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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #include <algorithm> | 10 #include <algorithm> |
11 #include <map> | 11 #include <map> |
12 #include <sstream> | 12 #include <sstream> |
13 #include <string> | 13 #include <string> |
14 | 14 |
15 #include "testing/gtest/include/gtest/gtest.h" | 15 #include "testing/gtest/include/gtest/gtest.h" |
16 | 16 |
17 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
18 #include "webrtc/base/event.h" | 18 #include "webrtc/base/event.h" |
19 #include "webrtc/base/scoped_ptr.h" | 19 #include "webrtc/base/scoped_ptr.h" |
20 #include "webrtc/call.h" | 20 #include "webrtc/call.h" |
21 #include "webrtc/call/transport_adapter.h" | 21 #include "webrtc/call/transport_adapter.h" |
22 #include "webrtc/frame_callback.h" | 22 #include "webrtc/frame_callback.h" |
23 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 23 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
24 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| 25 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" |
25 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | 26 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
26 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" | 27 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
27 #include "webrtc/modules/video_coding/include/video_coding_defines.h" | 28 #include "webrtc/modules/video_coding/include/video_coding_defines.h" |
28 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 29 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
29 #include "webrtc/system_wrappers/include/metrics.h" | 30 #include "webrtc/system_wrappers/include/metrics.h" |
30 #include "webrtc/system_wrappers/include/sleep.h" | 31 #include "webrtc/system_wrappers/include/sleep.h" |
31 #include "webrtc/test/call_test.h" | 32 #include "webrtc/test/call_test.h" |
32 #include "webrtc/test/direct_transport.h" | 33 #include "webrtc/test/direct_transport.h" |
33 #include "webrtc/test/encoder_settings.h" | 34 #include "webrtc/test/encoder_settings.h" |
34 #include "webrtc/test/fake_audio_device.h" | 35 #include "webrtc/test/fake_audio_device.h" |
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281 | 282 |
282 private: | 283 private: |
283 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; | 284 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; |
284 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; | 285 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; |
285 int frame_counter_; | 286 int frame_counter_; |
286 } test; | 287 } test; |
287 | 288 |
288 RunBaseTest(&test, FakeNetworkPipe::Config()); | 289 RunBaseTest(&test, FakeNetworkPipe::Config()); |
289 } | 290 } |
290 | 291 |
| 292 #if defined(WEBRTC_THIRD_PARTY_H264) |
| 293 |
291 TEST_F(EndToEndTest, SendsAndReceivesH264) { | 294 TEST_F(EndToEndTest, SendsAndReceivesH264) { |
292 class H264Observer : public test::EndToEndTest, public VideoRenderer { | 295 class H264Observer : public test::EndToEndTest, public VideoRenderer { |
293 public: | 296 public: |
294 H264Observer() | 297 H264Observer() |
295 : EndToEndTest(2 * kDefaultTimeoutMs), | 298 : EndToEndTest(2 * kDefaultTimeoutMs), |
296 fake_encoder_(Clock::GetRealTimeClock()), | 299 encoder_(VideoEncoder::Create(VideoEncoder::kH264)), |
| 300 decoder_(H264Decoder::Create()), |
297 frame_counter_(0) {} | 301 frame_counter_(0) {} |
298 | 302 |
299 void PerformTest() override { | 303 void PerformTest() override { |
300 EXPECT_TRUE(Wait()) | 304 EXPECT_TRUE(Wait()) |
301 << "Timed out while waiting for enough frames to be decoded."; | 305 << "Timed out while waiting for enough frames to be decoded."; |
302 } | 306 } |
303 | 307 |
304 void ModifyVideoConfigs( | 308 void ModifyVideoConfigs( |
305 VideoSendStream::Config* send_config, | 309 VideoSendStream::Config* send_config, |
306 std::vector<VideoReceiveStream::Config>* receive_configs, | 310 std::vector<VideoReceiveStream::Config>* receive_configs, |
307 VideoEncoderConfig* encoder_config) override { | 311 VideoEncoderConfig* encoder_config) override { |
308 send_config->rtp.nack.rtp_history_ms = | 312 send_config->rtp.nack.rtp_history_ms = |
309 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; | 313 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
310 send_config->encoder_settings.encoder = &fake_encoder_; | 314 send_config->encoder_settings.encoder = encoder_.get(); |
311 send_config->encoder_settings.payload_name = "H264"; | 315 send_config->encoder_settings.payload_name = "H264"; |
312 send_config->encoder_settings.payload_type = kFakeSendPayloadType; | 316 send_config->encoder_settings.payload_type = 126; |
313 encoder_config->streams[0].min_bitrate_bps = 50000; | 317 encoder_config->streams[0].min_bitrate_bps = 50000; |
314 encoder_config->streams[0].target_bitrate_bps = | 318 encoder_config->streams[0].target_bitrate_bps = |
315 encoder_config->streams[0].max_bitrate_bps = 2000000; | 319 encoder_config->streams[0].max_bitrate_bps = 2000000; |
316 | 320 |
317 (*receive_configs)[0].renderer = this; | 321 (*receive_configs)[0].renderer = this; |
318 (*receive_configs)[0].decoders.resize(1); | 322 (*receive_configs)[0].decoders.resize(1); |
319 (*receive_configs)[0].decoders[0].payload_type = | 323 (*receive_configs)[0].decoders[0].payload_type = |
320 send_config->encoder_settings.payload_type; | 324 send_config->encoder_settings.payload_type; |
321 (*receive_configs)[0].decoders[0].payload_name = | 325 (*receive_configs)[0].decoders[0].payload_name = |
322 send_config->encoder_settings.payload_name; | 326 send_config->encoder_settings.payload_name; |
323 (*receive_configs)[0].decoders[0].decoder = &fake_decoder_; | 327 (*receive_configs)[0].decoders[0].decoder = decoder_.get(); |
324 } | 328 } |
325 | 329 |
326 void RenderFrame(const VideoFrame& video_frame, | 330 void RenderFrame(const VideoFrame& video_frame, |
327 int time_to_render_ms) override { | 331 int time_to_render_ms) override { |
328 const int kRequiredFrames = 500; | 332 const int kRequiredFrames = 500; |
329 if (++frame_counter_ == kRequiredFrames) | 333 if (++frame_counter_ == kRequiredFrames) |
330 observation_complete_.Set(); | 334 observation_complete_.Set(); |
331 } | 335 } |
332 | 336 |
333 bool IsTextureSupported() const override { return false; } | 337 bool IsTextureSupported() const override { return false; } |
334 | 338 |
335 private: | 339 private: |
336 test::FakeH264Decoder fake_decoder_; | 340 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; |
337 test::FakeH264Encoder fake_encoder_; | 341 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; |
338 int frame_counter_; | 342 int frame_counter_; |
339 } test; | 343 } test; |
340 | 344 |
341 RunBaseTest(&test, FakeNetworkPipe::Config()); | 345 RunBaseTest(&test, FakeNetworkPipe::Config()); |
342 } | 346 } |
343 | 347 |
| 348 #endif // defined(WEBRTC_THIRD_PARTY_H264) |
| 349 |
344 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { | 350 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { |
345 class SyncRtcpObserver : public test::EndToEndTest { | 351 class SyncRtcpObserver : public test::EndToEndTest { |
346 public: | 352 public: |
347 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} | 353 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} |
348 | 354 |
349 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { | 355 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { |
350 RTCPUtility::RTCPParserV2 parser(packet, length, true); | 356 RTCPUtility::RTCPParserV2 parser(packet, length, true); |
351 EXPECT_TRUE(parser.IsValid()); | 357 EXPECT_TRUE(parser.IsValid()); |
352 uint32_t ssrc = 0; | 358 uint32_t ssrc = 0; |
353 ssrc |= static_cast<uint32_t>(packet[4]) << 24; | 359 ssrc |= static_cast<uint32_t>(packet[4]) << 24; |
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3244 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) | 3250 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) |
3245 << "Enabling RTX requires rtpmap: rtx negotiation."; | 3251 << "Enabling RTX requires rtpmap: rtx negotiation."; |
3246 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) | 3252 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) |
3247 << "Enabling RTP extensions require negotiation."; | 3253 << "Enabling RTP extensions require negotiation."; |
3248 | 3254 |
3249 VerifyEmptyNackConfig(default_receive_config.rtp.nack); | 3255 VerifyEmptyNackConfig(default_receive_config.rtp.nack); |
3250 VerifyEmptyFecConfig(default_receive_config.rtp.fec); | 3256 VerifyEmptyFecConfig(default_receive_config.rtp.fec); |
3251 } | 3257 } |
3252 | 3258 |
3253 } // namespace webrtc | 3259 } // namespace webrtc |
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