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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: All SetEventLog() functions have been removed, the pointer is now propagated through the constructo… Created 5 years, 4 months ago
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« webrtc/voice_engine/channel.h ('K') | « webrtc/voice_engine/channel.h ('k') | webrtc/voice_engine/channel_manager.h » ('j') | webrtc/voice_engine/channel_manager.cc » ('J')
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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 8992425b537b0e32d329a470d58fe50d2619171d..eeecc64acc20b131c960ada57e02f7573e1ce7eb 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,
@@ -713,86 +712,92 @@ Channel::RecordFileEnded(int32_t id)
Channel::Channel(int32_t channelId,
uint32_t instanceId,
- const Config& config) :
- _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
- _instanceId(instanceId),
- _channelId(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())),
- telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
- _outputAudioLevel(),
- _externalTransport(false),
- _inputFilePlayerPtr(NULL),
- _outputFilePlayerPtr(NULL),
- _outputFileRecorderPtr(NULL),
- // Avoid conflict with other channels by adding 1024 - 1026,
- // won't use as much as 1024 channels.
- _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
- _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
- _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
- _outputFileRecording(false),
- _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
- _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
- _outputExternalMedia(false),
- _inputExternalMediaCallbackPtr(NULL),
- _outputExternalMediaCallbackPtr(NULL),
- _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),
- playout_timestamp_rtp_(0),
- playout_timestamp_rtcp_(0),
- playout_delay_ms_(0),
- _numberOfDiscardedPackets(0),
- send_sequence_number_(0),
- ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
- rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
- capture_start_rtp_time_stamp_(-1),
- capture_start_ntp_time_ms_(-1),
- _engineStatisticsPtr(NULL),
- _outputMixerPtr(NULL),
- _transmitMixerPtr(NULL),
- _moduleProcessThreadPtr(NULL),
- _audioDeviceModulePtr(NULL),
- _voiceEngineObserverPtr(NULL),
- _callbackCritSectPtr(NULL),
- _transportPtr(NULL),
- _rxVadObserverPtr(NULL),
- _oldVadDecision(-1),
- _sendFrameType(0),
- _externalMixing(false),
- _mixFileWithMicrophone(false),
- _mute(false),
- _panLeft(1.0f),
- _panRight(1.0f),
- _outputGain(1.0f),
- _playOutbandDtmfEvent(false),
- _playInbandDtmfEvent(false),
- _lastLocalTimeStamp(0),
- _lastPayloadType(0),
- _includeAudioLevelIndication(false),
- _outputSpeechType(AudioFrame::kNormalSpeech),
- _average_jitter_buffer_delay_us(0),
- least_required_delay_ms_(0),
- _previousTimestamp(0),
- _recPacketDelayMs(20),
- _RxVadDetection(false),
- _rxAgcIsEnabled(false),
- _rxNsIsEnabled(false),
- restored_packet_in_use_(false),
- rtcp_observer_(new VoERtcpObserver(this)),
- network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
- assoc_send_channel_lock_(CriticalSectionWrapper::CreateCriticalSection()),
- associate_send_channel_(ChannelOwner(nullptr))
-{
+ RtcEventLog* const event_log,
+ const Config& config)
+ : _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ volume_settings_critsect_(
+ *CriticalSectionWrapper::CreateCriticalSection()),
+ _instanceId(instanceId),
+ _channelId(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())),
+ telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
+ _outputAudioLevel(),
+ _externalTransport(false),
+ _inputFilePlayerPtr(NULL),
+ _outputFilePlayerPtr(NULL),
+ _outputFileRecorderPtr(NULL),
+ // Avoid conflict with other channels by adding 1024 - 1026,
+ // won't use as much as 1024 channels.
+ _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
+ _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
+ _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
+ _outputFileRecording(false),
+ _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
+ _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
+ _outputExternalMedia(false),
+ _inputExternalMediaCallbackPtr(NULL),
+ _outputExternalMediaCallbackPtr(NULL),
+ _timeStamp(0), // This is just an offset, RTP module will add it's own
Henrik Grunell WebRTC 2015/08/12 15:29:53 Nit: Put the comment above, follows the style abov
+ // random offset
+ _sendTelephoneEventPayloadType(106),
+ ntp_estimator_(Clock::GetRealTimeClock()),
+ jitter_buffer_playout_timestamp_(0),
+ playout_timestamp_rtp_(0),
+ playout_timestamp_rtcp_(0),
+ playout_delay_ms_(0),
+ _numberOfDiscardedPackets(0),
+ send_sequence_number_(0),
+ ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
+ capture_start_rtp_time_stamp_(-1),
+ capture_start_ntp_time_ms_(-1),
+ _engineStatisticsPtr(NULL),
+ _outputMixerPtr(NULL),
+ _transmitMixerPtr(NULL),
+ _moduleProcessThreadPtr(NULL),
+ _audioDeviceModulePtr(NULL),
+ _voiceEngineObserverPtr(NULL),
+ _callbackCritSectPtr(NULL),
+ _transportPtr(NULL),
+ _rxVadObserverPtr(NULL),
+ _oldVadDecision(-1),
+ _sendFrameType(0),
+ _externalMixing(false),
+ _mixFileWithMicrophone(false),
+ _mute(false),
+ _panLeft(1.0f),
+ _panRight(1.0f),
+ _outputGain(1.0f),
+ _playOutbandDtmfEvent(false),
+ _playInbandDtmfEvent(false),
+ _lastLocalTimeStamp(0),
+ _lastPayloadType(0),
+ _includeAudioLevelIndication(false),
+ _outputSpeechType(AudioFrame::kNormalSpeech),
+ _average_jitter_buffer_delay_us(0),
+ least_required_delay_ms_(0),
+ _previousTimestamp(0),
+ _recPacketDelayMs(20),
+ _RxVadDetection(false),
+ _rxAgcIsEnabled(false),
+ _rxNsIsEnabled(false),
+ restored_packet_in_use_(false),
+ rtcp_observer_(new VoERtcpObserver(this)),
+ network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
+ assoc_send_channel_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ associate_send_channel_(ChannelOwner(nullptr)) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::Channel() - ctor");
AudioCodingModule::Config acm_config;
@@ -805,6 +810,7 @@ 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();
« webrtc/voice_engine/channel.h ('K') | « webrtc/voice_engine/channel.h ('k') | webrtc/voice_engine/channel_manager.h » ('j') | webrtc/voice_engine/channel_manager.cc » ('J')

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