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Revert "Add ability to set RTCP sender ssrc at construction time"

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This reverts commit 94c58fd815f0c7c6429aa53a79621ea9ef39c770.

Reason for revert: Speculative revert, as it looks like this one broke IOS debug perf bots: https://ci.chromium.org/p/webrtc-internal/builders/ci/iOS64%20Debug/18901

Original change's description:
> Add ability to set RTCP sender ssrc at construction time
> 
> Bug: webrtc:10774
> Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28506}

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

Change-Id: I3f377ca1c84a7448675e5d022cb2f86f9630dbaf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144564
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28508}
This commit is contained in:
Ilya Nikolaevskiy
2019-07-08 16:21:41 +00:00
committed by Commit Bot
parent 4a126e45c3
commit 34462f5dc3
6 changed files with 59 additions and 69 deletions
Download Patch File Download Diff File Expand all files Collapse all files

32
modules/rtp_rtcp/source/rtcp_sender.cc
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@ -48,8 +48,6 @@ namespace {
const uint32_t kRtcpAnyExtendedReports = kRtcpXrReceiverReferenceTime |
kRtcpXrDlrrReportBlock |
kRtcpXrTargetBitrate;
constexpr int32_t kDefaultVideoReportInterval = 1000;
constexpr int32_t kDefaultAudioReportInterval = 5000;
} // namespace
RTCPSender::FeedbackState::FeedbackState()
@ -114,25 +112,29 @@ class RTCPSender::RtcpContext {
const int64_t now_us_;
};
RTCPSender::RTCPSender(const RtpRtcp::Configuration& config)
: audio_(config.audio),
clock_(config.clock),
RTCPSender::RTCPSender(
bool audio,
Clock* clock,
ReceiveStatisticsProvider* receive_statistics,
RtcpPacketTypeCounterObserver* packet_type_counter_observer,
RtcEventLog* event_log,
Transport* outgoing_transport,
int report_interval_ms)
: audio_(audio),
clock_(clock),
random_(clock_->TimeInMicroseconds()),
method_(RtcpMode::kOff),
event_log_(config.event_log),
transport_(config.outgoing_transport),
report_interval_ms_(config.rtcp_report_interval_ms > 0
? config.rtcp_report_interval_ms
: (config.audio ? kDefaultAudioReportInterval
: kDefaultVideoReportInterval)),
event_log_(event_log),
transport_(outgoing_transport),
report_interval_ms_(report_interval_ms),
sending_(false),
next_time_to_send_rtcp_(0),
timestamp_offset_(0),
last_rtp_timestamp_(0),
last_frame_capture_time_ms_(-1),
ssrc_(config.media_send_ssrc.value_or(0)),
ssrc_(0),
remote_ssrc_(0),
receive_statistics_(config.receive_statistics),
receive_statistics_(receive_statistics),
sequence_number_fir_(0),
@ -148,7 +150,7 @@ RTCPSender::RTCPSender(const RtpRtcp::Configuration& config)
app_length_(0),
xr_send_receiver_reference_time_enabled_(false),
packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
packet_type_counter_observer_(packet_type_counter_observer),
send_video_bitrate_allocation_(false),
last_payload_type_(-1) {
RTC_DCHECK(transport_ != nullptr);
@ -305,7 +307,7 @@ uint32_t RTCPSender::SSRC() const {
void RTCPSender::SetSSRC(uint32_t ssrc) {
rtc::CritScope lock(&critical_section_rtcp_sender_);
if (ssrc != ssrc_) {
if (ssrc_ != 0) {
// not first SetSSRC, probably due to a collision
// schedule a new RTCP report
// make sure that we send a RTP packet