| Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
|
| index a0f1839e73393c4ad017c6d6d5393a740187f64d..c1c3cb3d2a2f9d1b28b6d7692777a81cc9871c47 100644
|
| --- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
|
| +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
|
| @@ -135,15 +135,6 @@ class AudioEncoder {
|
| virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
|
| rtc::ArrayView<const int16_t> audio,
|
| rtc::Buffer* encoded) = 0;
|
| -
|
| - private:
|
| - // This function is deprecated. It was used to return the maximum number of
|
| - // bytes that can be produced by the encoder at each Encode() call. Since the
|
| - // Encode interface was changed to use rtc::Buffer, this is no longer
|
| - // applicable. It is only kept in to avoid breaking subclasses that still have
|
| - // it implemented (with the override attribute). It will be removed as soon
|
| - // as these subclasses have been given a chance to change.
|
| - virtual size_t MaxEncodedBytes() const;
|
| };
|
| } // namespace webrtc
|
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
|
|
|