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 f7607c6521c2f434effb96ad1eaec1393bb2bc6d..0244882de628b704e733f93c47a4162f9f72b2a2 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; |
@@ -1273,6 +1275,14 @@ 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() |