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Enhanced documentation. Replaced DCHECK with CHECK.

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R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34309004

Cr-Commit-Position: refs/heads/master@{#8461}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8461 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
jmarusic@webrtc.org 2015-02-23 15:27:52 +00:00
parent 962c62475e
commit 2acec4cc32
3 changed files with 7 additions and 6 deletions
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2
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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@ -78,7 +78,7 @@ bool AudioEncoderPcm::EncodeInternal(uint32_t rtp_timestamp,
}
CHECK_EQ(speech_buffer_.size(), static_cast<size_t>(full_frame_samples_));
int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
DCHECK_GE(ret, 0);
CHECK_GE(ret, 0);
speech_buffer_.clear();
info->encoded_timestamp = first_timestamp_in_buffer_;
info->payload_type = payload_type_;

8
webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h
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@ -34,8 +34,8 @@ extern "C" {
* 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.
* Always equal to len input parameter.
*/
int16_t WebRtcG711_EncodeA(const int16_t* speechIn,
@ -55,8 +55,8 @@ int16_t WebRtcG711_EncodeA(const int16_t* speechIn,
* 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.
* Always equal to len input parameter.
*/
int16_t WebRtcG711_EncodeU(const int16_t* speechIn,

3
webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
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@ -32,7 +32,8 @@ extern "C" {
* Output:
* - encoded : Encoded data vector (big endian 16 bit)
*
* Returned value : Size in bytes of encoded
* Returned value : Length (in bytes) of coded data.
* Always equal to twice the len input parameter.
*/
int16_t WebRtcPcm16b_Encode(const int16_t* speech,