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
This commit is contained in:
tina.legrand@webrtc.org
2013-05-03 07:34:12 +00:00
parent 03efc89151
commit d5726a1286
34 changed files with 4385 additions and 5230 deletions

View File

@ -24,21 +24,18 @@ namespace webrtc {
#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 @@ class Receiver {
uint32_t _nextTime;
};
class EncodeDecodeTest: public ACMTest {
class EncodeDecodeTest : public ACMTest {
public:
EncodeDecodeTest();
EncodeDecodeTest(int testMode);
@ -109,6 +106,6 @@ class EncodeDecodeTest: public ACMTest {
Receiver _receiver;
};
} // namespace webrtc
} // namespace webrtc
#endif