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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/api/audio_codecs/builtin_audio_decoder_factory.h" | 15 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h" |
16 #include "webrtc/base/checks.h" | 16 #include "webrtc/base/checks.h" |
17 #include "webrtc/config.h" | 17 #include "webrtc/config.h" |
| 18 #include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h" |
18 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" | 19 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
19 #include "webrtc/test/testsupport/fileutils.h" | 20 #include "webrtc/test/testsupport/fileutils.h" |
20 #include "webrtc/voice_engine/include/voe_base.h" | 21 #include "webrtc/voice_engine/include/voe_base.h" |
21 | 22 |
22 namespace webrtc { | 23 namespace webrtc { |
23 namespace test { | 24 namespace test { |
24 | 25 |
25 namespace { | 26 namespace { |
26 const int kVideoRotationRtpExtensionId = 4; | 27 const int kVideoRotationRtpExtensionId = 4; |
27 } | 28 } |
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53 : clock_(Clock::GetRealTimeClock()), | 54 : clock_(Clock::GetRealTimeClock()), |
54 video_send_config_(nullptr), | 55 video_send_config_(nullptr), |
55 video_send_stream_(nullptr), | 56 video_send_stream_(nullptr), |
56 audio_send_config_(nullptr), | 57 audio_send_config_(nullptr), |
57 audio_send_stream_(nullptr), | 58 audio_send_stream_(nullptr), |
58 fake_encoder_(clock_), | 59 fake_encoder_(clock_), |
59 num_video_streams_(1), | 60 num_video_streams_(1), |
60 num_audio_streams_(0), | 61 num_audio_streams_(0), |
61 num_flexfec_streams_(0), | 62 num_flexfec_streams_(0), |
62 decoder_factory_(CreateBuiltinAudioDecoderFactory()), | 63 decoder_factory_(CreateBuiltinAudioDecoderFactory()), |
| 64 encoder_factory_(CreateBuiltinAudioEncoderFactory()), |
63 fake_send_audio_device_(nullptr), | 65 fake_send_audio_device_(nullptr), |
64 fake_recv_audio_device_(nullptr) {} | 66 fake_recv_audio_device_(nullptr) {} |
65 | 67 |
66 CallTest::~CallTest() { | 68 CallTest::~CallTest() { |
67 } | 69 } |
68 | 70 |
69 void CallTest::RunBaseTest(BaseTest* test) { | 71 void CallTest::RunBaseTest(BaseTest* test) { |
70 num_video_streams_ = test->GetNumVideoStreams(); | 72 num_video_streams_ = test->GetNumVideoStreams(); |
71 num_audio_streams_ = test->GetNumAudioStreams(); | 73 num_audio_streams_ = test->GetNumAudioStreams(); |
72 num_flexfec_streams_ = test->GetNumFlexfecStreams(); | 74 num_flexfec_streams_ = test->GetNumFlexfecStreams(); |
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247 for (size_t i = 0; i < num_video_streams; ++i) | 249 for (size_t i = 0; i < num_video_streams; ++i) |
248 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); | 250 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); |
249 video_send_config_.rtp.extensions.push_back(RtpExtension( | 251 video_send_config_.rtp.extensions.push_back(RtpExtension( |
250 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); | 252 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); |
251 } | 253 } |
252 | 254 |
253 if (num_audio_streams > 0) { | 255 if (num_audio_streams > 0) { |
254 audio_send_config_ = AudioSendStream::Config(send_transport); | 256 audio_send_config_ = AudioSendStream::Config(send_transport); |
255 audio_send_config_.voe_channel_id = voe_send_.channel_id; | 257 audio_send_config_.voe_channel_id = voe_send_.channel_id; |
256 audio_send_config_.rtp.ssrc = kAudioSendSsrc; | 258 audio_send_config_.rtp.ssrc = kAudioSendSsrc; |
257 audio_send_config_.send_codec_spec.codec_inst = | 259 audio_send_config_.send_codec_spec = |
258 CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000}; | 260 rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
| 261 {kAudioSendPayloadType, {"OPUS", 48000, 2, {{"stereo", "1"}}}}); |
| 262 audio_send_config_.encoder_factory = encoder_factory_; |
259 } | 263 } |
260 | 264 |
261 // TODO(brandtr): Update this when we support multistream protection. | 265 // TODO(brandtr): Update this when we support multistream protection. |
262 if (num_flexfec_streams > 0) { | 266 if (num_flexfec_streams > 0) { |
263 video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType; | 267 video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType; |
264 video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc; | 268 video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc; |
265 video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]}; | 269 video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]}; |
266 } | 270 } |
267 } | 271 } |
268 | 272 |
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574 | 578 |
575 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 579 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
576 } | 580 } |
577 | 581 |
578 bool EndToEndTest::ShouldCreateReceivers() const { | 582 bool EndToEndTest::ShouldCreateReceivers() const { |
579 return true; | 583 return true; |
580 } | 584 } |
581 | 585 |
582 } // namespace test | 586 } // namespace test |
583 } // namespace webrtc | 587 } // namespace webrtc |
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