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Unified Diff: webrtc/voice_engine/channel.cc

Issue 1267683002: Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years, 4 months ago
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Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index c42ac0a9278c6aad1f3968ddecb9a32949792df1..0a6a7526afd48a2b8ff7beb049b9036f0b0d4ee3 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -628,17 +628,16 @@ Channel::NeededFrequency(int32_t id)
return(highestNeeded);
}
-int32_t
-Channel::CreateChannel(Channel*& channel,
- int32_t channelId,
- uint32_t instanceId,
- const Config& config)
-{
+int32_t Channel::CreateChannel(Channel*& channel,
+ int32_t channelId,
+ uint32_t instanceId,
+ RtcEventLog* const event_log,
+ const Config& config) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)",
channelId, instanceId);
- channel = new Channel(channelId, instanceId, config);
+ channel = new Channel(channelId, instanceId, event_log, config);
if (channel == NULL)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice,
@@ -713,8 +712,9 @@ Channel::RecordFileEnded(int32_t id)
Channel::Channel(int32_t channelId,
uint32_t instanceId,
- const Config& config) :
- _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ RtcEventLog* const event_log,
+ const Config& config)
+ : _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
_instanceId(instanceId),
@@ -722,11 +722,15 @@ Channel::Channel(int32_t channelId,
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
- rtp_receive_statistics_(ReceiveStatistics::Create(
- Clock::GetRealTimeClock())),
- rtp_receiver_(RtpReceiver::CreateAudioReceiver(
- VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this,
- this, this, rtp_payload_registry_.get())),
+ rtp_receive_statistics_(
+ ReceiveStatistics::Create(Clock::GetRealTimeClock())),
+ rtp_receiver_(
+ RtpReceiver::CreateAudioReceiver(VoEModuleId(instanceId, channelId),
+ Clock::GetRealTimeClock(),
+ this,
+ this,
+ this,
+ rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
_outputAudioLevel(),
_externalTransport(false),
@@ -744,7 +748,8 @@ Channel::Channel(int32_t channelId,
_outputExternalMedia(false),
_inputExternalMediaCallbackPtr(NULL),
_outputExternalMediaCallbackPtr(NULL),
- _timeStamp(0), // This is just an offset, RTP module will add it's own random offset
+ _timeStamp(0), // This is just an offset, RTP module will add it's own
+ // random offset
_sendTelephoneEventPayloadType(106),
ntp_estimator_(Clock::GetRealTimeClock()),
jitter_buffer_playout_timestamp_(0),
@@ -791,8 +796,7 @@ Channel::Channel(int32_t channelId,
rtcp_observer_(new VoERtcpObserver(this)),
network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
assoc_send_channel_lock_(CriticalSectionWrapper::CreateCriticalSection()),
- associate_send_channel_(ChannelOwner(nullptr))
-{
+ associate_send_channel_(ChannelOwner(nullptr)) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::Channel() - ctor");
AudioCodingModule::Config acm_config;
@@ -805,6 +809,7 @@ Channel::Channel(int32_t channelId,
}
acm_config.neteq_config.enable_fast_accelerate =
config.Get<NetEqFastAccelerate>().enabled;
+ acm_config.event_log = event_log;
audio_coding_.reset(AudioCodingModule::Create(acm_config));
_inbandDtmfQueue.ResetDtmf();

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