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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 */ |
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26 namespace webrtc { | 26 namespace webrtc { |
27 | 27 |
28 // Converts a sample buffer emitted from the VideoToolbox encoder into a buffer | 28 // Converts a sample buffer emitted from the VideoToolbox encoder into a buffer |
29 // suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer | 29 // suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer |
30 // needs to be in Annex B format. Data is written directly to |annexb_buffer| | 30 // needs to be in Annex B format. Data is written directly to |annexb_buffer| |
31 // and a new RTPFragmentationHeader is returned in |out_header|. | 31 // and a new RTPFragmentationHeader is returned in |out_header|. |
32 bool H264CMSampleBufferToAnnexBBuffer( | 32 bool H264CMSampleBufferToAnnexBBuffer( |
33 CMSampleBufferRef avcc_sample_buffer, | 33 CMSampleBufferRef avcc_sample_buffer, |
34 bool is_keyframe, | 34 bool is_keyframe, |
35 rtc::Buffer* annexb_buffer, | 35 rtc::Buffer* annexb_buffer, |
36 webrtc::RTPFragmentationHeader** out_header); | 36 std::unique_ptr<RTPFragmentationHeader> *out_header); |
37 | 37 |
38 // Converts a buffer received from RTP into a sample buffer suitable for the | 38 // Converts a buffer received from RTP into a sample buffer suitable for the |
39 // VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample | 39 // VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample |
40 // buffer is in avcc format. | 40 // buffer is in avcc format. |
41 // If |is_keyframe| is true then |video_format| is ignored since the format will | 41 // If |is_keyframe| is true then |video_format| is ignored since the format will |
42 // be read from the buffer. Otherwise |video_format| must be provided. | 42 // be read from the buffer. Otherwise |video_format| must be provided. |
43 // Caller is responsible for releasing the created sample buffer. | 43 // Caller is responsible for releasing the created sample buffer. |
44 bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer, | 44 bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer, |
45 size_t annexb_buffer_size, | 45 size_t annexb_buffer_size, |
46 CMVideoFormatDescriptionRef video_format, | 46 CMVideoFormatDescriptionRef video_format, |
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103 | 103 |
104 private: | 104 private: |
105 uint8_t* const start_; | 105 uint8_t* const start_; |
106 size_t offset_; | 106 size_t offset_; |
107 const size_t length_; | 107 const size_t length_; |
108 }; | 108 }; |
109 | 109 |
110 } // namespace webrtc | 110 } // namespace webrtc |
111 | 111 |
112 #endif // WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_ | 112 #endif // WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_ |
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