| Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| index 3f11af1f9e0c665504520d7a485ecb5a544f487f..9294f2fed46313691d02ba5830c0cc4ef2204224 100644
|
| --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| @@ -54,7 +54,6 @@ class AudioEncoderOpus final : public AudioEncoder {
|
| explicit AudioEncoderOpus(const CodecInst& codec_inst);
|
| ~AudioEncoderOpus() override;
|
|
|
| - size_t MaxEncodedBytes() const override;
|
| int SampleRateHz() const override;
|
| size_t NumChannels() const override;
|
| size_t Num10MsFramesInNextPacket() const override;
|
| @@ -79,7 +78,7 @@ class AudioEncoderOpus final : public AudioEncoder {
|
| ApplicationMode application() const { return config_.application; }
|
| bool dtx_enabled() const { return config_.dtx_enabled; }
|
|
|
| -protected:
|
| + protected:
|
| EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
|
| rtc::ArrayView<const int16_t> audio,
|
| rtc::Buffer* encoded) override;
|
| @@ -87,6 +86,7 @@ protected:
|
| private:
|
| size_t Num10msFramesPerPacket() const;
|
| size_t SamplesPer10msFrame() const;
|
| + size_t ApproximateEncodedBytes() const;
|
| bool RecreateEncoderInstance(const Config& config);
|
|
|
| Config config_;
|
|
|