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

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

View File

@ -34,6 +34,7 @@ class MockRtpRtcpInterface : public RtpRtcpInterface {
(const uint8_t* incoming_packet, size_t packet_length),
(override));
MOCK_METHOD(void, SetRemoteSSRC, (uint32_t ssrc), (override));
MOCK_METHOD(void, SetLocalSsrc, (uint32_t ssrc), (override));
MOCK_METHOD(void, SetMaxRtpPacketSize, (size_t size), (override));
MOCK_METHOD(size_t, MaxRtpPacketSize, (), (const, override));
MOCK_METHOD(void,

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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_;

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@ -19,6 +19,7 @@
#include <vector>
#include "api/array_view.h"
#include "api/sequence_checker.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@ -27,11 +28,15 @@
#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
class ModuleRtpRtcpImpl2;
class VideoBitrateAllocationObserver;
namespace rtcp {
class CommonHeader;
class ReportBlock;
@ -57,6 +62,10 @@ class RTCPReceiver final {
RTCPReceiver(const RtpRtcpInterface::Configuration& config,
ModuleRtpRtcp* owner);
RTCPReceiver(const RtpRtcpInterface::Configuration& config,
ModuleRtpRtcpImpl2* owner);
~RTCPReceiver();
void IncomingPacket(const uint8_t* packet, size_t packet_size) {
@ -66,9 +75,14 @@ class RTCPReceiver final {
int64_t LastReceivedReportBlockMs() const;
void set_local_media_ssrc(uint32_t ssrc);
uint32_t local_media_ssrc() const;
void SetRemoteSSRC(uint32_t ssrc);
uint32_t RemoteSSRC() const;
bool receiver_only() const { return receiver_only_; }
// Get received NTP.
// The types for the arguments below derive from the specification:
// - `remote_sender_packet_count`: `RTCSentRtpStreamStats.packetsSent` [1]
@ -126,21 +140,48 @@ class RTCPReceiver final {
void NotifyTmmbrUpdated();
private:
// A lightweight inlined set of local SSRCs.
class RegisteredSsrcs {
#if RTC_DCHECK_IS_ON
class CustomSequenceChecker : public SequenceChecker {
public:
static constexpr size_t kMaxSsrcs = 3;
// Initializes the set of registered local SSRCS by extracting them from the
// provided `config`.
explicit RegisteredSsrcs(const RtpRtcpInterface::Configuration& config);
// Indicates if `ssrc` is in the set of registered local SSRCs.
bool contains(uint32_t ssrc) const {
return absl::c_linear_search(ssrcs_, ssrc);
explicit CustomSequenceChecker(bool disable_checks)
: disable_checks_(disable_checks) {}
bool IsCurrent() const {
if (disable_checks_)
return true;
return SequenceChecker::IsCurrent();
}
private:
absl::InlinedVector<uint32_t, kMaxSsrcs> ssrcs_;
const bool disable_checks_;
};
#else
class CustomSequenceChecker : public SequenceChecker {
public:
explicit CustomSequenceChecker(bool) {}
};
#endif
// A lightweight inlined set of local SSRCs.
class RegisteredSsrcs {
public:
static constexpr size_t kMediaSsrcIndex = 0;
static constexpr size_t kMaxSsrcs = 3;
// Initializes the set of registered local SSRCS by extracting them from the
// provided `config`. The `disable_sequence_checker` flag is a workaround
// to be able to use a sequence checker without breaking downstream
// code that currently doesn't follow the same threading rules as webrtc.
RegisteredSsrcs(bool disable_sequence_checker,
const RtpRtcpInterface::Configuration& config);
// Indicates if `ssrc` is in the set of registered local SSRCs.
bool contains(uint32_t ssrc) const;
uint32_t media_ssrc() const;
void set_media_ssrc(uint32_t ssrc);
private:
RTC_NO_UNIQUE_ADDRESS CustomSequenceChecker packet_sequence_checker_;
absl::InlinedVector<uint32_t, kMaxSsrcs> ssrcs_
RTC_GUARDED_BY(packet_sequence_checker_);
};
struct PacketInformation;
@ -290,7 +331,7 @@ class RTCPReceiver final {
ModuleRtpRtcp* const rtp_rtcp_;
const uint32_t main_ssrc_;
// The set of registered local SSRCs.
const RegisteredSsrcs registered_ssrcs_;
RegisteredSsrcs registered_ssrcs_;
RtcpBandwidthObserver* const rtcp_bandwidth_observer_;
RtcpIntraFrameObserver* const rtcp_intra_frame_observer_;

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@ -308,6 +308,16 @@ void RTCPSender::SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz) {
rtp_clock_rates_khz_[payload_type] = rtp_clock_rate_hz / 1000;
}
uint32_t RTCPSender::SSRC() const {
MutexLock lock(&mutex_rtcp_sender_);
return ssrc_;
}
void RTCPSender::SetSsrc(uint32_t ssrc) {
MutexLock lock(&mutex_rtcp_sender_);
ssrc_ = ssrc;
}
void RTCPSender::SetRemoteSSRC(uint32_t ssrc) {
MutexLock lock(&mutex_rtcp_sender_);
remote_ssrc_ = ssrc;

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@ -94,7 +94,8 @@ class RTCPSender final {
void SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
uint32_t SSRC() const { return ssrc_; }
uint32_t SSRC() const;
void SetSsrc(uint32_t ssrc);
void SetRemoteSSRC(uint32_t ssrc) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
@ -187,7 +188,11 @@ class RTCPSender final {
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
const bool audio_;
const uint32_t ssrc_;
// TODO(bugs.webrtc.org/11581): `mutex_rtcp_sender_` shouldn't be required if
// we consistently run network related operations on the network thread.
// This is currently not possible due to callbacks from the process thread in
// ModuleRtpRtcpImpl2.
uint32_t ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
Clock* const clock_;
Random random_ RTC_GUARDED_BY(mutex_rtcp_sender_);
RtcpMode method_ RTC_GUARDED_BY(mutex_rtcp_sender_);

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@ -683,6 +683,11 @@ void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
rtcp_receiver_.SetRemoteSSRC(ssrc);
}
void ModuleRtpRtcpImpl::SetLocalSsrc(uint32_t local_ssrc) {
rtcp_receiver_.set_local_media_ssrc(local_ssrc);
rtcp_sender_.SetSsrc(local_ssrc);
}
RtpSendRates ModuleRtpRtcpImpl::GetSendRates() const {
return rtp_sender_->packet_sender.GetSendRates();
}

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@ -63,6 +63,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
size_t incoming_packet_length) override;
void SetRemoteSSRC(uint32_t ssrc) override;
void SetLocalSsrc(uint32_t ssrc) override;
// Sender part.
void RegisterSendPayloadFrequency(int payload_type,

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@ -69,6 +69,7 @@ ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
rtt_ms_(0) {
RTC_DCHECK(worker_queue_);
process_thread_checker_.Detach();
packet_sequence_checker_.Detach();
if (!configuration.receiver_only) {
rtp_sender_ = std::make_unique<RtpSenderContext>(configuration);
// Make sure rtcp sender use same timestamp offset as rtp sender.
@ -169,6 +170,7 @@ absl::optional<uint32_t> ModuleRtpRtcpImpl2::FlexfecSsrc() const {
void ModuleRtpRtcpImpl2::IncomingRtcpPacket(const uint8_t* rtcp_packet,
const size_t length) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtcp_receiver_.IncomingPacket(rtcp_packet, length);
}
@ -219,6 +221,12 @@ RtpState ModuleRtpRtcpImpl2::GetRtxState() const {
return rtp_sender_->packet_generator.GetRtxRtpState();
}
uint32_t ModuleRtpRtcpImpl2::local_media_ssrc() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_EQ(rtcp_receiver_.local_media_ssrc(), rtcp_sender_.SSRC());
return rtcp_receiver_.local_media_ssrc();
}
void ModuleRtpRtcpImpl2::SetRid(const std::string& rid) {
if (rtp_sender_) {
rtp_sender_->packet_generator.SetRid(rid);
@ -650,6 +658,12 @@ void ModuleRtpRtcpImpl2::SetRemoteSSRC(const uint32_t ssrc) {
rtcp_receiver_.SetRemoteSSRC(ssrc);
}
void ModuleRtpRtcpImpl2::SetLocalSsrc(uint32_t local_ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtcp_receiver_.set_local_media_ssrc(local_ssrc);
rtcp_sender_.SetSsrc(local_ssrc);
}
RtpSendRates ModuleRtpRtcpImpl2::GetSendRates() const {
// Typically called on the `rtp_transport_queue_` owned by an
// RtpTransportControllerSendInterface instance.

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@ -77,6 +77,8 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
void SetRemoteSSRC(uint32_t ssrc) override;
void SetLocalSsrc(uint32_t local_ssrc) override;
// Sender part.
void RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) override;
@ -110,6 +112,11 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
uint32_t SSRC() const override { return rtcp_sender_.SSRC(); }
// Semantically identical to `SSRC()` but must be called on the packet
// delivery thread/tq and returns the ssrc that maps to
// RtpRtcpInterface::Configuration::local_media_ssrc.
uint32_t local_media_ssrc() const;
void SetRid(const std::string& rid) override;
void SetMid(const std::string& mid) override;
@ -284,6 +291,7 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
TaskQueueBase* const worker_queue_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker process_thread_checker_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_;
std::unique_ptr<RtpSenderContext> rtp_sender_;

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@ -180,6 +180,10 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
// Called when the local ssrc changes (post initialization) for receive
// streams to match with send. Called on the packet receive thread/tq.
virtual void SetLocalSsrc(uint32_t ssrc) = 0;
// **************************************************************************
// Sender
// **************************************************************************