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Side by Side Diff: modules/rtp_rtcp/source/rtp_sender.cc

Issue 2951033003: [EXPERIMENTAL] Generic stereo codec with index header sending single frames
Patch Set: Rebase and add external codec support. Created 3 years, 2 months ago
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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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342 } 342 }
343 if (payload_type_ == payload_type) { 343 if (payload_type_ == payload_type) {
344 if (!audio_configured_) { 344 if (!audio_configured_) {
345 *video_type = video_->VideoCodecType(); 345 *video_type = video_->VideoCodecType();
346 } 346 }
347 return 0; 347 return 0;
348 } 348 }
349 std::map<int8_t, RtpUtility::Payload*>::iterator it = 349 std::map<int8_t, RtpUtility::Payload*>::iterator it =
350 payload_type_map_.find(payload_type); 350 payload_type_map_.find(payload_type);
351 if (it == payload_type_map_.end()) { 351 if (it == payload_type_map_.end()) {
352 LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type) 352 LOG(LS_ERROR) << "Payload type " << static_cast<int>(payload_type)
353 << " not registered."; 353 << " not registered.";
354 return -1; 354 return -1;
355 } 355 }
356 SetSendPayloadType(payload_type); 356 SetSendPayloadType(payload_type);
357 RtpUtility::Payload* payload = it->second; 357 RtpUtility::Payload* payload = it->second;
358 RTC_DCHECK(payload); 358 RTC_DCHECK(payload);
359 if (!payload->audio && !audio_configured_) { 359 if (!payload->audio && !audio_configured_) {
360 video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType); 360 video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
361 *video_type = payload->typeSpecific.Video.videoCodecType; 361 *video_type = payload->typeSpecific.Video.videoCodecType;
362 } 362 }
363 return 0; 363 return 0;
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1306 rtc::CritScope lock(&send_critsect_); 1306 rtc::CritScope lock(&send_critsect_);
1307 packet->SetTimestamp(last_rtp_timestamp_); 1307 packet->SetTimestamp(last_rtp_timestamp_);
1308 packet->set_capture_time_ms(capture_time_ms_); 1308 packet->set_capture_time_ms(capture_time_ms_);
1309 } 1309 }
1310 AssignSequenceNumber(packet.get()); 1310 AssignSequenceNumber(packet.get());
1311 SendToNetwork(std::move(packet), StorageType::kDontRetransmit, 1311 SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
1312 RtpPacketSender::Priority::kLowPriority); 1312 RtpPacketSender::Priority::kLowPriority);
1313 } 1313 }
1314 1314
1315 } // namespace webrtc 1315 } // namespace webrtc
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