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webrtc_m130/webrtc/call/call.cc
nisse 0ffdcc51bc Delete unneeded includes of deprecated system_wrappers include files. 
Deletes left-over includes of trace.h and critical_section_wrapper.h.

BUG=webrtc:7035

Review-Url: https://codereview.webrtc.org/2784873002
Cr-Commit-Position: refs/heads/master@{#17460}
2017-03-30 07:31:15 +00:00

1335 lines
51 KiB
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/*
* 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 "webrtc/audio/audio_receive_stream.h"
#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/location.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/call/bitrate_allocator.h"
#include "webrtc/call/call.h"
#include "webrtc/call/flexfec_receive_stream_impl.h"
#include "webrtc/call/rtp_transport_controller_send.h"
#include "webrtc/config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/video/call_stats.h"
#include "webrtc/video/send_delay_stats.h"
#include "webrtc/video/stats_counter.h"
#include "webrtc/video/video_receive_stream.h"
#include "webrtc/video/video_send_stream.h"
#include "webrtc/video/vie_remb.h"
namespace webrtc {
const int Call::Config::kDefaultStartBitrateBps = 300000;
namespace {
// 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)
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);
}
class RtpTransportControllerSend : public RtpTransportControllerSendInterface {
public:
RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
void InitCongestionControl(SendSideCongestionController::Observer* observer);
PacketRouter* packet_router() override { return &packet_router_; }
SendSideCongestionController* send_side_cc() override {
return send_side_cc_.get();
}
TransportFeedbackObserver* transport_feedback_observer() override {
return send_side_cc_.get();
}
RtpPacketSender* packet_sender() override { return send_side_cc_->pacer(); }
private:
Clock* const clock_;
webrtc::RtcEventLog* const event_log_;
PacketRouter packet_router_;
// Construction delayed until InitCongestionControl, since the
// CongestionController wants its observer as a construction time
// argument, and setting it later seems non-trivial.
std::unique_ptr<SendSideCongestionController> send_side_cc_;
};
RtpTransportControllerSend::RtpTransportControllerSend(
Clock* clock,
webrtc::RtcEventLog* event_log)
: clock_(clock), event_log_(event_log) {}
void RtpTransportControllerSend::InitCongestionControl(
SendSideCongestionController::Observer* observer) {
// Must be called only once.
RTC_CHECK(!send_side_cc_);
send_side_cc_.reset(new SendSideCongestionController(
clock_, observer, event_log_, &packet_router_));
}
} // namespace
namespace internal {
class Call : public webrtc::Call,
public PacketReceiver,
public RecoveredPacketReceiver,
public SendSideCongestionController::Observer,
public BitrateAllocator::LimitObserver {
public:
Call(const Call::Config& config,
std::unique_ptr<RtpTransportControllerSend> transport_send);
virtual ~Call();
// 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;
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;
Stats GetStats() const override;
// Implements PacketReceiver.
DeliveryStatus DeliverPacket(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) override;
// Implements RecoveredPacketReceiver.
bool OnRecoveredPacket(const uint8_t* packet, size_t length) override;
void SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
void SignalChannelNetworkState(MediaType media, NetworkState state) override;
void OnTransportOverheadChanged(MediaType media,
int transport_overhead_per_packet) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
// Implements BitrateObserver.
void OnNetworkChanged(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt_ms,
int64_t probing_interval_ms) override;
// Implements BitrateAllocator::LimitObserver.
void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
uint32_t max_padding_bitrate_bps) override;
private:
DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
size_t length);
DeliveryStatus DeliverRtp(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time);
void ConfigureSync(const std::string& sync_group)
EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
MediaType media_type)
SHARED_LOCKS_REQUIRED(receive_crit_);
rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
size_t length,
const PacketTime& packet_time)
SHARED_LOCKS_REQUIRED(receive_crit_);
void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
void UpdateReceiveHistograms();
void UpdateHistograms();
void UpdateAggregateNetworkState();
Clock* const clock_;
const int num_cpu_cores_;
const std::unique_ptr<ProcessThread> module_process_thread_;
const std::unique_ptr<ProcessThread> pacer_thread_;
const std::unique_ptr<CallStats> call_stats_;
const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
Call::Config config_;
rtc::ThreadChecker configuration_thread_checker_;
NetworkState audio_network_state_;
NetworkState video_network_state_;
std::unique_ptr<RWLockWrapper> receive_crit_;
// Audio, Video, and FlexFEC receive streams are owned by the client that
// creates them.
std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
GUARDED_BY(receive_crit_);
std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
GUARDED_BY(receive_crit_);
std::set<VideoReceiveStream*> video_receive_streams_
GUARDED_BY(receive_crit_);
// Each media stream could conceivably be protected by multiple FlexFEC
// streams.
std::multimap<uint32_t, FlexfecReceiveStreamImpl*>
flexfec_receive_ssrcs_media_ GUARDED_BY(receive_crit_);
std::map<uint32_t, FlexfecReceiveStreamImpl*>
flexfec_receive_ssrcs_protection_ GUARDED_BY(receive_crit_);
std::set<FlexfecReceiveStreamImpl*> flexfec_receive_streams_
GUARDED_BY(receive_crit_);
std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
GUARDED_BY(receive_crit_);
// 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 {
ReceiveRtpConfig() = default; // Needed by std::map
ReceiveRtpConfig(const std::vector<RtpExtension>& extensions,
bool use_send_side_bwe)
: extensions(extensions), use_send_side_bwe(use_send_side_bwe) {}
// 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.
RtpHeaderExtensionMap extensions;
// Set if both RTP extension the RTCP feedback message needed for
// send side BWE are negotiated.
bool use_send_side_bwe = false;
};
std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
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_ GUARDED_BY(send_crit_);
std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
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.
int64_t first_packet_sent_ms_;
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_;
// 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_ GUARDED_BY(&bitrate_crit_);
uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
AvgCounter estimated_send_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
AvgCounter pacer_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
std::map<std::string, rtc::NetworkRoute> network_routes_;
std::unique_ptr<RtpTransportControllerSend> transport_send_;
VieRemb remb_;
ReceiveSideCongestionController receive_side_cc_;
const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
const int64_t start_ms_;
// TODO(perkj): |worker_queue_| is supposed to replace
// |module_process_thread_|.
// |worker_queue| is defined last to ensure all pending tasks are cancelled
// and deleted before any other members.
rtc::TaskQueue worker_queue_;
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
} // namespace internal
std::string Call::Stats::ToString(int64_t time_ms) const {
std::stringstream ss;
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, std::unique_ptr<RtpTransportControllerSend>(
new RtpTransportControllerSend(Clock::GetRealTimeClock(),
config.event_log)));
}
namespace internal {
Call::Call(const Call::Config& config,
std::unique_ptr<RtpTransportControllerSend> transport_send)
: clock_(Clock::GetRealTimeClock()),
num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
pacer_thread_(ProcessThread::Create("PacerThread")),
call_stats_(new CallStats(clock_)),
bitrate_allocator_(new BitrateAllocator(this)),
config_(config),
audio_network_state_(kNetworkDown),
video_network_state_(kNetworkDown),
receive_crit_(RWLockWrapper::CreateRWLock()),
send_crit_(RWLockWrapper::CreateRWLock()),
event_log_(config.event_log),
first_packet_sent_ms_(-1),
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),
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),
transport_send_(std::move(transport_send)),
remb_(clock_),
receive_side_cc_(clock_, &remb_, transport_send_->packet_router()),
video_send_delay_stats_(new SendDelayStats(clock_)),
start_ms_(clock_->TimeInMilliseconds()),
worker_queue_("call_worker_queue") {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK(config.event_log != nullptr);
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GT(config.bitrate_config.start_bitrate_bps,
config.bitrate_config.min_bitrate_bps);
if (config.bitrate_config.max_bitrate_bps != -1) {
RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
config.bitrate_config.start_bitrate_bps);
}
Trace::CreateTrace();
transport_send_->InitCongestionControl(this);
transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown);
transport_send_->send_side_cc()->SetBweBitrates(
config_.bitrate_config.min_bitrate_bps,
config_.bitrate_config.start_bitrate_bps,
config_.bitrate_config.max_bitrate_bps);
call_stats_->RegisterStatsObserver(&receive_side_cc_);
call_stats_->RegisterStatsObserver(transport_send_->send_side_cc());
module_process_thread_->Start();
module_process_thread_->RegisterModule(call_stats_.get(), RTC_FROM_HERE);
module_process_thread_->RegisterModule(&receive_side_cc_, RTC_FROM_HERE);
module_process_thread_->RegisterModule(transport_send_->send_side_cc(),
RTC_FROM_HERE);
pacer_thread_->RegisterModule(transport_send_->send_side_cc()->pacer(),
RTC_FROM_HERE);
pacer_thread_->RegisterModule(
receive_side_cc_.GetRemoteBitrateEstimator(true), RTC_FROM_HERE);
pacer_thread_->Start();
}
Call::~Call() {
RTC_DCHECK(!remb_.InUse());
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
RTC_CHECK(video_send_streams_.empty());
RTC_CHECK(audio_receive_ssrcs_.empty());
RTC_CHECK(video_receive_ssrcs_.empty());
RTC_CHECK(video_receive_streams_.empty());
pacer_thread_->Stop();
pacer_thread_->DeRegisterModule(transport_send_->send_side_cc()->pacer());
pacer_thread_->DeRegisterModule(
receive_side_cc_.GetRemoteBitrateEstimator(true));
module_process_thread_->DeRegisterModule(transport_send_->send_side_cc());
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_->send_side_cc());
// 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();
}
UpdateReceiveHistograms();
UpdateHistograms();
Trace::ReturnTrace();
}
rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
RtpPacketReceived parsed_packet;
if (!parsed_packet.Parse(packet, length))
return rtc::Optional<RtpPacketReceived>();
auto it = receive_rtp_config_.find(parsed_packet.Ssrc());
if (it != receive_rtp_config_.end())
parsed_packet.IdentifyExtensions(it->second.extensions);
int64_t arrival_time_ms;
if (packet_time.timestamp != -1) {
arrival_time_ms = (packet_time.timestamp + 500) / 1000;
} else {
arrival_time_ms = clock_->TimeInMilliseconds();
}
parsed_packet.set_arrival_time_ms(arrival_time_ms);
return rtc::Optional<RtpPacketReceived>(std::move(parsed_packet));
}
void Call::UpdateHistograms() {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.LifetimeInSeconds",
(clock_->TimeInMilliseconds() - start_ms_) / 1000);
}
void Call::UpdateSendHistograms() {
if (first_packet_sent_ms_ == -1)
return;
int64_t elapsed_sec =
(clock_->TimeInMilliseconds() - first_packet_sent_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);
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);
LOG(LS_INFO) << "WebRTC.Call.PacerBitrateInKbps, "
<< pacer_bitrate_stats.ToString();
}
}
void Call::UpdateReceiveHistograms() {
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);
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);
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);
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);
LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, "
<< recv_bytes_per_sec.ToStringWithMultiplier(8);
}
}
PacketReceiver* Call::Receiver() {
// TODO(solenberg): Some test cases in EndToEndTest use this from a different
// thread. Re-enable once that is fixed.
// RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
return this;
}
webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
event_log_->LogAudioSendStreamConfig(config);
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, &worker_queue_, transport_send_.get(),
bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());
{
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 (const auto& kv : audio_receive_ssrcs_) {
if (kv.second->config().rtp.local_ssrc == config.rtp.ssrc) {
kv.second->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(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK(send_stream != nullptr);
send_stream->Stop();
webrtc::internal::AudioSendStream* audio_send_stream =
static_cast<webrtc::internal::AudioSendStream*>(send_stream);
uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
{
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 (const auto& kv : audio_receive_ssrcs_) {
if (kv.second->config().rtp.local_ssrc == ssrc) {
kv.second->AssociateSendStream(nullptr);
}
}
}
UpdateAggregateNetworkState();
delete audio_send_stream;
}
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
event_log_->LogAudioReceiveStreamConfig(config);
AudioReceiveStream* receive_stream =
new AudioReceiveStream(transport_send_->packet_router(), config,
config_.audio_state, event_log_);
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
audio_receive_ssrcs_.end());
audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
receive_rtp_config_[config.rtp.remote_ssrc] =
ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
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(configuration_thread_checker_.CalledOnValidThread());
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);
size_t num_deleted = audio_receive_ssrcs_.erase(ssrc);
RTC_DCHECK(num_deleted == 1);
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;
}
webrtc::VideoSendStream* Call::CreateVideoSendStream(
webrtc::VideoSendStream::Config config,
VideoEncoderConfig encoder_config) {
TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
video_send_delay_stats_->AddSsrcs(config);
event_log_->LogVideoSendStreamConfig(config);
// 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;
VideoSendStream* send_stream = new VideoSendStream(
num_cpu_cores_, module_process_thread_.get(), &worker_queue_,
call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(),
video_send_delay_stats_.get(), &remb_, event_log_, std::move(config),
std::move(encoder_config), suspended_video_send_ssrcs_);
{
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);
}
send_stream->SignalNetworkState(video_network_state_);
UpdateAggregateNetworkState();
return send_stream;
}
void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
RTC_DCHECK(send_stream != nullptr);
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
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_state =
send_stream_impl->StopPermanentlyAndGetRtpStates();
for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin();
it != rtp_state.end(); ++it) {
suspended_video_send_ssrcs_[it->first] = it->second;
}
UpdateAggregateNetworkState();
delete send_stream_impl;
}
webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
webrtc::VideoReceiveStream::Config configuration) {
TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
VideoReceiveStream* receive_stream = new VideoReceiveStream(
num_cpu_cores_, transport_send_->packet_router(),
std::move(configuration), module_process_thread_.get(), call_stats_.get(),
&remb_);
const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
ReceiveRtpConfig receive_config(config.rtp.extensions,
UseSendSideBwe(config));
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
video_receive_ssrcs_.end());
video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
if (config.rtp.rtx_ssrc) {
video_receive_ssrcs_[config.rtp.rtx_ssrc] = receive_stream;
// 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_[config.rtp.rtx_ssrc] = receive_config;
}
receive_rtp_config_[config.rtp.remote_ssrc] = receive_config;
video_receive_streams_.insert(receive_stream);
ConfigureSync(config.sync_group);
}
receive_stream->SignalNetworkState(video_network_state_);
UpdateAggregateNetworkState();
event_log_->LogVideoReceiveStreamConfig(config);
return receive_stream;
}
void Call::DestroyVideoReceiveStream(
webrtc::VideoReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK(receive_stream != nullptr);
VideoReceiveStream* receive_stream_impl = nullptr;
{
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.
auto it = video_receive_ssrcs_.begin();
while (it != video_receive_ssrcs_.end()) {
if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
if (receive_stream_impl != nullptr)
RTC_DCHECK(receive_stream_impl == it->second);
receive_stream_impl = it->second;
receive_rtp_config_.erase(it->first);
it = video_receive_ssrcs_.erase(it);
} else {
++it;
}
}
video_receive_streams_.erase(receive_stream_impl);
RTC_CHECK(receive_stream_impl != nullptr);
ConfigureSync(receive_stream_impl->config().sync_group);
}
const VideoReceiveStream::Config& config = receive_stream_impl->config();
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(configuration_thread_checker_.CalledOnValidThread());
RecoveredPacketReceiver* recovered_packet_receiver = this;
FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl(
config, recovered_packet_receiver, call_stats_->rtcp_rtt_stats(),
module_process_thread_.get());
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(flexfec_receive_streams_.find(receive_stream) ==
flexfec_receive_streams_.end());
flexfec_receive_streams_.insert(receive_stream);
for (auto ssrc : config.protected_media_ssrcs)
flexfec_receive_ssrcs_media_.insert(std::make_pair(ssrc, receive_stream));
RTC_DCHECK(flexfec_receive_ssrcs_protection_.find(config.remote_ssrc) ==
flexfec_receive_ssrcs_protection_.end());
flexfec_receive_ssrcs_protection_[config.remote_ssrc] = receive_stream;
RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) ==
receive_rtp_config_.end());
receive_rtp_config_[config.remote_ssrc] =
ReceiveRtpConfig(config.rtp_header_extensions, UseSendSideBwe(config));
}
// TODO(brandtr): Store config in RtcEventLog here.
return receive_stream;
}
void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK(receive_stream != nullptr);
// There exist no other derived classes of FlexfecReceiveStream,
// so this downcast is safe.
FlexfecReceiveStreamImpl* receive_stream_impl =
static_cast<FlexfecReceiveStreamImpl*>(receive_stream);
{
WriteLockScoped write_lock(*receive_crit_);
const FlexfecReceiveStream::Config& config =
receive_stream_impl->GetConfig();
uint32_t ssrc = config.remote_ssrc;
receive_rtp_config_.erase(ssrc);
// Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
// destroyed.
auto prot_it = flexfec_receive_ssrcs_protection_.begin();
while (prot_it != flexfec_receive_ssrcs_protection_.end()) {
if (prot_it->second == receive_stream_impl)
prot_it = flexfec_receive_ssrcs_protection_.erase(prot_it);
else
++prot_it;
}
auto media_it = flexfec_receive_ssrcs_media_.begin();
while (media_it != flexfec_receive_ssrcs_media_.end()) {
if (media_it->second == receive_stream_impl)
media_it = flexfec_receive_ssrcs_media_.erase(media_it);
else
++media_it;
}
receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
->RemoveStream(ssrc);
flexfec_receive_streams_.erase(receive_stream_impl);
}
delete receive_stream_impl;
}
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(configuration_thread_checker_.CalledOnValidThread());
Stats stats;
// Fetch available send/receive bitrates.
uint32_t send_bandwidth = 0;
transport_send_->send_side_cc()->GetBitrateController()->AvailableBandwidth(
&send_bandwidth);
std::vector<unsigned int> ssrcs;
uint32_t recv_bandwidth = 0;
receive_side_cc_.GetRemoteBitrateEstimator(false)->LatestEstimate(
&ssrcs, &recv_bandwidth);
stats.send_bandwidth_bps = send_bandwidth;
stats.recv_bandwidth_bps = recv_bandwidth;
stats.pacer_delay_ms =
transport_send_->send_side_cc()->GetPacerQueuingDelayMs();
stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
{
rtc::CritScope cs(&bitrate_crit_);
stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_;
}
return stats;
}
void Call::SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) {
TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
if (bitrate_config.max_bitrate_bps != -1)
RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
if (config_.bitrate_config.min_bitrate_bps ==
bitrate_config.min_bitrate_bps &&
(bitrate_config.start_bitrate_bps <= 0 ||
config_.bitrate_config.start_bitrate_bps ==
bitrate_config.start_bitrate_bps) &&
config_.bitrate_config.max_bitrate_bps ==
bitrate_config.max_bitrate_bps) {
// Nothing new to set, early abort to avoid encoder reconfigurations.
return;
}
config_.bitrate_config.min_bitrate_bps = bitrate_config.min_bitrate_bps;
// Start bitrate of -1 means we should keep the old bitrate, which there is
// no point in remembering for the future.
if (bitrate_config.start_bitrate_bps > 0)
config_.bitrate_config.start_bitrate_bps = bitrate_config.start_bitrate_bps;
config_.bitrate_config.max_bitrate_bps = bitrate_config.max_bitrate_bps;
RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
transport_send_->send_side_cc()->SetBweBitrates(
bitrate_config.min_bitrate_bps, bitrate_config.start_bitrate_bps,
bitrate_config.max_bitrate_bps);
}
void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
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_);
}
for (auto& kv : video_send_ssrcs_) {
kv.second->SignalNetworkState(video_network_state_);
}
}
{
ReadLockScoped read_lock(*receive_crit_);
for (auto& kv : audio_receive_ssrcs_) {
kv.second->SignalNetworkState(audio_network_state_);
}
for (auto& kv : video_receive_ssrcs_) {
kv.second->SignalNetworkState(video_network_state_);
}
}
}
void Call::OnTransportOverheadChanged(MediaType media,
int transport_overhead_per_packet) {
switch (media) {
case MediaType::AUDIO: {
ReadLockScoped read_lock(*send_crit_);
for (auto& kv : audio_send_ssrcs_) {
kv.second->SetTransportOverhead(transport_overhead_per_packet);
}
break;
}
case MediaType::VIDEO: {
ReadLockScoped read_lock(*send_crit_);
for (auto& kv : video_send_ssrcs_) {
kv.second->SetTransportOverhead(transport_overhead_per_packet);
}
break;
}
case MediaType::ANY:
case MediaType::DATA:
RTC_NOTREACHED();
break;
}
}
// TODO(honghaiz): Add tests for this method.
void Call::OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
// Check if the network route is connected.
if (!network_route.connected) {
LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
// TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
// consider merging these two methods.
return;
}
// Check whether the network route has changed on each transport.
auto result =
network_routes_.insert(std::make_pair(transport_name, network_route));
auto kv = result.first;
bool inserted = result.second;
if (inserted) {
// No need to reset BWE if this is the first time the network connects.
return;
}
if (kv->second != network_route) {
kv->second = network_route;
LOG(LS_INFO) << "Network route changed on transport " << transport_name
<< ": new local network id " << network_route.local_network_id
<< " new remote network id " << network_route.remote_network_id
<< " Reset bitrates to min: "
<< config_.bitrate_config.min_bitrate_bps
<< " bps, start: " << config_.bitrate_config.start_bitrate_bps
<< " bps, max: " << config_.bitrate_config.start_bitrate_bps
<< " bps.";
RTC_DCHECK_GT(config_.bitrate_config.start_bitrate_bps, 0);
transport_send_->send_side_cc()->OnNetworkRouteChanged(
network_route, config_.bitrate_config.start_bitrate_bps,
config_.bitrate_config.min_bitrate_bps,
config_.bitrate_config.max_bitrate_bps);
}
}
void Call::UpdateAggregateNetworkState() {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
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_ssrcs_.size() > 0)
have_audio = true;
if (video_receive_ssrcs_.size() > 0)
have_video = true;
}
NetworkState aggregate_state = kNetworkDown;
if ((have_video && video_network_state_ == kNetworkUp) ||
(have_audio && audio_network_state_ == kNetworkUp)) {
aggregate_state = kNetworkUp;
}
LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="
<< (aggregate_state == kNetworkUp ? "up" : "down");
transport_send_->send_side_cc()->SignalNetworkState(aggregate_state);
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
if (first_packet_sent_ms_ == -1)
first_packet_sent_ms_ = clock_->TimeInMilliseconds();
video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
clock_->TimeInMilliseconds());
transport_send_->send_side_cc()->OnSentPacket(sent_packet);
}
void Call::OnNetworkChanged(uint32_t target_bitrate_bps,
uint8_t fraction_loss,
int64_t rtt_ms,
int64_t probing_interval_ms) {
// TODO(perkj): Consider making sure CongestionController operates on
// |worker_queue_|.
if (!worker_queue_.IsCurrent()) {
worker_queue_.PostTask(
[this, target_bitrate_bps, fraction_loss, rtt_ms, probing_interval_ms] {
OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt_ms,
probing_interval_ms);
});
return;
}
RTC_DCHECK_RUN_ON(&worker_queue_);
// For controlling the rate of feedback messages.
receive_side_cc_.OnBitrateChanged(target_bitrate_bps);
bitrate_allocator_->OnNetworkChanged(target_bitrate_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) {
transport_send_->send_side_cc()->SetAllocatedSendBitrateLimits(
min_send_bitrate_bps, max_padding_bitrate_bps);
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 (const auto& kv : audio_receive_ssrcs_) {
if (kv.second->config().sync_group == sync_group) {
if (sync_audio_stream != nullptr) {
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 = kv.second;
}
}
}
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
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 (auto& kv : audio_receive_ssrcs_) {
if (kv.second->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_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
}
PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
TRACE_EVENT0("webrtc", "Call::DeliverRtp");
RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO);
ReadLockScoped read_lock(*receive_crit_);
// TODO(nisse): We should parse the RTP header only here, and pass
// on parsed_packet to the receive streams.
rtc::Optional<RtpPacketReceived> parsed_packet =
ParseRtpPacket(packet, length, packet_time);
if (!parsed_packet)
return DELIVERY_PACKET_ERROR;
NotifyBweOfReceivedPacket(*parsed_packet, media_type);
uint32_t ssrc = parsed_packet->Ssrc();
if (media_type == MediaType::AUDIO) {
auto it = audio_receive_ssrcs_.find(ssrc);
if (it != audio_receive_ssrcs_.end()) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
it->second->OnRtpPacket(*parsed_packet);
event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
return DELIVERY_OK;
}
}
if (media_type == MediaType::VIDEO) {
auto it = video_receive_ssrcs_.find(ssrc);
if (it != video_receive_ssrcs_.end()) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
it->second->OnRtpPacket(*parsed_packet);
// Deliver media packets to FlexFEC subsystem.
auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc);
for (auto it = it_bounds.first; it != it_bounds.second; ++it)
it->second->OnRtpPacket(*parsed_packet);
event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
return DELIVERY_OK;
}
}
if (media_type == MediaType::VIDEO) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
// TODO(brandtr): Update here when FlexFEC supports protecting audio.
received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
auto it = flexfec_receive_ssrcs_protection_.find(ssrc);
if (it != flexfec_receive_ssrcs_protection_.end()) {
it->second->OnRtpPacket(*parsed_packet);
event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
return DELIVERY_OK;
}
}
return DELIVERY_UNKNOWN_SSRC;
}
PacketReceiver::DeliveryStatus Call::DeliverPacket(
MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
if (RtpHeaderParser::IsRtcp(packet, length))
return DeliverRtcp(media_type, packet, length);
return DeliverRtp(media_type, packet, length, packet_time);
}
// TODO(brandtr): Update this member function when we support protecting
// audio packets with FlexFEC.
bool Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
ReadLockScoped read_lock(*receive_crit_);
auto it = video_receive_ssrcs_.find(ssrc);
if (it == video_receive_ssrcs_.end())
return false;
return it->second->OnRecoveredPacket(packet, length);
}
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
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