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| 1 /* | 1 /* | 
| 2  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2  *  Copyright (c) 2014 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_FORMAT_H264_H_ | 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 
| 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 
| 13 | 13 | 
| 14 #include <queue> | 14 #include <queue> | 
| 15 #include <string> | 15 #include <string> | 
| 16 | 16 | 
|  | 17 #include "webrtc/base/constructormagic.h" | 
| 17 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" | 18 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" | 
| 18 | 19 | 
| 19 namespace webrtc { | 20 namespace webrtc { | 
| 20 | 21 | 
| 21 class RtpPacketizerH264 : public RtpPacketizer { | 22 class RtpPacketizerH264 : public RtpPacketizer { | 
| 22  public: | 23  public: | 
| 23   // Initialize with payload from encoder. | 24   // Initialize with payload from encoder. | 
| 24   // The payload_data must be exactly one encoded H264 frame. | 25   // The payload_data must be exactly one encoded H264 frame. | 
| 25   RtpPacketizerH264(FrameType frame_type, size_t max_payload_len); | 26   RtpPacketizerH264(FrameType frame_type, size_t max_payload_len); | 
| 26 | 27 | 
| (...skipping 65 matching lines...) Expand 10 before | Expand all | Expand 10 after  Loading... | 
| 92 class RtpDepacketizerH264 : public RtpDepacketizer { | 93 class RtpDepacketizerH264 : public RtpDepacketizer { | 
| 93  public: | 94  public: | 
| 94   virtual ~RtpDepacketizerH264() {} | 95   virtual ~RtpDepacketizerH264() {} | 
| 95 | 96 | 
| 96   bool Parse(ParsedPayload* parsed_payload, | 97   bool Parse(ParsedPayload* parsed_payload, | 
| 97              const uint8_t* payload_data, | 98              const uint8_t* payload_data, | 
| 98              size_t payload_data_length) override; | 99              size_t payload_data_length) override; | 
| 99 }; | 100 }; | 
| 100 }  // namespace webrtc | 101 }  // namespace webrtc | 
| 101 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 102 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 
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