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Adding Opus frame length test

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BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1193005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3672 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2013-03-15 13:29:17 +00:00
parent d613c207cc
commit 73222cff1a
11 changed files with 396 additions and 68 deletions
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52
webrtc/modules/audio_coding/main/test/opus_test.h Normal file
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
#include <math.h>
#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
namespace webrtc {
class OpusTest : public ACMTest {
public:
OpusTest();
~OpusTest();
void Perform();
private:
void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
int percent_loss = 0);
void OpenOutFile(int test_number);
AudioCodingModule* acm_receiver_;
TestPackStereo* channel_a2b_;
PCMFile in_file_stereo_;
PCMFile in_file_mono_;
PCMFile out_file_;
int counter_;
uint8_t payload_type_;
int rtp_timestamp_;
ACMResampler resampler_;
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_