commit d5726a1286ce53c47ebd2d21d61b2772fc24aaed
author tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Fri May 03 07:34:12 2013 +0000
committer tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Fri May 03 07:34:12 2013 +0000
tree 1c6cbce92b36f873cb33bfc367e0143b91c16d01
parent 03efc89151725efeee622fa86ff24c4affcf521d

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

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

diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.cc b/webrtc/modules/audio_coding/main/test/PCMFile.cc
index 0b61bbb..5c4b98e 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.cc
@@ -30,8 +30,8 @@
       rewinded_(false),
       read_stereo_(false),
       save_stereo_(false) {
-  timestamp_ = (((uint32_t)rand() & 0x0000FFFF) << 16) |
-      ((uint32_t)rand() & 0x0000FFFF);
+  timestamp_ = (((uint32_t) rand() & 0x0000FFFF) << 16) |
+      ((uint32_t) rand() & 0x0000FFFF);
 }
 
 PCMFile::PCMFile(uint32_t timestamp)
@@ -84,8 +84,7 @@
   return 0;
 }
 
-int16_t PCMFile::ChooseFile(std::string* file_name,
-                            int16_t max_len,
+int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len,
                             uint16_t* frequency_hz) {
   char tmp_name[MAX_FILE_NAME_LENGTH_BYTE];
 
@@ -158,10 +157,8 @@
     channels = 2;
   }
 
-  int32_t payload_size = (int32_t) fread(audio_frame.data_,
-                                                    sizeof(uint16_t),
-                                                    samples_10ms_ * channels,
-                                                    pcm_file_);
+  int32_t payload_size = (int32_t) fread(audio_frame.data_, sizeof(uint16_t),
+                                         samples_10ms_ * channels, pcm_file_);
   if (payload_size < samples_10ms_ * channels) {
     for (int k = payload_size; k < samples_10ms_ * channels; k++) {
       audio_frame.data_[k] = 0;
@@ -190,8 +187,7 @@
         return;
       }
     } else {
-      int16_t* stereo_audio =
-          new int16_t[2 * audio_frame.samples_per_channel_];
+      int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
       int k;
       for (k = 0; k < audio_frame.samples_per_channel_; k++) {
         stereo_audio[k << 1] = audio_frame.data_[k];
@@ -207,17 +203,17 @@
   } else {
     if (fwrite(audio_frame.data_, sizeof(int16_t),
                audio_frame.num_channels_ * audio_frame.samples_per_channel_,
-               pcm_file_) != static_cast<size_t>(
-            audio_frame.num_channels_ * audio_frame.samples_per_channel_)) {
+               pcm_file_) !=
+        static_cast<size_t>(audio_frame.num_channels_ *
+                            audio_frame.samples_per_channel_)) {
       return;
     }
   }
 }
 
-void PCMFile::Write10MsData(int16_t* playout_buffer,
-                            uint16_t length_smpls) {
-  if (fwrite(playout_buffer, sizeof(uint16_t),
-             length_smpls, pcm_file_) != length_smpls) {
+void PCMFile::Write10MsData(int16_t* playout_buffer, uint16_t length_smpls) {
+  if (fwrite(playout_buffer, sizeof(uint16_t), length_smpls, pcm_file_) !=
+      length_smpls) {
     return;
   }
 }