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Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/

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BUG=163
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1900004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4447 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-07-31 15:54:00 +00:00
parent e72428442d
commit 2d1a55caed
33 changed files with 443 additions and 351 deletions
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10
webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
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@@ -591,6 +591,10 @@ bool ACMGenericCodec::CanChangeEncodingParam(CodecInst& /*codec_inst*/) {
return true;
}
void ACMGenericCodec::CurrentRate(int32_t& /* rate_bps */) {
return;
}
int16_t ACMGenericCodec::InitDecoder(WebRtcACMCodecParams* codec_params,
bool force_initialization) {
WriteLockScoped lockCodc(codec_wrapper_lock_);
@@ -1159,6 +1163,10 @@ bool ACMGenericCodec::IsAudioBufferFresh() const {
return is_audio_buff_fresh_;
}
int16_t ACMGenericCodec::UpdateDecoderSampFreq(int16_t /* codec_id */) {
return 0;
}
// This function is replaced by codec specific functions for some codecs.
int16_t ACMGenericCodec::EncoderSampFreq(uint16_t& samp_freq_hz) {
int32_t f;
@@ -1231,4 +1239,6 @@ int16_t ACMGenericCodec::REDPayloadISAC(const int32_t /* isac_rate */,
return -1;
}
bool ACMGenericCodec::IsTrueStereoCodec() { return false; }
} // namespace webrtc