Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.
BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1227923005 .
Cr-Commit-Position: refs/heads/master@{#10073}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
new file mode 100644
index 0000000..21109c2
--- /dev/null
+++ b/webrtc/call/call.cc
@@ -0,0 +1,549 @@
+/*
+ * 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 <map>
+#include <vector>
+
+#include "webrtc/audio/audio_receive_stream.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/call.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/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/system_wrappers/interface/cpu_info.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/video/video_receive_stream.h"
+#include "webrtc/video/video_send_stream.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+
+namespace webrtc {
+
+const int Call::Config::kDefaultStartBitrateBps = 300000;
+
+namespace internal {
+
+class Call : public webrtc::Call, public PacketReceiver {
+ public:
+ explicit Call(const Call::Config& config);
+ virtual ~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(
+ const webrtc::VideoSendStream::Config& config,
+ const VideoEncoderConfig& encoder_config) override;
+ void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
+
+ webrtc::VideoReceiveStream* CreateVideoReceiveStream(
+ const webrtc::VideoReceiveStream::Config& config) override;
+ void DestroyVideoReceiveStream(
+ webrtc::VideoReceiveStream* receive_stream) override;
+
+ Stats GetStats() const override;
+
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) override;
+
+ void SetBitrateConfig(
+ const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
+ void SignalNetworkState(NetworkState state) 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 SetBitrateControllerConfig(
+ const webrtc::Call::Config::BitrateConfig& bitrate_config);
+
+ void ConfigureSync(const std::string& sync_group)
+ EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
+
+ const int num_cpu_cores_;
+ const rtc::scoped_ptr<ProcessThread> module_process_thread_;
+ const rtc::scoped_ptr<ChannelGroup> channel_group_;
+ volatile int next_channel_id_;
+ Call::Config config_;
+
+ // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
+ // ensures that we have a consistent network state signalled to all senders
+ // and receivers.
+ rtc::CriticalSection network_enabled_crit_;
+ bool network_enabled_ GUARDED_BY(network_enabled_crit_);
+
+ rtc::scoped_ptr<RWLockWrapper> receive_crit_;
+ 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_);
+ std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
+ GUARDED_BY(receive_crit_);
+
+ rtc::scoped_ptr<RWLockWrapper> 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_;
+
+ RtcEventLog* event_log_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(Call);
+};
+} // namespace internal
+
+Call* Call::Create(const Call::Config& config) {
+ return new internal::Call(config);
+}
+
+namespace internal {
+
+Call::Call(const Call::Config& config)
+ : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
+ module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
+ channel_group_(new ChannelGroup(module_process_thread_.get())),
+ next_channel_id_(0),
+ config_(config),
+ network_enabled_(true),
+ receive_crit_(RWLockWrapper::CreateRWLock()),
+ send_crit_(RWLockWrapper::CreateRWLock()),
+ event_log_(nullptr) {
+ 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);
+ if (config.bitrate_config.max_bitrate_bps != -1) {
+ RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
+ config.bitrate_config.start_bitrate_bps);
+ }
+ if (config.voice_engine) {
+ VoECodec* voe_codec = VoECodec::GetInterface(config.voice_engine);
+ if (voe_codec) {
+ event_log_ = voe_codec->GetEventLog();
+ voe_codec->Release();
+ }
+ }
+
+ Trace::CreateTrace();
+ module_process_thread_->Start();
+
+ SetBitrateControllerConfig(config_.bitrate_config);
+}
+
+Call::~Call() {
+ RTC_CHECK_EQ(0u, video_send_ssrcs_.size());
+ RTC_CHECK_EQ(0u, video_send_streams_.size());
+ RTC_CHECK_EQ(0u, audio_receive_ssrcs_.size());
+ RTC_CHECK_EQ(0u, video_receive_ssrcs_.size());
+ RTC_CHECK_EQ(0u, video_receive_streams_.size());
+
+ module_process_thread_->Stop();
+ Trace::ReturnTrace();
+}
+
+PacketReceiver* Call::Receiver() { return this; }
+
+webrtc::AudioSendStream* Call::CreateAudioSendStream(
+ const webrtc::AudioSendStream::Config& config) {
+ return nullptr;
+}
+
+void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+}
+
+webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
+ const webrtc::AudioReceiveStream::Config& config) {
+ TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
+ LOG(LS_INFO) << "CreateAudioReceiveStream: " << config.ToString();
+ AudioReceiveStream* receive_stream = new AudioReceiveStream(
+ channel_group_->GetRemoteBitrateEstimator(), config);
+ {
+ 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;
+ ConfigureSync(config.sync_group);
+ }
+ return receive_stream;
+}
+
+void Call::DestroyAudioReceiveStream(
+ webrtc::AudioReceiveStream* receive_stream) {
+ TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
+ RTC_DCHECK(receive_stream != nullptr);
+ AudioReceiveStream* audio_receive_stream =
+ static_cast<AudioReceiveStream*>(receive_stream);
+ {
+ WriteLockScoped write_lock(*receive_crit_);
+ size_t num_deleted = audio_receive_ssrcs_.erase(
+ audio_receive_stream->config().rtp.remote_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);
+ }
+ }
+ delete audio_receive_stream;
+}
+
+webrtc::VideoSendStream* Call::CreateVideoSendStream(
+ const webrtc::VideoSendStream::Config& config,
+ const VideoEncoderConfig& encoder_config) {
+ TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
+ LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
+ RTC_DCHECK(!config.rtp.ssrcs.empty());
+
+ // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
+ // the call has already started.
+ VideoSendStream* send_stream = new VideoSendStream(num_cpu_cores_,
+ module_process_thread_.get(), channel_group_.get(),
+ rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config,
+ suspended_video_send_ssrcs_);
+
+ // This needs to be taken before send_crit_ as both locks need to be held
+ // while changing network state.
+ rtc::CritScope lock(&network_enabled_crit_);
+ WriteLockScoped write_lock(*send_crit_);
+ for (uint32_t ssrc : config.rtp.ssrcs) {
+ RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
+ video_send_ssrcs_[ssrc] = send_stream;
+ }
+ video_send_streams_.insert(send_stream);
+
+ if (event_log_)
+ event_log_->LogVideoSendStreamConfig(config);
+
+ if (!network_enabled_)
+ send_stream->SignalNetworkState(kNetworkDown);
+ return send_stream;
+}
+
+void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
+ TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
+ RTC_DCHECK(send_stream != nullptr);
+
+ 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->GetRtpStates();
+
+ for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin();
+ it != rtp_state.end();
+ ++it) {
+ suspended_video_send_ssrcs_[it->first] = it->second;
+ }
+
+ delete send_stream_impl;
+}
+
+webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
+ const webrtc::VideoReceiveStream::Config& config) {
+ TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
+ LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
+ VideoReceiveStream* receive_stream = new VideoReceiveStream(
+ num_cpu_cores_, channel_group_.get(),
+ rtc::AtomicOps::Increment(&next_channel_id_), config,
+ config_.voice_engine);
+
+ // This needs to be taken before receive_crit_ as both locks need to be held
+ // while changing network state.
+ rtc::CritScope lock(&network_enabled_crit_);
+ 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;
+ // TODO(pbos): Configure different RTX payloads per receive payload.
+ VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
+ config.rtp.rtx.begin();
+ if (it != config.rtp.rtx.end())
+ video_receive_ssrcs_[it->second.ssrc] = receive_stream;
+ video_receive_streams_.insert(receive_stream);
+
+ ConfigureSync(config.sync_group);
+
+ if (!network_enabled_)
+ receive_stream->SignalNetworkState(kNetworkDown);
+
+ if (event_log_)
+ event_log_->LogVideoReceiveStreamConfig(config);
+
+ return receive_stream;
+}
+
+void Call::DestroyVideoReceiveStream(
+ webrtc::VideoReceiveStream* receive_stream) {
+ TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
+ 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;
+ 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);
+ }
+ delete receive_stream_impl;
+}
+
+Call::Stats Call::GetStats() const {
+ Stats stats;
+ // Fetch available send/receive bitrates.
+ uint32_t send_bandwidth = 0;
+ channel_group_->GetBitrateController()->AvailableBandwidth(&send_bandwidth);
+ std::vector<unsigned int> ssrcs;
+ uint32_t recv_bandwidth = 0;
+ channel_group_->GetRemoteBitrateEstimator()->LatestEstimate(&ssrcs,
+ &recv_bandwidth);
+ stats.send_bandwidth_bps = send_bandwidth;
+ stats.recv_bandwidth_bps = recv_bandwidth;
+ stats.pacer_delay_ms = channel_group_->GetPacerQueuingDelayMs();
+ {
+ ReadLockScoped read_lock(*send_crit_);
+ for (const auto& kv : video_send_ssrcs_) {
+ int rtt_ms = kv.second->GetRtt();
+ if (rtt_ms > 0)
+ stats.rtt_ms = rtt_ms;
+ }
+ }
+ return stats;
+}
+
+void Call::SetBitrateConfig(
+ const webrtc::Call::Config::BitrateConfig& bitrate_config) {
+ TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
+ 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 = bitrate_config;
+ SetBitrateControllerConfig(bitrate_config);
+}
+
+void Call::SetBitrateControllerConfig(
+ const webrtc::Call::Config::BitrateConfig& bitrate_config) {
+ BitrateController* bitrate_controller =
+ channel_group_->GetBitrateController();
+ if (bitrate_config.start_bitrate_bps > 0)
+ bitrate_controller->SetStartBitrate(bitrate_config.start_bitrate_bps);
+ bitrate_controller->SetMinMaxBitrate(bitrate_config.min_bitrate_bps,
+ bitrate_config.max_bitrate_bps);
+}
+
+void Call::SignalNetworkState(NetworkState state) {
+ // Take crit for entire function, it needs to be held while updating streams
+ // to guarantee a consistent state across streams.
+ rtc::CritScope lock(&network_enabled_crit_);
+ network_enabled_ = state == kNetworkUp;
+ {
+ ReadLockScoped write_lock(*send_crit_);
+ for (auto& kv : video_send_ssrcs_) {
+ kv.second->SignalNetworkState(state);
+ }
+ }
+ {
+ ReadLockScoped write_lock(*receive_crit_);
+ for (auto& kv : video_receive_ssrcs_) {
+ kv.second->SignalNetworkState(state);
+ }
+ }
+}
+
+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())
+ 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 (sync_audio_stream != nullptr && num_synced_streams == 1) {
+ video_stream->SetSyncChannel(config_.voice_engine,
+ sync_audio_stream->config().voe_channel_id);
+ } else {
+ video_stream->SetSyncChannel(config_.voice_engine, -1);
+ }
+ }
+}
+
+PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
+ const uint8_t* packet,
+ size_t length) {
+ // 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.
+ 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 (event_log_)
+ event_log_->LogRtcpPacket(true, media_type, packet, length);
+ }
+ }
+ }
+ 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 (event_log_)
+ event_log_->LogRtcpPacket(false, 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) {
+ // Minimum RTP header size.
+ if (length < 12)
+ return DELIVERY_PACKET_ERROR;
+
+ 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()) {
+ auto status = it->second->DeliverRtp(packet, length, packet_time)
+ ? DELIVERY_OK
+ : DELIVERY_PACKET_ERROR;
+ if (status == DELIVERY_OK && event_log_)
+ event_log_->LogRtpHeader(true, media_type, packet, length);
+ return status;
+ }
+ }
+ if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
+ auto it = video_receive_ssrcs_.find(ssrc);
+ if (it != video_receive_ssrcs_.end()) {
+ auto status = it->second->DeliverRtp(packet, length, packet_time)
+ ? DELIVERY_OK
+ : DELIVERY_PACKET_ERROR;
+ if (status == DELIVERY_OK && event_log_)
+ event_log_->LogRtpHeader(true, media_type, packet, length);
+ return status;
+ }
+ }
+ return DELIVERY_UNKNOWN_SSRC;
+}
+
+PacketReceiver::DeliveryStatus Call::DeliverPacket(
+ MediaType media_type,
+ const uint8_t* packet,
+ size_t length,
+ const PacketTime& packet_time) {
+ if (RtpHeaderParser::IsRtcp(packet, length))
+ return DeliverRtcp(media_type, packet, length);
+
+ return DeliverRtp(media_type, packet, length, packet_time);
+}
+
+} // namespace internal
+} // namespace webrtc
