Add tracing to interesting media-related methods.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 12 matching lines @@
 
 #include "webrtc/audio_sink.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/media/base/audioframe.h"
 #include "webrtc/media/base/audiorenderer.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ skipping 1323 matching lines @@
   }
   engine()->UnregisterChannel(this);
 }
 
 rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const {
   return kAudioDscpValue;
 }
 
 bool WebRtcVoiceMediaChannel::SetSendParameters(
     const AudioSendParameters& params) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSendParameters");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
                << params.ToString();
   // TODO(pthatcher): Refactor this to be more clean now that we have
   // all the information at once.
 
   if (!SetSendCodecs(params.codecs)) {
     return false;
   }
 
@@ skipping 11 matching lines @@
   }
 
   if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) {
     return false;
   }
   return SetOptions(params.options);
 }
 
 bool WebRtcVoiceMediaChannel::SetRecvParameters(
     const AudioRecvParameters& params) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetRecvParameters");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: "
                << params.ToString();
   // TODO(pthatcher): Refactor this to be more clean now that we have
   // all the information at once.
 
   if (!SetRecvCodecs(params.codecs)) {
     return false;
   }
 
@@ skipping 363 matching lines @@
 
 bool WebRtcVoiceMediaChannel::PausePlayout() {
   return ChangePlayout(false);
 }
 
 bool WebRtcVoiceMediaChannel::ResumePlayout() {
   return ChangePlayout(desired_playout_);
 }
 
 bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (playout_ == playout) {
     return true;
   }
 
   for (const auto& ch : recv_streams_) {
     if (!SetPlayout(ch.second->channel(), playout)) {
       LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
                     << ch.second->channel() << " failed";
       return false;
@@ skipping 13 matching lines @@
 
 bool WebRtcVoiceMediaChannel::PauseSend() {
   return ChangeSend(SEND_NOTHING);
 }
 
 bool WebRtcVoiceMediaChannel::ResumeSend() {
   return ChangeSend(desired_send_);
 }
 
 bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangeSend");
   if (send_ == send) {
     return true;
   }
 
   // Apply channel specific options when channel is enabled for sending.
   if (send == SEND_MICROPHONE) {
     engine()->ApplyOptions(options_);
   }
 
   // Change the settings on each send channel.
@@ skipping 62 matching lines @@
     LOG_RTCERR1(DeRegisterExternalTransport, channel);
   }
   if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
     LOG_RTCERR1(DeleteChannel, channel);
     return false;
   }
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
 
   uint32_t ssrc = sp.first_ssrc();
   RTC_DCHECK(0 != ssrc);
 
   if (GetSendChannelId(ssrc) != -1) {
     LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
     return false;
   }
@@ skipping 33 matching lines @@
       engine()->voe()->base()->AssociateSendChannel(recv_channel, channel);
       LOG(LS_INFO) << "VoiceEngine channel #" << recv_channel
                    << " is associated with channel #" << channel << ".";
     }
   }
 
   return ChangeSend(channel, desired_send_);
 }
 
 bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveSendStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
 
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
     LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
                     << " which doesn't exist.";
     return false;
   }
 
   int channel = it->second->channel();
   ChangeSend(channel, SEND_NOTHING);
 
   // Clean up and delete the send stream+channel.
   LOG(LS_INFO) << "Removing audio send stream " << ssrc
                << " with VoiceEngine channel #" << channel << ".";
   delete it->second;
   send_streams_.erase(it);
   if (!DeleteVoEChannel(channel)) {
     return false;
   }
   if (send_streams_.empty()) {
     ChangeSend(SEND_NOTHING);
   }
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddRecvStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
 
   if (!ValidateStreamParams(sp)) {
     return false;
   }
 
   const uint32_t ssrc = sp.first_ssrc();
   if (ssrc == 0) {
     LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
@@ skipping 59 matching lines @@
                                          transport_cc_enabled_, sp.sync_label,
                                          recv_rtp_extensions_, call_)));
 
   SetNack(channel, nack_enabled_);
   SetPlayout(channel, playout_);
 
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
 
   const auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
     LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
                     << " which doesn't exist.";
     return false;
   }
 
@@ skipping 318 matching lines @@
       LOG(LS_INFO) << "Failed to set codec " << codec.plname
                    << " to bitrate " << bps << " bps"
                    << ", requires at least " << codec.rate << " bps.";
       return false;
     }
     return true;
   }
 }
 
 bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(info);
 
   // 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;
     sinfo.add_ssrc(stats.local_ssrc);
     sinfo.bytes_sent = stats.bytes_sent;
@@ skipping 107 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE