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-rw-r--r--webrtc/call/call.cc209
1 files changed, 177 insertions, 32 deletions
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 594ddf5c97..5c46a48f14 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -15,27 +15,33 @@
#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/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/call.h"
+#include "webrtc/call/bitrate_allocator.h"
#include "webrtc/call/congestion_controller.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/config.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.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/video_receive_stream.h"
#include "webrtc/video/video_send_stream.h"
-#include "webrtc/video_engine/call_stats.h"
#include "webrtc/voice_engine/include/voe_codec.h"
namespace webrtc {
@@ -44,7 +50,8 @@ const int Call::Config::kDefaultStartBitrateBps = 300000;
namespace internal {
-class Call : public webrtc::Call, public PacketReceiver {
+class Call : public webrtc::Call, public PacketReceiver,
+ public BitrateObserver {
public:
explicit Call(const Call::Config& config);
virtual ~Call();
@@ -83,6 +90,10 @@ class Call : public webrtc::Call, public PacketReceiver {
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
+ // Implements BitrateObserver.
+ void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss,
+ int64_t rtt_ms) override;
+
private:
DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
size_t length);
@@ -94,14 +105,28 @@ class Call : public webrtc::Call, public PacketReceiver {
void ConfigureSync(const std::string& sync_group)
EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
+ VoiceEngine* voice_engine() {
+ internal::AudioState* audio_state =
+ static_cast<internal::AudioState*>(config_.audio_state.get());
+ if (audio_state)
+ return audio_state->voice_engine();
+ else
+ return nullptr;
+ }
+
+ void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
+ void UpdateReceiveHistograms();
+
+ Clock* const clock_;
+
const int num_cpu_cores_;
const rtc::scoped_ptr<ProcessThread> module_process_thread_;
const rtc::scoped_ptr<CallStats> call_stats_;
- const rtc::scoped_ptr<CongestionController> congestion_controller_;
+ const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
Call::Config config_;
rtc::ThreadChecker configuration_thread_checker_;
- bool network_enabled_;
+ bool network_enabled_;
rtc::scoped_ptr<RWLockWrapper> receive_crit_;
// Audio and Video receive streams are owned by the client that creates them.
@@ -123,7 +148,25 @@ class Call : public webrtc::Call, public PacketReceiver {
VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
RtcEventLog* event_log_ = nullptr;
- VoECodec* voe_codec_ = nullptr;
+
+ // The following members are only accessed (exclusively) from one thread and
+ // from the destructor, and therefore doesn't need any explicit
+ // synchronization.
+ int64_t received_video_bytes_;
+ int64_t received_audio_bytes_;
+ int64_t received_rtcp_bytes_;
+ int64_t first_rtp_packet_received_ms_;
+ int64_t last_rtp_packet_received_ms_;
+ int64_t first_packet_sent_ms_;
+
+ // TODO(holmer): Remove this lock once BitrateController no longer calls
+ // OnNetworkChanged from multiple threads.
+ rtc::CriticalSection bitrate_crit_;
+ int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
+ int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
+ int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
+
+ const rtc::scoped_ptr<CongestionController> congestion_controller_;
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
@@ -136,15 +179,29 @@ Call* Call::Create(const Call::Config& config) {
namespace internal {
Call::Call(const Call::Config& config)
- : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
+ : clock_(Clock::GetRealTimeClock()),
+ num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
- call_stats_(new CallStats()),
- congestion_controller_(new CongestionController(
- module_process_thread_.get(), call_stats_.get())),
+ call_stats_(new CallStats(clock_)),
+ bitrate_allocator_(new BitrateAllocator()),
config_(config),
network_enabled_(true),
receive_crit_(RWLockWrapper::CreateRWLock()),
- send_crit_(RWLockWrapper::CreateRWLock()) {
+ send_crit_(RWLockWrapper::CreateRWLock()),
+ received_video_bytes_(0),
+ received_audio_bytes_(0),
+ received_rtcp_bytes_(0),
+ first_rtp_packet_received_ms_(-1),
+ last_rtp_packet_received_ms_(-1),
+ first_packet_sent_ms_(-1),
+ estimated_send_bitrate_sum_kbits_(0),
+ pacer_bitrate_sum_kbits_(0),
+ num_bitrate_updates_(0),
+ congestion_controller_(
+ new CongestionController(module_process_thread_.get(),
+ call_stats_.get(),
+ this)) {
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
config.bitrate_config.min_bitrate_bps);
@@ -152,12 +209,9 @@ Call::Call(const Call::Config& config)
RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
config.bitrate_config.start_bitrate_bps);
}
- if (config.voice_engine) {
- // Keep a reference to VoECodec, so we're sure the VoiceEngine lives for the
- // duration of the call.
- voe_codec_ = VoECodec::GetInterface(config.voice_engine);
- if (voe_codec_)
- event_log_ = voe_codec_->GetEventLog();
+ if (config.audio_state.get()) {
+ ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
+ event_log_ = voe_codec->GetEventLog();
}
Trace::CreateTrace();
@@ -168,10 +222,14 @@ Call::Call(const Call::Config& config)
config_.bitrate_config.min_bitrate_bps,
config_.bitrate_config.start_bitrate_bps,
config_.bitrate_config.max_bitrate_bps);
+
+ congestion_controller_->GetBitrateController()->SetEventLog(event_log_);
}
Call::~Call() {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+ UpdateSendHistograms();
+ UpdateReceiveHistograms();
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
RTC_CHECK(video_send_streams_.empty());
@@ -182,9 +240,53 @@ Call::~Call() {
module_process_thread_->DeRegisterModule(call_stats_.get());
module_process_thread_->Stop();
Trace::ReturnTrace();
+}
+
+void Call::UpdateSendHistograms() {
+ if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
+ return;
+ int64_t elapsed_sec =
+ (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+ return;
+ int send_bitrate_kbps =
+ estimated_send_bitrate_sum_kbits_ / num_bitrate_updates_;
+ int pacer_bitrate_kbps = pacer_bitrate_sum_kbits_ / num_bitrate_updates_;
+ if (send_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
+ send_bitrate_kbps);
+ }
+ if (pacer_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.PacerBitrateInKbps",
+ pacer_bitrate_kbps);
+ }
+}
- if (voe_codec_)
- voe_codec_->Release();
+void Call::UpdateReceiveHistograms() {
+ if (first_rtp_packet_received_ms_ == -1)
+ return;
+ int64_t elapsed_sec =
+ (last_rtp_packet_received_ms_ - first_rtp_packet_received_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+ return;
+ int audio_bitrate_kbps = received_audio_bytes_ * 8 / elapsed_sec / 1000;
+ int video_bitrate_kbps = received_video_bytes_ * 8 / elapsed_sec / 1000;
+ int rtcp_bitrate_bps = received_rtcp_bytes_ * 8 / elapsed_sec;
+ if (video_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
+ video_bitrate_kbps);
+ }
+ if (audio_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
+ audio_bitrate_kbps);
+ }
+ if (rtcp_bitrate_bps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
+ rtcp_bitrate_bps);
+ }
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+ "WebRTC.Call.BitrateReceivedInKbps",
+ audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
}
PacketReceiver* Call::Receiver() {
@@ -198,8 +300,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
- AudioSendStream* send_stream =
- new AudioSendStream(config, config_.voice_engine);
+ AudioSendStream* send_stream = new AudioSendStream(
+ config, config_.audio_state, congestion_controller_.get());
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
{
@@ -234,8 +336,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioReceiveStream* receive_stream = new AudioReceiveStream(
- congestion_controller_->GetRemoteBitrateEstimator(false), config,
- config_.voice_engine);
+ congestion_controller_.get(), config, config_.audio_state);
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -279,8 +380,8 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
// the call has already started.
VideoSendStream* send_stream = new VideoSendStream(
num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
- congestion_controller_.get(), config, encoder_config,
- suspended_video_send_ssrcs_);
+ congestion_controller_.get(), bitrate_allocator_.get(), config,
+ encoder_config, suspended_video_send_ssrcs_);
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
@@ -338,7 +439,7 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
VideoReceiveStream* receive_stream = new VideoReceiveStream(
num_cpu_cores_, congestion_controller_.get(), config,
- config_.voice_engine, module_process_thread_.get(), call_stats_.get());
+ voice_engine(), module_process_thread_.get(), call_stats_.get());
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -463,12 +564,48 @@ void Call::SignalNetworkState(NetworkState state) {
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+ if (first_packet_sent_ms_ == -1)
+ first_packet_sent_ms_ = clock_->TimeInMilliseconds();
congestion_controller_->OnSentPacket(sent_packet);
}
+void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
+ int64_t rtt_ms) {
+ uint32_t allocated_bitrate_bps = bitrate_allocator_->OnNetworkChanged(
+ target_bitrate_bps, fraction_loss, rtt_ms);
+
+ int pad_up_to_bitrate_bps = 0;
+ {
+ ReadLockScoped read_lock(*send_crit_);
+ // No need to update as long as we're not sending.
+ if (video_send_streams_.empty())
+ return;
+
+ for (VideoSendStream* stream : video_send_streams_)
+ pad_up_to_bitrate_bps += stream->GetPaddingNeededBps();
+ }
+ // Allocated bitrate might be higher than bitrate estimate if enforcing min
+ // bitrate, or lower if estimate is higher than the sum of max bitrates, so
+ // set the pacer bitrate to the maximum of the two.
+ uint32_t pacer_bitrate_bps =
+ std::max(target_bitrate_bps, allocated_bitrate_bps);
+ {
+ rtc::CritScope lock(&bitrate_crit_);
+ // We only update these stats if we have send streams, and assume that
+ // OnNetworkChanged is called roughly with a fixed frequency.
+ estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000;
+ pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
+ ++num_bitrate_updates_;
+ }
+ congestion_controller_->UpdatePacerBitrate(
+ target_bitrate_bps / 1000,
+ PacedSender::kDefaultPaceMultiplier * pacer_bitrate_bps / 1000,
+ pad_up_to_bitrate_bps / 1000);
+}
+
void Call::ConfigureSync(const std::string& sync_group) {
// Set sync only if there was no previous one.
- if (config_.voice_engine == nullptr || sync_group.empty())
+ if (voice_engine() == nullptr || sync_group.empty())
return;
AudioReceiveStream* sync_audio_stream = nullptr;
@@ -506,10 +643,10 @@ void Call::ConfigureSync(const std::string& sync_group) {
}
// Only sync the first A/V pair within this sync group.
if (sync_audio_stream != nullptr && num_synced_streams == 1) {
- video_stream->SetSyncChannel(config_.voice_engine,
+ video_stream->SetSyncChannel(voice_engine(),
sync_audio_stream->config().voe_channel_id);
} else {
- video_stream->SetSyncChannel(config_.voice_engine, -1);
+ video_stream->SetSyncChannel(voice_engine(), -1);
}
}
}
@@ -517,10 +654,12 @@ void Call::ConfigureSync(const std::string& sync_group) {
PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
const uint8_t* packet,
size_t length) {
+ TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
// TODO(pbos): Figure out what channel needs it actually.
// Do NOT broadcast! Also make sure it's a valid packet.
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that
// there's no receiver of the packet.
+ received_rtcp_bytes_ += length;
bool rtcp_delivered = false;
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
ReadLockScoped read_lock(*receive_crit_);
@@ -549,16 +688,21 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
+ TRACE_EVENT0("webrtc", "Call::DeliverRtp");
// Minimum RTP header size.
if (length < 12)
return DELIVERY_PACKET_ERROR;
- uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
+ last_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
+ if (first_rtp_packet_received_ms_ == -1)
+ first_rtp_packet_received_ms_ = last_rtp_packet_received_ms_;
+ uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
ReadLockScoped read_lock(*receive_crit_);
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
auto it = audio_receive_ssrcs_.find(ssrc);
if (it != audio_receive_ssrcs_.end()) {
+ received_audio_bytes_ += length;
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
@@ -570,6 +714,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
auto it = video_receive_ssrcs_.find(ssrc);
if (it != video_receive_ssrcs_.end()) {
+ received_video_bytes_ += length;
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-19 22:25:39 +0000