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webrtc / src / cd5c25cb801fa0a1a2f1c7e453354ccb0e65beff / . / webrtc / video / video_send_stream.cc
blob: 40871af1211d7c5d94c73eea4f9fdf7cdd701995 [file] [log] [blame]
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/video/video_send_stream.h"
#include <algorithm>
#include <sstream>
#include <string>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/video/call_stats.h"
#include "webrtc/video/video_capture_input.h"
#include "webrtc/video/vie_remb.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
class RtcpIntraFrameObserver;
class TransportFeedbackObserver;
static const int kMinSendSidePacketHistorySize = 600;
namespace {
std::vector<RtpRtcp*> CreateRtpRtcpModules(
Transport* outgoing_transport,
RtcpIntraFrameObserver* intra_frame_callback,
RtcpBandwidthObserver* bandwidth_callback,
TransportFeedbackObserver* transport_feedback_callback,
RtcpRttStats* rtt_stats,
RtpPacketSender* paced_sender,
TransportSequenceNumberAllocator* transport_sequence_number_allocator,
SendStatisticsProxy* stats_proxy,
size_t num_modules) {
RTC_DCHECK_GT(num_modules, 0u);
RtpRtcp::Configuration configuration;
ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
configuration.audio = false;
configuration.receiver_only = false;
configuration.receive_statistics = null_receive_statistics;
configuration.outgoing_transport = outgoing_transport;
configuration.intra_frame_callback = intra_frame_callback;
configuration.rtt_stats = rtt_stats;
configuration.rtcp_packet_type_counter_observer = stats_proxy;
configuration.paced_sender = paced_sender;
configuration.transport_sequence_number_allocator =
transport_sequence_number_allocator;
configuration.send_bitrate_observer = stats_proxy;
configuration.send_frame_count_observer = stats_proxy;
configuration.send_side_delay_observer = stats_proxy;
configuration.bandwidth_callback = bandwidth_callback;
configuration.transport_feedback_callback = transport_feedback_callback;
std::vector<RtpRtcp*> modules;
for (size_t i = 0; i < num_modules; ++i) {
RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
modules.push_back(rtp_rtcp);
}
return modules;
}
} // namespace
std::string
VideoSendStream::Config::EncoderSettings::ToString() const {
std::stringstream ss;
ss << "{payload_name: " << payload_name;
ss << ", payload_type: " << payload_type;
ss << ", encoder: " << (encoder ? "(VideoEncoder)" : "nullptr");
ss << '}';
return ss.str();
}
std::string VideoSendStream::Config::Rtp::Rtx::ToString()
const {
std::stringstream ss;
ss << "{ssrcs: [";
for (size_t i = 0; i < ssrcs.size(); ++i) {
ss << ssrcs[i];
if (i != ssrcs.size() - 1)
ss << ", ";
}
ss << ']';
ss << ", payload_type: " << payload_type;
ss << '}';
return ss.str();
}
std::string VideoSendStream::Config::Rtp::ToString() const {
std::stringstream ss;
ss << "{ssrcs: [";
for (size_t i = 0; i < ssrcs.size(); ++i) {
ss << ssrcs[i];
if (i != ssrcs.size() - 1)
ss << ", ";
}
ss << ']';
ss << ", rtcp_mode: "
<< (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
: "RtcpMode::kReducedSize");
ss << ", max_packet_size: " << max_packet_size;
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
if (i != extensions.size() - 1)
ss << ", ";
}
ss << ']';
ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
ss << ", fec: " << fec.ToString();
ss << ", rtx: " << rtx.ToString();
ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
}
std::string VideoSendStream::Config::ToString() const {
std::stringstream ss;
ss << "{encoder_settings: " << encoder_settings.ToString();
ss << ", rtp: " << rtp.ToString();
ss << ", pre_encode_callback: "
<< (pre_encode_callback ? "(I420FrameCallback)" : "nullptr");
ss << ", post_encode_callback: "
<< (post_encode_callback ? "(EncodedFrameObserver)" : "nullptr");
ss << ", local_renderer: "
<< (local_renderer ? "(VideoRenderer)" : "nullptr");
ss << ", render_delay_ms: " << render_delay_ms;
ss << ", target_delay_ms: " << target_delay_ms;
ss << ", suspend_below_min_bitrate: " << (suspend_below_min_bitrate ? "on"
: "off");
ss << '}';
return ss.str();
}
namespace {
VideoCodecType PayloadNameToCodecType(const std::string& payload_name) {
if (payload_name == "VP8")
return kVideoCodecVP8;
if (payload_name == "VP9")
return kVideoCodecVP9;
if (payload_name == "H264")
return kVideoCodecH264;
return kVideoCodecGeneric;
}
bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
switch (PayloadNameToCodecType(payload_name)) {
case kVideoCodecVP8:
case kVideoCodecVP9:
return true;
case kVideoCodecH264:
case kVideoCodecGeneric:
return false;
case kVideoCodecI420:
case kVideoCodecRED:
case kVideoCodecULPFEC:
case kVideoCodecUnknown:
RTC_NOTREACHED();
return false;
}
RTC_NOTREACHED();
return false;
}
// TODO(pbos): Lower these thresholds (to closer to 100%) when we handle
// pipelining encoders better (multiple input frames before something comes
// out). This should effectively turn off CPU adaptations for systems that
// remotely cope with the load right now.
CpuOveruseOptions GetCpuOveruseOptions(bool full_overuse_time) {
CpuOveruseOptions options;
if (full_overuse_time) {
options.low_encode_usage_threshold_percent = 150;
options.high_encode_usage_threshold_percent = 200;
}
return options;
}
VideoCodec VideoEncoderConfigToVideoCodec(const VideoEncoderConfig& config,
const std::string& payload_name,
int payload_type) {
const std::vector<VideoStream>& streams = config.streams;
static const int kEncoderMinBitrateKbps = 30;
RTC_DCHECK(!streams.empty());
RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
VideoCodec video_codec;
memset(&video_codec, 0, sizeof(video_codec));
video_codec.codecType = PayloadNameToCodecType(payload_name);
switch (config.content_type) {
case VideoEncoderConfig::ContentType::kRealtimeVideo:
video_codec.mode = kRealtimeVideo;
break;
case VideoEncoderConfig::ContentType::kScreen:
video_codec.mode = kScreensharing;
if (config.streams.size() == 1 &&
config.streams[0].temporal_layer_thresholds_bps.size() == 1) {
video_codec.targetBitrate =
config.streams[0].temporal_layer_thresholds_bps[0] / 1000;
}
break;
}
switch (video_codec.codecType) {
case kVideoCodecVP8: {
if (config.encoder_specific_settings) {
video_codec.codecSpecific.VP8 = *reinterpret_cast<const VideoCodecVP8*>(
config.encoder_specific_settings);
} else {
video_codec.codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings();
}
video_codec.codecSpecific.VP8.numberOfTemporalLayers =
static_cast<unsigned char>(
streams.back().temporal_layer_thresholds_bps.size() + 1);
break;
}
case kVideoCodecVP9: {
if (config.encoder_specific_settings) {
video_codec.codecSpecific.VP9 = *reinterpret_cast<const VideoCodecVP9*>(
config.encoder_specific_settings);
if (video_codec.mode == kScreensharing) {
video_codec.codecSpecific.VP9.flexibleMode = true;
// For now VP9 screensharing use 1 temporal and 2 spatial layers.
RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfTemporalLayers,
1);
RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfSpatialLayers, 2);
}
} else {
video_codec.codecSpecific.VP9 = VideoEncoder::GetDefaultVp9Settings();
}
video_codec.codecSpecific.VP9.numberOfTemporalLayers =
static_cast<unsigned char>(
streams.back().temporal_layer_thresholds_bps.size() + 1);
break;
}
case kVideoCodecH264: {
if (config.encoder_specific_settings) {
video_codec.codecSpecific.H264 =
*reinterpret_cast<const VideoCodecH264*>(
config.encoder_specific_settings);
} else {
video_codec.codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
}
break;
}
default:
// TODO(pbos): Support encoder_settings codec-agnostically.
RTC_DCHECK(!config.encoder_specific_settings)
<< "Encoder-specific settings for codec type not wired up.";
break;
}
strncpy(video_codec.plName, payload_name.c_str(), kPayloadNameSize - 1);
video_codec.plName[kPayloadNameSize - 1] = '\0';
video_codec.plType = payload_type;
video_codec.numberOfSimulcastStreams =
static_cast<unsigned char>(streams.size());
video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
if (video_codec.minBitrate < kEncoderMinBitrateKbps)
video_codec.minBitrate = kEncoderMinBitrateKbps;
RTC_DCHECK_LE(streams.size(), static_cast<size_t>(kMaxSimulcastStreams));
if (video_codec.codecType == kVideoCodecVP9) {
// If the vector is empty, bitrates will be configured automatically.
RTC_DCHECK(config.spatial_layers.empty() ||
config.spatial_layers.size() ==
video_codec.codecSpecific.VP9.numberOfSpatialLayers);
RTC_DCHECK_LE(video_codec.codecSpecific.VP9.numberOfSpatialLayers,
kMaxSimulcastStreams);
for (size_t i = 0; i < config.spatial_layers.size(); ++i)
video_codec.spatialLayers[i] = config.spatial_layers[i];
}
for (size_t i = 0; i < streams.size(); ++i) {
SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
RTC_DCHECK_GT(streams[i].width, 0u);
RTC_DCHECK_GT(streams[i].height, 0u);
RTC_DCHECK_GT(streams[i].max_framerate, 0);
// Different framerates not supported per stream at the moment.
RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0);
RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
RTC_DCHECK_GE(streams[i].max_qp, 0);
sim_stream->width = static_cast<uint16_t>(streams[i].width);
sim_stream->height = static_cast<uint16_t>(streams[i].height);
sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
sim_stream->qpMax = streams[i].max_qp;
sim_stream->numberOfTemporalLayers = static_cast<unsigned char>(
streams[i].temporal_layer_thresholds_bps.size() + 1);
video_codec.width = std::max(video_codec.width,
static_cast<uint16_t>(streams[i].width));
video_codec.height = std::max(
video_codec.height, static_cast<uint16_t>(streams[i].height));
video_codec.minBitrate =
std::min(static_cast<uint16_t>(video_codec.minBitrate),
static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000));
video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
video_codec.qpMax = std::max(video_codec.qpMax,
static_cast<unsigned int>(streams[i].max_qp));
}
if (video_codec.maxBitrate == 0) {
// Unset max bitrate -> cap to one bit per pixel.
video_codec.maxBitrate =
(video_codec.width * video_codec.height * video_codec.maxFramerate) /
1000;
}
if (video_codec.maxBitrate < kEncoderMinBitrateKbps)
video_codec.maxBitrate = kEncoderMinBitrateKbps;
RTC_DCHECK_GT(streams[0].max_framerate, 0);
video_codec.maxFramerate = streams[0].max_framerate;
return video_codec;
}
} // namespace
namespace internal {
VideoSendStream::VideoSendStream(
int num_cpu_cores,
ProcessThread* module_process_thread,
CallStats* call_stats,
CongestionController* congestion_controller,
BitrateAllocator* bitrate_allocator,
VieRemb* remb,
const VideoSendStream::Config& config,
const VideoEncoderConfig& encoder_config,
const std::map<uint32_t, RtpState>& suspended_ssrcs)
: stats_proxy_(Clock::GetRealTimeClock(),
config,
encoder_config.content_type),
encoded_frame_proxy_(config.post_encode_callback),
config_(config),
suspended_ssrcs_(suspended_ssrcs),
module_process_thread_(module_process_thread),
call_stats_(call_stats),
congestion_controller_(congestion_controller),
bitrate_allocator_(bitrate_allocator),
remb_(remb),
encoder_thread_(EncoderThreadFunction, this, "EncoderThread"),
encoder_wakeup_event_(false, false),
stop_encoder_thread_(0),
overuse_detector_(
Clock::GetRealTimeClock(),
GetCpuOveruseOptions(config.encoder_settings.full_overuse_time),
this,
config.post_encode_callback,
&stats_proxy_),
vie_encoder_(
num_cpu_cores,
config_.rtp.ssrcs,
module_process_thread_,
&stats_proxy_,
config.pre_encode_callback,
&overuse_detector_,
congestion_controller_->pacer(),
&payload_router_,
config.post_encode_callback ? &encoded_frame_proxy_ : nullptr),
video_sender_(vie_encoder_.video_sender()),
bandwidth_observer_(congestion_controller_->GetBitrateController()
->CreateRtcpBandwidthObserver()),
rtp_rtcp_modules_(CreateRtpRtcpModules(
config.send_transport,
&encoder_feedback_,
bandwidth_observer_.get(),
congestion_controller_->GetTransportFeedbackObserver(),
call_stats_->rtcp_rtt_stats(),
congestion_controller_->pacer(),
congestion_controller_->packet_router(),
&stats_proxy_,
config_.rtp.ssrcs.size())),
payload_router_(rtp_rtcp_modules_, config.encoder_settings.payload_type),
input_(&encoder_wakeup_event_,
config_.local_renderer,
&stats_proxy_,
&overuse_detector_) {
LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
RTC_DCHECK(!config_.rtp.ssrcs.empty());
RTC_DCHECK(module_process_thread_);
RTC_DCHECK(call_stats_);
RTC_DCHECK(congestion_controller_);
RTC_DCHECK(remb_);
RTC_CHECK(vie_encoder_.Init());
encoder_feedback_.Init(config_.rtp.ssrcs, &vie_encoder_);
// RTP/RTCP initialization.
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
module_process_thread_->RegisterModule(rtp_rtcp);
congestion_controller_->packet_router()->AddRtpModule(rtp_rtcp);
}
video_sender_->RegisterProtectionCallback(this);
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
const std::string& extension = config_.rtp.extensions[i].name;
int id = config_.rtp.extensions[i].id;
// One-byte-extension local identifiers are in the range 1-14 inclusive.
RTC_DCHECK_GE(id, 1);
RTC_DCHECK_LE(id, 14);
RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
StringToRtpExtensionType(extension), id));
}
}
remb_->AddRembSender(rtp_rtcp_modules_[0]);
rtp_rtcp_modules_[0]->SetREMBStatus(true);
ConfigureProtection();
ConfigureSsrcs();
// TODO(pbos): Should we set CNAME on all RTP modules?
rtp_rtcp_modules_.front()->SetCNAME(config_.rtp.c_name.c_str());
// 28 to match packet overhead in ModuleRtpRtcpImpl.
static const size_t kRtpPacketSizeOverhead = 28;
RTC_DCHECK_LE(config_.rtp.max_packet_size, 0xFFFFu + kRtpPacketSizeOverhead);
const uint16_t mtu = static_cast<uint16_t>(config_.rtp.max_packet_size +
kRtpPacketSizeOverhead);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->RegisterRtcpStatisticsCallback(&stats_proxy_);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_);
rtp_rtcp->SetMaxTransferUnit(mtu);
rtp_rtcp->RegisterVideoSendPayload(
config_.encoder_settings.payload_type,
config_.encoder_settings.payload_name.c_str());
}
RTC_DCHECK(config.encoder_settings.encoder);
RTC_DCHECK_GE(config.encoder_settings.payload_type, 0);
RTC_DCHECK_LE(config.encoder_settings.payload_type, 127);
ReconfigureVideoEncoder(encoder_config);
module_process_thread_->RegisterModule(&overuse_detector_);
encoder_thread_.Start();
encoder_thread_.SetPriority(rtc::kHighPriority);
}
VideoSendStream::~VideoSendStream() {
LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString();
Stop();
// Stop the encoder thread permanently.
rtc::AtomicOps::ReleaseStore(&stop_encoder_thread_, 1);
encoder_wakeup_event_.Set();
encoder_thread_.Stop();
// This needs to happen after stopping the encoder thread,
// since the encoder thread calls AddObserver.
bitrate_allocator_->RemoveObserver(this);
module_process_thread_->DeRegisterModule(&overuse_detector_);
rtp_rtcp_modules_[0]->SetREMBStatus(false);
remb_->RemoveRembSender(rtp_rtcp_modules_[0]);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
congestion_controller_->packet_router()->RemoveRtpModule(rtp_rtcp);
module_process_thread_->DeRegisterModule(rtp_rtcp);
delete rtp_rtcp;
}
}
VideoCaptureInput* VideoSendStream::Input() {
return &input_;
}
void VideoSendStream::Start() {
if (payload_router_.active())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
vie_encoder_.Pause();
payload_router_.set_active(true);
// Was not already started, trigger a keyframe.
vie_encoder_.SendKeyFrame();
vie_encoder_.Restart();
}
void VideoSendStream::Stop() {
if (!payload_router_.active())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
// TODO(pbos): Make sure the encoder stops here.
payload_router_.set_active(false);
}
bool VideoSendStream::EncoderThreadFunction(void* obj) {
static_cast<VideoSendStream*>(obj)->EncoderProcess();
// We're done, return false to abort.
return false;
}
void VideoSendStream::EncoderProcess() {
RTC_CHECK_EQ(0, vie_encoder_.RegisterExternalEncoder(
config_.encoder_settings.encoder,
config_.encoder_settings.payload_type,
config_.encoder_settings.internal_source));
while (true) {
encoder_wakeup_event_.Wait(rtc::Event::kForever);
if (rtc::AtomicOps::AcquireLoad(&stop_encoder_thread_))
break;
rtc::Optional<EncoderSettings> encoder_settings;
{
rtc::CritScope lock(&encoder_settings_crit_);
if (pending_encoder_settings_) {
encoder_settings = pending_encoder_settings_;
pending_encoder_settings_ = rtc::Optional<EncoderSettings>();
}
}
if (encoder_settings) {
encoder_settings->video_codec.startBitrate =
bitrate_allocator_->AddObserver(
this, encoder_settings->video_codec.minBitrate * 1000,
encoder_settings->video_codec.maxBitrate * 1000) /
1000;
vie_encoder_.SetEncoder(encoder_settings->video_codec,
encoder_settings->min_transmit_bitrate_bps);
if (config_.suspend_below_min_bitrate) {
video_sender_->SuspendBelowMinBitrate();
bitrate_allocator_->EnforceMinBitrate(false);
}
// We might've gotten new settings while configuring the encoder settings,
// restart from the top to see if that's the case before trying to encode
// a frame (which might correspond to the last frame size).
encoder_wakeup_event_.Set();
continue;
}
VideoFrame frame;
if (input_.GetVideoFrame(&frame))
vie_encoder_.EncodeVideoFrame(frame);
}
vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type);
}
void VideoSendStream::ReconfigureVideoEncoder(
const VideoEncoderConfig& config) {
TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size());
VideoCodec video_codec = VideoEncoderConfigToVideoCodec(
config, config_.encoder_settings.payload_name,
config_.encoder_settings.payload_type);
{
rtc::CritScope lock(&encoder_settings_crit_);
pending_encoder_settings_ = rtc::Optional<EncoderSettings>(
{video_codec, config.min_transmit_bitrate_bps});
}
encoder_wakeup_event_.Set();
}
bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
rtp_rtcp->IncomingRtcpPacket(packet, length);
return true;
}
VideoSendStream::Stats VideoSendStream::GetStats() {
return stats_proxy_.GetStats();
}
void VideoSendStream::OveruseDetected() {
if (config_.overuse_callback)
config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
}
void VideoSendStream::NormalUsage() {
if (config_.overuse_callback)
config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse);
}
void VideoSendStream::ConfigureProtection() {
// Enable NACK, FEC or both.
const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
// Payload types without picture ID cannot determine that a stream is complete
// without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is
// a waste of bandwidth since FEC packets still have to be transmitted. Note
// that this is not the case with FLEXFEC.
if (enable_protection_nack &&
!PayloadTypeSupportsSkippingFecPackets(
config_.encoder_settings.payload_name)) {
LOG(LS_WARNING) << "Transmitting payload type without picture ID using"
"NACK+FEC is a waste of bandwidth since FEC packets "
"also have to be retransmitted. Disabling FEC.";
enable_protection_fec = false;
}
// Set to valid uint8_ts to be castable later without signed overflows.
uint8_t payload_type_red = 0;
uint8_t payload_type_fec = 0;
// TODO(changbin): Should set RTX for RED mapping in RTP sender in future.
// Validate payload types. If either RED or FEC payload types are set then
// both should be. If FEC is enabled then they both have to be set.
if (enable_protection_fec || config_.rtp.fec.red_payload_type != -1 ||
config_.rtp.fec.ulpfec_payload_type != -1) {
RTC_DCHECK_GE(config_.rtp.fec.red_payload_type, 0);
RTC_DCHECK_GE(config_.rtp.fec.ulpfec_payload_type, 0);
RTC_DCHECK_LE(config_.rtp.fec.red_payload_type, 127);
RTC_DCHECK_LE(config_.rtp.fec.ulpfec_payload_type, 127);
payload_type_red = static_cast<uint8_t>(config_.rtp.fec.red_payload_type);
payload_type_fec =
static_cast<uint8_t>(config_.rtp.fec.ulpfec_payload_type);
} else {
// Payload types unset.
RTC_DCHECK_EQ(config_.rtp.fec.red_payload_type, -1);
RTC_DCHECK_EQ(config_.rtp.fec.ulpfec_payload_type, -1);
}
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
// Set NACK.
rtp_rtcp->SetStorePacketsStatus(
enable_protection_nack || congestion_controller_->pacer(),
kMinSendSidePacketHistorySize);
// Set FEC.
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetGenericFECStatus(enable_protection_fec, payload_type_red,
payload_type_fec);
}
}
vie_encoder_.SetProtectionMethod(enable_protection_nack,
enable_protection_fec);
}
void VideoSendStream::ConfigureSsrcs() {
// Configure regular SSRCs.
for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
uint32_t ssrc = config_.rtp.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
rtp_rtcp->SetSSRC(ssrc);
// Restore RTP state if previous existed.
RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtpState(it->second);
}
// Set up RTX if available.
if (config_.rtp.rtx.ssrcs.empty())
return;
// Configure RTX SSRCs.
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];
RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
rtp_rtcp->SetRtxSsrc(ssrc);
RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtxState(it->second);
}
// Configure RTX payload types.
RTC_DCHECK_GE(config_.rtp.rtx.payload_type, 0);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetRtxSendPayloadType(config_.rtp.rtx.payload_type,
config_.encoder_settings.payload_type);
rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
}
if (config_.rtp.fec.red_payload_type != -1 &&
config_.rtp.fec.red_rtx_payload_type != -1) {
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type,
config_.rtp.fec.red_payload_type);
}
}
}
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];
RTC_DCHECK_EQ(ssrc, rtp_rtcp_modules_[i]->SSRC());
rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtpState();
}
for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtxState();
}
return rtp_states;
}
void VideoSendStream::SignalNetworkState(NetworkState state) {
// When network goes up, enable RTCP status before setting transmission 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) {
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode);
}
vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp);
if (state == kNetworkDown) {
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
rtp_rtcp->SetRTCPStatus(RtcpMode::kOff);
}
}
int VideoSendStream::GetPaddingNeededBps() const {
return vie_encoder_.GetPaddingNeededBps();
}
void VideoSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt) {
vie_encoder_.OnBitrateUpdated(bitrate_bps, fraction_loss, rtt);
}
int VideoSendStream::ProtectionRequest(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps) {
*sent_video_rate_bps = 0;
*sent_nack_rate_bps = 0;
*sent_fec_rate_bps = 0;
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
uint32_t not_used = 0;
uint32_t module_video_rate = 0;
uint32_t module_fec_rate = 0;
uint32_t module_nack_rate = 0;
rtp_rtcp->SetFecParameters(delta_params, key_params);
rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
&module_nack_rate);
*sent_video_rate_bps += module_video_rate;
*sent_nack_rate_bps += module_nack_rate;
*sent_fec_rate_bps += module_fec_rate;
}
return 0;
}
} // namespace internal
} // namespace webrtc