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

@ -78,8 +78,7 @@ int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
rtp_info.type.Audio.channel = 1;
memcpy(payload_data_, payload_data, payload_size);
status = receiver_acm_->IncomingPacket(payload_data_, payload_size,
rtp_info);
status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
payload_size_ = payload_size;
timestamp_diff_ = timestamp - last_in_timestamp_;
@ -127,8 +126,8 @@ TestAllCodecs::~TestAllCodecs() {
}
void TestAllCodecs::Perform() {
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
const std::string file_name = webrtc::test::ResourcePath(
"audio_coding/testfile32kHz", "pcm");
infile_a_.Open(file_name, 32000, "rb");
if (test_mode_ == 0) {
@ -725,9 +724,9 @@ void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
// packet. If variable rate codec (extra_byte == -1), set to -1 (65535).
if (extra_byte != -1) {
// Add 0.875 to always round up to a whole byte
packet_size_bytes_ =
static_cast<uint16_t>(static_cast<float>(packet_size * rate) /
static_cast<float>(sampling_freq_hz * 8) + 0.875) + extra_byte;
packet_size_bytes_ = static_cast<uint16_t>(static_cast<float>(packet_size
* rate) / static_cast<float>(sampling_freq_hz * 8) + 0.875)
+ extra_byte;
} else {
// Packets will have a variable size.
packet_size_bytes_ = -1;