| commit | abbdd520b0ddefa6b1d9a798843c0999bc6f1f25 | [log] [tgz] |
|---|---|---|
| author | jmarusic@webrtc.org <jmarusic@webrtc.org> | Fri Feb 27 09:20:01 2015 +0000 |
| committer | jmarusic@webrtc.org <jmarusic@webrtc.org> | Fri Feb 27 09:20:25 2015 +0000 |
| tree | 9d37978cfbd8d24bdfa90e52fce7abbfe60f2414 | |
| parent | ea89495786f29db5369e89a4c7ea59780e0c6787 [diff] |
AudioEncoder: documentation fix Follow-up to https://webrtc-codereview.appspot.com/38279004/ R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38309004 Cr-Commit-Position: refs/heads/master@{#8524} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8524 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 08cf66f..738669d 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -57,7 +57,7 @@ // Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 * // num_channels() samples). Multi-channel audio must be sample-interleaved. // The encoder produces zero or more bytes of output in |encoded|, - // and provides the number of encoded bytes in |encoded_bytes|. + // and provides additional encoding information in |info|. // The caller is responsible for making sure that |max_encoded_bytes| is // not smaller than the number of bytes actually produced by the encoder. void Encode(uint32_t rtp_timestamp,