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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 | 11 |
12 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_ | 12 #ifndef WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_H264_VIDEO_TOOLBOX_NALU_H_ |
13 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_ | 13 #define WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_H264_VIDEO_TOOLBOX_NALU_H_ |
14 | 14 |
15 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" | 15 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" |
16 | 16 |
17 #if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED) | |
18 | |
19 #include <CoreMedia/CoreMedia.h> | 17 #include <CoreMedia/CoreMedia.h> |
20 | 18 |
21 #include "webrtc/base/buffer.h" | 19 #include "webrtc/base/buffer.h" |
22 #include "webrtc/modules/include/module_common_types.h" | 20 #include "webrtc/modules/include/module_common_types.h" |
23 | 21 |
24 namespace webrtc { | 22 namespace webrtc { |
25 | 23 |
26 // Converts a sample buffer emitted from the VideoToolbox encoder into a buffer | 24 // Converts a sample buffer emitted from the VideoToolbox encoder into a buffer |
27 // suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer | 25 // suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer |
28 // needs to be in Annex B format. Data is written directly to |annexb_buffer| | 26 // needs to be in Annex B format. Data is written directly to |annexb_buffer| |
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100 size_t BytesRemaining() const; | 98 size_t BytesRemaining() const; |
101 | 99 |
102 private: | 100 private: |
103 uint8_t* const start_; | 101 uint8_t* const start_; |
104 size_t offset_; | 102 size_t offset_; |
105 const size_t length_; | 103 const size_t length_; |
106 }; | 104 }; |
107 | 105 |
108 } // namespace webrtc | 106 } // namespace webrtc |
109 | 107 |
110 #endif // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED) | 108 #endif // WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_H264_VIDEO_TOOLBOX_NALU_H_ |
111 #endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_ | |
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