Add tracing for slow paths in new video API.
Allows tracking what actually takes time in SetRemoteDescription and
SetLocalDescription.
BUG=1788
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38809004
Cr-Commit-Position: refs/heads/master@{#8202}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8202 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 7b60526..d77db23 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -45,6 +45,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/call.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
#include "webrtc/video_decoder.h"
#include "webrtc/video_encoder.h"
@@ -673,6 +674,7 @@
}
bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
+ TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvCodecs");
LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs);
if (!ValidateCodecFormats(codecs)) {
return false;
@@ -706,6 +708,7 @@
}
bool WebRtcVideoChannel2::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
+ TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendCodecs");
LOG(LS_INFO) << "SetSendCodecs: " << CodecVectorToString(codecs);
if (!ValidateCodecFormats(codecs)) {
return false;
@@ -1166,6 +1169,7 @@
bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
+ TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions");
LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
<< RtpExtensionsToString(extensions);
if (!ValidateRtpHeaderExtensionIds(extensions))
@@ -1190,6 +1194,7 @@
bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
+ TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendRtpHeaderExtensions");
LOG(LS_INFO) << "SetSendRtpHeaderExtensions: "
<< RtpExtensionsToString(extensions);
if (!ValidateRtpHeaderExtensionIds(extensions))
@@ -1229,6 +1234,7 @@
}
bool WebRtcVideoChannel2::SetOptions(const VideoOptions& options) {
+ TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetOptions");
LOG(LS_INFO) << "SetOptions: " << options.ToString();
VideoOptions old_options = options_;
options_.SetAll(options);
diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc
index b80d5c0..a11656e 100644
--- a/webrtc/video/call.cc
+++ b/webrtc/video/call.cc
@@ -27,6 +27,7 @@
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
#include "webrtc/video/video_receive_stream.h"
#include "webrtc/video/video_send_stream.h"
#include "webrtc/video_engine/include/vie_base.h"
@@ -221,6 +222,7 @@
VideoSendStream* Call::CreateVideoSendStream(
const VideoSendStream::Config& config,
const VideoEncoderConfig& encoder_config) {
+ TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
assert(config.rtp.ssrcs.size() > 0);
@@ -245,6 +247,7 @@
}
void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
+ TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
assert(send_stream != NULL);
send_stream->Stop();
@@ -277,6 +280,7 @@
VideoReceiveStream* Call::CreateVideoReceiveStream(
const VideoReceiveStream::Config& config) {
+ TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
VideoReceiveStream* receive_stream =
new VideoReceiveStream(video_engine_,
@@ -304,6 +308,7 @@
void Call::DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) {
+ TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
assert(receive_stream != NULL);
VideoReceiveStream* receive_stream_impl = NULL;
@@ -356,6 +361,7 @@
void Call::SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) {
+ TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
assert(bitrate_config.min_bitrate_bps >= 0);
assert(bitrate_config.max_bitrate_bps == -1 ||
bitrate_config.max_bitrate_bps > 0);
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index c9aefba..4c31f0c 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -17,6 +17,7 @@
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
#include "webrtc/video_engine/include/vie_base.h"
#include "webrtc/video_engine/include/vie_capture.h"
#include "webrtc/video_engine/include/vie_codec.h"
@@ -291,6 +292,7 @@
bool VideoSendStream::ReconfigureVideoEncoder(
const VideoEncoderConfig& config) {
+ TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
const std::vector<VideoStream>& streams = config.streams;
assert(!streams.empty());
