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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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 |
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
| 16 #include "webrtc/api/audio_codecs/audio_decoder.h" |
16 #include "webrtc/base/array_view.h" | 17 #include "webrtc/base/array_view.h" |
17 #include "webrtc/base/optional.h" | 18 #include "webrtc/base/optional.h" |
18 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" | |
19 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" | 19 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" |
20 | 20 |
21 namespace webrtc { | 21 namespace webrtc { |
22 namespace test { | 22 namespace test { |
23 | 23 |
24 // Provides an AudioDecoder implementation that delivers audio data from a file. | 24 // Provides an AudioDecoder implementation that delivers audio data from a file. |
25 // The "encoded" input should contain information about what RTP timestamp the | 25 // The "encoded" input should contain information about what RTP timestamp the |
26 // encoding represents, and how many samples the decoder should produce for that | 26 // encoding represents, and how many samples the decoder should produce for that |
27 // encoding. A helper method PrepareEncoded is provided to prepare such | 27 // encoding. A helper method PrepareEncoded is provided to prepare such |
28 // encodings. If packets are missing, as determined from the timestamps, the | 28 // encodings. If packets are missing, as determined from the timestamps, the |
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64 rtc::Optional<uint32_t> next_timestamp_from_input_; | 64 rtc::Optional<uint32_t> next_timestamp_from_input_; |
65 const int sample_rate_hz_; | 65 const int sample_rate_hz_; |
66 const bool stereo_; | 66 const bool stereo_; |
67 size_t last_decoded_length_ = 0; | 67 size_t last_decoded_length_ = 0; |
68 bool cng_mode_ = false; | 68 bool cng_mode_ = false; |
69 }; | 69 }; |
70 | 70 |
71 } // namespace test | 71 } // namespace test |
72 } // namespace webrtc | 72 } // namespace webrtc |
73 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ | 73 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |
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