| 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()
|
|
|