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_ |