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/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
index 1f68aca..c47a582 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -78,8 +78,7 @@
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 @@
}
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 @@
// 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;