Index: webrtc/modules/audio_coding/acm2/audio_coding_module.cc |
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc |
index 14958181859501459181f56380e79bf885e40376..b2b0c0bb0fa81d08352a50137eac99061eb119df 100644 |
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc |
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc |
@@ -204,6 +204,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule { |
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override; |
+ ANAStats GetANAStats() const override; |
+ |
private: |
struct InputData { |
uint32_t input_timestamp; |
@@ -1270,6 +1272,14 @@ void AudioCodingModuleImpl::GetDecodingCallStatistics( |
receiver_.GetDecodingCallStatistics(call_stats); |
} |
+ANAStats AudioCodingModuleImpl::GetANAStats() const { |
+ rtc::CritScope lock(&acm_crit_sect_); |
+ if (encoder_stack_) |
+ return encoder_stack_->GetANAStats(); |
+ // If no encoder is set, return default stats. |
+ return ANAStats(); |
+} |
+ |
} // namespace |
AudioCodingModule::Config::Config() |