Reland of Stop using hardcoded payload types for video codecs

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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 "webrtc/video/video_quality_test.h"
@@ skipping 11 matching lines @@
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_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/system_wrappers/include/cpu_info.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/layer_filtering_transport.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/vcm_capturer.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
@@ skipping 957 matching lines @@
 void VideoQualityTest::SetupVideo(Transport* send_transport,
                                   Transport* recv_transport) {
   if (params_.logs)
     trace_to_stderr_.reset(new test::TraceToStderr);
 
   size_t num_streams = params_.ss.streams.size();
   CreateSendConfig(num_streams, 0, 0, send_transport);
 
   int payload_type;
   if (params_.video.codec == "H264") {
-    video_encoder_.reset(VideoEncoder::Create(VideoEncoder::kH264));
+    video_encoder_.reset(H264Encoder::Create());
     payload_type = kPayloadTypeH264;
   } else if (params_.video.codec == "VP8") {
-    video_encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp8));
+    video_encoder_.reset(VP8Encoder::Create());
     payload_type = kPayloadTypeVP8;
   } else if (params_.video.codec == "VP9") {
-    video_encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp9));
+    video_encoder_.reset(VP9Encoder::Create());
     payload_type = kPayloadTypeVP9;
   } else {
     RTC_NOTREACHED() << "Codec not supported!";
     return;
   }
   video_send_config_.encoder_settings.encoder = video_encoder_.get();
   video_send_config_.encoder_settings.payload_name = params_.video.codec;
   video_send_config_.encoder_settings.payload_type = payload_type;
   video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
   video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
@@ skipping 456 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc