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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/scoped_ptr.h" | 18 #include "webrtc/base/scoped_ptr.h" |
19 #include "webrtc/call.h" | 19 #include "webrtc/call.h" |
20 #include "webrtc/frame_callback.h" | 20 #include "webrtc/frame_callback.h" |
21 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
22 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" | |
22 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | 23 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
23 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" | 24 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
24 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" | 25 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" |
25 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 26 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
26 #include "webrtc/system_wrappers/interface/event_wrapper.h" | 27 #include "webrtc/system_wrappers/interface/event_wrapper.h" |
27 #include "webrtc/system_wrappers/interface/metrics.h" | 28 #include "webrtc/system_wrappers/interface/metrics.h" |
28 #include "webrtc/system_wrappers/interface/sleep.h" | 29 #include "webrtc/system_wrappers/interface/sleep.h" |
29 #include "webrtc/test/call_test.h" | 30 #include "webrtc/test/call_test.h" |
30 #include "webrtc/test/direct_transport.h" | 31 #include "webrtc/test/direct_transport.h" |
31 #include "webrtc/test/encoder_settings.h" | 32 #include "webrtc/test/encoder_settings.h" |
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273 | 274 |
274 private: | 275 private: |
275 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; | 276 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; |
276 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; | 277 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; |
277 int frame_counter_; | 278 int frame_counter_; |
278 } test; | 279 } test; |
279 | 280 |
280 RunBaseTest(&test); | 281 RunBaseTest(&test); |
281 } | 282 } |
282 | 283 |
284 #if defined(WEBRTC_OPENH264) | |
285 | |
283 TEST_F(EndToEndTest, SendsAndReceivesH264) { | 286 TEST_F(EndToEndTest, SendsAndReceivesH264) { |
284 class H264Observer : public test::EndToEndTest, public VideoRenderer { | 287 class H264Observer : public test::EndToEndTest, public VideoRenderer { |
285 public: | 288 public: |
286 H264Observer() | 289 H264Observer() |
287 : EndToEndTest(2 * kDefaultTimeoutMs), | 290 : EndToEndTest(2 * kDefaultTimeoutMs), |
288 fake_encoder_(Clock::GetRealTimeClock()), | 291 encoder_(VideoEncoder::Create(VideoEncoder::kH264)), |
292 decoder_(H264Decoder::Create()), | |
289 frame_counter_(0) {} | 293 frame_counter_(0) {} |
290 | 294 |
291 void PerformTest() override { | 295 void PerformTest() override { |
292 EXPECT_EQ(kEventSignaled, Wait()) | 296 EXPECT_EQ(kEventSignaled, Wait()) |
293 << "Timed out while waiting for enough frames to be decoded."; | 297 << "Timed out while waiting for enough frames to be decoded."; |
294 } | 298 } |
295 | 299 |
296 void ModifyConfigs(VideoSendStream::Config* send_config, | 300 void ModifyConfigs(VideoSendStream::Config* send_config, |
297 std::vector<VideoReceiveStream::Config>* receive_configs, | 301 std::vector<VideoReceiveStream::Config>* receive_configs, |
298 VideoEncoderConfig* encoder_config) override { | 302 VideoEncoderConfig* encoder_config) override { |
299 send_config->rtp.nack.rtp_history_ms = | 303 send_config->encoder_settings.encoder = encoder_.get(); |
300 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; | |
stefan-webrtc
2015/09/28 11:19:02
I think you still want NACK to be enabled.
hbos
2015/09/30 15:35:19
Done.
| |
301 send_config->encoder_settings.encoder = &fake_encoder_; | |
302 send_config->encoder_settings.payload_name = "H264"; | 304 send_config->encoder_settings.payload_name = "H264"; |
303 send_config->encoder_settings.payload_type = kFakeSendPayloadType; | 305 send_config->encoder_settings.payload_type = 120; |
304 encoder_config->streams[0].min_bitrate_bps = 50000; | 306 encoder_config->streams[0].min_bitrate_bps = 50000; |
305 encoder_config->streams[0].target_bitrate_bps = | 307 encoder_config->streams[0].target_bitrate_bps = |
306 encoder_config->streams[0].max_bitrate_bps = 2000000; | 308 encoder_config->streams[0].max_bitrate_bps = 2000000; |
307 | 309 |
308 (*receive_configs)[0].renderer = this; | 310 (*receive_configs)[0].renderer = this; |
309 (*receive_configs)[0].decoders.resize(1); | 311 (*receive_configs)[0].decoders.resize(1); |
310 (*receive_configs)[0].decoders[0].payload_type = | 312 (*receive_configs)[0].decoders[0].payload_type = |
311 send_config->encoder_settings.payload_type; | 313 send_config->encoder_settings.payload_type; |
312 (*receive_configs)[0].decoders[0].payload_name = | 314 (*receive_configs)[0].decoders[0].payload_name = |
313 send_config->encoder_settings.payload_name; | 315 send_config->encoder_settings.payload_name; |
314 (*receive_configs)[0].decoders[0].decoder = &fake_decoder_; | 316 (*receive_configs)[0].decoders[0].decoder = decoder_.get(); |
315 } | 317 } |
316 | 318 |
317 void RenderFrame(const VideoFrame& video_frame, | 319 void RenderFrame(const VideoFrame& video_frame, |
318 int time_to_render_ms) override { | 320 int time_to_render_ms) override { |
319 const int kRequiredFrames = 500; | 321 const int kRequiredFrames = 500; |
320 if (++frame_counter_ == kRequiredFrames) | 322 if (++frame_counter_ == kRequiredFrames) |
321 observation_complete_->Set(); | 323 observation_complete_->Set(); |
322 } | 324 } |
323 | 325 |
324 bool IsTextureSupported() const override { return false; } | 326 bool IsTextureSupported() const override { return false; } |
325 | 327 |
326 private: | 328 private: |
327 test::FakeH264Decoder fake_decoder_; | 329 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; |
328 test::FakeH264Encoder fake_encoder_; | 330 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; |
329 int frame_counter_; | 331 int frame_counter_; |
330 } test; | 332 } test; |
331 | 333 |
332 RunBaseTest(&test); | 334 RunBaseTest(&test); |
333 } | 335 } |
334 | 336 |
337 #endif // defined(WEBRTC_OPENH264) | |
338 | |
335 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { | 339 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { |
336 class SyncRtcpObserver : public test::EndToEndTest { | 340 class SyncRtcpObserver : public test::EndToEndTest { |
337 public: | 341 public: |
338 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} | 342 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} |
339 | 343 |
340 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { | 344 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { |
341 RTCPUtility::RTCPParserV2 parser(packet, length, true); | 345 RTCPUtility::RTCPParserV2 parser(packet, length, true); |
342 EXPECT_TRUE(parser.IsValid()); | 346 EXPECT_TRUE(parser.IsValid()); |
343 uint32_t ssrc = 0; | 347 uint32_t ssrc = 0; |
344 ssrc |= static_cast<uint32_t>(packet[4]) << 24; | 348 ssrc |= static_cast<uint32_t>(packet[4]) << 24; |
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3047 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) | 3051 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) |
3048 << "Enabling RTX requires rtpmap: rtx negotiation."; | 3052 << "Enabling RTX requires rtpmap: rtx negotiation."; |
3049 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) | 3053 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) |
3050 << "Enabling RTP extensions require negotiation."; | 3054 << "Enabling RTP extensions require negotiation."; |
3051 | 3055 |
3052 VerifyEmptyNackConfig(default_receive_config.rtp.nack); | 3056 VerifyEmptyNackConfig(default_receive_config.rtp.nack); |
3053 VerifyEmptyFecConfig(default_receive_config.rtp.fec); | 3057 VerifyEmptyFecConfig(default_receive_config.rtp.fec); |
3054 } | 3058 } |
3055 | 3059 |
3056 } // namespace webrtc | 3060 } // namespace webrtc |
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