Google Git
Sign in
webrtc / src / bf9e01ab4e5c771596949edc91bdbb3cf7128874 / . / call / call.cc
blob: 5baf4250ef641d336a21b56a20d02f4d20826857 [file] [log] [blame]
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <string.h>
#include <algorithm>
#include <map>
#include <memory>
#include <set>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/transport/network_control.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "call/bitrate_allocator.h"
#include "call/call.h"
#include "call/flexfec_receive_stream_impl.h"
#include "call/receive_time_calculator.h"
#include "call/rtp_stream_receiver_controller.h"
#include "call/rtp_transport_controller_send.h"
#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/bitrate_controller/include/bitrate_controller.h"
#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
#include "modules/rtp_rtcp/include/flexfec_receiver.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/fec_controller_default.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/sequenced_task_checker.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/rw_lock_wrapper.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/cpu_info.h"
#include "system_wrappers/include/metrics.h"
#include "video/call_stats.h"
#include "video/send_delay_stats.h"
#include "video/stats_counter.h"
#include "video/video_receive_stream.h"
#include "video/video_send_stream.h"
namespace webrtc {
namespace {
bool SendFeedbackOnRequestOnly(const std::vector<RtpExtension>& extensions) {
for (const auto& extension : extensions) {
if (extension.uri == RtpExtension::kTransportSequenceNumberV2Uri)
return true;
}
return false;
}
// TODO(nisse): This really begs for a shared context struct.
bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
bool transport_cc) {
if (!transport_cc)
return false;
for (const auto& extension : extensions) {
if (extension.uri == RtpExtension::kTransportSequenceNumberUri ||
extension.uri == RtpExtension::kTransportSequenceNumberV2Uri)
return true;
}
return false;
}
bool UseSendSideBwe(const VideoReceiveStream::Config& config) {
return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
}
bool UseSendSideBwe(const AudioReceiveStream::Config& config) {
return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
}
bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
}
const int* FindKeyByValue(const std::map<int, int>& m, int v) {
for (const auto& kv : m) {
if (kv.second == v)
return &kv.first;
}
return nullptr;
}
std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
const VideoReceiveStream::Config& config) {
auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
rtclog_config->local_ssrc = config.rtp.local_ssrc;
rtclog_config->rtx_ssrc = config.rtp.rtx_ssrc;
rtclog_config->rtcp_mode = config.rtp.rtcp_mode;
rtclog_config->remb = config.rtp.remb;
rtclog_config->rtp_extensions = config.rtp.extensions;
for (const auto& d : config.decoders) {
const int* search =
FindKeyByValue(config.rtp.rtx_associated_payload_types, d.payload_type);
rtclog_config->codecs.emplace_back(d.video_format.name, d.payload_type,
search ? *search : 0);
}
return rtclog_config;
}
std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
const VideoSendStream::Config& config,
size_t ssrc_index) {
auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
rtclog_config->local_ssrc = config.rtp.ssrcs[ssrc_index];
if (ssrc_index < config.rtp.rtx.ssrcs.size()) {
rtclog_config->rtx_ssrc = config.rtp.rtx.ssrcs[ssrc_index];
}
rtclog_config->rtcp_mode = config.rtp.rtcp_mode;
rtclog_config->rtp_extensions = config.rtp.extensions;
rtclog_config->codecs.emplace_back(config.rtp.payload_name,
config.rtp.payload_type,
config.rtp.rtx.payload_type);
return rtclog_config;
}
std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
const AudioReceiveStream::Config& config) {
auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
rtclog_config->local_ssrc = config.rtp.local_ssrc;
rtclog_config->rtp_extensions = config.rtp.extensions;
return rtclog_config;
}
} // namespace
namespace internal {
class Call final : public webrtc::Call,
public PacketReceiver,
public RecoveredPacketReceiver,
public TargetTransferRateObserver,
public BitrateAllocator::LimitObserver {
public:
Call(const Call::Config& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
~Call() override;
// Implements webrtc::Call.
PacketReceiver* Receiver() override;
webrtc::AudioSendStream* CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) override;
void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
webrtc::AudioReceiveStream* CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) override;
void DestroyAudioReceiveStream(
webrtc::AudioReceiveStream* receive_stream) override;
webrtc::VideoSendStream* CreateVideoSendStream(
webrtc::VideoSendStream::Config config,
VideoEncoderConfig encoder_config) override;
webrtc::VideoSendStream* CreateVideoSendStream(
webrtc::VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
std::unique_ptr<FecController> fec_controller) override;
void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
webrtc::VideoReceiveStream* CreateVideoReceiveStream(
webrtc::VideoReceiveStream::Config configuration) override;
void DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) override;
FlexfecReceiveStream* CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config& config) override;
void DestroyFlexfecReceiveStream(
FlexfecReceiveStream* receive_stream) override;
RtpTransportControllerSendInterface* GetTransportControllerSend() override;
Stats GetStats() const override;
// Implements PacketReceiver.
DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
// Implements RecoveredPacketReceiver.
void OnRecoveredPacket(const uint8_t* packet, size_t length) override;
void SetBitrateAllocationStrategy(
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy) override;
void SignalChannelNetworkState(MediaType media, NetworkState state) override;
void OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
// Implements TargetTransferRateObserver,
void OnTargetTransferRate(TargetTransferRate msg) override;
void OnStartRateUpdate(DataRate start_rate) override;
// Implements BitrateAllocator::LimitObserver.
void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
uint32_t max_padding_bitrate_bps,
uint32_t total_bitrate_bps) override;
// This method is invoked when the media transport is created and when the
// media transport is being destructed.
// We only allow one media transport per connection.
//
// It should be called with non-null argument at most once, and if it was
// called with non-null argument, it has to be called with a null argument
// at least once after that.
void MediaTransportChange(MediaTransportInterface* media_transport) override;
private:
DeliveryStatus DeliverRtcp(MediaType media_type,
const uint8_t* packet,
size_t length);
DeliveryStatus DeliverRtp(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us);
void ConfigureSync(const std::string& sync_group)
RTC_EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
MediaType media_type)
RTC_SHARED_LOCKS_REQUIRED(receive_crit_);
void UpdateSendHistograms(int64_t first_sent_packet_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
void UpdateReceiveHistograms();
void UpdateHistograms();
void UpdateAggregateNetworkState();
// If |media_transport| is not null, it registers the rate observer for the
// media transport.
void RegisterRateObserver() RTC_LOCKS_EXCLUDED(target_observer_crit_);
// Intended for DCHECKs, to avoid locking in production builds.
MediaTransportInterface* media_transport()
RTC_LOCKS_EXCLUDED(target_observer_crit_);
Clock* const clock_;
const int num_cpu_cores_;
const std::unique_ptr<ProcessThread> module_process_thread_;
const std::unique_ptr<CallStats> call_stats_;
const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
Call::Config config_;
rtc::SequencedTaskChecker configuration_sequence_checker_;
NetworkState audio_network_state_;
NetworkState video_network_state_;
rtc::CriticalSection aggregate_network_up_crit_;
bool aggregate_network_up_ RTC_GUARDED_BY(aggregate_network_up_crit_);
std::unique_ptr<RWLockWrapper> receive_crit_;
// Audio, Video, and FlexFEC receive streams are owned by the client that
// creates them.
std::set<AudioReceiveStream*> audio_receive_streams_
RTC_GUARDED_BY(receive_crit_);
std::set<VideoReceiveStream*> video_receive_streams_
RTC_GUARDED_BY(receive_crit_);
std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
RTC_GUARDED_BY(receive_crit_);
// TODO(nisse): Should eventually be injected at creation,
// with a single object in the bundled case.
RtpStreamReceiverController audio_receiver_controller_;
RtpStreamReceiverController video_receiver_controller_;
// This extra map is used for receive processing which is
// independent of media type.
// TODO(nisse): In the RTP transport refactoring, we should have a
// single mapping from ssrc to a more abstract receive stream, with
// accessor methods for all configuration we need at this level.
struct ReceiveRtpConfig {
explicit ReceiveRtpConfig(const webrtc::AudioReceiveStream::Config& config)
: extensions(config.rtp.extensions),
use_send_side_bwe(UseSendSideBwe(config)) {}
explicit ReceiveRtpConfig(const webrtc::VideoReceiveStream::Config& config)
: extensions(config.rtp.extensions),
use_send_side_bwe(UseSendSideBwe(config)) {}
explicit ReceiveRtpConfig(const FlexfecReceiveStream::Config& config)
: extensions(config.rtp_header_extensions),
use_send_side_bwe(UseSendSideBwe(config)) {}
// Registered RTP header extensions for each stream. Note that RTP header
// extensions are negotiated per track ("m= line") in the SDP, but we have
// no notion of tracks at the Call level. We therefore store the RTP header
// extensions per SSRC instead, which leads to some storage overhead.
const RtpHeaderExtensionMap extensions;
// Set if both RTP extension the RTCP feedback message needed for
// send side BWE are negotiated.
const bool use_send_side_bwe;
};
std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
RTC_GUARDED_BY(receive_crit_);
std::unique_ptr<RWLockWrapper> send_crit_;
// Audio and Video send streams are owned by the client that creates them.
std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_
RTC_GUARDED_BY(send_crit_);
std::map<uint32_t, VideoSendStream*> video_send_ssrcs_
RTC_GUARDED_BY(send_crit_);
std::set<VideoSendStream*> video_send_streams_ RTC_GUARDED_BY(send_crit_);
using RtpStateMap = std::map<uint32_t, RtpState>;
RtpStateMap suspended_audio_send_ssrcs_
RTC_GUARDED_BY(configuration_sequence_checker_);
RtpStateMap suspended_video_send_ssrcs_
RTC_GUARDED_BY(configuration_sequence_checker_);
using RtpPayloadStateMap = std::map<uint32_t, RtpPayloadState>;
RtpPayloadStateMap suspended_video_payload_states_
RTC_GUARDED_BY(configuration_sequence_checker_);
webrtc::RtcEventLog* event_log_;
// The following members are only accessed (exclusively) from one thread and
// from the destructor, and therefore doesn't need any explicit
// synchronization.
RateCounter received_bytes_per_second_counter_;
RateCounter received_audio_bytes_per_second_counter_;
RateCounter received_video_bytes_per_second_counter_;
RateCounter received_rtcp_bytes_per_second_counter_;
absl::optional<int64_t> first_received_rtp_audio_ms_;
absl::optional<int64_t> last_received_rtp_audio_ms_;
absl::optional<int64_t> first_received_rtp_video_ms_;
absl::optional<int64_t> last_received_rtp_video_ms_;
rtc::CriticalSection last_bandwidth_bps_crit_;
uint32_t last_bandwidth_bps_ RTC_GUARDED_BY(&last_bandwidth_bps_crit_);
// TODO(holmer): Remove this lock once BitrateController no longer calls
// OnNetworkChanged from multiple threads.
rtc::CriticalSection bitrate_crit_;
uint32_t min_allocated_send_bitrate_bps_ RTC_GUARDED_BY(&bitrate_crit_);
uint32_t configured_max_padding_bitrate_bps_ RTC_GUARDED_BY(&bitrate_crit_);
AvgCounter estimated_send_bitrate_kbps_counter_
RTC_GUARDED_BY(&bitrate_crit_);
AvgCounter pacer_bitrate_kbps_counter_ RTC_GUARDED_BY(&bitrate_crit_);
ReceiveSideCongestionController receive_side_cc_;
const std::unique_ptr<ReceiveTimeCalculator> receive_time_calculator_;
const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
const int64_t start_ms_;
// Caches transport_send_.get(), to avoid racing with destructor.
// Note that this is declared before transport_send_ to ensure that it is not
// invalidated until no more tasks can be running on the transport_send_ task
// queue.
RtpTransportControllerSendInterface* transport_send_ptr_;
// Declared last since it will issue callbacks from a task queue. Declaring it
// last ensures that it is destroyed first and any running tasks are finished.
std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
// This is a precaution, since |MediaTransportChange| is not guaranteed to be
// invoked on a particular thread.
rtc::CriticalSection target_observer_crit_;
bool is_target_rate_observer_registered_
RTC_GUARDED_BY(&target_observer_crit_) = false;
MediaTransportInterface* media_transport_
RTC_GUARDED_BY(&target_observer_crit_) = nullptr;
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
} // namespace internal
std::string Call::Stats::ToString(int64_t time_ms) const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "Call stats: " << time_ms << ", {";
ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
ss << "rtt_ms: " << rtt_ms;
ss << '}';
return ss.str();
}
Call* Call::Create(const Call::Config& config) {
return new internal::Call(
config, absl::make_unique<RtpTransportControllerSend>(
Clock::GetRealTimeClock(), config.event_log,
config.network_controller_factory, config.bitrate_config));
}
Call* Call::Create(
const Call::Config& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send) {
return new internal::Call(config, std::move(transport_send));
}
// This method here to avoid subclasses has to implement this method.
// Call perf test will use Internal::Call::CreateVideoSendStream() to inject
// FecController.
VideoSendStream* Call::CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
std::unique_ptr<FecController> fec_controller) {
return nullptr;
}
namespace internal {
Call::Call(const Call::Config& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send)
: clock_(Clock::GetRealTimeClock()),
num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
call_stats_(new CallStats(clock_, module_process_thread_.get())),
bitrate_allocator_(new BitrateAllocator(this)),
config_(config),
audio_network_state_(kNetworkDown),
video_network_state_(kNetworkDown),
aggregate_network_up_(false),
receive_crit_(RWLockWrapper::CreateRWLock()),
send_crit_(RWLockWrapper::CreateRWLock()),
event_log_(config.event_log),
received_bytes_per_second_counter_(clock_, nullptr, true),
received_audio_bytes_per_second_counter_(clock_, nullptr, true),
received_video_bytes_per_second_counter_(clock_, nullptr, true),
received_rtcp_bytes_per_second_counter_(clock_, nullptr, true),
last_bandwidth_bps_(0),
min_allocated_send_bitrate_bps_(0),
configured_max_padding_bitrate_bps_(0),
estimated_send_bitrate_kbps_counter_(clock_, nullptr, true),
pacer_bitrate_kbps_counter_(clock_, nullptr, true),
receive_side_cc_(clock_, transport_send->packet_router()),
receive_time_calculator_(ReceiveTimeCalculator::CreateFromFieldTrial()),
video_send_delay_stats_(new SendDelayStats(clock_)),
start_ms_(clock_->TimeInMilliseconds()) {
RTC_DCHECK(config.event_log != nullptr);
transport_send_ = std::move(transport_send);
transport_send_ptr_ = transport_send_.get();
}
Call::~Call() {
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
RTC_CHECK(video_send_streams_.empty());
RTC_CHECK(audio_receive_streams_.empty());
RTC_CHECK(video_receive_streams_.empty());
if (!media_transport_) {
module_process_thread_->DeRegisterModule(
receive_side_cc_.GetRemoteBitrateEstimator(true));
module_process_thread_->DeRegisterModule(&receive_side_cc_);
module_process_thread_->DeRegisterModule(call_stats_.get());
module_process_thread_->Stop();
call_stats_->DeregisterStatsObserver(&receive_side_cc_);
call_stats_->DeregisterStatsObserver(
transport_send_->GetCallStatsObserver());
}
int64_t first_sent_packet_ms = transport_send_->GetFirstPacketTimeMs();
// Only update histograms after process threads have been shut down, so that
// they won't try to concurrently update stats.
{
rtc::CritScope lock(&bitrate_crit_);
UpdateSendHistograms(first_sent_packet_ms);
}
UpdateReceiveHistograms();
UpdateHistograms();
}
void Call::RegisterRateObserver() {
rtc::CritScope lock(&target_observer_crit_);
if (is_target_rate_observer_registered_) {
return;
}
is_target_rate_observer_registered_ = true;
if (media_transport_) {
// TODO(bugs.webrtc.org/9719): We should report call_stats_ from
// media transport (at least Rtt). We should extend media transport
// interface to include "receive_side bwe" if needed.
media_transport_->AddTargetTransferRateObserver(this);
} else {
transport_send_ptr_->RegisterTargetTransferRateObserver(this);
call_stats_->RegisterStatsObserver(&receive_side_cc_);
call_stats_->RegisterStatsObserver(transport_send_->GetCallStatsObserver());
module_process_thread_->RegisterModule(
receive_side_cc_.GetRemoteBitrateEstimator(true), RTC_FROM_HERE);
module_process_thread_->RegisterModule(call_stats_.get(), RTC_FROM_HERE);
module_process_thread_->RegisterModule(&receive_side_cc_, RTC_FROM_HERE);
module_process_thread_->Start();
}
}
MediaTransportInterface* Call::media_transport() {
rtc::CritScope lock(&target_observer_crit_);
return media_transport_;
}
void Call::MediaTransportChange(MediaTransportInterface* media_transport) {
rtc::CritScope lock(&target_observer_crit_);
if (is_target_rate_observer_registered_) {
// Only used to unregister rate observer from media transport. Registration
// happens when the stream is created.
if (!media_transport && media_transport_) {
media_transport_->RemoveTargetTransferRateObserver(this);
media_transport_ = nullptr;
is_target_rate_observer_registered_ = false;
}
} else if (media_transport) {
RTC_DCHECK(media_transport_ == nullptr ||
media_transport_ == media_transport)
<< "media_transport_=" << (media_transport_ != nullptr)
<< ", (media_transport_==media_transport)="
<< (media_transport_ == media_transport);
media_transport_ = media_transport;
}
}
void Call::UpdateHistograms() {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.LifetimeInSeconds",
(clock_->TimeInMilliseconds() - start_ms_) / 1000);
}
void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) {
if (first_sent_packet_ms == -1)
return;
int64_t elapsed_sec =
(clock_->TimeInMilliseconds() - first_sent_packet_ms) / 1000;
if (elapsed_sec < metrics::kMinRunTimeInSeconds)
return;
const int kMinRequiredPeriodicSamples = 5;
AggregatedStats send_bitrate_stats =
estimated_send_bitrate_kbps_counter_.ProcessAndGetStats();
if (send_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
send_bitrate_stats.average);
RTC_LOG(LS_INFO) << "WebRTC.Call.EstimatedSendBitrateInKbps, "
<< send_bitrate_stats.ToString();
}
AggregatedStats pacer_bitrate_stats =
pacer_bitrate_kbps_counter_.ProcessAndGetStats();
if (pacer_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
pacer_bitrate_stats.average);
RTC_LOG(LS_INFO) << "WebRTC.Call.PacerBitrateInKbps, "
<< pacer_bitrate_stats.ToString();
}
}
void Call::UpdateReceiveHistograms() {
if (first_received_rtp_audio_ms_) {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds",
(*last_received_rtp_audio_ms_ - *first_received_rtp_audio_ms_) / 1000);
}
if (first_received_rtp_video_ms_) {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds",
(*last_received_rtp_video_ms_ - *first_received_rtp_video_ms_) / 1000);
}
const int kMinRequiredPeriodicSamples = 5;
AggregatedStats video_bytes_per_sec =
received_video_bytes_per_second_counter_.GetStats();
if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
video_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBps, "
<< video_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats audio_bytes_per_sec =
received_audio_bytes_per_second_counter_.GetStats();
if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
audio_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInBps, "
<< audio_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats rtcp_bytes_per_sec =
received_rtcp_bytes_per_second_counter_.GetStats();
if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
rtcp_bytes_per_sec.average * 8);
RTC_LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBps, "
<< rtcp_bytes_per_sec.ToStringWithMultiplier(8);
}
AggregatedStats recv_bytes_per_sec =
received_bytes_per_second_counter_.GetStats();
if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
recv_bytes_per_sec.average * 8 / 1000);
RTC_LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, "
<< recv_bytes_per_sec.ToStringWithMultiplier(8);
}
}
PacketReceiver* Call::Receiver() {
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
return this;
}
webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(media_transport() == config.media_transport);
RegisterRateObserver();
// Stream config is logged in AudioSendStream::ConfigureStream, as it may
// change during the stream's lifetime.
absl::optional<RtpState> suspended_rtp_state;
{
const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
if (iter != suspended_audio_send_ssrcs_.end()) {
suspended_rtp_state.emplace(iter->second);
}
}
// TODO(srte): AudioSendStream should call GetWorkerQueue directly rather than
// having it injected.
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, transport_send_ptr_->GetWorkerQueue(),
module_process_thread_.get(), transport_send_ptr_,
bitrate_allocator_.get(), event_log_, call_stats_.get(),
suspended_rtp_state);
{
WriteLockScoped write_lock(*send_crit_);
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
audio_send_ssrcs_.end());
audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
}
{
ReadLockScoped read_lock(*receive_crit_);
for (AudioReceiveStream* stream : audio_receive_streams_) {
if (stream->config().rtp.local_ssrc == config.rtp.ssrc) {
stream->AssociateSendStream(send_stream);
}
}
}
send_stream->SignalNetworkState(audio_network_state_);
UpdateAggregateNetworkState();
return send_stream;
}
void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(send_stream != nullptr);
send_stream->Stop();
const uint32_t ssrc = send_stream->GetConfig().rtp.ssrc;
webrtc::internal::AudioSendStream* audio_send_stream =
static_cast<webrtc::internal::AudioSendStream*>(send_stream);
suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState();
{
WriteLockScoped write_lock(*send_crit_);
size_t num_deleted = audio_send_ssrcs_.erase(ssrc);
RTC_DCHECK_EQ(1, num_deleted);
}
{
ReadLockScoped read_lock(*receive_crit_);
for (AudioReceiveStream* stream : audio_receive_streams_) {
if (stream->config().rtp.local_ssrc == ssrc) {
stream->AssociateSendStream(nullptr);
}
}
}
UpdateAggregateNetworkState();
delete send_stream;
}
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RegisterRateObserver();
event_log_->Log(absl::make_unique<RtcEventAudioReceiveStreamConfig>(
CreateRtcLogStreamConfig(config)));
AudioReceiveStream* receive_stream = new AudioReceiveStream(
&audio_receiver_controller_, transport_send_ptr_->packet_router(),
module_process_thread_.get(), config, config_.audio_state, event_log_);
{
WriteLockScoped write_lock(*receive_crit_);
receive_rtp_config_.emplace(config.rtp.remote_ssrc,
ReceiveRtpConfig(config));
audio_receive_streams_.insert(receive_stream);
ConfigureSync(config.sync_group);
}
{
ReadLockScoped read_lock(*send_crit_);
auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
if (it != audio_send_ssrcs_.end()) {
receive_stream->AssociateSendStream(it->second);
}
}
receive_stream->SignalNetworkState(audio_network_state_);
UpdateAggregateNetworkState();
return receive_stream;
}
void Call::DestroyAudioReceiveStream(
webrtc::AudioReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
webrtc::internal::AudioReceiveStream* audio_receive_stream =
static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
{
WriteLockScoped write_lock(*receive_crit_);
const AudioReceiveStream::Config& config = audio_receive_stream->config();
uint32_t ssrc = config.rtp.remote_ssrc;
receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
->RemoveStream(ssrc);
audio_receive_streams_.erase(audio_receive_stream);
const std::string& sync_group = audio_receive_stream->config().sync_group;
const auto it = sync_stream_mapping_.find(sync_group);
if (it != sync_stream_mapping_.end() &&
it->second == audio_receive_stream) {
sync_stream_mapping_.erase(it);
ConfigureSync(sync_group);
}
receive_rtp_config_.erase(ssrc);
}
UpdateAggregateNetworkState();
delete audio_receive_stream;
}
// This method can be used for Call tests with external fec controller factory.
webrtc::VideoSendStream* Call::CreateVideoSendStream(
webrtc::VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
std::unique_ptr<FecController> fec_controller) {
TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(media_transport() == config.media_transport);
RegisterRateObserver();
video_send_delay_stats_->AddSsrcs(config);
for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
++ssrc_index) {
event_log_->Log(absl::make_unique<RtcEventVideoSendStreamConfig>(
CreateRtcLogStreamConfig(config, ssrc_index)));
}
// TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
// the call has already started.
// Copy ssrcs from |config| since |config| is moved.
std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
// TODO(srte): VideoSendStream should call GetWorkerQueue directly rather than
// having it injected.
VideoSendStream* send_stream = new VideoSendStream(
num_cpu_cores_, module_process_thread_.get(),
transport_send_ptr_->GetWorkerQueue(), call_stats_.get(),
transport_send_ptr_, bitrate_allocator_.get(),
video_send_delay_stats_.get(), event_log_, std::move(config),
std::move(encoder_config), suspended_video_send_ssrcs_,
suspended_video_payload_states_, std::move(fec_controller));
{
WriteLockScoped write_lock(*send_crit_);
for (uint32_t ssrc : ssrcs) {
RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
video_send_ssrcs_[ssrc] = send_stream;
}
video_send_streams_.insert(send_stream);
}
UpdateAggregateNetworkState();
return send_stream;
}
webrtc::VideoSendStream* Call::CreateVideoSendStream(
webrtc::VideoSendStream::Config config,
VideoEncoderConfig encoder_config) {
if (config_.fec_controller_factory) {
RTC_LOG(LS_INFO) << "External FEC Controller will be used.";
}
std::unique_ptr<FecController> fec_controller =
config_.fec_controller_factory
? config_.fec_controller_factory->CreateFecController()
: absl::make_unique<FecControllerDefault>(Clock::GetRealTimeClock());
return CreateVideoSendStream(std::move(config), std::move(encoder_config),
std::move(fec_controller));
}
void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
RTC_DCHECK(send_stream != nullptr);
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
send_stream->Stop();
VideoSendStream* send_stream_impl = nullptr;
{
WriteLockScoped write_lock(*send_crit_);
auto it = video_send_ssrcs_.begin();
while (it != video_send_ssrcs_.end()) {
if (it->second == static_cast<VideoSendStream*>(send_stream)) {
send_stream_impl = it->second;
video_send_ssrcs_.erase(it++);
} else {
++it;
}
}
video_send_streams_.erase(send_stream_impl);
}
RTC_CHECK(send_stream_impl != nullptr);
VideoSendStream::RtpStateMap rtp_states;
VideoSendStream::RtpPayloadStateMap rtp_payload_states;
send_stream_impl->StopPermanentlyAndGetRtpStates(&rtp_states,
&rtp_payload_states);
for (const auto& kv : rtp_states) {
suspended_video_send_ssrcs_[kv.first] = kv.second;
}
for (const auto& kv : rtp_payload_states) {
suspended_video_payload_states_[kv.first] = kv.second;
}
UpdateAggregateNetworkState();
delete send_stream_impl;
}
webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
webrtc::VideoReceiveStream::Config configuration) {
TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
receive_side_cc_.SetSendFeedbackOnRequestOnly(
SendFeedbackOnRequestOnly(configuration.rtp.extensions));
RegisterRateObserver();
VideoReceiveStream* receive_stream = new VideoReceiveStream(
&video_receiver_controller_, num_cpu_cores_,
transport_send_ptr_->packet_router(), std::move(configuration),
module_process_thread_.get(), call_stats_.get());
const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
{
WriteLockScoped write_lock(*receive_crit_);
if (config.rtp.rtx_ssrc) {
// We record identical config for the rtx stream as for the main
// stream. Since the transport_send_cc negotiation is per payload
// type, we may get an incorrect value for the rtx stream, but
// that is unlikely to matter in practice.
receive_rtp_config_.emplace(config.rtp.rtx_ssrc,
ReceiveRtpConfig(config));
}
receive_rtp_config_.emplace(config.rtp.remote_ssrc,
ReceiveRtpConfig(config));
video_receive_streams_.insert(receive_stream);
ConfigureSync(config.sync_group);
}
receive_stream->SignalNetworkState(video_network_state_);
UpdateAggregateNetworkState();
event_log_->Log(absl::make_unique<RtcEventVideoReceiveStreamConfig>(
CreateRtcLogStreamConfig(config)));
return receive_stream;
}
void Call::DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
VideoReceiveStream* receive_stream_impl =
static_cast<VideoReceiveStream*>(receive_stream);
const VideoReceiveStream::Config& config = receive_stream_impl->config();
{
WriteLockScoped write_lock(*receive_crit_);
// Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
// separate SSRC there can be either one or two.
receive_rtp_config_.erase(config.rtp.remote_ssrc);
if (config.rtp.rtx_ssrc) {
receive_rtp_config_.erase(config.rtp.rtx_ssrc);
}
video_receive_streams_.erase(receive_stream_impl);
ConfigureSync(config.sync_group);
}
receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
->RemoveStream(config.rtp.remote_ssrc);
UpdateAggregateNetworkState();
delete receive_stream_impl;
}
FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RecoveredPacketReceiver* recovered_packet_receiver = this;
FlexfecReceiveStreamImpl* receive_stream;
{
WriteLockScoped write_lock(*receive_crit_);
// Unlike the video and audio receive streams,
// FlexfecReceiveStream implements RtpPacketSinkInterface itself,
// and hence its constructor passes its |this| pointer to
// video_receiver_controller_->CreateStream(). Calling the
// constructor while holding |receive_crit_| ensures that we don't
// call OnRtpPacket until the constructor is finished and the
// object is in a valid state.
// TODO(nisse): Fix constructor so that it can be moved outside of
// this locked scope.
receive_stream = new FlexfecReceiveStreamImpl(
&video_receiver_controller_, config, recovered_packet_receiver,
call_stats_.get(), module_process_thread_.get());
RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) ==
receive_rtp_config_.end());
receive_rtp_config_.emplace(config.remote_ssrc, ReceiveRtpConfig(config));
}
// TODO(brandtr): Store config in RtcEventLog here.
return receive_stream;
}
void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
RTC_DCHECK(receive_stream != nullptr);
{
WriteLockScoped write_lock(*receive_crit_);
const FlexfecReceiveStream::Config& config = receive_stream->GetConfig();
uint32_t ssrc = config.remote_ssrc;
receive_rtp_config_.erase(ssrc);
// Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
// destroyed.
receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
->RemoveStream(ssrc);
}
delete receive_stream;
}
RtpTransportControllerSendInterface* Call::GetTransportControllerSend() {
return transport_send_ptr_;
}
Call::Stats Call::GetStats() const {
// TODO(solenberg): Some test cases in EndToEndTest use this from a different
// thread. Re-enable once that is fixed.
// RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
Stats stats;
// Fetch available send/receive bitrates.
std::vector<unsigned int> ssrcs;
uint32_t recv_bandwidth = 0;
receive_side_cc_.GetRemoteBitrateEstimator(false)->LatestEstimate(
&ssrcs, &recv_bandwidth);
{
rtc::CritScope cs(&last_bandwidth_bps_crit_);
stats.send_bandwidth_bps = last_bandwidth_bps_;
}
stats.recv_bandwidth_bps = recv_bandwidth;
// TODO(srte): It is unclear if we only want to report queues if network is
// available.
{
rtc::CritScope cs(&aggregate_network_up_crit_);
stats.pacer_delay_ms = aggregate_network_up_
? transport_send_ptr_->GetPacerQueuingDelayMs()
: 0;
}
stats.rtt_ms = call_stats_->LastProcessedRtt();
{
rtc::CritScope cs(&bitrate_crit_);
stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_;
}
return stats;
}
void Call::SetBitrateAllocationStrategy(
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy) {
// TODO(srte): This function should be moved to RtpTransportControllerSend
// when BitrateAllocator is moved there.
struct Functor {
void operator()() {
bitrate_allocator_->SetBitrateAllocationStrategy(
std::move(bitrate_allocation_strategy_));
}
BitrateAllocator* bitrate_allocator_;
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy_;
};
transport_send_ptr_->GetWorkerQueue()->PostTask(Functor{
bitrate_allocator_.get(), std::move(bitrate_allocation_strategy)});
}
void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
switch (media) {
case MediaType::AUDIO:
audio_network_state_ = state;
break;
case MediaType::VIDEO:
video_network_state_ = state;
break;
case MediaType::ANY:
case MediaType::DATA:
RTC_NOTREACHED();
break;
}
UpdateAggregateNetworkState();
{
ReadLockScoped read_lock(*send_crit_);
for (auto& kv : audio_send_ssrcs_) {
kv.second->SignalNetworkState(audio_network_state_);
}
}
{
ReadLockScoped read_lock(*receive_crit_);
for (AudioReceiveStream* audio_receive_stream : audio_receive_streams_) {
audio_receive_stream->SignalNetworkState(audio_network_state_);
}
for (VideoReceiveStream* video_receive_stream : video_receive_streams_) {
video_receive_stream->SignalNetworkState(video_network_state_);
}
}
}
void Call::OnAudioTransportOverheadChanged(int transport_overhead_per_packet) {
ReadLockScoped read_lock(*send_crit_);
for (auto& kv : audio_send_ssrcs_) {
kv.second->SetTransportOverhead(transport_overhead_per_packet);
}
}
void Call::UpdateAggregateNetworkState() {
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
bool have_audio = false;
bool have_video = false;
{
ReadLockScoped read_lock(*send_crit_);
if (audio_send_ssrcs_.size() > 0)
have_audio = true;
if (video_send_ssrcs_.size() > 0)
have_video = true;
}
{
ReadLockScoped read_lock(*receive_crit_);
if (audio_receive_streams_.size() > 0)
have_audio = true;
if (video_receive_streams_.size() > 0)
have_video = true;
}
bool aggregate_network_up =
((have_video && video_network_state_ == kNetworkUp) ||
(have_audio && audio_network_state_ == kNetworkUp));
RTC_LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="
<< (aggregate_network_up ? "up" : "down");
{
rtc::CritScope cs(&aggregate_network_up_crit_);
aggregate_network_up_ = aggregate_network_up;
}
transport_send_ptr_->OnNetworkAvailability(aggregate_network_up);
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
clock_->TimeInMilliseconds());
transport_send_ptr_->OnSentPacket(sent_packet);
}
void Call::OnStartRateUpdate(DataRate start_rate) {
if (!transport_send_ptr_->GetWorkerQueue()->IsCurrent()) {
transport_send_ptr_->GetWorkerQueue()->PostTask(
[this, start_rate] { this->OnStartRateUpdate(start_rate); });
return;
}
bitrate_allocator_->UpdateStartRate(start_rate.bps<uint32_t>());
}
void Call::OnTargetTransferRate(TargetTransferRate msg) {
// TODO(bugs.webrtc.org/9719)
// Call::OnTargetTransferRate requires that on target transfer rate is invoked
// from the worker queue (because bitrate_allocator_ requires it). Media
// transport does not guarantee the callback on the worker queue.
// When the threading model for MediaTransportInterface is update, reconsider
// changing this implementation.
if (!transport_send_ptr_->GetWorkerQueue()->IsCurrent()) {
transport_send_ptr_->GetWorkerQueue()->PostTask(
[this, msg] { this->OnTargetTransferRate(msg); });
return;
}
uint32_t target_bitrate_bps = msg.target_rate.bps();
int loss_ratio_255 = msg.network_estimate.loss_rate_ratio * 255;
uint8_t fraction_loss =
rtc::dchecked_cast<uint8_t>(rtc::SafeClamp(loss_ratio_255, 0, 255));
int64_t rtt_ms = msg.network_estimate.round_trip_time.ms();
int64_t probing_interval_ms = msg.network_estimate.bwe_period.ms();
uint32_t bandwidth_bps = msg.network_estimate.bandwidth.bps();
{
rtc::CritScope cs(&last_bandwidth_bps_crit_);
last_bandwidth_bps_ = bandwidth_bps;
}
// For controlling the rate of feedback messages.
receive_side_cc_.OnBitrateChanged(target_bitrate_bps);
bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, bandwidth_bps,
fraction_loss, rtt_ms,
probing_interval_ms);
// Ignore updates if bitrate is zero (the aggregate network state is down).
if (target_bitrate_bps == 0) {
rtc::CritScope lock(&bitrate_crit_);
estimated_send_bitrate_kbps_counter_.ProcessAndPause();
pacer_bitrate_kbps_counter_.ProcessAndPause();
return;
}
bool sending_video;
{
ReadLockScoped read_lock(*send_crit_);
sending_video = !video_send_streams_.empty();
}
rtc::CritScope lock(&bitrate_crit_);
if (!sending_video) {
// Do not update the stats if we are not sending video.
estimated_send_bitrate_kbps_counter_.ProcessAndPause();
pacer_bitrate_kbps_counter_.ProcessAndPause();
return;
}
estimated_send_bitrate_kbps_counter_.Add(target_bitrate_bps / 1000);
// Pacer bitrate may be higher than bitrate estimate if enforcing min bitrate.
uint32_t pacer_bitrate_bps =
std::max(target_bitrate_bps, min_allocated_send_bitrate_bps_);
pacer_bitrate_kbps_counter_.Add(pacer_bitrate_bps / 1000);
}
void Call::OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
uint32_t max_padding_bitrate_bps,
uint32_t total_bitrate_bps) {
transport_send_ptr_->SetAllocatedSendBitrateLimits(
min_send_bitrate_bps, max_padding_bitrate_bps, total_bitrate_bps);
{
rtc::CritScope lock(&target_observer_crit_);
if (media_transport_) {
MediaTransportAllocatedBitrateLimits limits;
limits.min_pacing_rate = DataRate::bps(min_send_bitrate_bps);
limits.max_padding_bitrate = DataRate::bps(max_padding_bitrate_bps);
limits.max_total_allocated_bitrate = DataRate::bps(total_bitrate_bps);
media_transport_->SetAllocatedBitrateLimits(limits);
}
}
rtc::CritScope lock(&bitrate_crit_);
min_allocated_send_bitrate_bps_ = min_send_bitrate_bps;
configured_max_padding_bitrate_bps_ = max_padding_bitrate_bps;
}
void Call::ConfigureSync(const std::string& sync_group) {
// Set sync only if there was no previous one.
if (sync_group.empty())
return;
AudioReceiveStream* sync_audio_stream = nullptr;
// Find existing audio stream.
const auto it = sync_stream_mapping_.find(sync_group);
if (it != sync_stream_mapping_.end()) {
sync_audio_stream = it->second;
} else {
// No configured audio stream, see if we can find one.
for (AudioReceiveStream* stream : audio_receive_streams_) {
if (stream->config().sync_group == sync_group) {
if (sync_audio_stream != nullptr) {
RTC_LOG(LS_WARNING)
<< "Attempting to sync more than one audio stream "
"within the same sync group. This is not "
"supported in the current implementation.";
break;
}
sync_audio_stream = stream;
}
}
}
if (sync_audio_stream)
sync_stream_mapping_[sync_group] = sync_audio_stream;
size_t num_synced_streams = 0;
for (VideoReceiveStream* video_stream : video_receive_streams_) {
if (video_stream->config().sync_group != sync_group)
continue;
++num_synced_streams;
if (num_synced_streams > 1) {
// TODO(pbos): Support synchronizing more than one A/V pair.
// https://code.google.com/p/webrtc/issues/detail?id=4762
RTC_LOG(LS_WARNING)
<< "Attempting to sync more than one audio/video pair "
"within the same sync group. This is not supported in "
"the current implementation.";
}
// Only sync the first A/V pair within this sync group.
if (num_synced_streams == 1) {
// sync_audio_stream may be null and that's ok.
video_stream->SetSync(sync_audio_stream);
} else {
video_stream->SetSync(nullptr);
}
}
}
PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
const uint8_t* packet,
size_t length) {
TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
// TODO(pbos): Make sure it's a valid packet.
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that
// there's no receiver of the packet.
if (received_bytes_per_second_counter_.HasSample()) {
// First RTP packet has been received.
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_rtcp_bytes_per_second_counter_.Add(static_cast<int>(length));
}
bool rtcp_delivered = false;
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
ReadLockScoped read_lock(*receive_crit_);
for (VideoReceiveStream* stream : video_receive_streams_) {
if (stream->DeliverRtcp(packet, length))
rtcp_delivered = true;
}
}
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
ReadLockScoped read_lock(*receive_crit_);
for (AudioReceiveStream* stream : audio_receive_streams_) {
if (stream->DeliverRtcp(packet, length))
rtcp_delivered = true;
}
}
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
ReadLockScoped read_lock(*send_crit_);
for (VideoSendStream* stream : video_send_streams_) {
if (stream->DeliverRtcp(packet, length))
rtcp_delivered = true;
}
}
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
ReadLockScoped read_lock(*send_crit_);
for (auto& kv : audio_send_ssrcs_) {
if (kv.second->DeliverRtcp(packet, length))
rtcp_delivered = true;
}
}
if (rtcp_delivered) {
event_log_->Log(absl::make_unique<RtcEventRtcpPacketIncoming>(
rtc::MakeArrayView(packet, length)));
}
return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
}
PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
TRACE_EVENT0("webrtc", "Call::DeliverRtp");
RtpPacketReceived parsed_packet;
if (!parsed_packet.Parse(std::move(packet)))
return DELIVERY_PACKET_ERROR;
if (packet_time_us != -1) {
if (receive_time_calculator_) {
// Repair packet_time_us for clock resets by comparing a new read of
// the same clock (TimeUTCMicros) to a monotonic clock reading.
packet_time_us = receive_time_calculator_->ReconcileReceiveTimes(
packet_time_us, rtc::TimeUTCMicros(), clock_->TimeInMicroseconds());
}
parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
} else {
parsed_packet.set_arrival_time_ms(clock_->TimeInMilliseconds());
}
// We might get RTP keep-alive packets in accordance with RFC6263 section 4.6.
// These are empty (zero length payload) RTP packets with an unsignaled
// payload type.
const bool is_keep_alive_packet = parsed_packet.payload_size() == 0;
RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO ||
is_keep_alive_packet);
ReadLockScoped read_lock(*receive_crit_);
auto it = receive_rtp_config_.find(parsed_packet.Ssrc());
if (it == receive_rtp_config_.end()) {
RTC_LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc "
<< parsed_packet.Ssrc();
// Destruction of the receive stream, including deregistering from the
// RtpDemuxer, is not protected by the |receive_crit_| lock. But
// deregistering in the |receive_rtp_config_| map is protected by that lock.
// So by not passing the packet on to demuxing in this case, we prevent
// incoming packets to be passed on via the demuxer to a receive stream
// which is being torned down.
return DELIVERY_UNKNOWN_SSRC;
}
parsed_packet.IdentifyExtensions(it->second.extensions);
NotifyBweOfReceivedPacket(parsed_packet, media_type);
// RateCounters expect input parameter as int, save it as int,
// instead of converting each time it is passed to RateCounter::Add below.
int length = static_cast<int>(parsed_packet.size());
if (media_type == MediaType::AUDIO) {
if (audio_receiver_controller_.OnRtpPacket(parsed_packet)) {
received_bytes_per_second_counter_.Add(length);
received_audio_bytes_per_second_counter_.Add(length);
event_log_->Log(
absl::make_unique<RtcEventRtpPacketIncoming>(parsed_packet));
const int64_t arrival_time_ms = parsed_packet.arrival_time_ms();
if (!first_received_rtp_audio_ms_) {
first_received_rtp_audio_ms_.emplace(arrival_time_ms);
}
last_received_rtp_audio_ms_.emplace(arrival_time_ms);
return DELIVERY_OK;
}
} else if (media_type == MediaType::VIDEO) {
parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
if (video_receiver_controller_.OnRtpPacket(parsed_packet)) {
received_bytes_per_second_counter_.Add(length);
received_video_bytes_per_second_counter_.Add(length);
event_log_->Log(
absl::make_unique<RtcEventRtpPacketIncoming>(parsed_packet));
const int64_t arrival_time_ms = parsed_packet.arrival_time_ms();
if (!first_received_rtp_video_ms_) {
first_received_rtp_video_ms_.emplace(arrival_time_ms);
}
last_received_rtp_video_ms_.emplace(arrival_time_ms);
return DELIVERY_OK;
}
}
return DELIVERY_UNKNOWN_SSRC;
}
PacketReceiver::DeliveryStatus Call::DeliverPacket(
MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size()))
return DeliverRtcp(media_type, packet.cdata(), packet.size());
return DeliverRtp(media_type, std::move(packet), packet_time_us);
}
void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
RtpPacketReceived parsed_packet;
if (!parsed_packet.Parse(packet, length))
return;
parsed_packet.set_recovered(true);
ReadLockScoped read_lock(*receive_crit_);
auto it = receive_rtp_config_.find(parsed_packet.Ssrc());
if (it == receive_rtp_config_.end()) {
RTC_LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc "
<< parsed_packet.Ssrc();
// Destruction of the receive stream, including deregistering from the
// RtpDemuxer, is not protected by the |receive_crit_| lock. But
// deregistering in the |receive_rtp_config_| map is protected by that lock.
// So by not passing the packet on to demuxing in this case, we prevent
// incoming packets to be passed on via the demuxer to a receive stream
// which is being torn down.
return;
}
parsed_packet.IdentifyExtensions(it->second.extensions);
// TODO(brandtr): Update here when we support protecting audio packets too.
parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
video_receiver_controller_.OnRtpPacket(parsed_packet);
}
void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
MediaType media_type) {
auto it = receive_rtp_config_.find(packet.Ssrc());
bool use_send_side_bwe =
(it != receive_rtp_config_.end()) && it->second.use_send_side_bwe;
RTPHeader header;
packet.GetHeader(&header);
if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) {
// Inconsistent configuration of send side BWE. Do nothing.
// TODO(nisse): Without this check, we may produce RTCP feedback
// packets even when not negotiated. But it would be cleaner to
// move the check down to RTCPSender::SendFeedbackPacket, which
// would also help the PacketRouter to select an appropriate rtp
// module in the case that some, but not all, have RTCP feedback
// enabled.
return;
}
// For audio, we only support send side BWE.
if (media_type == MediaType::VIDEO ||
(use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
receive_side_cc_.OnReceivedPacket(
packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
header);
}
}
} // namespace internal
} // namespace webrtc