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g722: Enhanced documentation. Added CHECK.

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R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43409004

Cr-Commit-Position: refs/heads/master@{#8462}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8462 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
jmarusic@webrtc.org 2015-02-23 15:41:30 +00:00
parent 2acec4cc32
commit f3a306b5bc
2 changed files with 2 additions and 2 deletions
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1
webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
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@ -101,6 +101,7 @@ bool AudioEncoderG722::EncodeInternal(uint32_t rtp_timestamp,
const int encoded = WebRtcG722_Encode(
encoders_[i].encoder, encoders_[i].speech_buffer.get(),
samples_per_channel, encoders_[i].encoded_buffer.get());
CHECK_GE(encoded, 0);
CHECK_EQ(encoded, samples_per_channel / 2);
}

3
webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
View File

@ -91,8 +91,7 @@ int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
* Output:
* - encoded : The encoded data vector
*
* Return value : >0 - Length (in bytes) of coded data
* -1 - Error
* Return value : Length (in bytes) of coded data
*/
int16_t WebRtcG722_Encode(G722EncInst* G722enc_inst,