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 254c2f420bf097c333527f82285094594bfc7c45..f00b2432f2430a47bfb67be58c276539107f9a53 100644 |
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc |
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc |
@@ -104,7 +104,6 @@ void ConvertEncodedInfoToFragmentationHeader( |
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(); } |
@@ -120,13 +119,6 @@ class RawAudioEncoderWrapper final : public AudioEncoder { |
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); } |
bool SetDtx(bool enable) override { return enc_->SetDtx(enable); } |