoSendStream(stream_);
@@ -1671,6 +1749,29 @@
last_height_ = height;
}
+VideoReceiverInfo
+WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo() {
+ VideoReceiverInfo info;
+ info.add_ssrc(config_.rtp.remote_ssrc);
+ webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
+ info.bytes_rcvd = stats.rtp_stats.bytes + stats.rtp_stats.header_bytes +
+ stats.rtp_stats.padding_bytes;
+ info.packets_rcvd = stats.rtp_stats.packets;
+
+ info.framerate_rcvd = stats.network_frame_rate;
+ info.framerate_decoded = stats.decode_frame_rate;
+ info.framerate_output = stats.render_frame_rate;
+
+ talk_base::CritScope frame_cs(&renderer_lock_);
+ info.frame_width = last_width_;
+ info.frame_height = last_height_;
+
+ // TODO(pbos): Support or remove the following stats.
+ info.packets_concealed = -1;
+
+ return info;
+}
+
WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings()
: rtx_payload_type(-1) {}
diff --git a/media/webrtc/webrtcvideoengine2.h b/media/webrtc/webrtcvideoengine2.h
index fe798b5..e93f571 100644
--- a/media/webrtc/webrtcvideoengine2.h
+++ b/media/webrtc/webrtcvideoengine2.h
@@ -80,6 +80,7 @@
class WebRtcVideoEngine2;
class WebRtcVideoChannel2;
+class WebRtcVideoRenderer;
class WebRtcVideoEncoderFactory2 {
public:
@@ -277,6 +278,8 @@
void Start();
void Stop();
+ VideoSenderInfo GetVideoSenderInfo();
+
private:
// Parameters needed to reconstruct the underlying stream.
// webrtc::VideoSendStream doesn't support setting a lot of options on the
@@ -337,6 +340,8 @@
void SetRenderer(cricket::VideoRenderer* renderer);
cricket::VideoRenderer* GetRenderer();
+ VideoReceiverInfo GetVideoReceiverInfo();
+
private:
void SetSize(int width, int height);
void RecreateWebRtcStream();
@@ -348,8 +353,8 @@
talk_base::CriticalSection renderer_lock_;
cricket::VideoRenderer* renderer_ GUARDED_BY(renderer_lock_);
- int last_width_;
- int last_height_;
+ int last_width_ GUARDED_BY(renderer_lock_);
+ int last_height_ GUARDED_BY(renderer_lock_);
};
void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
@@ -365,6 +370,10 @@
std::vector<VideoCodecSettings> FilterSupportedCodecs(
const std::vector<VideoCodecSettings>& mapped_codecs);
+ void FillSenderStats(VideoMediaInfo* info);
+ void FillReceiverStats(VideoMediaInfo* info);
+ void FillBandwidthEstimationStats(VideoMediaInfo* info);
+
uint32_t rtcp_receiver_report_ssrc_;
bool sending_;
talk_base::scoped_ptr<webrtc::Call> call_;
diff --git a/media/webrtc/webrtcvideoengine2_unittest.cc b/media/webrtc/webrtcvideoengine2_unittest.cc
index d272929..0954bcb 100644
--- a/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -451,15 +451,13 @@
Base::SetSendSetsTransportBufferSizes();
}
-// TODO(juberti): Fix this test to tolerate missing stats.
-TEST_F(WebRtcVideoChannel2BaseTest, DISABLED_GetStats) { Base::GetStats(); }
+TEST_F(WebRtcVideoChannel2BaseTest, GetStats) { Base::GetStats(); }
-// TODO(juberti): Fix this test to tolerate missing stats.
-TEST_F(WebRtcVideoChannel2BaseTest, DISABLED_GetStatsMultipleRecvStreams) {
+TEST_F(WebRtcVideoChannel2BaseTest, GetStatsMultipleRecvStreams) {
Base::GetStatsMultipleRecvStreams();
}
-TEST_F(WebRtcVideoChannel2BaseTest, DISABLED_GetStatsMultipleSendStreams) {
+TEST_F(WebRtcVideoChannel2BaseTest, GetStatsMultipleSendStreams) {
Base::GetStatsMultipleSendStreams();
}
