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			58 lines
		
	
	
		
			1.7 KiB
		
	
	
	
		
			C
		
	
| 
											10 years ago
										 | /*
 | ||
|  |  *  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/acm2/acm_opus.h"
 | ||
|  | #include "webrtc/modules/audio_coding/main/acm2/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"
 | ||
|  | #include "webrtc/system_wrappers/interface/scoped_ptr.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); | ||
|  | 
 | ||
|  |   scoped_ptr<AudioCodingModule> acm_receiver_; | ||
|  |   TestPackStereo* channel_a2b_; | ||
|  |   PCMFile in_file_stereo_; | ||
|  |   PCMFile in_file_mono_; | ||
|  |   PCMFile out_file_; | ||
|  |   PCMFile out_file_standalone_; | ||
|  |   int counter_; | ||
|  |   uint8_t payload_type_; | ||
|  |   int rtp_timestamp_; | ||
|  |   acm2::ACMResampler resampler_; | ||
|  |   WebRtcOpusEncInst* opus_mono_encoder_; | ||
|  |   WebRtcOpusEncInst* opus_stereo_encoder_; | ||
|  |   WebRtcOpusDecInst* opus_mono_decoder_; | ||
|  |   WebRtcOpusDecInst* opus_stereo_decoder_; | ||
|  | }; | ||
|  | 
 | ||
|  | }  // namespace webrtc
 | ||
|  | 
 | ||
|  | #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
 |