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

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This reverts commit 17608dc4592fe25c1effdd75bf856f4af251942e.

Reason for revert: Breaks downstream build

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,sprang@webrtc.org

Change-Id: Idc60f26f34dd0456a40c72375ae829e25b28621f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157046
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29483}
This commit is contained in:
Erik Språng
2019-10-15 09:43:11 +00:00
committed by Commit Bot
parent 8718afb283
commit 4ed0b52c12
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,