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Side by Side Diff: webrtc/modules/audio_coding/main/test/opus_test.h

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years ago
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1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
13
14 #include <math.h>
15
16 #include "webrtc/base/scoped_ptr.h"
17 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
18 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
19 #include "webrtc/modules/audio_coding/main/test/ACMTest.h"
20 #include "webrtc/modules/audio_coding/main/test/Channel.h"
21 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
22 #include "webrtc/modules/audio_coding/main/test/TestStereo.h"
23
24 namespace webrtc {
25
26 class OpusTest : public ACMTest {
27 public:
28 OpusTest();
29 ~OpusTest();
30
31 void Perform();
32
33 private:
34 void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
35 int percent_loss = 0);
36
37 void OpenOutFile(int test_number);
38
39 rtc::scoped_ptr<AudioCodingModule> acm_receiver_;
40 TestPackStereo* channel_a2b_;
41 PCMFile in_file_stereo_;
42 PCMFile in_file_mono_;
43 PCMFile out_file_;
44 PCMFile out_file_standalone_;
45 int counter_;
46 uint8_t payload_type_;
47 int rtp_timestamp_;
48 acm2::ACMResampler resampler_;
49 WebRtcOpusEncInst* opus_mono_encoder_;
50 WebRtcOpusEncInst* opus_stereo_encoder_;
51 WebRtcOpusDecInst* opus_mono_decoder_;
52 WebRtcOpusDecInst* opus_stereo_decoder_;
53 };
54
55 } // namespace webrtc
56
57 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
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