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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc

Issue 2772033002: Add content type information to encoded images and corresponding rtp extension header (Closed)
Patch Set: Fix typo, leading to failed video catpure test Created 3 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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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208 RTC_DCHECK_LE(0, playout_delay.min_ms); 208 RTC_DCHECK_LE(0, playout_delay.min_ms);
209 RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms); 209 RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms);
210 RTC_DCHECK_LE(playout_delay.max_ms, kMaxMs); 210 RTC_DCHECK_LE(playout_delay.max_ms, kMaxMs);
211 // Convert MS to value to be sent on extension header. 211 // Convert MS to value to be sent on extension header.
212 uint32_t min_delay = playout_delay.min_ms / kGranularityMs; 212 uint32_t min_delay = playout_delay.min_ms / kGranularityMs;
213 uint32_t max_delay = playout_delay.max_ms / kGranularityMs; 213 uint32_t max_delay = playout_delay.max_ms / kGranularityMs;
214 ByteWriter<uint32_t, 3>::WriteBigEndian(data, (min_delay << 12) | max_delay); 214 ByteWriter<uint32_t, 3>::WriteBigEndian(data, (min_delay << 12) | max_delay);
215 return true; 215 return true;
216 } 216 }
217 217
218 // Video Content Type.
219 //
220 // E.g. default video or screenshare.
221 //
222 // 0 1
223 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
224 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
225 // | ID | len=0 | Content type |
226 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
227 constexpr RTPExtensionType VideoContentTypeExtension::kId;
228 constexpr uint8_t VideoContentTypeExtension::kValueSizeBytes;
229 constexpr const char* VideoContentTypeExtension::kUri;
230
231 bool VideoContentTypeExtension::Parse(rtc::ArrayView<const uint8_t> data,
232 VideoContentType* content_type) {
233 if (data.size() == 1 &&
234 data[0] < static_cast<uint8_t>(VideoContentType::TOTAL_CONTENT_TYPES)) {
235 *content_type = static_cast<VideoContentType>(data[0]);
236 return true;
237 }
238 return false;
239 }
240
241 bool VideoContentTypeExtension::Write(uint8_t* data,
242 VideoContentType content_type) {
243 data[0] = static_cast<uint8_t>(content_type);
244 return true;
245 }
246
218 } // namespace webrtc 247 } // namespace webrtc
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