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Add missing tracing to RtpSender objects.

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BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1873793002 .

Cr-Commit-Position: refs/heads/master@{#12311}
This commit is contained in:
Peter Boström
2016-04-11 11:45:14 +02:00
parent 18d3d1e466
commit dabc9449b7
3 changed files with 15 additions and 0 deletions
Download Patch File Download Diff File Expand all files Collapse all files

11
webrtc/api/rtpsender.cc
View File

@ -13,6 +13,7 @@
#include "webrtc/api/localaudiosource.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/trace_event.h"
namespace webrtc {
@ -86,6 +87,7 @@ AudioRtpSender::~AudioRtpSender() {
}
void AudioRtpSender::OnChanged() {
TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
@ -96,6 +98,7 @@ void AudioRtpSender::OnChanged() {
}
bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetTrack");
if (stopped_) {
LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
return false;
@ -140,6 +143,7 @@ bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
}
void AudioRtpSender::SetSsrc(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
return;
}
@ -161,6 +165,7 @@ void AudioRtpSender::SetSsrc(uint32_t ssrc) {
}
void AudioRtpSender::Stop() {
TRACE_EVENT0("webrtc", "AudioRtpSender::Stop");
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (stopped_) {
return;
@ -204,6 +209,7 @@ RtpParameters AudioRtpSender::GetParameters() const {
}
bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
return provider_->SetAudioRtpParameters(ssrc_, parameters);
}
@ -240,6 +246,7 @@ VideoRtpSender::~VideoRtpSender() {
}
void VideoRtpSender::OnChanged() {
TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
@ -250,6 +257,7 @@ void VideoRtpSender::OnChanged() {
}
bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetTrack");
if (stopped_) {
LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
return false;
@ -292,6 +300,7 @@ bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
}
void VideoRtpSender::SetSsrc(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
return;
}
@ -308,6 +317,7 @@ void VideoRtpSender::SetSsrc(uint32_t ssrc) {
}
void VideoRtpSender::Stop() {
TRACE_EVENT0("webrtc", "VideoRtpSender::Stop");
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (stopped_) {
return;
@ -338,6 +348,7 @@ RtpParameters VideoRtpSender::GetParameters() const {
}
bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
return provider_->SetVideoRtpParameters(ssrc_, parameters);
}

2
webrtc/media/engine/webrtcvideoengine2.cc
View File

@ -883,6 +883,7 @@ webrtc::RtpParameters WebRtcVideoChannel2::GetRtpParameters(
bool WebRtcVideoChannel2::SetRtpParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpParameters");
rtc::CritScope stream_lock(&stream_crit_);
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
@ -985,6 +986,7 @@ bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
}
bool WebRtcVideoChannel2::SetSend(bool send) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSend");
LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
if (send && !send_codec_) {
LOG(LS_ERROR) << "SetSend(true) called before setting codec.";

2
webrtc/video/video_send_stream.cc
View File

@ -355,6 +355,7 @@ VideoCaptureInput* VideoSendStream::Input() {
void VideoSendStream::Start() {
if (payload_router_.active())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
vie_encoder_.Pause();
payload_router_.set_active(true);
// Was not already started, trigger a keyframe.
@ -366,6 +367,7 @@ void VideoSendStream::Start() {
void VideoSendStream::Stop() {
if (!payload_router_.active())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
// TODO(pbos): Make sure the encoder stops here.
payload_router_.set_active(false);
vie_receiver_->StopReceive();