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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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40 int16_t payloadType; | 40 int16_t payloadType; |
41 size_t lastPayloadLenByte; | 41 size_t lastPayloadLenByte; |
42 uint32_t lastTimestamp; | 42 uint32_t lastTimestamp; |
43 ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES]; | 43 ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES]; |
44 }; | 44 }; |
45 | 45 |
46 class Channel : public AudioPacketizationCallback { | 46 class Channel : public AudioPacketizationCallback { |
47 public: | 47 public: |
48 | 48 |
49 Channel(int16_t chID = -1); | 49 Channel(int16_t chID = -1); |
50 ~Channel(); | 50 ~Channel() override; |
51 | 51 |
52 int32_t SendData(FrameType frameType, | 52 int32_t SendData(FrameType frameType, |
53 uint8_t payloadType, | 53 uint8_t payloadType, |
54 uint32_t timeStamp, | 54 uint32_t timeStamp, |
55 const uint8_t* payloadData, | 55 const uint8_t* payloadData, |
56 size_t payloadSize, | 56 size_t payloadSize, |
57 const RTPFragmentationHeader* fragmentation) override; | 57 const RTPFragmentationHeader* fragmentation) override; |
58 | 58 |
59 void RegisterReceiverACM(AudioCodingModule *acm); | 59 void RegisterReceiverACM(AudioCodingModule *acm); |
60 | 60 |
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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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