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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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| 28 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 28 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| 29 #include "webrtc/modules/rtp_rtcp/source/ulpfec_generator.h" | 29 #include "webrtc/modules/rtp_rtcp/source/ulpfec_generator.h" |
| 30 #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h" | 30 #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h" |
| 31 #include "webrtc/typedefs.h" | 31 #include "webrtc/typedefs.h" |
| 32 | 32 |
| 33 namespace webrtc { | 33 namespace webrtc { |
| 34 class RtpPacketToSend; | 34 class RtpPacketToSend; |
| 35 | 35 |
| 36 class RTPSenderVideo { | 36 class RTPSenderVideo { |
| 37 public: | 37 public: |
| 38 RTPSenderVideo(Clock* clock, RTPSender* rtpSender); | 38 RTPSenderVideo(Clock* clock, |
| 39 RTPSender* rtpSender, |
| 40 FlexfecSender* flexfec_sender); |
| 39 virtual ~RTPSenderVideo(); | 41 virtual ~RTPSenderVideo(); |
| 40 | 42 |
| 41 virtual RtpVideoCodecTypes VideoCodecType() const; | 43 virtual RtpVideoCodecTypes VideoCodecType() const; |
| 42 | 44 |
| 43 size_t FecPacketOverhead() const; | 45 size_t FecPacketOverhead() const; |
| 44 | 46 |
| 45 static RtpUtility::Payload* CreateVideoPayload( | 47 static RtpUtility::Payload* CreateVideoPayload( |
| 46 const char payload_name[RTP_PAYLOAD_NAME_SIZE], | 48 const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| 47 int8_t payload_type); | 49 int8_t payload_type); |
| 48 | 50 |
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| 125 // and any padding overhead. | 127 // and any padding overhead. |
| 126 RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_); | 128 RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_); |
| 127 // Bitrate used for video payload and RTP headers. | 129 // Bitrate used for video payload and RTP headers. |
| 128 RateStatistics video_bitrate_ GUARDED_BY(stats_crit_); | 130 RateStatistics video_bitrate_ GUARDED_BY(stats_crit_); |
| 129 OneTimeEvent first_frame_sent_; | 131 OneTimeEvent first_frame_sent_; |
| 130 }; | 132 }; |
| 131 | 133 |
| 132 } // namespace webrtc | 134 } // namespace webrtc |
| 133 | 135 |
| 134 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ | 136 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ |
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