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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/rtcp_utility.h" | 23 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
24 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" | |
24 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | 25 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
25 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" | 26 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
26 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" | 27 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" |
27 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 28 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
28 #include "webrtc/system_wrappers/interface/event_wrapper.h" | 29 #include "webrtc/system_wrappers/interface/event_wrapper.h" |
29 #include "webrtc/system_wrappers/interface/metrics.h" | 30 #include "webrtc/system_wrappers/interface/metrics.h" |
30 #include "webrtc/system_wrappers/interface/sleep.h" | 31 #include "webrtc/system_wrappers/interface/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" |
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274 | 275 |
275 private: | 276 private: |
276 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; | 277 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; |
277 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; | 278 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; |
278 int frame_counter_; | 279 int frame_counter_; |
279 } test; | 280 } test; |
280 | 281 |
281 RunBaseTest(&test); | 282 RunBaseTest(&test); |
282 } | 283 } |
283 | 284 |
285 #if defined(WEBRTC_OPENH264) | |
hta - Chromium
2015/10/01 09:45:51
I don't like #ifdefs much - I think the style guid
hbos
2015/10/01 12:19:46
I don't like #ifdefs either, and it will compile i
| |
286 | |
284 TEST_F(EndToEndTest, SendsAndReceivesH264) { | 287 TEST_F(EndToEndTest, SendsAndReceivesH264) { |
285 class H264Observer : public test::EndToEndTest, public VideoRenderer { | 288 class H264Observer : public test::EndToEndTest, public VideoRenderer { |
286 public: | 289 public: |
287 H264Observer() | 290 H264Observer() |
288 : EndToEndTest(2 * kDefaultTimeoutMs), | 291 : EndToEndTest(2 * kDefaultTimeoutMs), |
289 fake_encoder_(Clock::GetRealTimeClock()), | 292 encoder_(VideoEncoder::Create(VideoEncoder::kH264)), |
293 decoder_(H264Decoder::Create()), | |
290 frame_counter_(0) {} | 294 frame_counter_(0) {} |
291 | 295 |
292 void PerformTest() override { | 296 void PerformTest() override { |
293 EXPECT_EQ(kEventSignaled, Wait()) | 297 EXPECT_EQ(kEventSignaled, Wait()) |
294 << "Timed out while waiting for enough frames to be decoded."; | 298 << "Timed out while waiting for enough frames to be decoded."; |
295 } | 299 } |
296 | 300 |
297 void ModifyConfigs(VideoSendStream::Config* send_config, | 301 void ModifyConfigs(VideoSendStream::Config* send_config, |
298 std::vector<VideoReceiveStream::Config>* receive_configs, | 302 std::vector<VideoReceiveStream::Config>* receive_configs, |
299 VideoEncoderConfig* encoder_config) override { | 303 VideoEncoderConfig* encoder_config) override { |
300 send_config->rtp.nack.rtp_history_ms = | 304 send_config->rtp.nack.rtp_history_ms = |
301 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; | 305 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
302 send_config->encoder_settings.encoder = &fake_encoder_; | 306 send_config->encoder_settings.encoder = encoder_.get(); |
303 send_config->encoder_settings.payload_name = "H264"; | 307 send_config->encoder_settings.payload_name = "H264"; |
304 send_config->encoder_settings.payload_type = kFakeSendPayloadType; | 308 send_config->encoder_settings.payload_type = 126; |
305 encoder_config->streams[0].min_bitrate_bps = 50000; | 309 encoder_config->streams[0].min_bitrate_bps = 50000; |
306 encoder_config->streams[0].target_bitrate_bps = | 310 encoder_config->streams[0].target_bitrate_bps = |
307 encoder_config->streams[0].max_bitrate_bps = 2000000; | 311 encoder_config->streams[0].max_bitrate_bps = 2000000; |
308 | 312 |
309 (*receive_configs)[0].renderer = this; | 313 (*receive_configs)[0].renderer = this; |
310 (*receive_configs)[0].decoders.resize(1); | 314 (*receive_configs)[0].decoders.resize(1); |
311 (*receive_configs)[0].decoders[0].payload_type = | 315 (*receive_configs)[0].decoders[0].payload_type = |
312 send_config->encoder_settings.payload_type; | 316 send_config->encoder_settings.payload_type; |
313 (*receive_configs)[0].decoders[0].payload_name = | 317 (*receive_configs)[0].decoders[0].payload_name = |
314 send_config->encoder_settings.payload_name; | 318 send_config->encoder_settings.payload_name; |
315 (*receive_configs)[0].decoders[0].decoder = &fake_decoder_; | 319 (*receive_configs)[0].decoders[0].decoder = decoder_.get(); |
316 } | 320 } |
317 | 321 |
318 void RenderFrame(const VideoFrame& video_frame, | 322 void RenderFrame(const VideoFrame& video_frame, |
319 int time_to_render_ms) override { | 323 int time_to_render_ms) override { |
320 const int kRequiredFrames = 500; | 324 const int kRequiredFrames = 500; |
321 if (++frame_counter_ == kRequiredFrames) | 325 if (++frame_counter_ == kRequiredFrames) |
322 observation_complete_->Set(); | 326 observation_complete_->Set(); |
323 } | 327 } |
324 | 328 |
325 bool IsTextureSupported() const override { return false; } | 329 bool IsTextureSupported() const override { return false; } |
326 | 330 |
327 private: | 331 private: |
328 test::FakeH264Decoder fake_decoder_; | 332 rtc::scoped_ptr<webrtc::VideoEncoder> encoder_; |
329 test::FakeH264Encoder fake_encoder_; | 333 rtc::scoped_ptr<webrtc::VideoDecoder> decoder_; |
330 int frame_counter_; | 334 int frame_counter_; |
331 } test; | 335 } test; |
332 | 336 |
333 RunBaseTest(&test); | 337 RunBaseTest(&test); |
334 } | 338 } |
335 | 339 |
340 #endif // defined(WEBRTC_OPENH264) | |
341 | |
336 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { | 342 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { |
337 class SyncRtcpObserver : public test::EndToEndTest { | 343 class SyncRtcpObserver : public test::EndToEndTest { |
338 public: | 344 public: |
339 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} | 345 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} |
340 | 346 |
341 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { | 347 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { |
342 RTCPUtility::RTCPParserV2 parser(packet, length, true); | 348 RTCPUtility::RTCPParserV2 parser(packet, length, true); |
343 EXPECT_TRUE(parser.IsValid()); | 349 EXPECT_TRUE(parser.IsValid()); |
344 uint32_t ssrc = 0; | 350 uint32_t ssrc = 0; |
345 ssrc |= static_cast<uint32_t>(packet[4]) << 24; | 351 ssrc |= static_cast<uint32_t>(packet[4]) << 24; |
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3122 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) | 3128 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) |
3123 << "Enabling RTX requires rtpmap: rtx negotiation."; | 3129 << "Enabling RTX requires rtpmap: rtx negotiation."; |
3124 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) | 3130 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) |
3125 << "Enabling RTP extensions require negotiation."; | 3131 << "Enabling RTP extensions require negotiation."; |
3126 | 3132 |
3127 VerifyEmptyNackConfig(default_receive_config.rtp.nack); | 3133 VerifyEmptyNackConfig(default_receive_config.rtp.nack); |
3128 VerifyEmptyFecConfig(default_receive_config.rtp.fec); | 3134 VerifyEmptyFecConfig(default_receive_config.rtp.fec); |
3129 } | 3135 } |
3130 | 3136 |
3131 } // namespace webrtc | 3137 } // namespace webrtc |
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