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..8900659f48eb812ca6b214b481daac16e808e8ff 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 SufficientOutputBufferSize() const; |
bool RecreateEncoderInstance(const Config& config); |
Config config_; |