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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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24 namespace webrtc { | 24 namespace webrtc { |
25 | 25 |
26 class OpusTest : public ACMTest { | 26 class OpusTest : public ACMTest { |
27 public: | 27 public: |
28 OpusTest(); | 28 OpusTest(); |
29 ~OpusTest(); | 29 ~OpusTest(); |
30 | 30 |
31 void Perform(); | 31 void Perform(); |
32 | 32 |
33 private: | 33 private: |
34 void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length, | 34 void Run(TestPackStereo* channel, |
| 35 int channels, |
| 36 int bitrate, |
| 37 size_t frame_length, |
35 int percent_loss = 0); | 38 int percent_loss = 0); |
36 | 39 |
37 void OpenOutFile(int test_number); | 40 void OpenOutFile(int test_number); |
38 | 41 |
39 rtc::scoped_ptr<AudioCodingModule> acm_receiver_; | 42 rtc::scoped_ptr<AudioCodingModule> acm_receiver_; |
40 TestPackStereo* channel_a2b_; | 43 TestPackStereo* channel_a2b_; |
41 PCMFile in_file_stereo_; | 44 PCMFile in_file_stereo_; |
42 PCMFile in_file_mono_; | 45 PCMFile in_file_mono_; |
43 PCMFile out_file_; | 46 PCMFile out_file_; |
44 PCMFile out_file_standalone_; | 47 PCMFile out_file_standalone_; |
45 int counter_; | 48 int counter_; |
46 uint8_t payload_type_; | 49 uint8_t payload_type_; |
47 int rtp_timestamp_; | 50 uint32_t rtp_timestamp_; |
48 acm2::ACMResampler resampler_; | 51 acm2::ACMResampler resampler_; |
49 WebRtcOpusEncInst* opus_mono_encoder_; | 52 WebRtcOpusEncInst* opus_mono_encoder_; |
50 WebRtcOpusEncInst* opus_stereo_encoder_; | 53 WebRtcOpusEncInst* opus_stereo_encoder_; |
51 WebRtcOpusDecInst* opus_mono_decoder_; | 54 WebRtcOpusDecInst* opus_mono_decoder_; |
52 WebRtcOpusDecInst* opus_stereo_decoder_; | 55 WebRtcOpusDecInst* opus_stereo_decoder_; |
53 }; | 56 }; |
54 | 57 |
55 } // namespace webrtc | 58 } // namespace webrtc |
56 | 59 |
57 #endif // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_ | 60 #endif // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_ |
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