commit 0946a56023d821e0deca04029bb016ae1f23aa82
author pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Tue Apr 09 00:28:06 2013 +0000
committer pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Tue Apr 09 00:28:06 2013 +0000
tree 504a7ac85144a61e871e5858ffdae54fddd99815
parent 6faf71d27b89ce4fc29ebd8148fc2ffd90983a61

WebRtc_Word32 => int32_t etc. in audio_coding/

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

webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc

diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index c4f9a47..58e6299 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -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() { }
 
-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 @@
 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::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 @@
 
   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 @@
   }
 
   _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 @@
       }
    }
 
-   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 @@
 }
 
 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) {