Formatting ACM tests
Pure formatting of all files located in /webrtc/modules/audio_coding/main/test/
Smaller manual modifications done after using Eclipse formatting tool, like wrapping long lines (mostly comments).
BUG=issue1024
Review URL: https://webrtc-codereview.appspot.com/1342004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3946 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
index 9b58d4d..4c0216a 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -24,21 +24,18 @@
#define MAX_INCOMING_PAYLOAD 8096
// TestPacketization callback which writes the encoded payloads to file
-class TestPacketization: public AudioPacketizationCallback {
+class TestPacketization : public AudioPacketizationCallback {
public:
TestPacketization(RTPStream *rtpStream, uint16_t frequency);
~TestPacketization();
- virtual int32_t SendData(const FrameType frameType,
- const uint8_t payloadType,
- const uint32_t timeStamp,
- const uint8_t* payloadData,
+ virtual int32_t SendData(const FrameType frameType, const uint8_t payloadType,
+ const uint32_t timeStamp, const uint8_t* payloadData,
const uint16_t payloadSize,
const RTPFragmentationHeader* fragmentation);
private:
static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
- int16_t seqNo, uint32_t timeStamp,
- uint32_t ssrc);
+ int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
RTPStream* _rtpStream;
int32_t _frequency;
int16_t _seqNo;
@@ -92,7 +89,7 @@
uint32_t _nextTime;
};
-class EncodeDecodeTest: public ACMTest {
+class EncodeDecodeTest : public ACMTest {
public:
EncodeDecodeTest();
EncodeDecodeTest(int testMode);
@@ -109,6 +106,6 @@
Receiver _receiver;
};
-} // namespace webrtc
+} // namespace webrtc
#endif