Index: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc |
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc |
index f3e47edfd9edb9c2ad5c19f1919a888d7f5047f8..d30daaaf3407f6a221d987eca57ba3917429fee8 100644 |
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc |
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc |
@@ -104,6 +104,7 @@ |
class RawAudioEncoderWrapper final : public AudioEncoder { |
public: |
RawAudioEncoderWrapper(AudioEncoder* enc) : enc_(enc) {} |
+ size_t MaxEncodedBytes() const override { return enc_->MaxEncodedBytes(); } |
int SampleRateHz() const override { return enc_->SampleRateHz(); } |
size_t NumChannels() const override { return enc_->NumChannels(); } |
int RtpTimestampRateHz() const override { return enc_->RtpTimestampRateHz(); } |
@@ -118,6 +119,13 @@ |
rtc::ArrayView<const int16_t> audio, |
rtc::Buffer* encoded) override { |
return enc_->Encode(rtp_timestamp, audio, encoded); |
+ } |
+ EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
+ rtc::ArrayView<const int16_t> audio, |
+ size_t max_encoded_bytes, |
+ uint8_t* encoded) override { |
+ return enc_->EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, |
+ encoded); |
} |
void Reset() override { return enc_->Reset(); } |
bool SetFec(bool enable) override { return enc_->SetFec(enable); } |