WebRtc_Word32 -> int32_t in utility/
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
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index f023d22..85fb698 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -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 @@
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 @@
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 @@
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 @@
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 @@
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;
}