H.264 video codec support using OpenH264/FFmpeg

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>
 #include <map>
 #include <sstream>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_decoder.h"
@@ skipping 244 matching lines @@
 
    private:
     rtc::scoped_ptr<webrtc::VideoEncoder> encoder_;
     rtc::scoped_ptr<webrtc::VideoDecoder> decoder_;
     int frame_counter_;
   } test;
 
   RunBaseTest(&test);
 }
 
+#if defined(WEBRTC_END_TO_END_H264_TESTS)
+
 TEST_F(EndToEndTest, SendsAndReceivesH264) {
   class H264Observer : public test::EndToEndTest, public VideoRenderer {
    public:
     H264Observer()
         : EndToEndTest(2 * kDefaultTimeoutMs),
-          fake_encoder_(Clock::GetRealTimeClock()),
+          encoder_(VideoEncoder::Create(VideoEncoder::kH264)),
+          decoder_(H264Decoder::Create()),
           frame_counter_(0) {}
 
     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for enough frames to be decoded.";
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms =
           (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-      send_config->encoder_settings.encoder = &fake_encoder_;
+      send_config->encoder_settings.encoder = encoder_.get();
       send_config->encoder_settings.payload_name = "H264";
-      send_config->encoder_settings.payload_type = kFakeVideoSendPayloadType;
+      send_config->encoder_settings.payload_type = 126;
       encoder_config->streams[0].min_bitrate_bps = 50000;
       encoder_config->streams[0].target_bitrate_bps =
           encoder_config->streams[0].max_bitrate_bps = 2000000;
 
       (*receive_configs)[0].renderer = this;
       (*receive_configs)[0].decoders.resize(1);
       (*receive_configs)[0].decoders[0].payload_type =
           send_config->encoder_settings.payload_type;
       (*receive_configs)[0].decoders[0].payload_name =
           send_config->encoder_settings.payload_name;
-      (*receive_configs)[0].decoders[0].decoder = &fake_decoder_;
+      (*receive_configs)[0].decoders[0].decoder = decoder_.get();
     }
 
     void RenderFrame(const VideoFrame& video_frame,
                      int time_to_render_ms) override {
       const int kRequiredFrames = 500;
       if (++frame_counter_ == kRequiredFrames)
         observation_complete_.Set();
     }
 
     bool IsTextureSupported() const override { return false; }
 
    private:
-    test::FakeH264Decoder fake_decoder_;
-    test::FakeH264Encoder fake_encoder_;
+    rtc::scoped_ptr<webrtc::VideoEncoder> encoder_;
+    rtc::scoped_ptr<webrtc::VideoDecoder> decoder_;
     int frame_counter_;
   } test;
 
   RunBaseTest(&test);
 }
 
+#endif  // defined(WEBRTC_END_TO_END_H264_TESTS)
+
 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
   class SyncRtcpObserver : public test::EndToEndTest {
    public:
     SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
       uint32_t ssrc = 0;
       ssrc |= static_cast<uint32_t>(packet[4]) << 24;
@@ skipping 3004 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc