Add media related stats (audio level etc.) to unsignaled streams.
The media related stats wasn't working for unsignaled stream because there
is no mapping between the receiver_info and unsignaled tracks.
This CL fixes the issue by adding some special logic to the TrackMediaInfoMap
which would create the mapping.
BUG=b/37836881
BUG=webrtc:7685
TBR=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/2883943003
Cr-Commit-Position: refs/heads/master@{#18217}
diff --git a/webrtc/pc/peerconnection_integrationtest.cc b/webrtc/pc/peerconnection_integrationtest.cc
index ad1a12c..e6c3cf1 100644
--- a/webrtc/pc/peerconnection_integrationtest.cc
+++ b/webrtc/pc/peerconnection_integrationtest.cc
@@ -116,6 +116,18 @@
desc->set_msid_supported(false);
}
+int FindFirstMediaStatsIndexByKind(
+ const std::string& kind,
+ const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
+ media_stats_vec) {
+ for (size_t i = 0; i < media_stats_vec.size(); i++) {
+ if (media_stats_vec[i]->kind.ValueToString() == kind) {
+ return i;
+ }
+ }
+ return -1;
+}
+
class SignalingMessageReceiver {
public:
virtual void ReceiveSdpMessage(const std::string& type,
@@ -1926,9 +1938,31 @@
ASSERT_EQ(1U, inbound_stream_stats.size());
ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
- // TODO(deadbeef): Test that track_id is defined. This is not currently
- // working since SSRCs are used to match RtpReceivers (and their tracks) with
- // received stream stats in TrackMediaInfoMap.
+ ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined());
+}
+
+// Test that we can successfully get the media related stats (audio level
+// etc.) for the unsignaled stream.
+TEST_F(PeerConnectionIntegrationTest,
+ GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
+ ASSERT_TRUE(CreatePeerConnectionWrappers());
+ ConnectFakeSignaling();
+ caller()->AddAudioVideoMediaStream();
+ // Remove SSRCs and MSIDs from the received offer SDP.
+ callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+ // Wait for one audio frame to be received by the callee.
+ ExpectNewFramesReceivedWithWait(0, 0, 1, 1, kMaxWaitForFramesMs);
+
+ rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
+ callee()->NewGetStats();
+ ASSERT_NE(nullptr, report);
+
+ auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
+ auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats);
+ ASSERT_GE(audio_index, 0);
+ EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined());
}
// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
diff --git a/webrtc/pc/trackmediainfomap.cc b/webrtc/pc/trackmediainfomap.cc
index 33880b9..b5c5ac6 100644
--- a/webrtc/pc/trackmediainfomap.cc
+++ b/webrtc/pc/trackmediainfomap.cc
@@ -34,7 +34,9 @@
std::map<uint32_t, AudioTrackInterface*>* local_audio_track_by_ssrc,
std::map<uint32_t, VideoTrackInterface*>* local_video_track_by_ssrc,
std::map<uint32_t, AudioTrackInterface*>* remote_audio_track_by_ssrc,
- std::map<uint32_t, VideoTrackInterface*>* remote_video_track_by_ssrc) {
+ std::map<uint32_t, VideoTrackInterface*>* remote_video_track_by_ssrc,
+ AudioTrackInterface** unsignaled_audio_track,
+ VideoTrackInterface** unsignaled_video_track) {
RTC_DCHECK(local_audio_track_by_ssrc->empty());
RTC_DCHECK(local_video_track_by_ssrc->empty());
RTC_DCHECK(remote_audio_track_by_ssrc->empty());
@@ -80,6 +82,12 @@
RtpParameters params = rtp_receiver->GetParameters();
for (const RtpEncodingParameters& encoding : params.encodings) {
if (!encoding.ssrc) {
+ if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ *unsignaled_audio_track = static_cast<AudioTrackInterface*>(track);
+ } else {
+ RTC_DCHECK(media_type == cricket::MEDIA_TYPE_VIDEO);
+ *unsignaled_video_track = static_cast<VideoTrackInterface*>(track);
+ }
continue;
}
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
@@ -110,12 +118,13 @@
std::map<uint32_t, VideoTrackInterface*> local_video_track_by_ssrc;
std::map<uint32_t, AudioTrackInterface*> remote_audio_track_by_ssrc;
std::map<uint32_t, VideoTrackInterface*> remote_video_track_by_ssrc;
- GetAudioAndVideoTrackBySsrc(rtp_senders,
- rtp_receivers,
- &local_audio_track_by_ssrc,
- &local_video_track_by_ssrc,
- &remote_audio_track_by_ssrc,
- &remote_video_track_by_ssrc);
+ AudioTrackInterface* unsignaled_audio_track = nullptr;
+ VideoTrackInterface* unsignaled_video_track = nullptr;
+ GetAudioAndVideoTrackBySsrc(
+ rtp_senders, rtp_receivers, &local_audio_track_by_ssrc,
+ &local_video_track_by_ssrc, &remote_audio_track_by_ssrc,
+ &remote_video_track_by_ssrc, &unsignaled_audio_track,
+ &unsignaled_video_track);
if (voice_media_info_) {
for (auto& sender_info : voice_media_info_->senders) {
AudioTrackInterface* associated_track =
@@ -137,6 +146,9 @@
RTC_DCHECK(voice_info_by_remote_track_.find(associated_track) ==
voice_info_by_remote_track_.end());
voice_info_by_remote_track_[associated_track] = &receiver_info;
+ } else if (unsignaled_audio_track) {
+ audio_track_by_receiver_info_[&receiver_info] = unsignaled_audio_track;
+ voice_info_by_remote_track_[unsignaled_audio_track] = &receiver_info;
}
}
}
@@ -161,6 +173,9 @@
RTC_DCHECK(video_info_by_remote_track_.find(associated_track) ==
video_info_by_remote_track_.end());
video_info_by_remote_track_[associated_track] = &receiver_info;
+ } else if (unsignaled_video_track) {
+ video_track_by_receiver_info_[&receiver_info] = unsignaled_video_track;
+ video_info_by_remote_track_[unsignaled_video_track] = &receiver_info;
}
}
}
