48af652ea5
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2. BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2206004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
58 lines
1.8 KiB
C++
58 lines
1.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
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#include <math.h>
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#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
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#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
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#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
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#include "webrtc/modules/audio_coding/main/test/Channel.h"
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#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
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#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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class OpusTest : public ACMTest {
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public:
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OpusTest();
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~OpusTest();
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void Perform();
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private:
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void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
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int percent_loss = 0);
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void OpenOutFile(int test_number);
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scoped_ptr<AudioCodingModule> acm_receiver_;
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TestPackStereo* channel_a2b_;
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PCMFile in_file_stereo_;
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PCMFile in_file_mono_;
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PCMFile out_file_;
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PCMFile out_file_standalone_;
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int counter_;
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uint8_t payload_type_;
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int rtp_timestamp_;
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acm1::ACMResampler resampler_;
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WebRtcOpusEncInst* opus_mono_encoder_;
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WebRtcOpusEncInst* opus_stereo_encoder_;
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WebRtcOpusDecInst* opus_mono_decoder_;
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WebRtcOpusDecInst* opus_stereo_decoder_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
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