Revert "Allows FEC generation after pacer step."

This reverts commit 75fd127640bdf1729af6b4a25875e6d01f1570e0.

Reason for revert: Breaks downstream test

Original change's description:
> Allows FEC generation after pacer step.
> 
> Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> This CL enables FEC packets to be generated as media packets are sent,
> rather than generated, i.e. media packets are inserted into the fec
> generator after the pacing stage rather than at packetization time.
> 
> This may have some small impact of performance. FEC packets are
> typically only generated when a new packet with a marker bit is added,
> which means FEC packets protecting a frame will now be sent after all
> of the media packets, rather than (potentially) interleaved with them.
> Therefore this feature is currently behind a flag so we can examine the
> impact. Once we are comfortable with the behavior we'll make it default
> and remove the old code.
> 
> Note that this change does not include the "protect all header
> extensions" part of the original CL - that will be a follow-up.
> 
> Bug: webrtc:11340
> Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31558}

TBR=sprang@webrtc.org,srte@webrtc.org

Change-Id: Ie714e5f68580cbd57560e086c9dc7292a052de5f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177983
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31559}

2 files changed, 8 insertions, 28 deletions

diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 854a18aa54..5f8d2df965 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -197,7 +197,6 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
     FrameEncryptorInterface* frame_encryptor,
     const CryptoOptions& crypto_options,
     rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
-    bool use_deferred_fec,
     const WebRtcKeyValueConfig& trials) {
   RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
 
@@ -245,9 +244,7 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
     std::unique_ptr<VideoFecGenerator> fec_generator =
         MaybeCreateFecGenerator(clock, rtp_config, suspended_ssrcs, i, trials);
     configuration.fec_generator = fec_generator.get();
-    if (!use_deferred_fec) {
-      video_config.fec_generator = fec_generator.get();
-    }
+    video_config.fec_generator = fec_generator.get();
 
     configuration.rtx_send_ssrc =
         rtp_config.GetRtxSsrcAssociatedWithMediaSsrc(rtp_config.ssrcs[i]);
@@ -341,9 +338,6 @@ RtpVideoSender::RtpVideoSender(
           field_trials_.Lookup("WebRTC-UseEarlyLossDetection"),
           "Disabled")),
       has_packet_feedback_(TransportSeqNumExtensionConfigured(rtp_config)),
-      use_deferred_fec_(
-          absl::StartsWith(field_trials_.Lookup("WebRTC-DeferredFecGeneration"),
-                           "Enabled")),
       active_(false),
       module_process_thread_(nullptr),
       suspended_ssrcs_(std::move(suspended_ssrcs)),
@@ -362,7 +356,6 @@ RtpVideoSender::RtpVideoSender(
                                           frame_encryptor,
                                           crypto_options,
                                           std::move(frame_transformer),
-                                          use_deferred_fec_,
                                           field_trials_)),
       rtp_config_(rtp_config),
       codec_type_(GetVideoCodecType(rtp_config)),
@@ -855,26 +848,14 @@ int RtpVideoSender::ProtectionRequest(const FecProtectionParams* delta_params,
   *sent_nack_rate_bps = 0;
   *sent_fec_rate_bps = 0;
   for (const RtpStreamSender& stream : rtp_streams_) {
-    if (use_deferred_fec_) {
-      stream.rtp_rtcp->SetFecProtectionParams(*delta_params, *key_params);
-
-      auto send_bitrate = stream.rtp_rtcp->GetSendRates();
-      *sent_video_rate_bps += send_bitrate[RtpPacketMediaType::kVideo].bps();
-      *sent_fec_rate_bps +=
-          send_bitrate[RtpPacketMediaType::kForwardErrorCorrection].bps();
-      *sent_nack_rate_bps +=
-          send_bitrate[RtpPacketMediaType::kRetransmission].bps();
-    } else {
-      if (stream.fec_generator) {
-        stream.fec_generator->SetProtectionParameters(*delta_params,
-                                                      *key_params);
-        *sent_fec_rate_bps += stream.fec_generator->CurrentFecRate().bps();
-      }
-      *sent_video_rate_bps += stream.sender_video->VideoBitrateSent();
-      *sent_nack_rate_bps +=
-          stream.rtp_rtcp->GetSendRates()[RtpPacketMediaType::kRetransmission]
-              .bps<uint32_t>();
+    if (stream.fec_generator) {
+      stream.fec_generator->SetProtectionParameters(*delta_params, *key_params);
+      *sent_fec_rate_bps += stream.fec_generator->CurrentFecRate().bps();
     }
+    *sent_video_rate_bps += stream.sender_video->VideoBitrateSent();
+    *sent_nack_rate_bps +=
+        stream.rtp_rtcp->GetSendRates()[RtpPacketMediaType::kRetransmission]
+            .bps<uint32_t>();
   }
   return 0;
 }
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index e364729236..0c277d6aa7 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -176,7 +176,6 @@ class RtpVideoSender : public RtpVideoSenderInterface,
   const bool account_for_packetization_overhead_;
   const bool use_early_loss_detection_;
   const bool has_packet_feedback_;
-  const bool use_deferred_fec_;
 
   // TODO(holmer): Remove crit_ once RtpVideoSender runs on the
   // transport task queue.