Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(2376)

Side by Side Diff: webrtc/test/call_test.cc

Issue 2772033002: Add content type information to encoded images and corresponding rtp extension header (Closed)
Patch Set: Set EncodedImage content_type from vie_encoder Created 3 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 184 matching lines...) Expand 10 before | Expand all | Expand 10 after
195 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); 195 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
196 if (num_video_streams > 0) { 196 if (num_video_streams > 0) {
197 video_send_config_ = VideoSendStream::Config(send_transport); 197 video_send_config_ = VideoSendStream::Config(send_transport);
198 video_send_config_.encoder_settings.encoder = &fake_encoder_; 198 video_send_config_.encoder_settings.encoder = &fake_encoder_;
199 video_send_config_.encoder_settings.payload_name = "FAKE"; 199 video_send_config_.encoder_settings.payload_name = "FAKE";
200 video_send_config_.encoder_settings.payload_type = 200 video_send_config_.encoder_settings.payload_type =
201 kFakeVideoSendPayloadType; 201 kFakeVideoSendPayloadType;
202 video_send_config_.rtp.extensions.push_back( 202 video_send_config_.rtp.extensions.push_back(
203 RtpExtension(RtpExtension::kTransportSequenceNumberUri, 203 RtpExtension(RtpExtension::kTransportSequenceNumberUri,
204 kTransportSequenceNumberExtensionId)); 204 kTransportSequenceNumberExtensionId));
205 video_send_config_.rtp.extensions.push_back(
206 RtpExtension(RtpExtension::kVideoContentTypeUri,
207 kVideoContentTypeExtensionId));
205 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); 208 FillEncoderConfiguration(num_video_streams, &video_encoder_config_);
206 209
207 for (size_t i = 0; i < num_video_streams; ++i) 210 for (size_t i = 0; i < num_video_streams; ++i)
208 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); 211 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
209 video_send_config_.rtp.extensions.push_back(RtpExtension( 212 video_send_config_.rtp.extensions.push_back(RtpExtension(
210 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); 213 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
211 } 214 }
212 215
213 if (num_audio_streams > 0) { 216 if (num_audio_streams > 0) {
214 audio_send_config_ = AudioSendStream::Config(send_transport); 217 audio_send_config_ = AudioSendStream::Config(send_transport);
(...skipping 283 matching lines...) Expand 10 before | Expand all | Expand 10 after
498 501
499 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { 502 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
500 } 503 }
501 504
502 bool EndToEndTest::ShouldCreateReceivers() const { 505 bool EndToEndTest::ShouldCreateReceivers() const {
503 return true; 506 return true;
504 } 507 }
505 508
506 } // namespace test 509 } // namespace test
507 } // namespace webrtc 510 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698