| 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 d30daaaf3407f6a221d987eca57ba3917429fee8..f3e47edfd9edb9c2ad5c19f1919a888d7f5047f8 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); }
|
|
|