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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 |
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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)); |
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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 |
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