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WebRtc_Word32 => int32_t etc. in audio_coding/

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BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-09 00:28:06 +00:00
parent 6faf71d27b
commit 0946a56023
382 changed files with 8469 additions and 8488 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@ -28,7 +28,7 @@
namespace webrtc {
TestPacketization::TestPacketization(RTPStream *rtpStream,
WebRtc_UWord16 frequency)
uint16_t frequency)
: _rtpStream(rtpStream),
_frequency(frequency),
_seqNo(0) {
@ -36,12 +36,12 @@ TestPacketization::TestPacketization(RTPStream *rtpStream,
TestPacketization::~TestPacketization() { }
WebRtc_Word32 TestPacketization::SendData(
int32_t TestPacketization::SendData(
const FrameType /* frameType */,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const uint16_t payloadSize,
const RTPFragmentationHeader* /* fragmentation */) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
_frequency);
@ -103,7 +103,7 @@ void Sender::Teardown() {
bool Sender::Add10MsData() {
if (!_pcmFile.EndOfFile()) {
_pcmFile.Read10MsData(_audioFrame);
WebRtc_Word32 ok = _acm->Add10MsData(_audioFrame);
int32_t ok = _acm->Add10MsData(_audioFrame);
if (ok != 0) {
printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
exit(1);
@ -114,7 +114,7 @@ bool Sender::Add10MsData() {
}
bool Sender::Process() {
WebRtc_Word32 ok = _acm->Process();
int32_t ok = _acm->Process();
if (ok < 0) {
printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
exit(1);
@ -145,7 +145,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
noOfCodecs = acm->NumberOfCodecs();
for (int i = 0; i < noOfCodecs; i++) {
acm->Codec((WebRtc_UWord8) i, &recvCodec);
acm->Codec((uint8_t) i, &recvCodec);
if (acm->RegisterReceiveCodec(recvCodec) != 0) {
printf("Unable to register codec: for run: codecId: %d\n", codeId);
exit(1);
@ -177,7 +177,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
}
_realPayloadSizeBytes = 0;
_playoutBuffer = new WebRtc_Word16[WEBRTC_10MS_PCM_AUDIO];
_playoutBuffer = new int16_t[WEBRTC_10MS_PCM_AUDIO];
_frequency = playSampFreq;
_acm = acm;
_firstTime = true;
@ -207,7 +207,7 @@ bool Receiver::IncomingPacket() {
}
}
WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload,
int32_t ok = _acm->IncomingPacket(_incomingPayload,
_realPayloadSizeBytes, _rtpInfo);
if (ok != 0) {
printf("Error when inserting packet to ACM, for run: codecId: %d\n",
@ -239,8 +239,8 @@ bool Receiver::PlayoutData() {
}
void Receiver::Run() {
WebRtc_UWord8 counter500Ms = 50;
WebRtc_UWord32 clock = 0;
uint8_t counter500Ms = 50;
uint32_t clock = 0;
while (counter500Ms > 0) {
if (clock == 0 || clock >= _nextTime) {