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

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

View File

@ -18,39 +18,42 @@ namespace webrtc {
class ACMPCMU : public ACMGenericCodec {
public:
explicit ACMPCMU(int16_t codec_id);
~ACMPCMU();
virtual ~ACMPCMU();
// for FEC
ACMGenericCodec* CreateInstance(void);
virtual ACMGenericCodec* CreateInstance(void) OVERRIDE;
int16_t InternalEncode(uint8_t* bitstream,
int16_t* bitstream_len_byte);
virtual int16_t InternalEncode(uint8_t* bitstream,
int16_t* bitstream_len_byte) OVERRIDE;
int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
virtual int16_t InternalInitEncoder(
WebRtcACMCodecParams* codec_params) OVERRIDE;
int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
virtual int16_t InternalInitDecoder(
WebRtcACMCodecParams* codec_params) OVERRIDE;
protected:
int16_t DecodeSafe(uint8_t* bitstream,
int16_t bitstream_len_byte,
int16_t* audio,
int16_t* audio_samples,
int8_t* speech_type);
virtual int16_t DecodeSafe(uint8_t* bitstream,
int16_t bitstream_len_byte,
int16_t* audio,
int16_t* audio_samples,
int8_t* speech_type) OVERRIDE;
int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
const CodecInst& codec_inst);
virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
const CodecInst& codec_inst) OVERRIDE;
void DestructEncoderSafe();
virtual void DestructEncoderSafe() OVERRIDE;
void DestructDecoderSafe();
virtual void DestructDecoderSafe() OVERRIDE;
int16_t InternalCreateEncoder();
virtual int16_t InternalCreateEncoder() OVERRIDE;
int16_t InternalCreateDecoder();
virtual int16_t InternalCreateDecoder() OVERRIDE;
void InternalDestructEncoderInst(void* ptr_inst);
virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
void SplitStereoPacket(uint8_t* payload, int32_t* payload_length);
virtual void SplitStereoPacket(uint8_t* payload,
int32_t* payload_length) OVERRIDE;
};
} // namespace webrtc