Rename internal AudioEncoder::Encode method to EncodeInternal
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index c0790a2..15d6bea 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -45,12 +45,12 @@
EncodedInfo* info) {
CHECK_EQ(num_samples_per_channel,
static_cast<size_t>(sample_rate_hz() / 100));
- bool ret = Encode(timestamp,
- audio,
- max_encoded_bytes,
- encoded,
- encoded_bytes,
- info);
+ bool ret = EncodeInternal(timestamp,
+ audio,
+ max_encoded_bytes,
+ encoded,
+ encoded_bytes,
+ info);
CHECK_LE(*encoded_bytes, max_encoded_bytes);
return ret;
}
@@ -68,12 +68,12 @@
virtual int Num10MsFramesInNextPacket() const = 0;
protected:
- virtual bool Encode(uint32_t timestamp,
- const int16_t* audio,
- size_t max_encoded_bytes,
- uint8_t* encoded,
- size_t* encoded_bytes,
- EncodedInfo* info) = 0;
+ virtual bool EncodeInternal(uint32_t timestamp,
+ const int16_t* audio,
+ size_t max_encoded_bytes,
+ uint8_t* encoded,
+ size_t* encoded_bytes,
+ EncodedInfo* info) = 0;
};
} // namespace webrtc