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Revert "Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const""

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This reverts commit c9348218cfe0cff6d0d3a383f7d1d6cfce4b1262.

Reason for revert: Downstream tests are relying on incorrect behavior which this CL explicitly checks...

Original change's description:
> Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"
> 
> This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e
> 
> Downstream fixed, relanding.
> 
> Original change's description:
> > RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
> >
> > Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> > remove them, make the members const, and remove now unnecessary locking.
> >
> > Bug: webrtc:10774
> > Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29475}
> 
> TBR=nisse@webrtc.org
> 
> Bug: webrtc:10774
> Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29486}

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

Change-Id: I168fb3738a04dfdbd1581ddd8c3276ede9f72322
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29488}
This commit is contained in:
Erik Språng
2019-10-15 11:54:23 +00:00
committed by Commit Bot
parent 844600e8a4
commit e8a6bc3f25
14 changed files with 215 additions and 44 deletions
Download Patch File Download Diff File Expand all files Collapse all files

31
modules/rtp_rtcp/source/rtcp_receiver.cc
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@ -65,18 +65,6 @@ const size_t kMaxNumberOfStoredRrtrs = 200;
constexpr int32_t kDefaultVideoReportInterval = 1000;
constexpr int32_t kDefaultAudioReportInterval = 5000;
std::set<uint32_t> GetRegisteredSsrcs(const RtpRtcp::Configuration& config) {
std::set<uint32_t> ssrcs;
ssrcs.insert(config.local_media_ssrc);
if (config.rtx_send_ssrc) {
ssrcs.insert(*config.rtx_send_ssrc);
}
if (config.flexfec_sender) {
ssrcs.insert(config.flexfec_sender->ssrc());
}
return ssrcs;
}
} // namespace
struct RTCPReceiver::PacketInformation {
@ -138,8 +126,6 @@ RTCPReceiver::RTCPReceiver(const RtpRtcp::Configuration& config,
: clock_(config.clock),
receiver_only_(config.receiver_only),
rtp_rtcp_(owner),
main_ssrc_(config.local_media_ssrc),
registered_ssrcs_(GetRegisteredSsrcs(config)),
rtcp_bandwidth_observer_(config.bandwidth_callback),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
@ -151,6 +137,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcp::Configuration& config,
: (config.audio ? kDefaultAudioReportInterval
: kDefaultVideoReportInterval)),
// TODO(bugs.webrtc.org/10774): Remove fallback.
main_ssrc_(config.local_media_ssrc.value_or(0)),
remote_ssrc_(0),
remote_sender_rtp_time_(0),
xr_rrtr_status_(false),
@ -165,6 +152,15 @@ RTCPReceiver::RTCPReceiver(const RtpRtcp::Configuration& config,
num_skipped_packets_(0),
last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) {
RTC_DCHECK(owner);
if (config.local_media_ssrc) {
registered_ssrcs_.insert(*config.local_media_ssrc);
}
if (config.rtx_send_ssrc) {
registered_ssrcs_.insert(*config.rtx_send_ssrc);
}
if (config.flexfec_sender) {
registered_ssrcs_.insert(config.flexfec_sender->ssrc());
}
}
RTCPReceiver::~RTCPReceiver() {}
@ -198,6 +194,13 @@ uint32_t RTCPReceiver::RemoteSSRC() const {
return remote_ssrc_;
}
void RTCPReceiver::SetSsrcs(uint32_t main_ssrc,
const std::set<uint32_t>& registered_ssrcs) {
rtc::CritScope lock(&rtcp_receiver_lock_);
main_ssrc_ = main_ssrc;
registered_ssrcs_ = registered_ssrcs;
}
int32_t RTCPReceiver::RTT(uint32_t remote_ssrc,
int64_t* last_rtt_ms,
int64_t* avg_rtt_ms,