Reland "Reland "Move rtp-specific config out of EncoderSettings.""

This reverts commit 6c2c13af06b32778b86950681758a7970d1c5d9e.

Reason for revert: Intend to investigate and fix perf problems.

Original change's description:
> Revert "Reland "Move rtp-specific config out of EncoderSettings.""
> 
> This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
> 
> Reason for revert: Regression in ramp up perf tests.
> 
> Original change's description:
> > Reland "Move rtp-specific config out of EncoderSettings."
> >
> > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
> >
> > Original change's description:
> > > Move rtp-specific config out of EncoderSettings.
> > >
> > > In VideoSendStream::Config, move payload_name and payload_type from
> > > EncoderSettings to Rtp.
> > >
> > > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > > and should perhaps be renamed in a follow up cl. It's no longer
> > > passed as an argument to VideoCodecInitializer::SetupCodec.
> > >
> > > The latter then needs a different way to know the codec type,
> > > which is provided by a new codec_type member in VideoEncoderConfig.
> > >
> > > Bug: webrtc:8830
> > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22532}
> >
> > Bug: webrtc:8830
> > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> > Reviewed-on: https://webrtc-review.googlesource.com/63721
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22595}
> 
> TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
> 
> Bug: webrtc:8830,chromium:827080
> Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
> Reviewed-on: https://webrtc-review.googlesource.com/65520
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22677}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8830, chromium:827080
Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f
Reviewed-on: https://webrtc-review.googlesource.com/66862
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22751}

3 files changed, 8 insertions, 6 deletions

diff --git a/media/engine/fakewebrtccall.cc b/media/engine/fakewebrtccall.cc
index 4bc01f0dd6..0aa9cb0967 100644
--- a/media/engine/fakewebrtccall.cc
+++ b/media/engine/fakewebrtccall.cc
@@ -229,19 +229,19 @@ void FakeVideoSendStream::ReconfigureVideoEncoder(
   if (config.encoder_specific_settings != NULL) {
     const unsigned char num_temporal_layers = static_cast<unsigned char>(
         video_streams_.back().num_temporal_layers.value_or(1));
-    if (config_.encoder_settings.payload_name == "VP8") {
+    if (config_.rtp.payload_name == "VP8") {
       config.encoder_specific_settings->FillVideoCodecVp8(&vpx_settings_.vp8);
       if (!video_streams_.empty()) {
         vpx_settings_.vp8.numberOfTemporalLayers = num_temporal_layers;
       }
-    } else if (config_.encoder_settings.payload_name == "VP9") {
+    } else if (config_.rtp.payload_name == "VP9") {
       config.encoder_specific_settings->FillVideoCodecVp9(&vpx_settings_.vp9);
       if (!video_streams_.empty()) {
         vpx_settings_.vp9.numberOfTemporalLayers = num_temporal_layers;
       }
     } else {
       ADD_FAILURE() << "Unsupported encoder payload: "
-                    << config_.encoder_settings.payload_name;
+                    << config_.rtp.payload_name;
     }
   }
   codec_settings_set_ = config.encoder_specific_settings != NULL;
diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index 8de05dafc8..3cf72cf732 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -1757,8 +1757,8 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetCodec(
   parameters_.config.encoder_settings.internal_source =
       info.has_internal_source;
 
-  parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
-  parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
+  parameters_.config.rtp.payload_name = codec_settings.codec.name;
+  parameters_.config.rtp.payload_type = codec_settings.codec.id;
   parameters_.config.rtp.ulpfec = codec_settings.ulpfec;
   parameters_.config.rtp.flexfec.payload_type =
       codec_settings.flexfec_payload_type;
@@ -1914,6 +1914,8 @@ WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(
     const VideoCodec& codec) const {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   webrtc::VideoEncoderConfig encoder_config;
+  encoder_config.codec_type = webrtc::PayloadStringToCodecType(codec.name);
+
   bool is_screencast = parameters_.options.is_screencast.value_or(false);
   if (is_screencast) {
     encoder_config.min_transmit_bitrate_bps =
diff --git a/media/engine/webrtcvideoengine_unittest.cc b/media/engine/webrtcvideoengine_unittest.cc
index ae85fc0c73..ed46e3dde2 100644
--- a/media/engine/webrtcvideoengine_unittest.cc
+++ b/media/engine/webrtcvideoengine_unittest.cc
@@ -133,7 +133,7 @@ rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateBlackFrameBuffer(
 void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config,
                                  const std::map<int, int>& rtx_types) {
   std::map<int, int>::const_iterator it;
-  it = rtx_types.find(config.encoder_settings.payload_type);
+  it = rtx_types.find(config.rtp.payload_type);
   EXPECT_TRUE(it != rtx_types.end() &&
               it->second == config.rtp.rtx.payload_type);