Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(219)

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

Issue 2542413002: Generalize FlexfecReceiveStream::Config. (CL1) (Closed)
Patch Set: Rebase. Created 4 years 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
« no previous file with comments | « webrtc/media/engine/webrtcvideoengine2_unittest.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
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
11 #include "webrtc/test/call_test.h" 11 #include "webrtc/test/call_test.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 14
15 #include "webrtc/base/checks.h" 15 #include "webrtc/base/checks.h"
16 #include "webrtc/config.h" 16 #include "webrtc/config.h"
17 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" 17 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
18 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" 18 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
19 #include "webrtc/test/call_test.h"
20 #include "webrtc/test/testsupport/fileutils.h" 19 #include "webrtc/test/testsupport/fileutils.h"
21 #include "webrtc/voice_engine/include/voe_base.h" 20 #include "webrtc/voice_engine/include/voe_base.h"
22 21
23 namespace webrtc { 22 namespace webrtc {
24 namespace test { 23 namespace test {
25 24
26 namespace { 25 namespace {
27 const int kVideoRotationRtpExtensionId = 4; 26 const int kVideoRotationRtpExtensionId = 4;
28 } 27 }
29 28
(...skipping 239 matching lines...) Expand 10 before | Expand all | Expand 10 after
269 audio_config.rtcp_send_transport = rtcp_send_transport; 268 audio_config.rtcp_send_transport = rtcp_send_transport;
270 audio_config.voe_channel_id = voe_recv_.channel_id; 269 audio_config.voe_channel_id = voe_recv_.channel_id;
271 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; 270 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
272 audio_config.decoder_factory = decoder_factory_; 271 audio_config.decoder_factory = decoder_factory_;
273 audio_receive_configs_.push_back(audio_config); 272 audio_receive_configs_.push_back(audio_config);
274 } 273 }
275 274
276 // TODO(brandtr): Update this when we support multistream protection. 275 // TODO(brandtr): Update this when we support multistream protection.
277 RTC_DCHECK(num_flexfec_streams_ <= 1); 276 RTC_DCHECK(num_flexfec_streams_ <= 1);
278 if (num_flexfec_streams_ == 1) { 277 if (num_flexfec_streams_ == 1) {
279 FlexfecReceiveStream::Config flexfec_config; 278 FlexfecReceiveStream::Config config;
280 flexfec_config.flexfec_payload_type = kFlexfecPayloadType; 279 config.payload_type = kFlexfecPayloadType;
281 flexfec_config.flexfec_ssrc = kFlexfecSendSsrc; 280 config.remote_ssrc = kFlexfecSendSsrc;
282 flexfec_config.protected_media_ssrcs = {kVideoSendSsrcs[0]}; 281 config.protected_media_ssrcs = {kVideoSendSsrcs[0]};
283 flexfec_receive_configs_.push_back(flexfec_config); 282 flexfec_receive_configs_.push_back(config);
284 } 283 }
285 } 284 }
286 285
287 void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock, 286 void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
288 float speed, 287 float speed,
289 int framerate, 288 int framerate,
290 int width, 289 int width,
291 int height) { 290 int height) {
292 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create( 291 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
293 width, height, framerate * speed, clock)); 292 width, height, framerate * speed, clock));
(...skipping 213 matching lines...) Expand 10 before | Expand all | Expand 10 after
507 506
508 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { 507 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
509 } 508 }
510 509
511 bool EndToEndTest::ShouldCreateReceivers() const { 510 bool EndToEndTest::ShouldCreateReceivers() const {
512 return true; 511 return true;
513 } 512 }
514 513
515 } // namespace test 514 } // namespace test
516 } // namespace webrtc 515 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/media/engine/webrtcvideoengine2_unittest.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698