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AudioEncoder: documentation fix

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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
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jmarusic@webrtc.org 2015-02-27 09:20:01 +00:00
parent ea89495786
commit abbdd520b0
1 changed files with 1 additions and 1 deletions
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2
webrtc/modules/audio_coding/codecs/audio_encoder.h
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@ -57,7 +57,7 @@ class AudioEncoder {
// 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,