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WebRtc_Word32 -> int32_t in utility/

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BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3797 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-09 13:32:55 +00:00
parent 0ea11c1768
commit c75102eba7
19 changed files with 360 additions and 366 deletions
Download Patch File Download Diff File Expand all files Collapse all files

46
webrtc/modules/utility/source/coder.cc
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@ -20,7 +20,7 @@
#endif
namespace webrtc {
AudioCoder::AudioCoder(WebRtc_UWord32 instanceID)
AudioCoder::AudioCoder(uint32_t instanceID)
: _acm(AudioCodingModule::Create(instanceID)),
_receiveCodec(),
_encodeTimestamp(0),
@ -38,8 +38,8 @@ AudioCoder::~AudioCoder()
AudioCodingModule::Destroy(_acm);
}
WebRtc_Word32 AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
ACMAMRPackingFormat amrFormat)
int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
ACMAMRPackingFormat amrFormat)
{
if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1)
{
@ -48,8 +48,8 @@ WebRtc_Word32 AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
return 0;
}
WebRtc_Word32 AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
ACMAMRPackingFormat amrFormat)
int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
ACMAMRPackingFormat amrFormat)
{
if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1)
{
@ -59,16 +59,16 @@ WebRtc_Word32 AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
return 0;
}
WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio,
WebRtc_UWord32 sampFreqHz,
const WebRtc_Word8* incomingPayload,
WebRtc_Word32 payloadLength)
int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
uint32_t sampFreqHz,
const int8_t* incomingPayload,
int32_t payloadLength)
{
if (payloadLength > 0)
{
const WebRtc_UWord8 payloadType = _receiveCodec.pltype;
const uint8_t payloadType = _receiveCodec.pltype;
_decodeTimestamp += _receiveCodec.pacsize;
if(_acm->IncomingPayload((const WebRtc_UWord8*) incomingPayload,
if(_acm->IncomingPayload((const uint8_t*) incomingPayload,
payloadLength,
payloadType,
_decodeTimestamp) == -1)
@ -76,18 +76,18 @@ WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio,
return -1;
}
}
return _acm->PlayoutData10Ms((WebRtc_UWord16)sampFreqHz, &decodedAudio);
return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio);
}
WebRtc_Word32 AudioCoder::PlayoutData(AudioFrame& decodedAudio,
WebRtc_UWord16& sampFreqHz)
int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
uint16_t& sampFreqHz)
{
return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
}
WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio,
WebRtc_Word8* encodedData,
WebRtc_UWord32& encodedLengthInBytes)
int32_t AudioCoder::Encode(const AudioFrame& audio,
int8_t* encodedData,
uint32_t& encodedLengthInBytes)
{
// Fake a timestamp in case audio doesn't contain a correct timestamp.
// Make a local copy of the audio frame since audio is const
@ -112,15 +112,15 @@ WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio,
return 0;
}
WebRtc_Word32 AudioCoder::SendData(
int32_t AudioCoder::SendData(
FrameType /* frameType */,
WebRtc_UWord8 /* payloadType */,
WebRtc_UWord32 /* timeStamp */,
const WebRtc_UWord8* payloadData,
WebRtc_UWord16 payloadSize,
uint8_t /* payloadType */,
uint32_t /* timeStamp */,
const uint8_t* payloadData,
uint16_t payloadSize,
const RTPFragmentationHeader* /* fragmentation*/)
{
memcpy(_encodedData,payloadData,sizeof(WebRtc_UWord8) * payloadSize);
memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
_encodedLengthInBytes = payloadSize;
return 0;
}