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Removing AudioCoding duplicate tests

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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
This commit is contained in:
henrik.lundin@webrtc.org
2014-04-17 08:29:10 +00:00
parent 6cec07f6a7
commit adaf809612
23 changed files with 197 additions and 548 deletions
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23
webrtc/modules/audio_coding/main/test/opus_test.cc
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@ -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 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
}
// 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