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 b3f6965fff955fcaa4c3c01aed079d5d89d31a1c..dec87b2b7a4a44b4794879fef346b32fce4a4a24 100644 |
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h |
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h |
@@ -35,6 +35,7 @@ |
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; |
@@ -43,7 +44,7 @@ |
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; |