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Don't recreate the audio receive stream when updating the local_ssrc.

Browse Source 
Bug: webrtc:11993
Change-Id: Ic5d8a8a8b7c12fb1d906e0b3cbdf657fd9e8eafc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222042
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34299}
This commit is contained in:
Tommi
2021-06-15 23:01:57 +02:00
committed by WebRTC LUCI CQ
parent bc03259de7
commit 08be9baaa3
22 changed files with 266 additions and 34 deletions
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83
modules/rtp_rtcp/source/rtcp_receiver.cc
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@ -39,6 +39,7 @@
#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "modules/rtp_rtcp/source/tmmbr_help.h"
#include "rtc_base/checks.h"
@ -84,8 +85,14 @@ bool ResetTimestampIfExpired(const Timestamp now,
} // namespace
constexpr size_t RTCPReceiver::RegisteredSsrcs::kMediaSsrcIndex;
constexpr size_t RTCPReceiver::RegisteredSsrcs::kMaxSsrcs;
RTCPReceiver::RegisteredSsrcs::RegisteredSsrcs(
const RtpRtcpInterface::Configuration& config) {
bool disable_sequence_checker,
const RtpRtcpInterface::Configuration& config)
: packet_sequence_checker_(disable_sequence_checker) {
packet_sequence_checker_.Detach();
ssrcs_.push_back(config.local_media_ssrc);
if (config.rtx_send_ssrc) {
ssrcs_.push_back(*config.rtx_send_ssrc);
@ -100,6 +107,21 @@ RTCPReceiver::RegisteredSsrcs::RegisteredSsrcs(
RTC_DCHECK_LE(ssrcs_.size(), RTCPReceiver::RegisteredSsrcs::kMaxSsrcs);
}
bool RTCPReceiver::RegisteredSsrcs::contains(uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return absl::c_linear_search(ssrcs_, ssrc);
}
uint32_t RTCPReceiver::RegisteredSsrcs::media_ssrc() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return ssrcs_[kMediaSsrcIndex];
}
void RTCPReceiver::RegisteredSsrcs::set_media_ssrc(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
ssrcs_[kMediaSsrcIndex] = ssrc;
}
struct RTCPReceiver::PacketInformation {
uint32_t packet_type_flags = 0; // RTCPPacketTypeFlags bit field.
@ -117,12 +139,12 @@ struct RTCPReceiver::PacketInformation {
};
RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
ModuleRtpRtcp* owner)
ModuleRtpRtcpImpl2* owner)
: clock_(config.clock),
receiver_only_(config.receiver_only),
rtp_rtcp_(owner),
main_ssrc_(config.local_media_ssrc),
registered_ssrcs_(config),
registered_ssrcs_(false, config),
rtcp_bandwidth_observer_(config.bandwidth_callback),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
@ -150,6 +172,53 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
RTC_DCHECK(owner);
}
RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
ModuleRtpRtcp* owner)
: clock_(config.clock),
receiver_only_(config.receiver_only),
rtp_rtcp_(owner),
main_ssrc_(config.local_media_ssrc),
registered_ssrcs_(true, config),
rtcp_bandwidth_observer_(config.bandwidth_callback),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
network_state_estimate_observer_(config.network_state_estimate_observer),
transport_feedback_observer_(config.transport_feedback_callback),
bitrate_allocation_observer_(config.bitrate_allocation_observer),
report_interval_(config.rtcp_report_interval_ms > 0
? TimeDelta::Millis(config.rtcp_report_interval_ms)
: (config.audio ? kDefaultAudioReportInterval
: kDefaultVideoReportInterval)),
// TODO(bugs.webrtc.org/10774): Remove fallback.
remote_ssrc_(0),
remote_sender_rtp_time_(0),
remote_sender_packet_count_(0),
remote_sender_octet_count_(0),
remote_sender_reports_count_(0),
xr_rrtr_status_(config.non_sender_rtt_measurement),
xr_rr_rtt_ms_(0),
oldest_tmmbr_info_ms_(0),
cname_callback_(config.rtcp_cname_callback),
report_block_data_observer_(config.report_block_data_observer),
packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
num_skipped_packets_(0),
last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) {
RTC_DCHECK(owner);
// Dear reader - if you're here because of this log statement and are
// wondering what this is about, chances are that you are using an instance
// of RTCPReceiver without using the webrtc APIs. This creates a bit of a
// problem for WebRTC because this class is a part of an internal
// implementation that is constantly changing and being improved.
// The intention of this log statement is to give a heads up that changes
// are coming and encourage you to use the public APIs or be prepared that
// things might break down the line as more changes land. A thing you could
// try out for now is to replace the `CustomSequenceChecker` in the header
// with a regular `SequenceChecker` and see if that triggers an
// error in your code. If it does, chances are you have your own threading
// model that is not the same as WebRTC internally has.
RTC_LOG(LS_INFO) << "************** !!!DEPRECATION WARNING!! **************";
}
RTCPReceiver::~RTCPReceiver() {}
void RTCPReceiver::IncomingPacket(rtc::ArrayView<const uint8_t> packet) {
@ -178,6 +247,14 @@ void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
remote_ssrc_ = ssrc;
}
void RTCPReceiver::set_local_media_ssrc(uint32_t ssrc) {
registered_ssrcs_.set_media_ssrc(ssrc);
}
uint32_t RTCPReceiver::local_media_ssrc() const {
return registered_ssrcs_.media_ssrc();
}
uint32_t RTCPReceiver::RemoteSSRC() const {
MutexLock lock(&rtcp_receiver_lock_);
return remote_ssrc_;