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Remove the useless dummy state parameter to WebRtcG711_*

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R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2014-11-04 13:23:36 +00:00
parent b5d045e94d
commit c78cf97ecb
8 changed files with 22 additions and 61 deletions
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6
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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@@ -82,8 +82,7 @@ bool AudioEncoderPcm::Encode(uint32_t timestamp,
int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
return WebRtcG711_EncodeA(NULL,
const_cast<int16_t*>(audio),
return WebRtcG711_EncodeA(const_cast<int16_t*>(audio),
static_cast<int16_t>(input_len),
reinterpret_cast<int16_t*>(encoded));
}
@@ -91,8 +90,7 @@ int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
return WebRtcG711_EncodeU(NULL,
const_cast<int16_t*>(audio),
return WebRtcG711_EncodeU(const_cast<int16_t*>(audio),
static_cast<int16_t>(input_len),
reinterpret_cast<int16_t*>(encoded));
}