| 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); }
|
|
|