| Index: webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
|
| index dec87b2b7a4a44b4794879fef346b32fce4a4a24..b3f6965fff955fcaa4c3c01aed079d5d89d31a1c 100644
|
| --- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
|
| +++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
|
| @@ -35,7 +35,6 @@ class AudioEncoderG722 final : public AudioEncoder {
|
| explicit AudioEncoderG722(const CodecInst& codec_inst);
|
| ~AudioEncoderG722() override;
|
|
|
| - size_t MaxEncodedBytes() const override;
|
| int SampleRateHz() const override;
|
| size_t NumChannels() const override;
|
| int RtpTimestampRateHz() const override;
|
| @@ -44,7 +43,7 @@ class AudioEncoderG722 final : public AudioEncoder {
|
| int GetTargetBitrate() const override;
|
| void Reset() override;
|
|
|
| -protected:
|
| + protected:
|
| EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
|
| rtc::ArrayView<const int16_t> audio,
|
| rtc::Buffer* encoded) override;
|
|
|