| Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
|
| diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
|
| index 9c3183271b75bcbdaca46d20cd915d8ecee05323..6b5552e95dd5dead7f1c7b8a85c1482d59277cff 100644
|
| --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
|
| +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
|
| @@ -27,6 +27,7 @@
|
| #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
|
| #include "webrtc/system_wrappers/interface/trace.h"
|
| #include "webrtc/typedefs.h"
|
| +#include "webrtc/video/rtc_event_log.h"
|
|
|
| namespace webrtc {
|
|
|
| @@ -145,7 +146,9 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
|
| first_frame_(true),
|
| callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
|
| packetization_callback_(NULL),
|
| - vad_callback_(NULL) {
|
| + vad_callback_(NULL),
|
| + event_log_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
|
| + event_log_(nullptr) {
|
| if (InitializeReceiverSafe() < 0) {
|
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
|
| "Cannot initialize receiver");
|
| @@ -736,6 +739,11 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
|
| "PlayoutData failed, RecOut Failed");
|
| return -1;
|
| }
|
| + {
|
| + CriticalSectionScoped lock(event_log_crit_sect_);
|
| + if (event_log_)
|
| + event_log_->LogDebugEvent(RtcEventLog::DebugEvent::kAudioPlayout);
|
| + }
|
|
|
| audio_frame->id_ = id_;
|
| return 0;
|
| @@ -971,6 +979,11 @@ void AudioCodingModuleImpl::DisableNack() {
|
| receiver_.DisableNack();
|
| }
|
|
|
| +void AudioCodingModuleImpl::SetEventLog(RtcEventLog* event_log) {
|
| + CriticalSectionScoped lock(event_log_crit_sect_);
|
| + event_log_ = event_log;
|
| +}
|
| +
|
| std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
|
| int64_t round_trip_time_ms) const {
|
| return receiver_.GetNackList(round_trip_time_ms);
|
|
|