commit adaf80961280dfecd3fdcc23bd3fa0a312ce42de
author henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Apr 17 08:29:10 2014
committer henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Apr 17 08:29:10 2014
tree 6f9815b8be8345dc2d03bf59d191e9d23bb5e5a5
parent 6cec07f6a7ff5d90a55ba18c90cfad8da295aaa7

Removing AudioCoding duplicate tests

Reverting to using one version of ACM in ACM tests.

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 4adac7df-926f-26a2-2b94-8c16560cd09d

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

diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc
index 027aeb0..230e9f1a 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc
@@ -15,13 +15,12 @@
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"  // Config.
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_opus.h"
 #include "webrtc/modules/audio_coding/main/test/TestStereo.h"
 #include "webrtc/modules/audio_coding/main/test/utility.h"
 #include "webrtc/system_wrappers/interface/trace.h"
@@ -29,13 +28,12 @@
 
 namespace webrtc {
 
-OpusTest::OpusTest(const Config& config)
-    : acm_receiver_(config.Get<AudioCodingModuleFactory>().Create(0)),
+OpusTest::OpusTest()
+    : acm_receiver_(AudioCodingModule::Create(0)),
       channel_a2b_(NULL),
       counter_(0),
       payload_type_(255),
-      rtp_timestamp_(0) {
-}
+      rtp_timestamp_(0) {}
 
 OpusTest::~OpusTest() {
   if (channel_a2b_ != NULL) {
@@ -254,11 +252,12 @@
     }
 
     // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
-    EXPECT_EQ(480, resampler_.Resample10Msec(audio_frame.data_,
-                                             audio_frame.sample_rate_hz_,
-                                             &audio[written_samples],
-                                             48000,
-                                             channels));
+    EXPECT_EQ(480,
+              resampler_.Resample10Msec(audio_frame.data_,
+                                        audio_frame.sample_rate_hz_,
+                                        48000,
+                                        channels,
+                                        &audio[written_samples]));
     written_samples += 480 * channels;
 
     // Sometimes we need to loop over the audio vector to produce the right