blob: 4e22a487a2fb86324c25aa5cc4b2520f02ff3656 [file] [log] [blame]
/*
* Copyright (c) 2014 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/media/engine/webrtcvideoengine2.h"
#include <stdio.h>
#include <algorithm>
#include <set>
#include <string>
#include <utility>
#include "webrtc/base/copyonwritebuffer.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/call.h"
#include "webrtc/media/engine/constants.h"
#include "webrtc/media/engine/simulcast.h"
#include "webrtc/media/engine/webrtcmediaengine.h"
#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
#include "webrtc/media/engine/webrtcvideoframe.h"
#include "webrtc/media/engine/webrtcvoiceengine.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/video_decoder.h"
#include "webrtc/video_encoder.h"
namespace cricket {
namespace {
// Wrap cricket::WebRtcVideoEncoderFactory as a webrtc::VideoEncoderFactory.
class EncoderFactoryAdapter : public webrtc::VideoEncoderFactory {
public:
// EncoderFactoryAdapter doesn't take ownership of |factory|, which is owned
// by e.g. PeerConnectionFactory.
explicit EncoderFactoryAdapter(cricket::WebRtcVideoEncoderFactory* factory)
: factory_(factory) {}
virtual ~EncoderFactoryAdapter() {}
// Implement webrtc::VideoEncoderFactory.
webrtc::VideoEncoder* Create() override {
return factory_->CreateVideoEncoder(webrtc::kVideoCodecVP8);
}
void Destroy(webrtc::VideoEncoder* encoder) override {
return factory_->DestroyVideoEncoder(encoder);
}
private:
cricket::WebRtcVideoEncoderFactory* const factory_;
};
webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec(
const VideoCodec& codec) {
webrtc::Call::Config::BitrateConfig config;
int bitrate_kbps;
if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
config.min_bitrate_bps = bitrate_kbps * 1000;
} else {
config.min_bitrate_bps = 0;
}
if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
config.start_bitrate_bps = bitrate_kbps * 1000;
} else {
// Do not reconfigure start bitrate unless it's specified and positive.
config.start_bitrate_bps = -1;
}
if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
config.max_bitrate_bps = bitrate_kbps * 1000;
} else {
config.max_bitrate_bps = -1;
}
return config;
}
// An encoder factory that wraps Create requests for simulcastable codec types
// with a webrtc::SimulcastEncoderAdapter. Non simulcastable codec type
// requests are just passed through to the contained encoder factory.
class WebRtcSimulcastEncoderFactory
: public cricket::WebRtcVideoEncoderFactory {
public:
// WebRtcSimulcastEncoderFactory doesn't take ownership of |factory|, which is
// owned by e.g. PeerConnectionFactory.
explicit WebRtcSimulcastEncoderFactory(
cricket::WebRtcVideoEncoderFactory* factory)
: factory_(factory) {}
static bool UseSimulcastEncoderFactory(
const std::vector<VideoCodec>& codecs) {
// If any codec is VP8, use the simulcast factory. If asked to create a
// non-VP8 codec, we'll just return a contained factory encoder directly.
for (const auto& codec : codecs) {
if (codec.type == webrtc::kVideoCodecVP8) {
return true;
}
}
return false;
}
webrtc::VideoEncoder* CreateVideoEncoder(
webrtc::VideoCodecType type) override {
RTC_DCHECK(factory_ != NULL);
// If it's a codec type we can simulcast, create a wrapped encoder.
if (type == webrtc::kVideoCodecVP8) {
return new webrtc::SimulcastEncoderAdapter(
new EncoderFactoryAdapter(factory_));
}
webrtc::VideoEncoder* encoder = factory_->CreateVideoEncoder(type);
if (encoder) {
non_simulcast_encoders_.push_back(encoder);
}
return encoder;
}
const std::vector<VideoCodec>& codecs() const override {
return factory_->codecs();
}
bool EncoderTypeHasInternalSource(
webrtc::VideoCodecType type) const override {
return factory_->EncoderTypeHasInternalSource(type);
}
void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override {
// Check first to see if the encoder wasn't wrapped in a
// SimulcastEncoderAdapter. In that case, ask the factory to destroy it.
if (std::remove(non_simulcast_encoders_.begin(),
non_simulcast_encoders_.end(),
encoder) != non_simulcast_encoders_.end()) {
factory_->DestroyVideoEncoder(encoder);
return;
}
// Otherwise, SimulcastEncoderAdapter can be deleted directly, and will call
// DestroyVideoEncoder on the factory for individual encoder instances.
delete encoder;
}
private:
cricket::WebRtcVideoEncoderFactory* factory_;
// A list of encoders that were created without being wrapped in a
// SimulcastEncoderAdapter.
std::vector<webrtc::VideoEncoder*> non_simulcast_encoders_;
};
bool CodecIsInternallySupported(const std::string& codec_name) {
if (CodecNamesEq(codec_name, kVp8CodecName)) {
return true;
}
if (CodecNamesEq(codec_name, kVp9CodecName)) {
return webrtc::VP9Encoder::IsSupported() &&
webrtc::VP9Decoder::IsSupported();
}
if (CodecNamesEq(codec_name, kH264CodecName)) {
return webrtc::H264Encoder::IsSupported() &&
webrtc::H264Decoder::IsSupported();
}
return false;
}
void AddDefaultFeedbackParams(VideoCodec* codec) {
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir));
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
codec->AddFeedbackParam(
FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
}
static VideoCodec MakeVideoCodecWithDefaultFeedbackParams(int payload_type,
const char* name) {
VideoCodec codec(payload_type, name, kDefaultVideoMaxWidth,
kDefaultVideoMaxHeight, kDefaultVideoMaxFramerate);
AddDefaultFeedbackParams(&codec);
return codec;
}
static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) {
std::stringstream out;
out << '{';
for (size_t i = 0; i < codecs.size(); ++i) {
out << codecs[i].ToString();
if (i != codecs.size() - 1) {
out << ", ";
}
}
out << '}';
return out.str();
}
static bool ValidateCodecFormats(const std::vector<VideoCodec>& codecs) {
bool has_video = false;
for (size_t i = 0; i < codecs.size(); ++i) {
if (!codecs[i].ValidateCodecFormat()) {
return false;
}
if (codecs[i].GetCodecType() == VideoCodec::CODEC_VIDEO) {
has_video = true;
}
}
if (!has_video) {
LOG(LS_ERROR) << "Setting codecs without a video codec is invalid: "
<< CodecVectorToString(codecs);
return false;
}
return true;
}
static bool ValidateStreamParams(const StreamParams& sp) {
if (sp.ssrcs.empty()) {
LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
return false;
}
std::vector<uint32_t> primary_ssrcs;
sp.GetPrimarySsrcs(&primary_ssrcs);
std::vector<uint32_t> rtx_ssrcs;
sp.GetFidSsrcs(primary_ssrcs, &rtx_ssrcs);
for (uint32_t rtx_ssrc : rtx_ssrcs) {
bool rtx_ssrc_present = false;
for (uint32_t sp_ssrc : sp.ssrcs) {
if (sp_ssrc == rtx_ssrc) {
rtx_ssrc_present = true;
break;
}
}
if (!rtx_ssrc_present) {
LOG(LS_ERROR) << "RTX SSRC '" << rtx_ssrc
<< "' missing from StreamParams ssrcs: " << sp.ToString();
return false;
}
}
if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) {
LOG(LS_ERROR)
<< "RTX SSRCs exist, but don't cover all SSRCs (unsupported): "
<< sp.ToString();
return false;
}
return true;
}
inline bool ContainsHeaderExtension(
const std::vector<webrtc::RtpExtension>& extensions,
const std::string& uri) {
for (const auto& kv : extensions) {
if (kv.uri == uri) {
return true;
}
}
return false;
}
// Returns true if the given codec is disallowed from doing simulcast.
bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) {
return CodecNamesEq(codec_name, kH264CodecName) ||
CodecNamesEq(codec_name, kVp9CodecName);
}
// The selected thresholds for QVGA and VGA corresponded to a QP around 10.
// The change in QP declined above the selected bitrates.
static int GetMaxDefaultVideoBitrateKbps(int width, int height) {
if (width * height <= 320 * 240) {
return 600;
} else if (width * height <= 640 * 480) {
return 1700;
} else if (width * height <= 960 * 540) {
return 2000;
} else {
return 2500;
}
}
bool GetVp9LayersFromFieldTrialGroup(int* num_spatial_layers,
int* num_temporal_layers) {
std::string group = webrtc::field_trial::FindFullName("WebRTC-SupportVP9SVC");
if (group.empty())
return false;
if (sscanf(group.c_str(), "EnabledByFlag_%dSL%dTL", num_spatial_layers,
num_temporal_layers) != 2) {
return false;
}
const int kMaxSpatialLayers = 2;
if (*num_spatial_layers > kMaxSpatialLayers || *num_spatial_layers < 1)
return false;
const int kMaxTemporalLayers = 3;
if (*num_temporal_layers > kMaxTemporalLayers || *num_temporal_layers < 1)
return false;
return true;
}
int GetDefaultVp9SpatialLayers() {
int num_sl;
int num_tl;
if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) {
return num_sl;
}
return 1;
}
int GetDefaultVp9TemporalLayers() {
int num_sl;
int num_tl;
if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) {
return num_tl;
}
return 1;
}
class EncoderStreamFactory
: public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
public:
EncoderStreamFactory(std::string codec_name,
int max_qp,
int max_framerate,
bool is_screencast,
bool conference_mode)
: codec_name_(codec_name),
max_qp_(max_qp),
max_framerate_(max_framerate),
is_screencast_(is_screencast),
conference_mode_(conference_mode) {}
private:
std::vector<webrtc::VideoStream> CreateEncoderStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config) override {
RTC_DCHECK(encoder_config.number_of_streams > 1 ? !is_screencast_ : true);
if (encoder_config.number_of_streams > 1) {
return GetSimulcastConfig(encoder_config.number_of_streams, width, height,
encoder_config.max_bitrate_bps, max_qp_,
max_framerate_);
}
// For unset max bitrates set default bitrate for non-simulcast.
int max_bitrate_bps =
(encoder_config.max_bitrate_bps > 0)
? encoder_config.max_bitrate_bps
: GetMaxDefaultVideoBitrateKbps(width, height) * 1000;
webrtc::VideoStream stream;
stream.width = width;
stream.height = height;
stream.max_framerate = max_framerate_;
stream.min_bitrate_bps = kMinVideoBitrateKbps * 1000;
stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate_bps;
stream.max_qp = max_qp_;
// Conference mode screencast uses 2 temporal layers split at 100kbit.
if (conference_mode_ && is_screencast_) {
ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
// For screenshare in conference mode, tl0 and tl1 bitrates are
// piggybacked
// on the VideoCodec struct as target and max bitrates, respectively.
// See eg. webrtc::VP8EncoderImpl::SetRates().
stream.target_bitrate_bps = config.tl0_bitrate_kbps * 1000;
stream.max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
stream.temporal_layer_thresholds_bps.clear();
stream.temporal_layer_thresholds_bps.push_back(config.tl0_bitrate_kbps *
1000);
}
if (CodecNamesEq(codec_name_, kVp9CodecName) && !is_screencast_) {
stream.temporal_layer_thresholds_bps.resize(
GetDefaultVp9TemporalLayers() - 1);
}
std::vector<webrtc::VideoStream> streams;
streams.push_back(stream);
return streams;
}
const std::string codec_name_;
const int max_qp_;
const int max_framerate_;
const bool is_screencast_;
const bool conference_mode_;
};
} // namespace
// Constants defined in webrtc/media/engine/constants.h
// TODO(pbos): Move these to a separate constants.cc file.
const int kMinVideoBitrateKbps = 30;
const int kVideoMtu = 1200;
const int kVideoRtpBufferSize = 65536;
// This constant is really an on/off, lower-level configurable NACK history
// duration hasn't been implemented.
static const int kNackHistoryMs = 1000;
static const int kDefaultQpMax = 56;
static const int kDefaultRtcpReceiverReportSsrc = 1;
// Down grade resolution at most 2 times for CPU reasons.
static const int kMaxCpuDowngrades = 2;
// Minimum time interval for logging stats.
static const int64_t kStatsLogIntervalMs = 10000;
// Adds |codec| to |list|, and also adds an RTX codec if |codec|'s name is
// recognized.
// TODO(deadbeef): Should we add RTX codecs for external codecs whose names we
// don't recognize?
void AddCodecAndMaybeRtxCodec(const VideoCodec& codec,
std::vector<VideoCodec>* codecs) {
codecs->push_back(codec);
int rtx_payload_type = 0;
if (CodecNamesEq(codec.name, kVp8CodecName)) {
rtx_payload_type = kDefaultRtxVp8PlType;
} else if (CodecNamesEq(codec.name, kVp9CodecName)) {
rtx_payload_type = kDefaultRtxVp9PlType;
} else if (CodecNamesEq(codec.name, kH264CodecName)) {
rtx_payload_type = kDefaultRtxH264PlType;
} else if (CodecNamesEq(codec.name, kRedCodecName)) {
rtx_payload_type = kDefaultRtxRedPlType;
} else {
return;
}
codecs->push_back(VideoCodec::CreateRtxCodec(rtx_payload_type, codec.id));
}
std::vector<VideoCodec> DefaultVideoCodecList() {
std::vector<VideoCodec> codecs;
AddCodecAndMaybeRtxCodec(
MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType, kVp8CodecName),
&codecs);
if (CodecIsInternallySupported(kVp9CodecName)) {
AddCodecAndMaybeRtxCodec(MakeVideoCodecWithDefaultFeedbackParams(
kDefaultVp9PlType, kVp9CodecName),
&codecs);
}
if (CodecIsInternallySupported(kH264CodecName)) {
VideoCodec codec = MakeVideoCodecWithDefaultFeedbackParams(
kDefaultH264PlType, kH264CodecName);
// TODO(hta): Move all parameter generation for SDP into the codec
// implementation, for all codecs and parameters.
// TODO(hta): Move selection of profile-level-id to H.264 codec
// implementation.
// TODO(hta): Set FMTP parameters for all codecs of type H264.
codec.SetParam(kH264FmtpProfileLevelId,
kH264ProfileLevelConstrainedBaseline);
codec.SetParam(kH264FmtpLevelAsymmetryAllowed, "1");
codec.SetParam(kH264FmtpPacketizationMode, "1");
AddCodecAndMaybeRtxCodec(codec, &codecs);
}
AddCodecAndMaybeRtxCodec(VideoCodec(kDefaultRedPlType, kRedCodecName),
&codecs);
codecs.push_back(VideoCodec(kDefaultUlpfecType, kUlpfecCodecName));
return codecs;
}
rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
const VideoCodec& codec) {
RTC_DCHECK_RUN_ON(&thread_checker_);
bool is_screencast = parameters_.options.is_screencast.value_or(false);
// No automatic resizing when using simulcast or screencast.
bool automatic_resize =
!is_screencast && parameters_.config.rtp.ssrcs.size() == 1;
bool frame_dropping = !is_screencast;
bool denoising;
bool codec_default_denoising = false;
if (is_screencast) {
denoising = false;
} else {
// Use codec default if video_noise_reduction is unset.
codec_default_denoising = !parameters_.options.video_noise_reduction;
denoising = parameters_.options.video_noise_reduction.value_or(false);
}
if (CodecNamesEq(codec.name, kH264CodecName)) {
webrtc::VideoCodecH264 h264_settings =
webrtc::VideoEncoder::GetDefaultH264Settings();
h264_settings.frameDroppingOn = frame_dropping;
return new rtc::RefCountedObject<
webrtc::VideoEncoderConfig::H264EncoderSpecificSettings>(h264_settings);
}
if (CodecNamesEq(codec.name, kVp8CodecName)) {
webrtc::VideoCodecVP8 vp8_settings =
webrtc::VideoEncoder::GetDefaultVp8Settings();
vp8_settings.automaticResizeOn = automatic_resize;
// VP8 denoising is enabled by default.
vp8_settings.denoisingOn = codec_default_denoising ? true : denoising;
vp8_settings.frameDroppingOn = frame_dropping;
return new rtc::RefCountedObject<
webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings);
}
if (CodecNamesEq(codec.name, kVp9CodecName)) {
webrtc::VideoCodecVP9 vp9_settings =
webrtc::VideoEncoder::GetDefaultVp9Settings();
if (is_screencast) {
// TODO(asapersson): Set to 2 for now since there is a DCHECK in
// VideoSendStream::ReconfigureVideoEncoder.
vp9_settings.numberOfSpatialLayers = 2;
} else {
vp9_settings.numberOfSpatialLayers = GetDefaultVp9SpatialLayers();
}
// VP9 denoising is disabled by default.
vp9_settings.denoisingOn = codec_default_denoising ? false : denoising;
vp9_settings.frameDroppingOn = frame_dropping;
return new rtc::RefCountedObject<
webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
}
return nullptr;
}
DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler()
: default_recv_ssrc_(0), default_sink_(NULL) {}
UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
WebRtcVideoChannel2* channel,
uint32_t ssrc) {
if (default_recv_ssrc_ != 0) { // Already one default stream.
LOG(LS_WARNING) << "Unknown SSRC, but default receive stream already set.";
return kDropPacket;
}
StreamParams sp;
sp.ssrcs.push_back(ssrc);
LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
if (!channel->AddRecvStream(sp, true)) {
LOG(LS_WARNING) << "Could not create default receive stream.";
}
channel->SetSink(ssrc, default_sink_);
default_recv_ssrc_ = ssrc;
return kDeliverPacket;
}
rtc::VideoSinkInterface<VideoFrame>*
DefaultUnsignalledSsrcHandler::GetDefaultSink() const {
return default_sink_;
}
void DefaultUnsignalledSsrcHandler::SetDefaultSink(
VideoMediaChannel* channel,
rtc::VideoSinkInterface<VideoFrame>* sink) {
default_sink_ = sink;
if (default_recv_ssrc_ != 0) {
channel->SetSink(default_recv_ssrc_, default_sink_);
}
}
WebRtcVideoEngine2::WebRtcVideoEngine2()
: initialized_(false),
external_decoder_factory_(NULL),
external_encoder_factory_(NULL) {
LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
video_codecs_ = GetSupportedCodecs();
}
WebRtcVideoEngine2::~WebRtcVideoEngine2() {
LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
}
void WebRtcVideoEngine2::Init() {
LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
initialized_ = true;
}
WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options) {
RTC_DCHECK(initialized_);
LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString();
return new WebRtcVideoChannel2(call, config, options, video_codecs_,
external_encoder_factory_,
external_decoder_factory_);
}
const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const {
return video_codecs_;
}
RtpCapabilities WebRtcVideoEngine2::GetCapabilities() const {
RtpCapabilities capabilities;
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kTimestampOffsetUri,
webrtc::RtpExtension::kTimestampOffsetDefaultId));
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kAbsSendTimeDefaultId));
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kVideoRotationUri,
webrtc::RtpExtension::kVideoRotationDefaultId));
capabilities.header_extensions.push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kPlayoutDelayUri,
webrtc::RtpExtension::kPlayoutDelayDefaultId));
return capabilities;
}
void WebRtcVideoEngine2::SetExternalDecoderFactory(
WebRtcVideoDecoderFactory* decoder_factory) {
RTC_DCHECK(!initialized_);
external_decoder_factory_ = decoder_factory;
}
void WebRtcVideoEngine2::SetExternalEncoderFactory(
WebRtcVideoEncoderFactory* encoder_factory) {
RTC_DCHECK(!initialized_);
if (external_encoder_factory_ == encoder_factory)
return;
// No matter what happens we shouldn't hold on to a stale
// WebRtcSimulcastEncoderFactory.
simulcast_encoder_factory_.reset();
if (encoder_factory &&
WebRtcSimulcastEncoderFactory::UseSimulcastEncoderFactory(
encoder_factory->codecs())) {
simulcast_encoder_factory_.reset(
new WebRtcSimulcastEncoderFactory(encoder_factory));
encoder_factory = simulcast_encoder_factory_.get();
}
external_encoder_factory_ = encoder_factory;
video_codecs_ = GetSupportedCodecs();
}
std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const {
std::vector<VideoCodec> supported_codecs = DefaultVideoCodecList();
if (external_encoder_factory_ == NULL) {
LOG(LS_INFO) << "Supported codecs: "
<< CodecVectorToString(supported_codecs);
return supported_codecs;
}
std::stringstream out;
const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
external_encoder_factory_->codecs();
for (size_t i = 0; i < codecs.size(); ++i) {
out << codecs[i].name;
if (i != codecs.size() - 1) {
out << ", ";
}
// Don't add internally-supported codecs twice.
if (CodecIsInternallySupported(codecs[i].name)) {
continue;
}
// External video encoders are given payloads 120-127. This also means that
// we only support up to 8 external payload types.
// TODO(deadbeef): mediasession.cc already has code to dynamically
// determine a payload type. We should be able to just leave the payload
// type empty and let mediasession determine it. However, currently RTX
// codecs are associated to codecs by payload type, meaning we DO need
// to allocate unique payload types here. So to make this change we would
// need to make RTX codecs associated by name instead.
const int kExternalVideoPayloadTypeBase = 120;
size_t payload_type = kExternalVideoPayloadTypeBase + i;
RTC_DCHECK(payload_type < 128);
VideoCodec codec(static_cast<int>(payload_type), codecs[i].name,
codecs[i].max_width, codecs[i].max_height,
codecs[i].max_fps);
AddDefaultFeedbackParams(&codec);
AddCodecAndMaybeRtxCodec(codec, &supported_codecs);
}
LOG(LS_INFO) << "Supported codecs (incl. external codecs): "
<< CodecVectorToString(supported_codecs);
LOG(LS_INFO) << "Codecs supported by the external encoder factory: "
<< out.str();
return supported_codecs;
}
WebRtcVideoChannel2::WebRtcVideoChannel2(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const std::vector<VideoCodec>& recv_codecs,
WebRtcVideoEncoderFactory* external_encoder_factory,
WebRtcVideoDecoderFactory* external_decoder_factory)
: VideoMediaChannel(config),
call_(call),
unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
video_config_(config.video),
external_encoder_factory_(external_encoder_factory),
external_decoder_factory_(external_decoder_factory),
default_send_options_(options),
red_disabled_by_remote_side_(false),
last_stats_log_ms_(-1) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
sending_ = false;
RTC_DCHECK(ValidateCodecFormats(recv_codecs));
recv_codecs_ = FilterSupportedCodecs(MapCodecs(recv_codecs));
}
WebRtcVideoChannel2::~WebRtcVideoChannel2() {
for (auto& kv : send_streams_)
delete kv.second;
for (auto& kv : receive_streams_)
delete kv.second;
}
bool WebRtcVideoChannel2::CodecIsExternallySupported(
const std::string& name) const {
if (external_encoder_factory_ == NULL) {
return false;
}
const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs =
external_encoder_factory_->codecs();
for (size_t c = 0; c < external_codecs.size(); ++c) {
if (CodecNamesEq(name, external_codecs[c].name)) {
return true;
}
}
return false;
}
std::vector<WebRtcVideoChannel2::VideoCodecSettings>
WebRtcVideoChannel2::FilterSupportedCodecs(
const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs)
const {
std::vector<VideoCodecSettings> supported_codecs;
for (size_t i = 0; i < mapped_codecs.size(); ++i) {
const VideoCodecSettings& codec = mapped_codecs[i];
if (CodecIsInternallySupported(codec.codec.name) ||
CodecIsExternallySupported(codec.codec.name)) {
supported_codecs.push_back(codec);
}
}
return supported_codecs;
}
bool WebRtcVideoChannel2::ReceiveCodecsHaveChanged(
std::vector<VideoCodecSettings> before,
std::vector<VideoCodecSettings> after) {
if (before.size() != after.size()) {
return true;
}
// The receive codec order doesn't matter, so we sort the codecs before
// comparing. This is necessary because currently the
// only way to change the send codec is to munge SDP, which causes
// the receive codec list to change order, which causes the streams
// to be recreates which causes a "blink" of black video. In order
// to support munging the SDP in this way without recreating receive
// streams, we ignore the order of the received codecs so that
// changing the order doesn't cause this "blink".
auto comparison =
[](const VideoCodecSettings& codec1, const VideoCodecSettings& codec2) {
return codec1.codec.id > codec2.codec.id;
};
std::sort(before.begin(), before.end(), comparison);
std::sort(after.begin(), after.end(), comparison);
return before != after;
}
bool WebRtcVideoChannel2::GetChangedSendParameters(
const VideoSendParameters& params,
ChangedSendParameters* changed_params) const {
if (!ValidateCodecFormats(params.codecs) ||
!ValidateRtpExtensions(params.extensions)) {
return false;
}
// Handle send codec.
const std::vector<VideoCodecSettings> supported_codecs =
FilterSupportedCodecs(MapCodecs(params.codecs));
if (supported_codecs.empty()) {
LOG(LS_ERROR) << "No video codecs supported.";
return false;
}
if (!send_codec_ || supported_codecs.front() != *send_codec_) {
changed_params->codec =
rtc::Optional<VideoCodecSettings>(supported_codecs.front());
}
// Handle RTP header extensions.
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) {
changed_params->rtp_header_extensions =
rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
}
// Handle max bitrate.
if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps &&
params.max_bandwidth_bps >= 0) {
// 0 uncaps max bitrate (-1).
changed_params->max_bandwidth_bps = rtc::Optional<int>(
params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps);
}
// Handle conference mode.
if (params.conference_mode != send_params_.conference_mode) {
changed_params->conference_mode =
rtc::Optional<bool>(params.conference_mode);
}
// Handle RTCP mode.
if (params.rtcp.reduced_size != send_params_.rtcp.reduced_size) {
changed_params->rtcp_mode = rtc::Optional<webrtc::RtcpMode>(
params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound);
}
return true;
}
rtc::DiffServCodePoint WebRtcVideoChannel2::PreferredDscp() const {
return rtc::DSCP_AF41;
}
bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters");
LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
ChangedSendParameters changed_params;
if (!GetChangedSendParameters(params, &changed_params)) {
return false;
}
if (changed_params.codec) {
const VideoCodecSettings& codec_settings = *changed_params.codec;
send_codec_ = rtc::Optional<VideoCodecSettings>(codec_settings);
LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString();
}
if (changed_params.rtp_header_extensions) {
send_rtp_extensions_ = changed_params.rtp_header_extensions;
}
if (changed_params.codec || changed_params.max_bandwidth_bps) {
if (send_codec_) {
// TODO(holmer): Changing the codec parameters shouldn't necessarily mean
// that we change the min/max of bandwidth estimation. Reevaluate this.
bitrate_config_ = GetBitrateConfigForCodec(send_codec_->codec);
if (!changed_params.codec) {
// If the codec isn't changing, set the start bitrate to -1 which means
// "unchanged" so that BWE isn't affected.
bitrate_config_.start_bitrate_bps = -1;
}
}
if (params.max_bandwidth_bps >= 0) {
// Note that max_bandwidth_bps intentionally takes priority over the
// bitrate config for the codec. This allows FEC to be applied above the
// codec target bitrate.
// TODO(pbos): Figure out whether b=AS means max bitrate for this
// WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC),
// in which case this should not set a Call::BitrateConfig but rather
// reconfigure all senders.
bitrate_config_.max_bitrate_bps =
params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
}
call_->SetBitrateConfig(bitrate_config_);
}
{
rtc::CritScope stream_lock(&stream_crit_);
for (auto& kv : send_streams_) {
kv.second->SetSendParameters(changed_params);
}
if (changed_params.codec || changed_params.rtcp_mode) {
// Update receive feedback parameters from new codec or RTCP mode.
LOG(LS_INFO)
<< "SetFeedbackOptions on all the receive streams because the send "
"codec or RTCP mode has changed.";
for (auto& kv : receive_streams_) {
RTC_DCHECK(kv.second != nullptr);
kv.second->SetFeedbackParameters(
HasNack(send_codec_->codec), HasRemb(send_codec_->codec),
HasTransportCc(send_codec_->codec),
params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound);
}
}
if (changed_params.codec) {
bool red_was_disabled = red_disabled_by_remote_side_;
red_disabled_by_remote_side_ =
changed_params.codec->ulpfec.red_payload_type == -1;
if (red_was_disabled != red_disabled_by_remote_side_) {
for (auto& kv : receive_streams_) {
// In practice VideoChannel::SetRemoteContent appears to most of the
// time also call UpdateRemoteStreams, which recreates the receive
// streams. If that's always true this call isn't needed.
kv.second->SetUlpfecDisabledRemotely(red_disabled_by_remote_side_);
}
}
}
}
send_params_ = params;
return true;
}
webrtc::RtpParameters WebRtcVideoChannel2::GetRtpSendParameters(
uint32_t ssrc) const {
rtc::CritScope stream_lock(&stream_crit_);
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
webrtc::RtpParameters rtp_params = it->second->GetRtpParameters();
// Need to add the common list of codecs to the send stream-specific
// RTP parameters.
for (const VideoCodec& codec : send_params_.codecs) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
return rtp_params;
}
bool WebRtcVideoChannel2::SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpSendParameters");
rtc::CritScope stream_lock(&stream_crit_);
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return false;
}
// TODO(deadbeef): Handle setting parameters with a list of codecs in a
// different order (which should change the send codec).
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) {
LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
<< "is not currently supported.";
return false;
}
return it->second->SetRtpParameters(parameters);
}
webrtc::RtpParameters WebRtcVideoChannel2::GetRtpReceiveParameters(
uint32_t ssrc) const {
rtc::CritScope stream_lock(&stream_crit_);
auto it = receive_streams_.find(ssrc);
if (it == receive_streams_.end()) {
LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
// TODO(deadbeef): Return stream-specific parameters.
webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding();
for (const VideoCodec& codec : recv_params_.codecs) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
rtp_params.encodings[0].ssrc = it->second->GetFirstPrimarySsrc();
return rtp_params;
}
bool WebRtcVideoChannel2::SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpReceiveParameters");
rtc::CritScope stream_lock(&stream_crit_);
auto it = receive_streams_.find(ssrc);
if (it == receive_streams_.end()) {
LOG(LS_ERROR) << "Attempting to set RTP receive parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return false;
}
webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
if (current_parameters != parameters) {
LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
<< "unsupported.";
return false;
}
return true;
}
bool WebRtcVideoChannel2::GetChangedRecvParameters(
const VideoRecvParameters& params,
ChangedRecvParameters* changed_params) const {
if (!ValidateCodecFormats(params.codecs) ||
!ValidateRtpExtensions(params.extensions)) {
return false;
}
// Handle receive codecs.
const std::vector<VideoCodecSettings> mapped_codecs =
MapCodecs(params.codecs);
if (mapped_codecs.empty()) {
LOG(LS_ERROR) << "SetRecvParameters called without any video codecs.";
return false;
}
std::vector<VideoCodecSettings> supported_codecs =
FilterSupportedCodecs(mapped_codecs);
if (mapped_codecs.size() != supported_codecs.size()) {
LOG(LS_ERROR) << "SetRecvParameters called with unsupported video codecs.";
return false;
}
if (ReceiveCodecsHaveChanged(recv_codecs_, supported_codecs)) {
changed_params->codec_settings =
rtc::Optional<std::vector<VideoCodecSettings>>(supported_codecs);
}
// Handle RTP header extensions.
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false);
if (filtered_extensions != recv_rtp_extensions_) {
changed_params->rtp_header_extensions =
rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
}
return true;
}
bool WebRtcVideoChannel2::SetRecvParameters(const VideoRecvParameters& params) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvParameters");
LOG(LS_INFO) << "SetRecvParameters: " << params.ToString();
ChangedRecvParameters changed_params;
if (!GetChangedRecvParameters(params, &changed_params)) {
return false;
}
if (changed_params.rtp_header_extensions) {
recv_rtp_extensions_ = *changed_params.rtp_header_extensions;
}
if (changed_params.codec_settings) {
LOG(LS_INFO) << "Changing recv codecs from "
<< CodecSettingsVectorToString(recv_codecs_) << " to "
<< CodecSettingsVectorToString(*changed_params.codec_settings);
recv_codecs_ = *changed_params.codec_settings;
}
{
rtc::CritScope stream_lock(&stream_crit_);
for (auto& kv : receive_streams_) {
kv.second->SetRecvParameters(changed_params);
}
}
recv_params_ = params;
return true;
}
std::string WebRtcVideoChannel2::CodecSettingsVectorToString(
const std::vector<VideoCodecSettings>& codecs) {
std::stringstream out;
out << '{';
for (size_t i = 0; i < codecs.size(); ++i) {
out << codecs[i].codec.ToString();
if (i != codecs.size() - 1) {
out << ", ";
}
}
out << '}';
return out.str();
}
bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
if (!send_codec_) {
LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
return false;
}
*codec = send_codec_->codec;
return true;
}
bool WebRtcVideoChannel2::SetSend(bool send) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSend");
LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
if (send && !send_codec_) {
LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
return false;
}
{
rtc::CritScope stream_lock(&stream_crit_);
for (const auto& kv : send_streams_) {
kv.second->SetSend(send);
}
}
sending_ = send;
return true;
}
// TODO(nisse): The enable argument was used for mute logic which has
// been moved to VideoBroadcaster. So remove the argument from this
// method.
bool WebRtcVideoChannel2::SetVideoSend(
uint32_t ssrc,
bool enable,
const VideoOptions* options,
rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
TRACE_EVENT0("webrtc", "SetVideoSend");
RTC_DCHECK(ssrc != 0);
LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", enable = " << enable
<< ", options: " << (options ? options->ToString() : "nullptr")
<< ", source = " << (source ? "(source)" : "nullptr") << ")";
rtc::CritScope stream_lock(&stream_crit_);
const auto& kv = send_streams_.find(ssrc);
if (kv == send_streams_.end()) {
// Allow unknown ssrc only if source is null.
RTC_CHECK(source == nullptr);
LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
return false;
}
return kv->second->SetVideoSend(enable, options, source);
}
bool WebRtcVideoChannel2::ValidateSendSsrcAvailability(
const StreamParams& sp) const {
for (uint32_t ssrc : sp.ssrcs) {
if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) {
LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists.";
return false;
}
}
return true;
}
bool WebRtcVideoChannel2::ValidateReceiveSsrcAvailability(
const StreamParams& sp) const {
for (uint32_t ssrc : sp.ssrcs) {
if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) {
LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc
<< "' already exists.";
return false;
}
}
return true;
}
bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) {
LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
if (!ValidateStreamParams(sp))
return false;
rtc::CritScope stream_lock(&stream_crit_);
if (!ValidateSendSsrcAvailability(sp))
return false;
for (uint32_t used_ssrc : sp.ssrcs)
send_ssrcs_.insert(used_ssrc);
webrtc::VideoSendStream::Config config(this);
config.suspend_below_min_bitrate = video_config_.suspend_below_min_bitrate;
WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
call_, sp, std::move(config), default_send_options_,
external_encoder_factory_, video_config_.enable_cpu_overuse_detection,
bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_,
send_params_);
uint32_t ssrc = sp.first_ssrc();
RTC_DCHECK(ssrc != 0);
send_streams_[ssrc] = stream;
if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
rtcp_receiver_report_ssrc_ = ssrc;
LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added "
"a send stream.";
for (auto& kv : receive_streams_)
kv.second->SetLocalSsrc(ssrc);
}
if (sending_) {
stream->SetSend(true);
}
return true;
}
bool WebRtcVideoChannel2::RemoveSendStream(uint32_t ssrc) {
LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
WebRtcVideoSendStream* removed_stream;
{
rtc::CritScope stream_lock(&stream_crit_);
std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
send_streams_.find(ssrc);
if (it == send_streams_.end()) {
return false;
}
for (uint32_t old_ssrc : it->second->GetSsrcs())
send_ssrcs_.erase(old_ssrc);
removed_stream = it->second;
send_streams_.erase(it);
// Switch receiver report SSRCs, the one in use is no longer valid.
if (rtcp_receiver_report_ssrc_ == ssrc) {
rtcp_receiver_report_ssrc_ = send_streams_.empty()
? kDefaultRtcpReceiverReportSsrc
: send_streams_.begin()->first;
LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because the "
"previous local SSRC was removed.";
for (auto& kv : receive_streams_) {
kv.second->SetLocalSsrc(rtcp_receiver_report_ssrc_);
}
}
}
delete removed_stream;
return true;
}
void WebRtcVideoChannel2::DeleteReceiveStream(
WebRtcVideoChannel2::WebRtcVideoReceiveStream* stream) {
for (uint32_t old_ssrc : stream->GetSsrcs())
receive_ssrcs_.erase(old_ssrc);
delete stream;
}
bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp) {
return AddRecvStream(sp, false);
}
bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp,
bool default_stream) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "AddRecvStream" << (default_stream ? " (default stream)" : "")
<< ": " << sp.ToString();
if (!ValidateStreamParams(sp))
return false;
uint32_t ssrc = sp.first_ssrc();
RTC_DCHECK(ssrc != 0); // TODO(pbos): Is this ever valid?
rtc::CritScope stream_lock(&stream_crit_);
// Remove running stream if this was a default stream.
const auto& prev_stream = receive_streams_.find(ssrc);
if (prev_stream != receive_streams_.end()) {
if (default_stream || !prev_stream->second->IsDefaultStream()) {
LOG(LS_ERROR) << "Receive stream for SSRC '" << ssrc
<< "' already exists.";
return false;
}
DeleteReceiveStream(prev_stream->second);
receive_streams_.erase(prev_stream);
}
if (!ValidateReceiveSsrcAvailability(sp))
return false;
for (uint32_t used_ssrc : sp.ssrcs)
receive_ssrcs_.insert(used_ssrc);
webrtc::VideoReceiveStream::Config config(this);
ConfigureReceiverRtp(&config, sp);
// Set up A/V sync group based on sync label.
config.sync_group = sp.sync_label;
config.rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false;
config.rtp.transport_cc =
send_codec_ ? HasTransportCc(send_codec_->codec) : false;
config.disable_prerenderer_smoothing =
video_config_.disable_prerenderer_smoothing;
receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
call_, sp, std::move(config), external_decoder_factory_, default_stream,
recv_codecs_, red_disabled_by_remote_side_);
return true;
}
void WebRtcVideoChannel2::ConfigureReceiverRtp(
webrtc::VideoReceiveStream::Config* config,
const StreamParams& sp) const {
uint32_t ssrc = sp.first_ssrc();
config->rtp.remote_ssrc = ssrc;
config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
config->rtp.extensions = recv_rtp_extensions_;
// Whether or not the receive stream sends reduced size RTCP is determined
// by the send params.
// TODO(deadbeef): Once we change "send_params" to "sender_params" and
// "recv_params" to "receiver_params", we should get this out of
// receiver_params_.
config->rtp.rtcp_mode = send_params_.rtcp.reduced_size
? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound;
// TODO(pbos): This protection is against setting the same local ssrc as
// remote which is not permitted by the lower-level API. RTCP requires a
// corresponding sender SSRC. Figure out what to do when we don't have
// (receive-only) or know a good local SSRC.
if (config->rtp.remote_ssrc == config->rtp.local_ssrc) {
if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) {
config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc;
} else {
config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1;
}
}
for (size_t i = 0; i < recv_codecs_.size(); ++i) {
uint32_t rtx_ssrc;
if (recv_codecs_[i].rtx_payload_type != -1 &&
sp.GetFidSsrc(ssrc, &rtx_ssrc)) {
webrtc::VideoReceiveStream::Config::Rtp::Rtx& rtx =
config->rtp.rtx[recv_codecs_[i].codec.id];
rtx.ssrc = rtx_ssrc;
rtx.payload_type = recv_codecs_[i].rtx_payload_type;
}
}
}
bool WebRtcVideoChannel2::RemoveRecvStream(uint32_t ssrc) {
LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
if (ssrc == 0) {
LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported.";
return false;
}
rtc::CritScope stream_lock(&stream_crit_);
std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator stream =
receive_streams_.find(ssrc);
if (stream == receive_streams_.end()) {
LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc;
return false;
}
DeleteReceiveStream(stream->second);
receive_streams_.erase(stream);
return true;
}
bool WebRtcVideoChannel2::SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<VideoFrame>* sink) {
LOG(LS_INFO) << "SetSink: ssrc:" << ssrc << " "
<< (sink ? "(ptr)" : "nullptr");
if (ssrc == 0) {
default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink);
return true;
}
rtc::CritScope stream_lock(&stream_crit_);
std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.find(ssrc);
if (it == receive_streams_.end()) {
return false;
}
it->second->SetSink(sink);
return true;
}
bool WebRtcVideoChannel2::GetStats(VideoMediaInfo* info) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::GetStats");
// Log stats periodically.
bool log_stats = false;
int64_t now_ms = rtc::TimeMillis();
if (last_stats_log_ms_ == -1 ||
now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) {
last_stats_log_ms_ = now_ms;
log_stats = true;
}
info->Clear();
FillSenderStats(info, log_stats);
FillReceiverStats(info, log_stats);
webrtc::Call::Stats stats = call_->GetStats();
FillBandwidthEstimationStats(stats, info);
if (stats.rtt_ms != -1) {
for (size_t i = 0; i < info->senders.size(); ++i) {
info->senders[i].rtt_ms = stats.rtt_ms;
}
}
if (log_stats)
LOG(LS_INFO) << stats.ToString(now_ms);
return true;
}
void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info,
bool log_stats) {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end(); ++it) {
video_media_info->senders.push_back(
it->second->GetVideoSenderInfo(log_stats));
}
}
void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info,
bool log_stats) {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.begin();
it != receive_streams_.end(); ++it) {
video_media_info->receivers.push_back(
it->second->GetVideoReceiverInfo(log_stats));
}
}
void WebRtcVideoChannel2::FillBandwidthEstimationStats(
const webrtc::Call::Stats& stats,
VideoMediaInfo* video_media_info) {
BandwidthEstimationInfo bwe_info;
bwe_info.available_send_bandwidth = stats.send_bandwidth_bps;
bwe_info.available_recv_bandwidth = stats.recv_bandwidth_bps;
bwe_info.bucket_delay = stats.pacer_delay_ms;
// Get send stream bitrate stats.
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream =
send_streams_.begin();
stream != send_streams_.end(); ++stream) {
stream->second->FillBandwidthEstimationInfo(&bwe_info);
}
video_media_info->bw_estimations.push_back(bwe_info);
}
void WebRtcVideoChannel2::OnPacketReceived(
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
const webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(
webrtc::MediaType::VIDEO,
packet->cdata(), packet->size(),
webrtc_packet_time);
switch (delivery_result) {
case webrtc::PacketReceiver::DELIVERY_OK:
return;
case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR:
return;
case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC:
break;
}
uint32_t ssrc = 0;
if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
return;
}
int payload_type = 0;
if (!GetRtpPayloadType(packet->cdata(), packet->size(), &payload_type)) {
return;
}
// See if this payload_type is registered as one that usually gets its own
// SSRC (RTX) or at least is safe to drop either way (ULPFEC). If it is, and
// it wasn't handled above by DeliverPacket, that means we don't know what
// stream it associates with, and we shouldn't ever create an implicit channel
// for these.
for (auto& codec : recv_codecs_) {
if (payload_type == codec.rtx_payload_type ||
payload_type == codec.ulpfec.red_rtx_payload_type ||
payload_type == codec.ulpfec.ulpfec_payload_type) {
return;
}
}
switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) {
case UnsignalledSsrcHandler::kDropPacket:
return;
case UnsignalledSsrcHandler::kDeliverPacket:
break;
}
if (call_->Receiver()->DeliverPacket(
webrtc::MediaType::VIDEO,
packet->cdata(), packet->size(),
webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
return;
}
}
void WebRtcVideoChannel2::OnRtcpReceived(
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
// TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
// for both audio and video on the same path. Since BundleFilter doesn't
// filter RTCP anymore incoming RTCP packets could've been going to audio (so
// logging failures spam the log).
call_->Receiver()->DeliverPacket(
webrtc::MediaType::VIDEO,
packet->cdata(), packet->size(),
webrtc_packet_time);
}
void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
call_->SignalChannelNetworkState(
webrtc::MediaType::VIDEO,
ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
}
void WebRtcVideoChannel2::OnNetworkRouteChanged(
const std::string& transport_name,
const rtc::NetworkRoute& network_route) {
call_->OnNetworkRouteChanged(transport_name, network_route);
}
void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) {
MediaChannel::SetInterface(iface);
// Set the RTP recv/send buffer to a bigger size
MediaChannel::SetOption(NetworkInterface::ST_RTP,
rtc::Socket::OPT_RCVBUF,
kVideoRtpBufferSize);
// Speculative change to increase the outbound socket buffer size.
// In b/15152257, we are seeing a significant number of packets discarded
// due to lack of socket buffer space, although it's not yet clear what the
// ideal value should be.
MediaChannel::SetOption(NetworkInterface::ST_RTP,
rtc::Socket::OPT_SNDBUF,
kVideoRtpBufferSize);
}
bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
size_t len,
const webrtc::PacketOptions& options) {
rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
rtc::PacketOptions rtc_options;
rtc_options.packet_id = options.packet_id;
return MediaChannel::SendPacket(&packet, rtc_options);
}
bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
return MediaChannel::SendRtcp(&packet, rtc::PacketOptions());
}
WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters::
VideoSendStreamParameters(
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
int max_bitrate_bps,
const rtc::Optional<VideoCodecSettings>& codec_settings)
: config(std::move(config)),
options(options),
max_bitrate_bps(max_bitrate_bps),
conference_mode(false),
codec_settings(codec_settings) {}
WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder::AllocatedEncoder(
webrtc::VideoEncoder* encoder,
webrtc::VideoCodecType type,
bool external)
: encoder(encoder),
external_encoder(nullptr),
type(type),
external(external) {
if (external) {
external_encoder = encoder;
this->encoder =
new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder);
}
}
WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
WebRtcVideoEncoderFactory* external_encoder_factory,
bool enable_cpu_overuse_detection,
int max_bitrate_bps,
const rtc::Optional<VideoCodecSettings>& codec_settings,
const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
// TODO(deadbeef): Don't duplicate information between send_params,
// rtp_extensions, options, etc.
const VideoSendParameters& send_params)
: worker_thread_(rtc::Thread::Current()),
ssrcs_(sp.ssrcs),
ssrc_groups_(sp.ssrc_groups),
call_(call),
cpu_restricted_counter_(0),
number_of_cpu_adapt_changes_(0),
frame_count_(0),
cpu_restricted_frame_count_(0),
source_(nullptr),
external_encoder_factory_(external_encoder_factory),
stream_(nullptr),
encoder_sink_(nullptr),
parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
rtp_parameters_(CreateRtpParametersWithOneEncoding()),
allocated_encoder_(nullptr, webrtc::kVideoCodecUnknown, false),
sending_(false),
last_frame_timestamp_us_(0) {
parameters_.config.rtp.max_packet_size = kVideoMtu;
parameters_.conference_mode = send_params.conference_mode;
sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
&parameters_.config.rtp.rtx.ssrcs);
parameters_.config.rtp.c_name = sp.cname;
if (rtp_extensions) {
parameters_.config.rtp.extensions = *rtp_extensions;
}
parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound;
parameters_.config.overuse_callback =
enable_cpu_overuse_detection ? this : nullptr;
// Only request rotation at the source when we positively know that the remote
// side doesn't support the rotation extension. This allows us to prepare the
// encoder in the expectation that rotation is supported - which is the common
// case.
sink_wants_.rotation_applied =
rtp_extensions &&
!ContainsHeaderExtension(*rtp_extensions,
webrtc::RtpExtension::kVideoRotationUri);
if (codec_settings) {
SetCodec(*codec_settings);
}
}
WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
DisconnectSource();
if (stream_ != NULL) {
call_->DestroyVideoSendStream(stream_);
}
DestroyVideoEncoder(&allocated_encoder_);
UpdateHistograms();
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::UpdateHistograms() const {
const int kMinRequiredFrames = 200;
if (frame_count_ > kMinRequiredFrames) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.CpuLimitedResolutionInPercent",
cpu_restricted_frame_count_ * 100 / frame_count_);
}
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame(
const VideoFrame& frame) {
TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::OnFrame");
webrtc::VideoFrame video_frame(frame.video_frame_buffer(),
frame.rotation(),
frame.timestamp_us());
rtc::CritScope cs(&lock_);
if (video_frame.width() != last_frame_info_.width ||
video_frame.height() != last_frame_info_.height ||
video_frame.rotation() != last_frame_info_.rotation ||
video_frame.is_texture() != last_frame_info_.is_texture) {
last_frame_info_.width = video_frame.width();
last_frame_info_.height = video_frame.height();
last_frame_info_.rotation = video_frame.rotation();
last_frame_info_.is_texture = video_frame.is_texture();
LOG(LS_INFO) << "Video frame parameters changed: dimensions="
<< last_frame_info_.width << "x" << last_frame_info_.height
<< ", rotation=" << last_frame_info_.rotation
<< ", texture=" << last_frame_info_.is_texture;
}
if (encoder_sink_ == NULL) {
// Frame input before send codecs are configured, dropping frame.
return;
}
last_frame_timestamp_us_ = video_frame.timestamp_us();
++frame_count_;
if (cpu_restricted_counter_ > 0)
++cpu_restricted_frame_count_;
// Forward frame to the encoder regardless if we are sending or not. This is
// to ensure that the encoder can be reconfigured with the correct frame size
// as quickly as possible.
encoder_sink_->OnFrame(video_frame);
}
bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoSend(
bool enable,
const VideoOptions* options,
rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetVideoSend");
RTC_DCHECK_RUN_ON(&thread_checker_);
// Ignore |options| pointer if |enable| is false.
bool options_present = enable && options;
bool source_changing = source_ != source;
if (source_changing) {
DisconnectSource();
}
if (options_present) {
VideoOptions old_options = parameters_.options;
parameters_.options.SetAll(*options);
if (parameters_.options != old_options) {
ReconfigureEncoder();
}
}
if (source_changing) {
rtc::CritScope cs(&lock_);
if (source == nullptr && encoder_sink_ != nullptr &&
last_frame_info_.width > 0) {
LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
// Force this black frame not to be dropped due to timestamp order
// check. As IncomingCapturedFrame will drop the frame if this frame's
// timestamp is less than or equal to last frame's timestamp, it is
// necessary to give this black frame a larger timestamp than the
// previous one.
last_frame_timestamp_us_ += rtc::kNumMicrosecsPerMillisec;
rtc::scoped_refptr<webrtc::I420Buffer> black_buffer(
webrtc::I420Buffer::Create(last_frame_info_.width,
last_frame_info_.height));
black_buffer->SetToBlack();
encoder_sink_->OnFrame(webrtc::VideoFrame(
black_buffer, last_frame_info_.rotation, last_frame_timestamp_us_));
}
source_ = source;
}
if (source_changing && source_) {
// |source_->AddOrUpdateSink| may not be called while holding |lock_| since
// that might cause a lock order inversion.
source_->AddOrUpdateSink(this, sink_wants_);
}
return true;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectSource() {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (source_ == nullptr) {
return;
}
// |source_->RemoveSink| may not be called while holding |lock_| since
// that might cause a lock order inversion.
source_->RemoveSink(this);
source_ = nullptr;
// Reset |cpu_restricted_counter_| if the source is changed. It is not
// possible to know if the video resolution is restricted by CPU usage after
// the source is changed since the next source might be screen capture
// with another resolution and frame rate.
cpu_restricted_counter_ = 0;
}
const std::vector<uint32_t>&
WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
return ssrcs_;
}
webrtc::VideoCodecType CodecTypeFromName(const std::string& name) {
if (CodecNamesEq(name, kVp8CodecName)) {
return webrtc::kVideoCodecVP8;
} else if (CodecNamesEq(name, kVp9CodecName)) {
return webrtc::kVideoCodecVP9;
} else if (CodecNamesEq(name, kH264CodecName)) {
return webrtc::kVideoCodecH264;
}
return webrtc::kVideoCodecUnknown;
}
WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder(
const VideoCodec& codec) {
RTC_DCHECK_RUN_ON(&thread_checker_);
webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
// Do not re-create encoders of the same type.
if (type == allocated_encoder_.type && allocated_encoder_.encoder != NULL) {
return allocated_encoder_;
}
if (external_encoder_factory_ != NULL) {
webrtc::VideoEncoder* encoder =
external_encoder_factory_->CreateVideoEncoder(type);
if (encoder != NULL) {
return AllocatedEncoder(encoder, type, true);
}
}
if (type == webrtc::kVideoCodecVP8) {
return AllocatedEncoder(
webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp8), type, false);
} else if (type == webrtc::kVideoCodecVP9) {
return AllocatedEncoder(
webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp9), type, false);
} else if (type == webrtc::kVideoCodecH264) {
return AllocatedEncoder(
webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kH264), type, false);
}
// This shouldn't happen, we should not be trying to create something we don't
// support.
RTC_DCHECK(false);
return AllocatedEncoder(NULL, webrtc::kVideoCodecUnknown, false);
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
AllocatedEncoder* encoder) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (encoder->external) {
external_encoder_factory_->DestroyVideoEncoder(encoder->external_encoder);
}
delete encoder->encoder;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec(
const VideoCodecSettings& codec_settings) {
RTC_DCHECK_RUN_ON(&thread_checker_);
parameters_.encoder_config = CreateVideoEncoderConfig(codec_settings.codec);
RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0u);
AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec);
parameters_.config.encoder_settings.encoder = new_encoder.encoder;
parameters_.config.encoder_settings.full_overuse_time = new_encoder.external;
parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
if (new_encoder.external) {
webrtc::VideoCodecType type = CodecTypeFromName(codec_settings.codec.name);
parameters_.config.encoder_settings.internal_source =
external_encoder_factory_->EncoderTypeHasInternalSource(type);
}
parameters_.config.rtp.ulpfec = codec_settings.ulpfec;
// Set RTX payload type if RTX is enabled.
if (!parameters_.config.rtp.rtx.ssrcs.empty()) {
if (codec_settings.rtx_payload_type == -1) {
LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
"payload type. Ignoring.";
parameters_.config.rtp.rtx.ssrcs.clear();
} else {
parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
}
}
parameters_.config.rtp.nack.rtp_history_ms =
HasNack(codec_settings.codec) ? kNackHistoryMs : 0;
parameters_.codec_settings =
rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(codec_settings);
LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetCodec.";
RecreateWebRtcStream();
if (allocated_encoder_.encoder != new_encoder.encoder) {
DestroyVideoEncoder(&allocated_encoder_);
allocated_encoder_ = new_encoder;
}
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
const ChangedSendParameters& params) {
RTC_DCHECK_RUN_ON(&thread_checker_);
// |recreate_stream| means construction-time parameters have changed and the
// sending stream needs to be reset with the new config.
bool recreate_stream = false;
if (params.rtcp_mode) {
parameters_.config.rtp.rtcp_mode = *params.rtcp_mode;
recreate_stream = true;
}
if (params.rtp_header_extensions) {
parameters_.config.rtp.extensions = *params.rtp_header_extensions;
recreate_stream = true;
}
if (params.max_bandwidth_bps) {
parameters_.max_bitrate_bps = *params.max_bandwidth_bps;
ReconfigureEncoder();
}
if (params.conference_mode) {
parameters_.conference_mode = *params.conference_mode;
}
// Set codecs and options.
if (params.codec) {
SetCodec(*params.codec);
recreate_stream = false; // SetCodec has already recreated the stream.
} else if (params.conference_mode && parameters_.codec_settings) {
SetCodec(*parameters_.codec_settings);
recreate_stream = false; // SetCodec has already recreated the stream.
}
if (recreate_stream) {
LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
RecreateWebRtcStream();
}
// |source_->AddOrUpdateSink| may not be called while holding |lock_| since
// that might cause a lock order inversion.
if (params.rtp_header_extensions) {
sink_wants_.rotation_applied = !ContainsHeaderExtension(
*params.rtp_header_extensions, webrtc::RtpExtension::kVideoRotationUri);
if (source_) {
source_->AddOrUpdateSink(this, sink_wants_);
}
}
}
bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpParameters(
const webrtc::RtpParameters& new_parameters) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!ValidateRtpParameters(new_parameters)) {
return false;
}
bool reconfigure_encoder = new_parameters.encodings[0].max_bitrate_bps !=
rtp_parameters_.encodings[0].max_bitrate_bps;
rtp_parameters_ = new_parameters;
// Codecs are currently handled at the WebRtcVideoChannel2 level.
rtp_parameters_.codecs.clear();
if (reconfigure_encoder) {
ReconfigureEncoder();
}
// Encoding may have been activated/deactivated.
UpdateSendState();
return true;
}
webrtc::RtpParameters
WebRtcVideoChannel2::WebRtcVideoSendStream::GetRtpParameters() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
return rtp_parameters_;
}
bool WebRtcVideoChannel2::WebRtcVideoSendStream::ValidateRtpParameters(
const webrtc::RtpParameters& rtp_parameters) {
if (rtp_parameters.encodings.size() != 1) {
LOG(LS_ERROR)
<< "Attempted to set RtpParameters without exactly one encoding";
return false;
}
return true;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::UpdateSendState() {
RTC_DCHECK_RUN_ON(&thread_checker_);
// TODO(deadbeef): Need to handle more than one encoding in the future.
RTC_DCHECK(rtp_parameters_.encodings.size() == 1u);
if (sending_ && rtp_parameters_.encodings[0].active) {
RTC_DCHECK(stream_ != nullptr);
stream_->Start();
} else {
if (stream_ != nullptr) {
stream_->Stop();
}
}
}
webrtc::VideoEncoderConfig
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
const VideoCodec& codec) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
webrtc::VideoEncoderConfig encoder_config;
bool is_screencast = parameters_.options.is_screencast.value_or(false);
if (is_screencast) {
encoder_config.min_transmit_bitrate_bps =
1000 * parameters_.options.screencast_min_bitrate_kbps.value_or(0);
encoder_config.content_type =
webrtc::VideoEncoderConfig::ContentType::kScreen;
} else {
encoder_config.min_transmit_bitrate_bps = 0;
encoder_config.content_type =
webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;
}
// By default, the stream count for the codec configuration should match the
// number of negotiated ssrcs. But if the codec is blacklisted for simulcast
// or a screencast, only configure a single stream.
encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size();
if (IsCodecBlacklistedForSimulcast(codec.name) || is_screencast) {
encoder_config.number_of_streams = 1;
}
int stream_max_bitrate =
MinPositive(rtp_parameters_.encodings[0].max_bitrate_bps,
parameters_.max_bitrate_bps);
int codec_max_bitrate_kbps;
if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps)) {
stream_max_bitrate = codec_max_bitrate_kbps * 1000;
}
encoder_config.max_bitrate_bps = stream_max_bitrate;
int max_qp = kDefaultQpMax;
codec.GetParam(kCodecParamMaxQuantization, &max_qp);
int max_framerate =
codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate;
encoder_config.video_stream_factory =
new rtc::RefCountedObject<EncoderStreamFactory>(
codec.name, max_qp, max_framerate, is_screencast,
parameters_.conference_mode);
return encoder_config;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::ReconfigureEncoder() {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!stream_) {
// The webrtc::VideoSendStream |stream_|has not yet been created but other
// parameters has changed.
return;
}
RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0u);
RTC_CHECK(parameters_.codec_settings);
VideoCodecSettings codec_settings = *parameters_.codec_settings;
webrtc::VideoEncoderConfig encoder_config =
CreateVideoEncoderConfig(codec_settings.codec);
encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
codec_settings.codec);
stream_->ReconfigureVideoEncoder(encoder_config.Copy());
encoder_config.encoder_specific_settings = NULL;
parameters_.encoder_config = std::move(encoder_config);
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSend(bool send) {
RTC_DCHECK_RUN_ON(&thread_checker_);
sending_ = send;
UpdateSendState();
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::AddOrUpdateSink(
VideoSinkInterface<webrtc::VideoFrame>* sink,
const rtc::VideoSinkWants& wants) {
// TODO(perkj): Actually consider the encoder |wants| and remove
// WebRtcVideoSendStream::OnLoadUpdate(Load load).
rtc::CritScope cs(&lock_);
RTC_DCHECK(!encoder_sink_ || encoder_sink_ == sink);
encoder_sink_ = sink;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::RemoveSink(
VideoSinkInterface<webrtc::VideoFrame>* sink) {
rtc::CritScope cs(&lock_);
RTC_DCHECK_EQ(encoder_sink_, sink);
encoder_sink_ = nullptr;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) {
if (worker_thread_ != rtc::Thread::Current()) {
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
rtc::Bind(&WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate,
this, load));
return;
}
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!source_) {
return;
}
LOG(LS_INFO) << "OnLoadUpdate " << load << ", is_screencast: "
<< (parameters_.options.is_screencast
? (*parameters_.options.is_screencast ? "true" : "false")
: "unset");
// Do not adapt resolution for screen content as this will likely result in
// blurry and unreadable text.
if (parameters_.options.is_screencast.value_or(false))
return;
rtc::Optional<int> max_pixel_count;
rtc::Optional<int> max_pixel_count_step_up;
if (load == kOveruse) {
rtc::CritScope cs(&lock_);
if (cpu_restricted_counter_ >= kMaxCpuDowngrades) {
return;
}
// The input video frame size will have a resolution with less than or
// equal to |max_pixel_count| depending on how the source can scale the
// input frame size.
max_pixel_count = rtc::Optional<int>(
(last_frame_info_.height * last_frame_info_.width * 3) / 5);
// Increase |number_of_cpu_adapt_changes_| if
// sink_wants_.max_pixel_count will be changed since
// last time |source_->AddOrUpdateSink| was called. That is, this will
// result in a new request for the source to change resolution.
if (!sink_wants_.max_pixel_count ||
*sink_wants_.max_pixel_count > *max_pixel_count) {
++number_of_cpu_adapt_changes_;
++cpu_restricted_counter_;
}
} else {
RTC_DCHECK(load == kUnderuse);
rtc::CritScope cs(&lock_);
// The input video frame size will have a resolution with "one step up"
// pixels than |max_pixel_count_step_up| where "one step up" depends on
// how the source can scale the input frame size.
max_pixel_count_step_up =
rtc::Optional<int>(last_frame_info_.height * last_frame_info_.width);
// Increase |number_of_cpu_adapt_changes_| if
// sink_wants_.max_pixel_count_step_up will be changed since
// last time |source_->AddOrUpdateSink| was called. That is, this will
// result in a new request for the source to change resolution.
if (sink_wants_.max_pixel_count ||
(sink_wants_.max_pixel_count_step_up &&
*sink_wants_.max_pixel_count_step_up < *max_pixel_count_step_up)) {
++number_of_cpu_adapt_changes_;
--cpu_restricted_counter_;
}
}
sink_wants_.max_pixel_count = max_pixel_count;
sink_wants_.max_pixel_count_step_up = max_pixel_count_step_up;
// |source_->AddOrUpdateSink| may not be called while holding |lock_| since
// that might cause a lock order inversion.
source_->AddOrUpdateSink(this, sink_wants_);
}
VideoSenderInfo WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo(
bool log_stats) {
VideoSenderInfo info;
RTC_DCHECK_RUN_ON(&thread_checker_);
for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
info.add_ssrc(ssrc);
if (parameters_.codec_settings)
info.codec_name = parameters_.codec_settings->codec.name;
if (stream_ == NULL)
return info;
webrtc::VideoSendStream::Stats stats = stream_->GetStats();
if (log_stats)
LOG(LS_INFO) << stats.ToString(rtc::TimeMillis());
info.adapt_changes = number_of_cpu_adapt_changes_;
info.adapt_reason =
cpu_restricted_counter_ <= 0 ? ADAPTREASON_NONE : ADAPTREASON_CPU;
// Get bandwidth limitation info from stream_->GetStats().
// Input resolution (output from video_adapter) can be further scaled down or
// higher video layer(s) can be dropped due to bitrate constraints.
// Note, adapt_changes only include changes from the video_adapter.
if (stats.bw_limited_resolution)
info.adapt_reason |= ADAPTREASON_BANDWIDTH;
info.encoder_implementation_name = stats.encoder_implementation_name;
info.ssrc_groups = ssrc_groups_;
info.framerate_input = stats.input_frame_rate;
info.framerate_sent = stats.encode_frame_rate;
info.avg_encode_ms = stats.avg_encode_time_ms;
info.encode_usage_percent = stats.encode_usage_percent;
info.nominal_bitrate = stats.media_bitrate_bps;
info.preferred_bitrate = stats.preferred_media_bitrate_bps;
info.send_frame_width = 0;
info.send_frame_height = 0;
for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
stats.substreams.begin();
it != stats.substreams.end(); ++it) {
// TODO(pbos): Wire up additional stats, such as padding bytes.
webrtc::VideoSendStream::StreamStats stream_stats = it->second;
info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
stream_stats.rtp_stats.transmitted.header_bytes +
stream_stats.rtp_stats.transmitted.padding_bytes;
info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
info.packets_lost += stream_stats.rtcp_stats.cumulative_lost;
if (stream_stats.width > info.send_frame_width)
info.send_frame_width = stream_stats.width;
if (stream_stats.height > info.send_frame_height)
info.send_frame_height = stream_stats.height;
info.firs_rcvd += stream_stats.rtcp_packet_type_counts.fir_packets;
info.nacks_rcvd += stream_stats.rtcp_packet_type_counts.nack_packets;
info.plis_rcvd += stream_stats.rtcp_packet_type_counts.pli_packets;
}
if (!stats.substreams.empty()) {
// TODO(pbos): Report fraction lost per SSRC.
webrtc::VideoSendStream::StreamStats first_stream_stats =
stats.substreams.begin()->second;
info.fraction_lost =
static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
(1 << 8);
}
return info;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBandwidthEstimationInfo(
BandwidthEstimationInfo* bwe_info) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (stream_ == NULL) {
return;
}
webrtc::VideoSendStream::Stats stats = stream_->GetStats();
for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
stats.substreams.begin();
it != stats.substreams.end(); ++it) {
bwe_info->transmit_bitrate += it->second.total_bitrate_bps;
bwe_info->retransmit_bitrate += it->second.retransmit_bitrate_bps;
}
bwe_info->target_enc_bitrate += stats.target_media_bitrate_bps;
bwe_info->actual_enc_bitrate += stats.media_bitrate_bps;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (stream_ != NULL) {
call_->DestroyVideoSendStream(stream_);
}
RTC_CHECK(parameters_.codec_settings);
RTC_DCHECK_EQ((parameters_.encoder_config.content_type ==
webrtc::VideoEncoderConfig::ContentType::kScreen),
parameters_.options.is_screencast.value_or(false))
<< "encoder content type inconsistent with screencast option";
parameters_.encoder_config.encoder_specific_settings =
ConfigureVideoEncoderSettings(parameters_.codec_settings->codec);
webrtc::VideoSendStream::Config config = parameters_.config.Copy();
if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) {
LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
"payload type the set codec. Ignoring RTX.";
config.rtp.rtx.ssrcs.clear();
}
stream_ = call_->CreateVideoSendStream(std::move(config),
parameters_.encoder_config.Copy());
stream_->SetSource(this);
parameters_.encoder_config.encoder_specific_settings = NULL;
// Call stream_->Start() if necessary conditions are met.
UpdateSendState();
}
WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoReceiveStream::Config config,
WebRtcVideoDecoderFactory* external_decoder_factory,
bool default_stream,
const std::vector<VideoCodecSettings>& recv_codecs,
bool red_disabled_by_remote_side)
: call_(call),
stream_params_(sp),
stream_(NULL),
default_stream_(default_stream),
config_(std::move(config)),
red_disabled_by_remote_side_(red_disabled_by_remote_side),
external_decoder_factory_(external_decoder_factory),
sink_(NULL),
first_frame_timestamp_(-1),
estimated_remote_start_ntp_time_ms_(0) {
config_.renderer = this;
std::vector<AllocatedDecoder> old_decoders;
ConfigureCodecs(recv_codecs, &old_decoders);
RecreateWebRtcStream();
RTC_DCHECK(old_decoders.empty());
}
WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder::
AllocatedDecoder(webrtc::VideoDecoder* decoder,
webrtc::VideoCodecType type,
bool external)
: decoder(decoder),
external_decoder(nullptr),
type(type),
external(external) {
if (external) {
external_decoder = decoder;
this->decoder =
new webrtc::VideoDecoderSoftwareFallbackWrapper(type, external_decoder);
}
}
WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() {
call_->DestroyVideoReceiveStream(stream_);
ClearDecoders(&allocated_decoders_);
}
const std::vector<uint32_t>&
WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetSsrcs() const {
return stream_params_.ssrcs;
}
rtc::Optional<uint32_t>
WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetFirstPrimarySsrc() const {
std::vector<uint32_t> primary_ssrcs;
stream_params_.GetPrimarySsrcs(&primary_ssrcs);
if (primary_ssrcs.empty()) {
LOG(LS_WARNING) << "Empty primary ssrcs vector, returning empty optional";
return rtc::Optional<uint32_t>();
} else {
return rtc::Optional<uint32_t>(primary_ssrcs[0]);
}
}
WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder
WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder(
std::vector<AllocatedDecoder>* old_decoders,
const VideoCodec& codec) {
webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
for (size_t i = 0; i < old_decoders->size(); ++i) {
if ((*old_decoders)[i].type == type) {
AllocatedDecoder decoder = (*old_decoders)[i];
(*old_decoders)[i] = old_decoders->back();
old_decoders->pop_back();
return decoder;
}
}
if (external_decoder_factory_ != NULL) {
webrtc::VideoDecoder* decoder =
external_decoder_factory_->CreateVideoDecoderWithParams(
type, {stream_params_.id});
if (decoder != NULL) {
return AllocatedDecoder(decoder, type, true);
}
}
if (type == webrtc::kVideoCodecVP8) {
return AllocatedDecoder(
webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp8), type, false);
}
if (type == webrtc::kVideoCodecVP9) {
return AllocatedDecoder(
webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp9), type, false);
}
if (type == webrtc::kVideoCodecH264) {
return AllocatedDecoder(
webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kH264), type, false);
}
return AllocatedDecoder(
webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kUnsupportedCodec),
webrtc::kVideoCodecUnknown, false);
}
void ConfigureDecoderSpecifics(webrtc::VideoReceiveStream::Decoder* decoder,
const cricket::VideoCodec& recv_video_codec) {
if (recv_video_codec.name.compare("H264") == 0) {
auto it = recv_video_codec.params.find("sprop-parameter-sets");
if (it != recv_video_codec.params.end()) {
decoder->decoder_specific.h264_extra_settings =
rtc::Optional<webrtc::VideoDecoderH264Settings>(
webrtc::VideoDecoderH264Settings());
decoder->decoder_specific.h264_extra_settings->sprop_parameter_sets =
it->second;
}
}
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ConfigureCodecs(
const std::vector<VideoCodecSettings>& recv_codecs,
std::vector<AllocatedDecoder>* old_decoders) {
*old_decoders = allocated_decoders_;
allocated_decoders_.clear();
config_.decoders.clear();
for (size_t i = 0; i < recv_codecs.size(); ++i) {
AllocatedDecoder allocated_decoder =
CreateOrReuseVideoDecoder(old_decoders, recv_codecs[i].codec);
allocated_decoders_.push_back(allocated_decoder);
webrtc::VideoReceiveStream::Decoder decoder;
decoder.decoder = allocated_decoder.decoder;
decoder.payload_type = recv_codecs[i].codec.id;
decoder.payload_name = recv_codecs[i].codec.name;
ConfigureDecoderSpecifics(&decoder, recv_codecs[i].codec);
config_.decoders.push_back(decoder);
}
// TODO(pbos): Reconfigure RTX based on incoming recv_codecs.
config_.rtp.ulpfec = recv_codecs.front().ulpfec;
config_.rtp.nack.rtp_history_ms =
HasNack(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetLocalSsrc(
uint32_t local_ssrc) {
// TODO(pbos): Consider turning this sanity check into a RTC_DCHECK. You
// should not be able to create a sender with the same SSRC as a receiver, but
// right now this can't be done due to unittests depending on receiving what
// they are sending from the same MediaChannel.
if (local_ssrc == config_.rtp.remote_ssrc) {
LOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
"unchanged; local_ssrc=" << local_ssrc;
return;
}
config_.rtp.local_ssrc = local_ssrc;
LOG(LS_INFO)
<< "RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc="
<< local_ssrc;
RecreateWebRtcStream();
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetFeedbackParameters(
bool nack_enabled,
bool remb_enabled,
bool transport_cc_enabled,
webrtc::RtcpMode rtcp_mode) {
int nack_history_ms = nack_enabled ? kNackHistoryMs : 0;
if (config_.rtp.nack.rtp_history_ms == nack_history_ms &&
config_.rtp.remb == remb_enabled &&
config_.rtp.transport_cc == transport_cc_enabled &&
config_.rtp.rtcp_mode == rtcp_mode) {
LOG(LS_INFO)
<< "Ignoring call to SetFeedbackParameters because parameters are "
"unchanged; nack="
<< nack_enabled << ", remb=" << remb_enabled
<< ", transport_cc=" << transport_cc_enabled;
return;
}
config_.rtp.remb = remb_enabled;
config_.rtp.nack.rtp_history_ms = nack_history_ms;
config_.rtp.transport_cc = transport_cc_enabled;
config_.rtp.rtcp_mode = rtcp_mode;
LOG(LS_INFO)
<< "RecreateWebRtcStream (recv) because of SetFeedbackParameters; nack="
<< nack_enabled << ", remb=" << remb_enabled
<< ", transport_cc=" << transport_cc_enabled;
RecreateWebRtcStream();
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvParameters(
const ChangedRecvParameters& params) {
bool needs_recreation = false;
std::vector<AllocatedDecoder> old_decoders;
if (params.codec_settings) {
ConfigureCodecs(*params.codec_settings, &old_decoders);
needs_recreation = true;
}
if (params.rtp_header_extensions) {
config_.rtp.extensions = *params.rtp_header_extensions;
needs_recreation = true;
}
if (needs_recreation) {
LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRecvParameters";
RecreateWebRtcStream();
ClearDecoders(&old_decoders);
}
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() {
if (stream_ != NULL) {
call_->DestroyVideoReceiveStream(stream_);
}
webrtc::VideoReceiveStream::Config config = config_.Copy();
if (red_disabled_by_remote_side_) {
config.rtp.ulpfec.red_payload_type = -1;
config.rtp.ulpfec.ulpfec_payload_type = -1;
config.rtp.ulpfec.red_rtx_payload_type = -1;
}
stream_ = call_->CreateVideoReceiveStream(std::move(config));
stream_->Start();
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders(
std::vector<AllocatedDecoder>* allocated_decoders) {
for (size_t i = 0; i < allocated_decoders->size(); ++i) {
if ((*allocated_decoders)[i].external) {
external_decoder_factory_->DestroyVideoDecoder(
(*allocated_decoders)[i].external_decoder);
}
delete (*allocated_decoders)[i].decoder;
}
allocated_decoders->clear();
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::OnFrame(
const webrtc::VideoFrame& frame) {
rtc::CritScope crit(&sink_lock_);
if (first_frame_timestamp_ < 0)
first_frame_timestamp_ = frame.timestamp();
int64_t rtp_time_elapsed_since_first_frame =
(timestamp_wraparound_handler_.Unwrap(frame.timestamp()) -
first_frame_timestamp_);
int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame /
(cricket::kVideoCodecClockrate / 1000);
if (frame.ntp_time_ms() > 0)
estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms;
if (sink_ == NULL) {
LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoSink.";
return;
}
WebRtcVideoFrame render_frame(
frame.video_frame_buffer(), frame.rotation(),
frame.render_time_ms() * rtc::kNumNanosecsPerMicrosec, frame.timestamp());
sink_->OnFrame(render_frame);
}
bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsDefaultStream() const {
return default_stream_;
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSink(
rtc::VideoSinkInterface<cricket::VideoFrame>* sink) {
rtc::CritScope crit(&sink_lock_);
sink_ = sink;
}
std::string
WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetCodecNameFromPayloadType(
int payload_type) {
for (const webrtc::VideoReceiveStream::Decoder& decoder : config_.decoders) {
if (decoder.payload_type == payload_type) {
return decoder.payload_name;
}
}
return "";
}
VideoReceiverInfo
WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
bool log_stats) {
VideoReceiverInfo info;
info.ssrc_groups = stream_params_.ssrc_groups;
info.add_ssrc(config_.rtp.remote_ssrc);
webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
info.decoder_implementation_name = stats.decoder_implementation_name;
info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
stats.rtp_stats.transmitted.header_bytes +
stats.rtp_stats.transmitted.padding_bytes;
info.packets_rcvd = stats.rtp_stats.transmitted.packets;
info.packets_lost = stats.rtcp_stats.cumulative_lost;
info.fraction_lost =
static_cast<float>(stats.rtcp_stats.fraction_lost) / (1 << 8);
info.framerate_rcvd = stats.network_frame_rate;
info.framerate_decoded = stats.decode_frame_rate;
info.framerate_output = stats.render_frame_rate;
info.frame_width = stats.width;