Cleans up code related to legacy pre-pacing fec generation.

Bug: webrtc:11340
Change-Id: If3493db9fafdd3ad041f78999e304c8714be517f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186562
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32349}
This commit is contained in:
Erik Språng
2020-10-05 14:20:51 +02:00
committed by Commit Bot
parent 0d1b044db8
commit b6477858ac
9 changed files with 27 additions and 167 deletions

View File

@ -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 +244,6 @@ 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();
}
configuration.rtx_send_ssrc =
rtp_config.GetRtxSsrcAssociatedWithMediaSsrc(rtp_config.ssrcs[i]);
@ -335,9 +331,6 @@ RtpVideoSender::RtpVideoSender(
field_trials_.Lookup("WebRTC-SendSideBwe-WithOverhead"),
"Enabled")),
has_packet_feedback_(TransportSeqNumExtensionConfigured(rtp_config)),
use_deferred_fec_(!absl::StartsWith(
field_trials_.Lookup("WebRTC-DeferredFecGeneration"),
"Disabled")),
active_(false),
module_process_thread_(nullptr),
suspended_ssrcs_(std::move(suspended_ssrcs)),
@ -356,7 +349,6 @@ RtpVideoSender::RtpVideoSender(
frame_encryptor,
crypto_options,
std::move(frame_transformer),
use_deferred_fec_,
field_trials_)),
rtp_config_(rtp_config),
codec_type_(GetVideoCodecType(rtp_config)),
@ -844,7 +836,6 @@ 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();
@ -853,17 +844,6 @@ int RtpVideoSender::ProtectionRequest(const FecProtectionParams* delta_params,
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>();
}
}
return 0;
}

View File

@ -172,7 +172,6 @@ class RtpVideoSender : public RtpVideoSenderInterface,
const FieldTrialBasedConfig field_trials_;
const bool send_side_bwe_with_overhead_;
const bool has_packet_feedback_;
const bool use_deferred_fec_;
// TODO(holmer): Remove mutex_ once RtpVideoSender runs on the
// transport task queue.