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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 |
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ |
13 | 13 |
14 #include <stdio.h> | 14 #include <stdio.h> |
15 | 15 |
16 #include "webrtc/base/criticalsection.h" | |
17 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 16 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
18 #include "webrtc/modules/include/module_common_types.h" | 17 #include "webrtc/modules/include/module_common_types.h" |
| 18 #include "webrtc/rtc_base/criticalsection.h" |
19 #include "webrtc/typedefs.h" | 19 #include "webrtc/typedefs.h" |
20 | 20 |
21 namespace webrtc { | 21 namespace webrtc { |
22 | 22 |
23 #define MAX_NUM_PAYLOADS 50 | 23 #define MAX_NUM_PAYLOADS 50 |
24 #define MAX_NUM_FRAMESIZES 6 | 24 #define MAX_NUM_FRAMESIZES 6 |
25 | 25 |
26 // TODO(turajs): Write constructor for this structure. | 26 // TODO(turajs): Write constructor for this structure. |
27 struct ACMTestFrameSizeStats { | 27 struct ACMTestFrameSizeStats { |
28 uint16_t frameSizeSample; | 28 uint16_t frameSizeSample; |
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120 // External timing info, defaulted to -1. Only used if they are | 120 // External timing info, defaulted to -1. Only used if they are |
121 // non-negative. | 121 // non-negative. |
122 int64_t external_send_timestamp_; | 122 int64_t external_send_timestamp_; |
123 int32_t external_sequence_number_; | 123 int32_t external_sequence_number_; |
124 int num_packets_to_drop_; | 124 int num_packets_to_drop_; |
125 }; | 125 }; |
126 | 126 |
127 } // namespace webrtc | 127 } // namespace webrtc |
128 | 128 |
129 #endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ | 129 #endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ |
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