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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_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ | 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ |
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ | 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ |
13 | 13 |
14 #include <set> | 14 #include <set> |
15 | 15 |
| 16 #include "webrtc/base/onetimeevent.h" |
16 #include "webrtc/base/scoped_ptr.h" | 17 #include "webrtc/base/scoped_ptr.h" |
17 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" | 18 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 19 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
19 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" | 20 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" |
20 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
21 #include "webrtc/typedefs.h" | 22 #include "webrtc/typedefs.h" |
22 | 23 |
23 namespace webrtc { | 24 namespace webrtc { |
24 | 25 |
25 class CriticalSectionWrapper; | 26 class CriticalSectionWrapper; |
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111 int8_t cng_fb_payload_type_; | 112 int8_t cng_fb_payload_type_; |
112 int8_t cng_payload_type_; | 113 int8_t cng_payload_type_; |
113 | 114 |
114 // G722 is special since it use the wrong number of RTP samples in timestamp | 115 // G722 is special since it use the wrong number of RTP samples in timestamp |
115 // VS. number of samples in the frame | 116 // VS. number of samples in the frame |
116 int8_t g722_payload_type_; | 117 int8_t g722_payload_type_; |
117 bool last_received_g722_; | 118 bool last_received_g722_; |
118 | 119 |
119 uint8_t num_energy_; | 120 uint8_t num_energy_; |
120 uint8_t current_remote_energy_[kRtpCsrcSize]; | 121 uint8_t current_remote_energy_[kRtpCsrcSize]; |
| 122 |
| 123 ThreadUnsafeOneTimeEvent first_packet_received_; |
121 }; | 124 }; |
122 } // namespace webrtc | 125 } // namespace webrtc |
123 | 126 |
124 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ | 127 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ |
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