| Index: webrtc/video/video_send_stream.cc
|
| diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
|
| index 927a978aad77bf053bb59f1819bf0151184e9975..9a4c106e285a6713618a7e6a52a8a89baf320822 100644
|
| --- a/webrtc/video/video_send_stream.cc
|
| +++ b/webrtc/video/video_send_stream.cc
|
| @@ -24,10 +24,7 @@
|
| #include "webrtc/modules/pacing/packet_router.h"
|
| #include "webrtc/modules/utility/include/process_thread.h"
|
| #include "webrtc/video/call_stats.h"
|
| -#include "webrtc/video/encoder_state_feedback.h"
|
| #include "webrtc/video/video_capture_input.h"
|
| -#include "webrtc/video/vie_channel.h"
|
| -#include "webrtc/video/vie_encoder.h"
|
| #include "webrtc/video/vie_remb.h"
|
| #include "webrtc/video_send_stream.h"
|
|
|
| @@ -148,8 +145,33 @@ VideoSendStream::VideoSendStream(
|
| this,
|
| config.post_encode_callback,
|
| &stats_proxy_),
|
| - encoder_feedback_(new EncoderStateFeedback()),
|
| - use_config_bitrate_(true) {
|
| + vie_encoder_(num_cpu_cores,
|
| + module_process_thread_,
|
| + &stats_proxy_,
|
| + config.pre_encode_callback,
|
| + &overuse_detector_,
|
| + congestion_controller_->pacer(),
|
| + &payload_router_,
|
| + bitrate_allocator),
|
| + vcm_(vie_encoder_.vcm()),
|
| + vie_channel_(config.send_transport,
|
| + module_process_thread_,
|
| + &payload_router_,
|
| + nullptr,
|
| + encoder_feedback_.GetRtcpIntraFrameObserver(),
|
| + congestion_controller_->GetBitrateController()
|
| + ->CreateRtcpBandwidthObserver(),
|
| + congestion_controller_->GetTransportFeedbackObserver(),
|
| + nullptr,
|
| + call_stats_->rtcp_rtt_stats(),
|
| + congestion_controller_->pacer(),
|
| + congestion_controller_->packet_router(),
|
| + config_.rtp.ssrcs.size(),
|
| + true),
|
| + input_(&vie_encoder_,
|
| + config_.local_renderer,
|
| + &stats_proxy_,
|
| + &overuse_detector_) {
|
| LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
|
|
|
| RTC_DCHECK(!config_.rtp.ssrcs.empty());
|
| @@ -158,42 +180,14 @@ VideoSendStream::VideoSendStream(
|
| RTC_DCHECK(congestion_controller_);
|
| RTC_DCHECK(remb_);
|
|
|
| - // Set up Call-wide sequence numbers, if configured for this send stream.
|
| - TransportFeedbackObserver* transport_feedback_observer = nullptr;
|
| - for (const RtpExtension& extension : config.rtp.extensions) {
|
| - if (extension.name == RtpExtension::kTransportSequenceNumber) {
|
| - transport_feedback_observer =
|
| - congestion_controller_->GetTransportFeedbackObserver();
|
| - break;
|
| - }
|
| - }
|
| -
|
| - const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs;
|
| -
|
| - vie_encoder_.reset(new ViEEncoder(
|
| - num_cpu_cores, module_process_thread_, &stats_proxy_,
|
| - config.pre_encode_callback, &overuse_detector_,
|
| - congestion_controller_->pacer(), &payload_router_, bitrate_allocator));
|
| - vcm_ = vie_encoder_->vcm();
|
| - RTC_CHECK(vie_encoder_->Init());
|
| + RTC_CHECK(vie_encoder_.Init());
|
| + RTC_CHECK(vie_channel_.Init() == 0);
|
|
|
| - vie_channel_.reset(new ViEChannel(
|
| - config.send_transport, module_process_thread_, &payload_router_, nullptr,
|
| - encoder_feedback_->GetRtcpIntraFrameObserver(),
|
| - congestion_controller_->GetBitrateController()
|
| - ->CreateRtcpBandwidthObserver(),
|
| - transport_feedback_observer,
|
| - congestion_controller_->GetRemoteBitrateEstimator(false),
|
| - call_stats_->rtcp_rtt_stats(), congestion_controller_->pacer(),
|
| - congestion_controller_->packet_router(), ssrcs.size(), true));
|
| - RTC_CHECK(vie_channel_->Init() == 0);
|
| + vcm_->RegisterProtectionCallback(vie_channel_.vcm_protection_callback());
|
|
|
| - vcm_->RegisterProtectionCallback(vie_channel_->vcm_protection_callback());
|
| + call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
|
|
|
| - call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver());
|
| -
|
| - std::vector<uint32_t> first_ssrc(1, ssrcs[0]);
|
| - vie_encoder_->SetSsrcs(first_ssrc);
|
| + vie_encoder_.SetSsrcs(std::vector<uint32_t>(1, config_.rtp.ssrcs[0]));
|
|
|
| for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
|
| const std::string& extension = config_.rtp.extensions[i].name;
|
| @@ -202,19 +196,19 @@ VideoSendStream::VideoSendStream(
|
| RTC_DCHECK_GE(id, 1);
|
| RTC_DCHECK_LE(id, 14);
|
| if (extension == RtpExtension::kTOffset) {
|
| - RTC_CHECK_EQ(0, vie_channel_->SetSendTimestampOffsetStatus(true, id));
|
| + RTC_CHECK_EQ(0, vie_channel_.SetSendTimestampOffsetStatus(true, id));
|
| } else if (extension == RtpExtension::kAbsSendTime) {
|
| - RTC_CHECK_EQ(0, vie_channel_->SetSendAbsoluteSendTimeStatus(true, id));
|
| + RTC_CHECK_EQ(0, vie_channel_.SetSendAbsoluteSendTimeStatus(true, id));
|
| } else if (extension == RtpExtension::kVideoRotation) {
|
| - RTC_CHECK_EQ(0, vie_channel_->SetSendVideoRotationStatus(true, id));
|
| + RTC_CHECK_EQ(0, vie_channel_.SetSendVideoRotationStatus(true, id));
|
| } else if (extension == RtpExtension::kTransportSequenceNumber) {
|
| - RTC_CHECK_EQ(0, vie_channel_->SetSendTransportSequenceNumber(true, id));
|
| + RTC_CHECK_EQ(0, vie_channel_.SetSendTransportSequenceNumber(true, id));
|
| } else {
|
| RTC_NOTREACHED() << "Registering unsupported RTP extension.";
|
| }
|
| }
|
|
|
| - RtpRtcp* rtp_module = vie_channel_->rtp_rtcp();
|
| + RtpRtcp* rtp_module = vie_channel_.rtp_rtcp();
|
| remb_->AddRembSender(rtp_module);
|
| rtp_module->SetREMBStatus(true);
|
|
|
| @@ -222,49 +216,45 @@ VideoSendStream::VideoSendStream(
|
| const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
|
| const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
|
| // TODO(changbin): Should set RTX for RED mapping in RTP sender in future.
|
| - vie_channel_->SetProtectionMode(enable_protection_nack, enable_protection_fec,
|
| + vie_channel_.SetProtectionMode(enable_protection_nack, enable_protection_fec,
|
| config_.rtp.fec.red_payload_type,
|
| config_.rtp.fec.ulpfec_payload_type);
|
| - vie_encoder_->SetProtectionMethod(enable_protection_nack,
|
| + vie_encoder_.SetProtectionMethod(enable_protection_nack,
|
| enable_protection_fec);
|
|
|
| ConfigureSsrcs();
|
|
|
| - vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str());
|
| -
|
| - input_.reset(new internal::VideoCaptureInput(
|
| - vie_encoder_.get(), config_.local_renderer, &stats_proxy_,
|
| - &overuse_detector_));
|
| + vie_channel_.SetRTCPCName(config_.rtp.c_name.c_str());
|
|
|
| // 28 to match packet overhead in ModuleRtpRtcpImpl.
|
| RTC_DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28));
|
| - vie_channel_->SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28));
|
| + vie_channel_.SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28));
|
|
|
| RTC_DCHECK(config.encoder_settings.encoder != nullptr);
|
| RTC_DCHECK_GE(config.encoder_settings.payload_type, 0);
|
| RTC_DCHECK_LE(config.encoder_settings.payload_type, 127);
|
| - RTC_CHECK_EQ(0, vie_encoder_->RegisterExternalEncoder(
|
| + RTC_CHECK_EQ(0, vie_encoder_.RegisterExternalEncoder(
|
| config.encoder_settings.encoder,
|
| config.encoder_settings.payload_type,
|
| config.encoder_settings.internal_source));
|
|
|
| RTC_CHECK(ReconfigureVideoEncoder(encoder_config));
|
|
|
| - vie_channel_->RegisterSendSideDelayObserver(&stats_proxy_);
|
| + vie_channel_.RegisterSendSideDelayObserver(&stats_proxy_);
|
|
|
| if (config_.post_encode_callback)
|
| - vie_encoder_->RegisterPostEncodeImageCallback(&encoded_frame_proxy_);
|
| + vie_encoder_.RegisterPostEncodeImageCallback(&encoded_frame_proxy_);
|
|
|
| if (config_.suspend_below_min_bitrate)
|
| - vie_encoder_->SuspendBelowMinBitrate();
|
| + vie_encoder_.SuspendBelowMinBitrate();
|
|
|
| - encoder_feedback_->AddEncoder(ssrcs, vie_encoder_.get());
|
| + encoder_feedback_.AddEncoder(config_.rtp.ssrcs, &vie_encoder_);
|
|
|
| - vie_channel_->RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_);
|
| - vie_channel_->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_);
|
| - vie_channel_->RegisterRtcpPacketTypeCounterObserver(&stats_proxy_);
|
| - vie_channel_->RegisterSendBitrateObserver(&stats_proxy_);
|
| - vie_channel_->RegisterSendFrameCountObserver(&stats_proxy_);
|
| + vie_channel_.RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_);
|
| + vie_channel_.RegisterSendChannelRtpStatisticsCallback(&stats_proxy_);
|
| + vie_channel_.RegisterRtcpPacketTypeCounterObserver(&stats_proxy_);
|
| + vie_channel_.RegisterSendBitrateObserver(&stats_proxy_);
|
| + vie_channel_.RegisterSendFrameCountObserver(&stats_proxy_);
|
|
|
| module_process_thread_->RegisterModule(&overuse_detector_);
|
| }
|
| @@ -276,53 +266,49 @@ VideoSendStream::~VideoSendStream() {
|
| // ViEChannel. vcm_ is owned by ViEEncoder and the registered callback does
|
| // not outlive it.
|
| vcm_->RegisterProtectionCallback(nullptr);
|
| - vie_channel_->RegisterSendFrameCountObserver(nullptr);
|
| - vie_channel_->RegisterSendBitrateObserver(nullptr);
|
| - vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr);
|
| - vie_channel_->RegisterSendChannelRtpStatisticsCallback(nullptr);
|
| - vie_channel_->RegisterSendChannelRtcpStatisticsCallback(nullptr);
|
| + vie_channel_.RegisterSendFrameCountObserver(nullptr);
|
| + vie_channel_.RegisterSendBitrateObserver(nullptr);
|
| + vie_channel_.RegisterRtcpPacketTypeCounterObserver(nullptr);
|
| + vie_channel_.RegisterSendChannelRtpStatisticsCallback(nullptr);
|
| + vie_channel_.RegisterSendChannelRtcpStatisticsCallback(nullptr);
|
|
|
| - // Remove capture input (thread) so that it's not running after the current
|
| - // channel is deleted.
|
| - input_.reset();
|
| -
|
| - vie_encoder_->DeRegisterExternalEncoder(
|
| + vie_encoder_.DeRegisterExternalEncoder(
|
| config_.encoder_settings.payload_type);
|
|
|
| - call_stats_->DeregisterStatsObserver(vie_channel_->GetStatsObserver());
|
| + call_stats_->DeregisterStatsObserver(vie_channel_.GetStatsObserver());
|
|
|
| - RtpRtcp* rtp_module = vie_channel_->rtp_rtcp();
|
| + RtpRtcp* rtp_module = vie_channel_.rtp_rtcp();
|
| rtp_module->SetREMBStatus(false);
|
| remb_->RemoveRembSender(rtp_module);
|
|
|
| // Remove the feedback, stop all encoding threads and processing. This must be
|
| // done before deleting the channel.
|
| - encoder_feedback_->RemoveEncoder(vie_encoder_.get());
|
| + encoder_feedback_.RemoveEncoder(&vie_encoder_);
|
|
|
| - uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC();
|
| + uint32_t remote_ssrc = vie_channel_.GetRemoteSSRC();
|
| congestion_controller_->GetRemoteBitrateEstimator(false)->RemoveStream(
|
| remote_ssrc);
|
| }
|
|
|
| VideoCaptureInput* VideoSendStream::Input() {
|
| - return input_.get();
|
| + return &input_;
|
| }
|
|
|
| void VideoSendStream::Start() {
|
| transport_adapter_.Enable();
|
| - vie_encoder_->Pause();
|
| - if (vie_channel_->StartSend() == 0) {
|
| + vie_encoder_.Pause();
|
| + if (vie_channel_.StartSend() == 0) {
|
| // Was not already started, trigger a keyframe.
|
| - vie_encoder_->SendKeyFrame();
|
| + vie_encoder_.SendKeyFrame();
|
| }
|
| - vie_encoder_->Restart();
|
| - vie_channel_->StartReceive();
|
| + vie_encoder_.Restart();
|
| + vie_channel_.StartReceive();
|
| }
|
|
|
| void VideoSendStream::Stop() {
|
| // TODO(pbos): Make sure the encoder stops here.
|
| - vie_channel_->StopSend();
|
| - vie_channel_->StopReceive();
|
| + vie_channel_.StopSend();
|
| + vie_channel_.StopReceive();
|
| transport_adapter_.Disable();
|
| }
|
|
|
| @@ -472,15 +458,14 @@ bool VideoSendStream::ReconfigureVideoEncoder(
|
| stats_proxy_.SetContentType(config.content_type);
|
|
|
| RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
|
| - vie_encoder_->SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
|
| + vie_encoder_.SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
|
|
|
| encoder_config_ = config;
|
| - use_config_bitrate_ = false;
|
| return true;
|
| }
|
|
|
| bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
|
| - return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
|
| + return vie_channel_.ReceivedRTCPPacket(packet, length) == 0;
|
| }
|
|
|
| VideoSendStream::Stats VideoSendStream::GetStats() {
|
| @@ -498,14 +483,14 @@ void VideoSendStream::NormalUsage() {
|
| }
|
|
|
| void VideoSendStream::ConfigureSsrcs() {
|
| - vie_channel_->SetSSRC(config_.rtp.ssrcs.front(), kViEStreamTypeNormal, 0);
|
| + vie_channel_.SetSSRC(config_.rtp.ssrcs.front(), kViEStreamTypeNormal, 0);
|
| for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
|
| uint32_t ssrc = config_.rtp.ssrcs[i];
|
| - vie_channel_->SetSSRC(ssrc, kViEStreamTypeNormal,
|
| + vie_channel_.SetSSRC(ssrc, kViEStreamTypeNormal,
|
| static_cast<unsigned char>(i));
|
| RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
|
| if (it != suspended_ssrcs_.end())
|
| - vie_channel_->SetRtpStateForSsrc(ssrc, it->second);
|
| + vie_channel_.SetRtpStateForSsrc(ssrc, it->second);
|
| }
|
|
|
| if (config_.rtp.rtx.ssrcs.empty()) {
|
| @@ -516,19 +501,19 @@ void VideoSendStream::ConfigureSsrcs() {
|
| RTC_DCHECK_EQ(config_.rtp.rtx.ssrcs.size(), config_.rtp.ssrcs.size());
|
| for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
|
| uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
|
| - vie_channel_->SetSSRC(config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx,
|
| + vie_channel_.SetSSRC(config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx,
|
| static_cast<unsigned char>(i));
|
| RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
|
| if (it != suspended_ssrcs_.end())
|
| - vie_channel_->SetRtpStateForSsrc(ssrc, it->second);
|
| + vie_channel_.SetRtpStateForSsrc(ssrc, it->second);
|
| }
|
|
|
| RTC_DCHECK_GE(config_.rtp.rtx.payload_type, 0);
|
| - vie_channel_->SetRtxSendPayloadType(config_.rtp.rtx.payload_type,
|
| + vie_channel_.SetRtxSendPayloadType(config_.rtp.rtx.payload_type,
|
| config_.encoder_settings.payload_type);
|
| if (config_.rtp.fec.red_payload_type != -1 &&
|
| config_.rtp.fec.red_rtx_payload_type != -1) {
|
| - vie_channel_->SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type,
|
| + vie_channel_.SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type,
|
| config_.rtp.fec.red_payload_type);
|
| }
|
| }
|
| @@ -537,12 +522,12 @@ std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
|
| std::map<uint32_t, RtpState> rtp_states;
|
| for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
|
| uint32_t ssrc = config_.rtp.ssrcs[i];
|
| - rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
|
| + rtp_states[ssrc] = vie_channel_.GetRtpStateForSsrc(ssrc);
|
| }
|
|
|
| for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
|
| uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
|
| - rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
|
| + rtp_states[ssrc] = vie_channel_.GetRtpStateForSsrc(ssrc);
|
| }
|
|
|
| return rtp_states;
|
| @@ -553,10 +538,10 @@ void VideoSendStream::SignalNetworkState(NetworkState state) {
|
| // When it goes down, disable RTCP afterwards. This ensures that any packets
|
| // sent due to the network state changed will not be dropped.
|
| if (state == kNetworkUp)
|
| - vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);
|
| - vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp);
|
| + vie_channel_.SetRTCPMode(config_.rtp.rtcp_mode);
|
| + vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp);
|
| if (state == kNetworkDown)
|
| - vie_channel_->SetRTCPMode(RtcpMode::kOff);
|
| + vie_channel_.SetRTCPMode(RtcpMode::kOff);
|
| }
|
|
|
| int64_t VideoSendStream::GetRtt() const {
|
| @@ -566,7 +551,7 @@ int64_t VideoSendStream::GetRtt() const {
|
| uint32_t extended_max_sequence_number;
|
| uint32_t jitter;
|
| int64_t rtt_ms;
|
| - if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
|
| + if (vie_channel_.GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
|
| &extended_max_sequence_number,
|
| &jitter, &rtt_ms) == 0) {
|
| return rtt_ms;
|
| @@ -575,7 +560,7 @@ int64_t VideoSendStream::GetRtt() const {
|
| }
|
|
|
| int VideoSendStream::GetPaddingNeededBps() const {
|
| - return vie_encoder_->GetPaddingNeededBps();
|
| + return vie_encoder_.GetPaddingNeededBps();
|
| }
|
|
|
| bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
|
| @@ -593,14 +578,14 @@ bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
|
| video_codec.maxBitrate = kEncoderMinBitrate;
|
|
|
| // Stop the media flow while reconfiguring.
|
| - vie_encoder_->Pause();
|
| + vie_encoder_.Pause();
|
|
|
| - if (vie_encoder_->SetEncoder(video_codec) != 0) {
|
| + if (vie_encoder_.SetEncoder(video_codec) != 0) {
|
| LOG(LS_ERROR) << "Failed to set encoder.";
|
| return false;
|
| }
|
|
|
| - if (vie_channel_->SetSendCodec(video_codec, false) != 0) {
|
| + if (vie_channel_.SetSendCodec(video_codec, false) != 0) {
|
| LOG(LS_ERROR) << "Failed to set send codec.";
|
| return false;
|
| }
|
| @@ -609,13 +594,12 @@ bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
|
| // to send on all SSRCs at once etc.)
|
| std::vector<uint32_t> used_ssrcs = config_.rtp.ssrcs;
|
| used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
|
| - vie_encoder_->SetSsrcs(used_ssrcs);
|
| + vie_encoder_.SetSsrcs(used_ssrcs);
|
|
|
| // Restart the media flow
|
| - vie_encoder_->Restart();
|
| + vie_encoder_.Restart();
|
|
|
| return true;
|
| }
|
| -
|
| } // namespace internal
|
| } // namespace webrtc
|
|
|