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webrtc / src / 06aea31d10f860ae4236e3422252557762d39188 / . / media / engine / webrtc_video_engine.cc
blob: 5b3ecae728fe40fdebdeadf0cea04b4c86d9dfba [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 "media/engine/webrtc_video_engine.h"
#include <stdio.h>
#include <algorithm>
#include <set>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "api/media_stream_interface.h"
#include "api/units/data_rate.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_codec_type.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/call.h"
#include "media/engine/simulcast.h"
#include "media/engine/webrtc_media_engine.h"
#include "media/engine/webrtc_voice_engine.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "modules/video_coding/svc/scalability_mode_util.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/field_trial_units.h"
#include "rtc_base/experiments/min_video_bitrate_experiment.h"
#include "rtc_base/experiments/normalize_simulcast_size_experiment.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
namespace cricket {
namespace {
using ::webrtc::ParseRtpPayloadType;
using ::webrtc::ParseRtpSsrc;
const int kMinLayerSize = 16;
constexpr int64_t kUnsignaledSsrcCooldownMs = rtc::kNumMillisecsPerSec / 2;
// TODO(bugs.webrtc.org/13166): Remove AV1X when backwards compatibility is not
// needed.
constexpr char kAv1xCodecName[] = "AV1X";
int ScaleDownResolution(int resolution,
double scale_down_by,
int min_resolution) {
// Resolution is never scalied down to smaller than min_resolution.
// If the input resolution is already smaller than min_resolution,
// no scaling should be done at all.
if (resolution <= min_resolution)
return resolution;
return std::max(static_cast<int>(resolution / scale_down_by + 0.5),
min_resolution);
}
const char* StreamTypeToString(
webrtc::VideoSendStream::StreamStats::StreamType type) {
switch (type) {
case webrtc::VideoSendStream::StreamStats::StreamType::kMedia:
return "kMedia";
case webrtc::VideoSendStream::StreamStats::StreamType::kRtx:
return "kRtx";
case webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec:
return "kFlexfec";
}
return nullptr;
}
bool IsEnabled(const webrtc::FieldTrialsView& trials, absl::string_view name) {
return absl::StartsWith(trials.Lookup(name), "Enabled");
}
bool IsDisabled(const webrtc::FieldTrialsView& trials, absl::string_view name) {
return absl::StartsWith(trials.Lookup(name), "Disabled");
}
bool PowerOfTwo(int value) {
return (value > 0) && ((value & (value - 1)) == 0);
}
bool IsScaleFactorsPowerOfTwo(const webrtc::VideoEncoderConfig& config) {
for (const auto& layer : config.simulcast_layers) {
double scale = std::max(layer.scale_resolution_down_by, 1.0);
if (std::round(scale) != scale || !PowerOfTwo(scale)) {
return false;
}
}
return true;
}
void AddDefaultFeedbackParams(VideoCodec* codec,
const webrtc::FieldTrialsView& trials) {
// Don't add any feedback params for RED and ULPFEC.
if (codec->name == kRedCodecName || codec->name == kUlpfecCodecName)
return;
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
codec->AddFeedbackParam(
FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
// Don't add any more feedback params for FLEXFEC.
if (codec->name == kFlexfecCodecName)
return;
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir));
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
if (codec->name == kVp8CodecName &&
IsEnabled(trials, "WebRTC-RtcpLossNotification")) {
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamLntf, kParamValueEmpty));
}
}
// Helper function to determine whether a codec should use the [35, 63] range.
// Should be used when adding new codecs (or variants).
bool IsCodecValidForLowerRange(const VideoCodec& codec) {
if (absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName) ||
absl::EqualsIgnoreCase(codec.name, kAv1CodecName) ||
absl::EqualsIgnoreCase(codec.name, kAv1xCodecName)) {
return true;
} else if (absl::EqualsIgnoreCase(codec.name, kH264CodecName)) {
std::string profileLevelId;
std::string packetizationMode;
if (codec.GetParam(kH264FmtpProfileLevelId, &profileLevelId)) {
if (absl::StartsWithIgnoreCase(profileLevelId, "4d00")) {
if (codec.GetParam(kH264FmtpPacketizationMode, &packetizationMode)) {
return packetizationMode == "0";
}
}
// H264 with YUV444.
return absl::StartsWithIgnoreCase(profileLevelId, "f400");
}
}
return false;
}
// This function will assign dynamic payload types (in the range [96, 127]
// and then [35, 63]) to the input codecs, and also add ULPFEC, RED, FlexFEC,
// and associated RTX codecs for recognized codecs (VP8, VP9, H264, and RED).
// It will also add default feedback params to the codecs.
// is_decoder_factory is needed to keep track of the implict assumption that any
// H264 decoder also supports constrained base line profile.
// Also, is_decoder_factory is used to decide whether FlexFEC video format
// should be advertised as supported.
// TODO(kron): Perhaps it is better to move the implicit knowledge to the place
// where codecs are negotiated.
template <class T>
std::vector<VideoCodec> GetPayloadTypesAndDefaultCodecs(
const T* factory,
bool is_decoder_factory,
bool include_rtx,
const webrtc::FieldTrialsView& trials) {
if (!factory) {
return {};
}
std::vector<webrtc::SdpVideoFormat> supported_formats =
factory->GetSupportedFormats();
if (is_decoder_factory) {
AddH264ConstrainedBaselineProfileToSupportedFormats(&supported_formats);
}
if (supported_formats.empty())
return std::vector<VideoCodec>();
supported_formats.push_back(webrtc::SdpVideoFormat(kRedCodecName));
supported_formats.push_back(webrtc::SdpVideoFormat(kUlpfecCodecName));
// flexfec-03 is supported as
// - receive codec unless WebRTC-FlexFEC-03-Advertised is disabled
// - send codec if WebRTC-FlexFEC-03-Advertised is enabled
if ((is_decoder_factory &&
!IsDisabled(trials, "WebRTC-FlexFEC-03-Advertised")) ||
(!is_decoder_factory &&
IsEnabled(trials, "WebRTC-FlexFEC-03-Advertised"))) {
webrtc::SdpVideoFormat flexfec_format(kFlexfecCodecName);
// This value is currently arbitrarily set to 10 seconds. (The unit
// is microseconds.) This parameter MUST be present in the SDP, but
// we never use the actual value anywhere in our code however.
// TODO(brandtr): Consider honouring this value in the sender and receiver.
flexfec_format.parameters = {{kFlexfecFmtpRepairWindow, "10000000"}};
supported_formats.push_back(flexfec_format);
}
// Due to interoperability issues with old Chrome/WebRTC versions that
// ignore the [35, 63] range prefer the lower range for new codecs.
static const int kFirstDynamicPayloadTypeLowerRange = 35;
static const int kLastDynamicPayloadTypeLowerRange = 63;
static const int kFirstDynamicPayloadTypeUpperRange = 96;
static const int kLastDynamicPayloadTypeUpperRange = 127;
int payload_type_upper = kFirstDynamicPayloadTypeUpperRange;
int payload_type_lower = kFirstDynamicPayloadTypeLowerRange;
std::vector<VideoCodec> output_codecs;
for (const webrtc::SdpVideoFormat& format : supported_formats) {
VideoCodec codec(format);
bool isFecCodec = absl::EqualsIgnoreCase(codec.name, kUlpfecCodecName) ||
absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName);
// Check if we ran out of payload types.
if (payload_type_lower > kLastDynamicPayloadTypeLowerRange) {
// TODO(https://bugs.chromium.org/p/webrtc/issues/detail?id=12248):
// return an error.
RTC_LOG(LS_ERROR) << "Out of dynamic payload types [35,63] after "
"fallback from [96, 127], skipping the rest.";
RTC_DCHECK_EQ(payload_type_upper, kLastDynamicPayloadTypeUpperRange);
break;
}
// Lower range gets used for "new" codecs or when running out of payload
// types in the upper range.
if (IsCodecValidForLowerRange(codec) ||
payload_type_upper >= kLastDynamicPayloadTypeUpperRange) {
codec.id = payload_type_lower++;
} else {
codec.id = payload_type_upper++;
}
AddDefaultFeedbackParams(&codec, trials);
output_codecs.push_back(codec);
// Add associated RTX codec for non-FEC codecs.
if (include_rtx) {
if (!isFecCodec) {
// Check if we ran out of payload types.
if (payload_type_lower > kLastDynamicPayloadTypeLowerRange) {
// TODO(https://bugs.chromium.org/p/webrtc/issues/detail?id=12248):
// return an error.
RTC_LOG(LS_ERROR) << "Out of dynamic payload types [35,63] after "
"fallback from [96, 127], skipping the rest.";
RTC_DCHECK_EQ(payload_type_upper, kLastDynamicPayloadTypeUpperRange);
break;
}
if (IsCodecValidForLowerRange(codec) ||
payload_type_upper >= kLastDynamicPayloadTypeUpperRange) {
output_codecs.push_back(
VideoCodec::CreateRtxCodec(payload_type_lower++, codec.id));
} else {
output_codecs.push_back(
VideoCodec::CreateRtxCodec(payload_type_upper++, codec.id));
}
}
}
}
return output_codecs;
}
bool IsTemporalLayersSupported(const std::string& codec_name) {
return absl::EqualsIgnoreCase(codec_name, kVp8CodecName) ||
absl::EqualsIgnoreCase(codec_name, kVp9CodecName) ||
absl::EqualsIgnoreCase(codec_name, kAv1CodecName);
}
static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) {
rtc::StringBuilder out;
out << "{";
for (size_t i = 0; i < codecs.size(); ++i) {
out << codecs[i].ToString();
if (i != codecs.size() - 1) {
out << ", ";
}
}
out << "}";
return out.Release();
}
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) {
RTC_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()) {
RTC_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) {
RTC_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()) {
RTC_LOG(LS_ERROR)
<< "RTX SSRCs exist, but don't cover all SSRCs (unsupported): "
<< sp.ToString();
return false;
}
return true;
}
// Returns true if the given codec is disallowed from doing simulcast.
bool IsCodecDisabledForSimulcast(const std::string& codec_name,
const webrtc::FieldTrialsView& trials) {
if (absl::EqualsIgnoreCase(codec_name, kVp9CodecName) ||
absl::EqualsIgnoreCase(codec_name, kAv1CodecName)) {
return true;
}
if (absl::EqualsIgnoreCase(codec_name, kH264CodecName)) {
return absl::StartsWith(trials.Lookup("WebRTC-H264Simulcast"), "Disabled");
}
return false;
}
// 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,
bool is_screenshare) {
int max_bitrate;
if (width * height <= 320 * 240) {
max_bitrate = 600;
} else if (width * height <= 640 * 480) {
max_bitrate = 1700;
} else if (width * height <= 960 * 540) {
max_bitrate = 2000;
} else {
max_bitrate = 2500;
}
if (is_screenshare)
max_bitrate = std::max(max_bitrate, 1200);
return max_bitrate;
}
// Returns its smallest positive argument. If neither argument is positive,
// returns an arbitrary nonpositive value.
int MinPositive(int a, int b) {
if (a <= 0) {
return b;
}
if (b <= 0) {
return a;
}
return std::min(a, b);
}
bool IsLayerActive(const webrtc::RtpEncodingParameters& layer) {
return layer.active &&
(!layer.max_bitrate_bps || *layer.max_bitrate_bps > 0) &&
(!layer.max_framerate || *layer.max_framerate > 0);
}
size_t FindRequiredActiveLayers(
const webrtc::VideoEncoderConfig& encoder_config) {
// Need enough layers so that at least the first active one is present.
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
if (encoder_config.simulcast_layers[i].active) {
return i + 1;
}
}
return 0;
}
int NumActiveStreams(const webrtc::RtpParameters& rtp_parameters) {
int res = 0;
for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
if (rtp_parameters.encodings[i].active) {
++res;
}
}
return res;
}
absl::optional<int> NumSpatialLayersFromEncoding(
const webrtc::RtpParameters& rtp_parameters,
size_t idx) {
if (idx >= rtp_parameters.encodings.size())
return absl::nullopt;
absl::optional<webrtc::ScalabilityMode> scalability_mode =
webrtc::ScalabilityModeFromString(
rtp_parameters.encodings[idx].scalability_mode.value_or(""));
return scalability_mode
? absl::optional<int>(
ScalabilityModeToNumSpatialLayers(*scalability_mode))
: absl::nullopt;
}
std::map<uint32_t, webrtc::VideoSendStream::StreamStats>
MergeInfoAboutOutboundRtpSubstreams(
const std::map<uint32_t, webrtc::VideoSendStream::StreamStats>&
substreams) {
std::map<uint32_t, webrtc::VideoSendStream::StreamStats> rtp_substreams;
// Add substreams for all RTP media streams.
for (const auto& pair : substreams) {
uint32_t ssrc = pair.first;
const webrtc::VideoSendStream::StreamStats& substream = pair.second;
switch (substream.type) {
case webrtc::VideoSendStream::StreamStats::StreamType::kMedia:
break;
case webrtc::VideoSendStream::StreamStats::StreamType::kRtx:
case webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec:
continue;
}
rtp_substreams.insert(std::make_pair(ssrc, substream));
}
// Complement the kMedia substream stats with the associated kRtx and kFlexfec
// substream stats.
for (const auto& pair : substreams) {
switch (pair.second.type) {
case webrtc::VideoSendStream::StreamStats::StreamType::kMedia:
continue;
case webrtc::VideoSendStream::StreamStats::StreamType::kRtx:
case webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec:
break;
}
// The associated substream is an RTX or FlexFEC substream that is
// referencing an RTP media substream.
const webrtc::VideoSendStream::StreamStats& associated_substream =
pair.second;
RTC_DCHECK(associated_substream.referenced_media_ssrc.has_value());
uint32_t media_ssrc = associated_substream.referenced_media_ssrc.value();
if (substreams.find(media_ssrc) == substreams.end()) {
RTC_LOG(LS_WARNING) << "Substream [ssrc: " << pair.first << ", type: "
<< StreamTypeToString(associated_substream.type)
<< "] is associated with a media ssrc (" << media_ssrc
<< ") that does not have StreamStats. Ignoring its "
<< "RTP stats.";
continue;
}
webrtc::VideoSendStream::StreamStats& rtp_substream =
rtp_substreams[media_ssrc];
// We only merge `rtp_stats`. All other metrics are not applicable for RTX
// and FlexFEC.
// TODO(hbos): kRtx and kFlexfec stats should use a separate struct to make
// it clear what is or is not applicable.
rtp_substream.rtp_stats.Add(associated_substream.rtp_stats);
}
return rtp_substreams;
}
} // namespace
// 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 kDefaultRtcpReceiverReportSsrc = 1;
// Minimum time interval for logging stats.
static const int64_t kStatsLogIntervalMs = 10000;
std::map<uint32_t, webrtc::VideoSendStream::StreamStats>
MergeInfoAboutOutboundRtpSubstreamsForTesting(
const std::map<uint32_t, webrtc::VideoSendStream::StreamStats>&
substreams) {
return MergeInfoAboutOutboundRtpSubstreams(substreams);
}
rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
WebRtcVideoChannel::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, or when
// disabled by field trial flag.
bool automatic_resize = !disable_automatic_resize_ && !is_screencast &&
(parameters_.config.rtp.ssrcs.size() == 1 ||
NumActiveStreams(rtp_parameters_) == 1);
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 (absl::EqualsIgnoreCase(codec.name, kH264CodecName)) {
return nullptr;
}
if (absl::EqualsIgnoreCase(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;
return rtc::make_ref_counted<
webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings);
}
if (absl::EqualsIgnoreCase(codec.name, kVp9CodecName)) {
webrtc::VideoCodecVP9 vp9_settings =
webrtc::VideoEncoder::GetDefaultVp9Settings();
vp9_settings.numberOfSpatialLayers = std::min<unsigned char>(
parameters_.config.rtp.ssrcs.size(), kConferenceMaxNumSpatialLayers);
vp9_settings.numberOfTemporalLayers =
std::min<unsigned char>(parameters_.config.rtp.ssrcs.size() > 1
? kConferenceDefaultNumTemporalLayers
: 1,
kConferenceMaxNumTemporalLayers);
// VP9 denoising is disabled by default.
vp9_settings.denoisingOn = codec_default_denoising ? true : denoising;
// Disable automatic resize if more than one spatial layer is requested.
bool vp9_automatic_resize = automatic_resize;
absl::optional<int> num_spatial_layers =
NumSpatialLayersFromEncoding(rtp_parameters_, /*idx=*/0);
if (num_spatial_layers && *num_spatial_layers > 1) {
vp9_automatic_resize = false;
}
vp9_settings.automaticResizeOn = vp9_automatic_resize;
if (!is_screencast) {
webrtc::FieldTrialFlag interlayer_pred_experiment_enabled("Enabled");
webrtc::FieldTrialEnum<webrtc::InterLayerPredMode> inter_layer_pred_mode(
"inter_layer_pred_mode", webrtc::InterLayerPredMode::kOnKeyPic,
{{"off", webrtc::InterLayerPredMode::kOff},
{"on", webrtc::InterLayerPredMode::kOn},
{"onkeypic", webrtc::InterLayerPredMode::kOnKeyPic}});
webrtc::FieldTrialFlag force_flexible_mode("FlexibleMode");
webrtc::ParseFieldTrial(
{&interlayer_pred_experiment_enabled, &inter_layer_pred_mode,
&force_flexible_mode},
call_->trials().Lookup("WebRTC-Vp9InterLayerPred"));
if (interlayer_pred_experiment_enabled) {
vp9_settings.interLayerPred = inter_layer_pred_mode;
} else {
// Limit inter-layer prediction to key pictures by default.
vp9_settings.interLayerPred = webrtc::InterLayerPredMode::kOnKeyPic;
}
vp9_settings.flexibleMode = force_flexible_mode.Get();
} else {
// Multiple spatial layers vp9 screenshare needs flexible mode.
vp9_settings.flexibleMode = vp9_settings.numberOfSpatialLayers > 1;
vp9_settings.interLayerPred = webrtc::InterLayerPredMode::kOn;
}
return rtc::make_ref_counted<
webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
}
return nullptr;
}
DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler()
: default_sink_(nullptr) {}
UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
WebRtcVideoChannel* channel,
uint32_t ssrc) {
absl::optional<uint32_t> default_recv_ssrc =
channel->GetDefaultReceiveStreamSsrc();
if (default_recv_ssrc) {
RTC_LOG(LS_INFO) << "Destroying old default receive stream for SSRC="
<< ssrc << ".";
channel->RemoveRecvStream(*default_recv_ssrc);
}
StreamParams sp = channel->unsignaled_stream_params();
sp.ssrcs.push_back(ssrc);
RTC_LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc
<< ".";
if (!channel->AddRecvStream(sp, /*default_stream=*/true)) {
RTC_LOG(LS_WARNING) << "Could not create default receive stream.";
}
// SSRC 0 returns default_recv_base_minimum_delay_ms.
const int unsignaled_ssrc = 0;
int default_recv_base_minimum_delay_ms =
channel->GetBaseMinimumPlayoutDelayMs(unsignaled_ssrc).value_or(0);
// Set base minimum delay if it was set before for the default receive stream.
channel->SetBaseMinimumPlayoutDelayMs(ssrc,
default_recv_base_minimum_delay_ms);
channel->SetSink(ssrc, default_sink_);
return kDeliverPacket;
}
rtc::VideoSinkInterface<webrtc::VideoFrame>*
DefaultUnsignalledSsrcHandler::GetDefaultSink() const {
return default_sink_;
}
void DefaultUnsignalledSsrcHandler::SetDefaultSink(
WebRtcVideoChannel* channel,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
default_sink_ = sink;
absl::optional<uint32_t> default_recv_ssrc =
channel->GetDefaultReceiveStreamSsrc();
if (default_recv_ssrc) {
channel->SetSink(*default_recv_ssrc, default_sink_);
}
}
WebRtcVideoEngine::WebRtcVideoEngine(
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
const webrtc::FieldTrialsView& trials)
: decoder_factory_(std::move(video_decoder_factory)),
encoder_factory_(std::move(video_encoder_factory)),
trials_(trials) {
RTC_DLOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine()";
}
WebRtcVideoEngine::~WebRtcVideoEngine() {
RTC_DLOG(LS_INFO) << "WebRtcVideoEngine::~WebRtcVideoEngine";
}
VideoMediaChannel* WebRtcVideoEngine::CreateMediaChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
RTC_LOG(LS_INFO) << "CreateMediaChannel. Options: " << options.ToString();
return new WebRtcVideoChannel(call, config, options, crypto_options,
encoder_factory_.get(), decoder_factory_.get(),
video_bitrate_allocator_factory);
}
std::vector<VideoCodec> WebRtcVideoEngine::send_codecs(bool include_rtx) const {
return GetPayloadTypesAndDefaultCodecs(encoder_factory_.get(),
/*is_decoder_factory=*/false,
include_rtx, trials_);
}
std::vector<VideoCodec> WebRtcVideoEngine::recv_codecs(bool include_rtx) const {
return GetPayloadTypesAndDefaultCodecs(decoder_factory_.get(),
/*is_decoder_factory=*/true,
include_rtx, trials_);
}
std::vector<webrtc::RtpHeaderExtensionCapability>
WebRtcVideoEngine::GetRtpHeaderExtensions() const {
std::vector<webrtc::RtpHeaderExtensionCapability> result;
int id = 1;
for (const auto& uri :
{webrtc::RtpExtension::kTimestampOffsetUri,
webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kVideoRotationUri,
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kPlayoutDelayUri,
webrtc::RtpExtension::kVideoContentTypeUri,
webrtc::RtpExtension::kVideoTimingUri,
webrtc::RtpExtension::kColorSpaceUri, webrtc::RtpExtension::kMidUri,
webrtc::RtpExtension::kRidUri, webrtc::RtpExtension::kRepairedRidUri}) {
result.emplace_back(uri, id++, webrtc::RtpTransceiverDirection::kSendRecv);
}
result.emplace_back(webrtc::RtpExtension::kGenericFrameDescriptorUri00, id++,
IsEnabled(trials_, "WebRTC-GenericDescriptorAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
result.emplace_back(
webrtc::RtpExtension::kDependencyDescriptorUri, id++,
IsEnabled(trials_, "WebRTC-DependencyDescriptorAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
result.emplace_back(
webrtc::RtpExtension::kVideoLayersAllocationUri, id++,
IsEnabled(trials_, "WebRTC-VideoLayersAllocationAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
result.emplace_back(
webrtc::RtpExtension::kVideoFrameTrackingIdUri, id++,
IsEnabled(trials_, "WebRTC-VideoFrameTrackingIdAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
return result;
}
WebRtcVideoChannel::WebRtcVideoChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoEncoderFactory* encoder_factory,
webrtc::VideoDecoderFactory* decoder_factory,
webrtc::VideoBitrateAllocatorFactory* bitrate_allocator_factory)
: VideoMediaChannel(call->network_thread(), config.enable_dscp),
worker_thread_(call->worker_thread()),
call_(call),
unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
video_config_(config.video),
encoder_factory_(encoder_factory),
decoder_factory_(decoder_factory),
bitrate_allocator_factory_(bitrate_allocator_factory),
default_send_options_(options),
last_stats_log_ms_(-1),
discard_unknown_ssrc_packets_(
IsEnabled(call_->trials(),
"WebRTC-Video-DiscardPacketsWithUnknownSsrc")),
crypto_options_(crypto_options),
unknown_ssrc_packet_buffer_(
IsEnabled(call_->trials(),
"WebRTC-Video-BufferPacketsWithUnknownSsrc")
? new UnhandledPacketsBuffer()
: nullptr) {
RTC_DCHECK_RUN_ON(&thread_checker_);
network_thread_checker_.Detach();
rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
sending_ = false;
recv_codecs_ = MapCodecs(GetPayloadTypesAndDefaultCodecs(
decoder_factory_, /*is_decoder_factory=*/true,
/*include_rtx=*/true, call_->trials()));
recv_flexfec_payload_type_ =
recv_codecs_.empty() ? 0 : recv_codecs_.front().flexfec_payload_type;
}
WebRtcVideoChannel::~WebRtcVideoChannel() {
for (auto& kv : send_streams_)
delete kv.second;
for (auto& kv : receive_streams_)
delete kv.second;
}
std::vector<WebRtcVideoChannel::VideoCodecSettings>
WebRtcVideoChannel::SelectSendVideoCodecs(
const std::vector<VideoCodecSettings>& remote_mapped_codecs) const {
std::vector<webrtc::SdpVideoFormat> sdp_formats =
encoder_factory_ ? encoder_factory_->GetImplementations()
: std::vector<webrtc::SdpVideoFormat>();
// The returned vector holds the VideoCodecSettings in term of preference.
// They are orderd by receive codec preference first and local implementation
// preference second.
std::vector<VideoCodecSettings> encoders;
for (const VideoCodecSettings& remote_codec : remote_mapped_codecs) {
for (auto format_it = sdp_formats.begin();
format_it != sdp_formats.end();) {
// For H264, we will limit the encode level to the remote offered level
// regardless if level asymmetry is allowed or not. This is strictly not
// following the spec in https://tools.ietf.org/html/rfc6184#section-8.2.2
// since we should limit the encode level to the lower of local and remote
// level when level asymmetry is not allowed.
if (format_it->IsSameCodec(
{remote_codec.codec.name, remote_codec.codec.params})) {
encoders.push_back(remote_codec);
// To allow the VideoEncoderFactory to keep information about which
// implementation to instantitate when CreateEncoder is called the two
// parmeter sets are merged.
encoders.back().codec.params.insert(format_it->parameters.begin(),
format_it->parameters.end());
format_it = sdp_formats.erase(format_it);
} else {
++format_it;
}
}
}
return encoders;
}
bool WebRtcVideoChannel::NonFlexfecReceiveCodecsHaveChanged(
std::vector<VideoCodecSettings> before,
std::vector<VideoCodecSettings> after) {
// 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;
};
absl::c_sort(before, comparison);
absl::c_sort(after, comparison);
// Changes in FlexFEC payload type are handled separately in
// WebRtcVideoChannel::GetChangedRecvParameters, so disregard FlexFEC in the
// comparison here.
return !absl::c_equal(before, after,
VideoCodecSettings::EqualsDisregardingFlexfec);
}
bool WebRtcVideoChannel::GetChangedSendParameters(
const VideoSendParameters& params,
ChangedSendParameters* changed_params) const {
if (!ValidateCodecFormats(params.codecs) ||
!ValidateRtpExtensions(params.extensions, send_rtp_extensions_)) {
return false;
}
std::vector<VideoCodecSettings> negotiated_codecs =
SelectSendVideoCodecs(MapCodecs(params.codecs));
// We should only fail here if send direction is enabled.
if (params.is_stream_active && negotiated_codecs.empty()) {
RTC_LOG(LS_ERROR) << "No video codecs supported.";
return false;
}
// Never enable sending FlexFEC, unless we are in the experiment.
if (!IsEnabled(call_->trials(), "WebRTC-FlexFEC-03")) {
for (VideoCodecSettings& codec : negotiated_codecs)
codec.flexfec_payload_type = -1;
}
if (negotiated_codecs_ != negotiated_codecs) {
if (negotiated_codecs.empty()) {
changed_params->send_codec = absl::nullopt;
} else if (send_codec_ != negotiated_codecs.front()) {
changed_params->send_codec = negotiated_codecs.front();
}
changed_params->negotiated_codecs = std::move(negotiated_codecs);
}
// Handle RTP header extensions.
if (params.extmap_allow_mixed != ExtmapAllowMixed()) {
changed_params->extmap_allow_mixed = params.extmap_allow_mixed;
}
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true,
call_->trials());
if (send_rtp_extensions_ != filtered_extensions) {
changed_params->rtp_header_extensions =
absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
}
if (params.mid != send_params_.mid) {
changed_params->mid = params.mid;
}
// Handle max bitrate.
if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps &&
params.max_bandwidth_bps >= -1) {
// 0 or -1 uncaps max bitrate.
// TODO(pbos): Reconsider how 0 should be treated. It is not mentioned as a
// special value and might very well be used for stopping sending.
changed_params->max_bandwidth_bps =
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 = params.conference_mode;
}
// Handle RTCP mode.
if (params.rtcp.reduced_size != send_params_.rtcp.reduced_size) {
changed_params->rtcp_mode = params.rtcp.reduced_size
? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound;
}
return true;
}
bool WebRtcVideoChannel::SetSendParameters(const VideoSendParameters& params) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSendParameters");
RTC_LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
ChangedSendParameters changed_params;
if (!GetChangedSendParameters(params, &changed_params)) {
return false;
}
if (changed_params.negotiated_codecs) {
for (const auto& send_codec : *changed_params.negotiated_codecs)
RTC_LOG(LS_INFO) << "Negotiated codec: " << send_codec.codec.ToString();
}
send_params_ = params;
return ApplyChangedParams(changed_params);
}
void WebRtcVideoChannel::RequestEncoderFallback() {
if (!worker_thread_->IsCurrent()) {
worker_thread_->PostTask(
ToQueuedTask(task_safety_, [this] { RequestEncoderFallback(); }));
return;
}
RTC_DCHECK_RUN_ON(&thread_checker_);
if (negotiated_codecs_.size() <= 1) {
RTC_LOG(LS_WARNING) << "Encoder failed but no fallback codec is available";
return;
}
ChangedSendParameters params;
params.negotiated_codecs = negotiated_codecs_;
params.negotiated_codecs->erase(params.negotiated_codecs->begin());
params.send_codec = params.negotiated_codecs->front();
ApplyChangedParams(params);
}
void WebRtcVideoChannel::RequestEncoderSwitch(
const webrtc::SdpVideoFormat& format,
bool allow_default_fallback) {
if (!worker_thread_->IsCurrent()) {
worker_thread_->PostTask(
ToQueuedTask(task_safety_, [this, format, allow_default_fallback] {
RequestEncoderSwitch(format, allow_default_fallback);
}));
return;
}
RTC_DCHECK_RUN_ON(&thread_checker_);
for (const VideoCodecSettings& codec_setting : negotiated_codecs_) {
if (format.IsSameCodec(
{codec_setting.codec.name, codec_setting.codec.params})) {
VideoCodecSettings new_codec_setting = codec_setting;
for (const auto& kv : format.parameters) {
new_codec_setting.codec.params[kv.first] = kv.second;
}
if (send_codec_ == new_codec_setting) {
// Already using this codec, no switch required.
return;
}
ChangedSendParameters params;
params.send_codec = new_codec_setting;
ApplyChangedParams(params);
return;
}
}
RTC_LOG(LS_WARNING) << "Failed to switch encoder to: " << format.ToString()
<< ". Is default fallback allowed: "
<< allow_default_fallback;
if (allow_default_fallback) {
RequestEncoderFallback();
}
}
bool WebRtcVideoChannel::ApplyChangedParams(
const ChangedSendParameters& changed_params) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (changed_params.negotiated_codecs)
negotiated_codecs_ = *changed_params.negotiated_codecs;
if (changed_params.send_codec)
send_codec_ = changed_params.send_codec;
if (changed_params.extmap_allow_mixed) {
SetExtmapAllowMixed(*changed_params.extmap_allow_mixed);
}
if (changed_params.rtp_header_extensions) {
send_rtp_extensions_ = *changed_params.rtp_header_extensions;
}
if (changed_params.send_codec || changed_params.max_bandwidth_bps) {
if (send_params_.max_bandwidth_bps == -1) {
// Unset the global max bitrate (max_bitrate_bps) if max_bandwidth_bps is
// -1, which corresponds to no "b=AS" attribute in SDP. Note that the
// global max bitrate may be set below in GetBitrateConfigForCodec, from
// the codec max bitrate.
// TODO(pbos): This should be reconsidered (codec max bitrate should
// probably not affect global call max bitrate).
bitrate_config_.max_bitrate_bps = -1;
}
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.send_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 (send_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
// WebRtcVideoChannel (in which case we're good), or per sender (SSRC),
// in which case this should not set a BitrateConstraints but rather
// reconfigure all senders.
bitrate_config_.max_bitrate_bps = send_params_.max_bandwidth_bps == 0
? -1
: send_params_.max_bandwidth_bps;
}
call_->GetTransportControllerSend()->SetSdpBitrateParameters(
bitrate_config_);
}
for (auto& kv : send_streams_) {
kv.second->SetSendParameters(changed_params);
}
if (changed_params.send_codec || changed_params.rtcp_mode) {
// Update receive feedback parameters from new codec or RTCP mode.
RTC_LOG(LS_INFO)
<< "SetFeedbackParameters 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(
HasLntf(send_codec_->codec), HasNack(send_codec_->codec),
HasTransportCc(send_codec_->codec),
send_params_.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound,
send_codec_->rtx_time);
}
}
return true;
}
webrtc::RtpParameters WebRtcVideoChannel::GetRtpSendParameters(
uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_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) {
if (send_codec_ && send_codec_->codec.id == codec.id) {
// Put the current send codec to the front of the codecs list.
RTC_DCHECK_EQ(codec.name, send_codec_->codec.name);
rtp_params.codecs.insert(rtp_params.codecs.begin(),
codec.ToCodecParameters());
} else {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
}
return rtp_params;
}
webrtc::RTCError WebRtcVideoChannel::SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRtpSendParameters");
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream "
"with ssrc "
<< ssrc << " which doesn't exist.";
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
// 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) {
RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
"is not currently supported.";
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
if (!parameters.encodings.empty()) {
// Note that these values come from:
// https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-16#section-5
// TODO(deadbeef): Change values depending on whether we are sending a
// keyframe or non-keyframe.
rtc::DiffServCodePoint new_dscp = rtc::DSCP_DEFAULT;
switch (parameters.encodings[0].network_priority) {
case webrtc::Priority::kVeryLow:
new_dscp = rtc::DSCP_CS1;
break;
case webrtc::Priority::kLow:
new_dscp = rtc::DSCP_DEFAULT;
break;
case webrtc::Priority::kMedium:
new_dscp = rtc::DSCP_AF42;
break;
case webrtc::Priority::kHigh:
new_dscp = rtc::DSCP_AF41;
break;
}
SetPreferredDscp(new_dscp);
}
return it->second->SetRtpParameters(parameters);
}
webrtc::RtpParameters WebRtcVideoChannel::GetRtpReceiveParameters(
uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
webrtc::RtpParameters rtp_params;
auto it = receive_streams_.find(ssrc);
if (it == receive_streams_.end()) {
RTC_LOG(LS_WARNING)
<< "Attempting to get RTP receive parameters for stream "
"with SSRC "
<< ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
rtp_params = it->second->GetRtpParameters();
// Add codecs, which any stream is prepared to receive.
for (const VideoCodec& codec : recv_params_.codecs) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
return rtp_params;
}
webrtc::RtpParameters WebRtcVideoChannel::GetDefaultRtpReceiveParameters()
const {
RTC_DCHECK_RUN_ON(&thread_checker_);
webrtc::RtpParameters rtp_params;
if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) {
// Getting parameters on a default, unsignaled video receive stream but
// because we've not configured to receive such a stream, `encodings` is
// empty.
return rtp_params;
}
rtp_params.encodings.emplace_back();
// Add codecs, which any stream is prepared to receive.
for (const VideoCodec& codec : recv_params_.codecs) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
return rtp_params;
}
bool WebRtcVideoChannel::GetChangedRecvParameters(
const VideoRecvParameters& params,
ChangedRecvParameters* changed_params) const {
if (!ValidateCodecFormats(params.codecs) ||
!ValidateRtpExtensions(params.extensions, recv_rtp_extensions_)) {
return false;
}
// Handle receive codecs.
const std::vector<VideoCodecSettings> mapped_codecs =
MapCodecs(params.codecs);
if (mapped_codecs.empty()) {
RTC_LOG(LS_ERROR)
<< "GetChangedRecvParameters called without any video codecs.";
return false;
}
// Verify that every mapped codec is supported locally.
if (params.is_stream_active) {
const std::vector<VideoCodec> local_supported_codecs =
GetPayloadTypesAndDefaultCodecs(decoder_factory_,
/*is_decoder_factory=*/true,
/*include_rtx=*/true, call_->trials());
for (const VideoCodecSettings& mapped_codec : mapped_codecs) {
if (!FindMatchingCodec(local_supported_codecs, mapped_codec.codec)) {
RTC_LOG(LS_ERROR)
<< "GetChangedRecvParameters called with unsupported video codec: "
<< mapped_codec.codec.ToString();
return false;
}
}
}
if (NonFlexfecReceiveCodecsHaveChanged(recv_codecs_, mapped_codecs)) {
changed_params->codec_settings =
absl::optional<std::vector<VideoCodecSettings>>(mapped_codecs);
}
// Handle RTP header extensions.
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false,
call_->trials());
if (filtered_extensions != recv_rtp_extensions_) {
changed_params->rtp_header_extensions =
absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
}
int flexfec_payload_type = mapped_codecs.front().flexfec_payload_type;
if (flexfec_payload_type != recv_flexfec_payload_type_) {
changed_params->flexfec_payload_type = flexfec_payload_type;
}
return true;
}
bool WebRtcVideoChannel::SetRecvParameters(const VideoRecvParameters& params) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRecvParameters");
RTC_LOG(LS_INFO) << "SetRecvParameters: " << params.ToString();
ChangedRecvParameters changed_params;
if (!GetChangedRecvParameters(params, &changed_params)) {
return false;
}
if (changed_params.flexfec_payload_type) {
RTC_DLOG(LS_INFO) << "Changing FlexFEC payload type (recv) from "
<< recv_flexfec_payload_type_ << " to "
<< *changed_params.flexfec_payload_type;
recv_flexfec_payload_type_ = *changed_params.flexfec_payload_type;
}
if (changed_params.rtp_header_extensions) {
recv_rtp_extensions_ = *changed_params.rtp_header_extensions;
}
if (changed_params.codec_settings) {
RTC_DLOG(LS_INFO) << "Changing recv codecs from "
<< CodecSettingsVectorToString(recv_codecs_) << " to "
<< CodecSettingsVectorToString(
*changed_params.codec_settings);
recv_codecs_ = *changed_params.codec_settings;
}
for (auto& kv : receive_streams_) {
kv.second->SetRecvParameters(changed_params);
}
recv_params_ = params;
return true;
}
std::string WebRtcVideoChannel::CodecSettingsVectorToString(
const std::vector<VideoCodecSettings>& codecs) {
rtc::StringBuilder 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.Release();
}
// RTC_RUN_ON(&thread_checker_)
void WebRtcVideoChannel::SetReceiverReportSsrc(uint32_t ssrc) {
if (ssrc == rtcp_receiver_report_ssrc_)
return;
rtcp_receiver_report_ssrc_ = ssrc;
for (auto& [unused, receive_stream] : receive_streams_)
receive_stream->SetLocalSsrc(ssrc);
}
bool WebRtcVideoChannel::GetSendCodec(VideoCodec* codec) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!send_codec_) {
RTC_LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
return false;
}
*codec = send_codec_->codec;
return true;
}
bool WebRtcVideoChannel::SetSend(bool send) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSend");
RTC_LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
if (send && !send_codec_) {
RTC_DLOG(LS_ERROR) << "SetSend(true) called before setting codec.";
return false;
}
for (const auto& kv : send_streams_) {
kv.second->SetSend(send);
}
sending_ = send;
return true;
}
bool WebRtcVideoChannel::SetVideoSend(
uint32_t ssrc,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "SetVideoSend");
RTC_DCHECK(ssrc != 0);
RTC_LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", options: "
<< (options ? options->ToString() : "nullptr")
<< ", source = " << (source ? "(source)" : "nullptr") << ")";
const auto& kv = send_streams_.find(ssrc);
if (kv == send_streams_.end()) {
// Allow unknown ssrc only if source is null.
RTC_CHECK(source == nullptr);
RTC_LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
return false;
}
return kv->second->SetVideoSend(options, source);
}
bool WebRtcVideoChannel::ValidateSendSsrcAvailability(
const StreamParams& sp) const {
for (uint32_t ssrc : sp.ssrcs) {
if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) {
RTC_LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc
<< "' already exists.";
return false;
}
}
return true;
}
bool WebRtcVideoChannel::ValidateReceiveSsrcAvailability(
const StreamParams& sp) const {
for (uint32_t ssrc : sp.ssrcs) {
if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) {
RTC_LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc
<< "' already exists.";
return false;
}
}
return true;
}
bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
if (!ValidateStreamParams(sp))
return false;
if (!ValidateSendSsrcAvailability(sp))
return false;
for (uint32_t used_ssrc : sp.ssrcs)
send_ssrcs_.insert(used_ssrc);
webrtc::VideoSendStream::Config config(this);
for (const RidDescription& rid : sp.rids()) {
config.rtp.rids.push_back(rid.rid);
}
config.suspend_below_min_bitrate = video_config_.suspend_below_min_bitrate;
config.periodic_alr_bandwidth_probing =
video_config_.periodic_alr_bandwidth_probing;
config.encoder_settings.experiment_cpu_load_estimator =
video_config_.experiment_cpu_load_estimator;
config.encoder_settings.encoder_factory = encoder_factory_;
config.encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_;
config.encoder_settings.encoder_switch_request_callback = this;
config.crypto_options = crypto_options_;
config.rtp.extmap_allow_mixed = ExtmapAllowMixed();
config.rtcp_report_interval_ms = video_config_.rtcp_report_interval_ms;
WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
call_, sp, std::move(config), default_send_options_,
video_config_.enable_cpu_adaptation, 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) {
SetReceiverReportSsrc(ssrc);
}
if (sending_) {
stream->SetSend(true);
}
return true;
}
bool WebRtcVideoChannel::RemoveSendStream(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
WebRtcVideoSendStream* removed_stream;
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) {
SetReceiverReportSsrc(send_streams_.empty() ? kDefaultRtcpReceiverReportSsrc
: send_streams_.begin()->first);
}
delete removed_stream;
return true;
}
void WebRtcVideoChannel::DeleteReceiveStream(
WebRtcVideoChannel::WebRtcVideoReceiveStream* stream) {
for (uint32_t old_ssrc : stream->GetSsrcs())
receive_ssrcs_.erase(old_ssrc);
delete stream;
}
bool WebRtcVideoChannel::AddRecvStream(const StreamParams& sp) {
return AddRecvStream(sp, false);
}
bool WebRtcVideoChannel::AddRecvStream(const StreamParams& sp,
bool default_stream) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "AddRecvStream"
<< (default_stream ? " (default stream)" : "") << ": "
<< sp.ToString();
if (!sp.has_ssrcs()) {
// This is a StreamParam with unsignaled SSRCs. Store it, so it can be used
// later when we know the SSRC on the first packet arrival.
unsignaled_stream_params_ = sp;
return true;
}
if (!ValidateStreamParams(sp))
return false;
for (uint32_t ssrc : sp.ssrcs) {
// 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()) {
RTC_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::VideoReceiveStreamInterface::Config config(this, decoder_factory_);
webrtc::FlexfecReceiveStream::Config flexfec_config(this);
ConfigureReceiverRtp(&config, &flexfec_config, sp);
config.crypto_options = crypto_options_;
config.enable_prerenderer_smoothing =
video_config_.enable_prerenderer_smoothing;
if (!sp.stream_ids().empty()) {
config.sync_group = sp.stream_ids()[0];
}
if (unsignaled_frame_transformer_ && !config.frame_transformer)
config.frame_transformer = unsignaled_frame_transformer_;
receive_streams_[sp.first_ssrc()] = new WebRtcVideoReceiveStream(
this, call_, sp, std::move(config), default_stream, recv_codecs_,
flexfec_config);
return true;
}
void WebRtcVideoChannel::ConfigureReceiverRtp(
webrtc::VideoReceiveStreamInterface::Config* config,
webrtc::FlexfecReceiveStream::Config* flexfec_config,
const StreamParams& sp) const {
uint32_t ssrc = sp.first_ssrc();
config->rtp.remote_ssrc = ssrc;
config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
// 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;
}
}
// 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;
// rtx-time (RFC 4588) is a declarative attribute similar to rtcp-rsize and
// determined by the sender / send codec.
if (send_codec_ && send_codec_->rtx_time != -1) {
config->rtp.nack.rtp_history_ms = send_codec_->rtx_time;
}
config->rtp.transport_cc =
send_codec_ ? HasTransportCc(send_codec_->codec) : false;
sp.GetFidSsrc(ssrc, &config->rtp.rtx_ssrc);
config->rtp.extensions = recv_rtp_extensions_;
// TODO(brandtr): Generalize when we add support for multistream protection.
flexfec_config->payload_type = recv_flexfec_payload_type_;
if (!IsDisabled(call_->trials(), "WebRTC-FlexFEC-03-Advertised") &&
sp.GetFecFrSsrc(ssrc, &flexfec_config->rtp.remote_ssrc)) {
flexfec_config->protected_media_ssrcs = {ssrc};
flexfec_config->rtp.local_ssrc = config->rtp.local_ssrc;
flexfec_config->rtcp_mode = config->rtp.rtcp_mode;
// TODO(brandtr): We should be spec-compliant and set `transport_cc` here
// based on the rtcp-fb for the FlexFEC codec, not the media codec.
flexfec_config->rtp.transport_cc = config->rtp.transport_cc;
flexfec_config->rtp.extensions = config->rtp.extensions;
}
}
bool WebRtcVideoChannel::RemoveRecvStream(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator stream =
receive_streams_.find(ssrc);
if (stream == receive_streams_.end()) {
RTC_LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc;
return false;
}
DeleteReceiveStream(stream->second);
receive_streams_.erase(stream);
return true;
}
void WebRtcVideoChannel::ResetUnsignaledRecvStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "ResetUnsignaledRecvStream.";
unsignaled_stream_params_ = StreamParams();
last_unsignalled_ssrc_creation_time_ms_ = absl::nullopt;
// Delete any created default streams. This is needed to avoid SSRC collisions
// in Call's RtpDemuxer, in the case that `this` has created a default video
// receiver, and then some other WebRtcVideoChannel gets the SSRC signaled
// in the corresponding Unified Plan "m=" section.
auto it = receive_streams_.begin();
while (it != receive_streams_.end()) {
if (it->second->IsDefaultStream()) {
DeleteReceiveStream(it->second);
receive_streams_.erase(it++);
} else {
++it;
}
}
}
void WebRtcVideoChannel::OnDemuxerCriteriaUpdatePending() {
RTC_DCHECK_RUN_ON(&thread_checker_);
++demuxer_criteria_id_;
}
void WebRtcVideoChannel::OnDemuxerCriteriaUpdateComplete() {
RTC_DCHECK_RUN_ON(&thread_checker_);
++demuxer_criteria_completed_id_;
}
bool WebRtcVideoChannel::SetSink(
uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "SetSink: ssrc:" << ssrc << " "
<< (sink ? "(ptr)" : "nullptr");
std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.find(ssrc);
if (it == receive_streams_.end()) {
return false;
}
it->second->SetSink(sink);
return true;
}
void WebRtcVideoChannel::SetDefaultSink(
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "SetDefaultSink: " << (sink ? "(ptr)" : "nullptr");
default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink);
}
bool WebRtcVideoChannel::GetStats(VideoMediaInfo* info) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "WebRtcVideoChannel::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);
FillSendAndReceiveCodecStats(info);
// TODO(holmer): We should either have rtt available as a metric on
// VideoSend/ReceiveStreams, or we should remove rtt from VideoSenderInfo.
webrtc::Call::Stats stats = call_->GetStats();
if (stats.rtt_ms != -1) {
for (size_t i = 0; i < info->senders.size(); ++i) {
info->senders[i].rtt_ms = stats.rtt_ms;
}
for (size_t i = 0; i < info->aggregated_senders.size(); ++i) {
info->aggregated_senders[i].rtt_ms = stats.rtt_ms;
}
}
if (log_stats)
RTC_LOG(LS_INFO) << stats.ToString(now_ms);
return true;
}
void WebRtcVideoChannel::FillSenderStats(VideoMediaInfo* video_media_info,
bool log_stats) {
for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end(); ++it) {
auto infos = it->second->GetPerLayerVideoSenderInfos(log_stats);
if (infos.empty())
continue;
video_media_info->aggregated_senders.push_back(
it->second->GetAggregatedVideoSenderInfo(infos));
for (auto&& info : infos) {
video_media_info->senders.push_back(info);
}
}
}
void WebRtcVideoChannel::FillReceiverStats(VideoMediaInfo* video_media_info,
bool log_stats) {
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 WebRtcVideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
RTC_DCHECK_RUN_ON(&thread_checker_);
for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream =
send_streams_.begin();
stream != send_streams_.end(); ++stream) {
stream->second->FillBitrateInfo(bwe_info);
}
}
void WebRtcVideoChannel::FillSendAndReceiveCodecStats(
VideoMediaInfo* video_media_info) {
for (const VideoCodec& codec : send_params_.codecs) {
webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
video_media_info->send_codecs.insert(
std::make_pair(codec_params.payload_type, std::move(codec_params)));
}
for (const VideoCodec& codec : recv_params_.codecs) {
webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
video_media_info->receive_codecs.insert(
std::make_pair(codec_params.payload_type, std::move(codec_params)));
}
}
void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
RTC_DCHECK_RUN_ON(&network_thread_checker_);
// TODO(bugs.webrtc.org/11993): This code is very similar to what
// WebRtcVoiceMediaChannel::OnPacketReceived does. For maintainability and
// consistency it would be good to move the interaction with call_->Receiver()
// to a common implementation and provide a callback on the worker thread
// for the exception case (DELIVERY_UNKNOWN_SSRC) and how retry is attempted.
worker_thread_->PostTask(
ToQueuedTask(task_safety_, [this, packet, packet_time_us] {
RTC_DCHECK_RUN_ON(&thread_checker_);
const webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet,
packet_time_us);
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 = ParseRtpSsrc(packet);
if (unknown_ssrc_packet_buffer_) {
unknown_ssrc_packet_buffer_->AddPacket(ssrc, packet_time_us, packet);
return;
}
if (discard_unknown_ssrc_packets_) {
return;
}
int payload_type = ParseRtpPayloadType(packet);
// 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 (FEC). 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;
}
}
if (payload_type == recv_flexfec_payload_type_) {
return;
}
// Ignore unknown ssrcs if there is a demuxer criteria update pending.
// During a demuxer update we may receive ssrcs that were recently
// removed or we may receve ssrcs that were recently configured for a
// different video channel.
if (demuxer_criteria_id_ != demuxer_criteria_completed_id_) {
return;
}
// Ignore unknown ssrcs if we recently created an unsignalled receive
// stream since this shouldn't happen frequently. Getting into a state
// of creating decoders on every packet eats up processing time (e.g.
// https://crbug.com/1069603) and this cooldown prevents that.
if (last_unsignalled_ssrc_creation_time_ms_.has_value()) {
int64_t now_ms = rtc::TimeMillis();
if (now_ms - last_unsignalled_ssrc_creation_time_ms_.value() <
kUnsignaledSsrcCooldownMs) {
// We've already created an unsignalled ssrc stream within the last
// 0.5 s, ignore with a warning.
RTC_LOG(LS_WARNING)
<< "Another unsignalled ssrc packet arrived shortly after the "
<< "creation of an unsignalled ssrc stream. Dropping packet.";
return;
}
}
// Let the unsignalled ssrc handler decide whether to drop or deliver.
switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) {
case UnsignalledSsrcHandler::kDropPacket:
return;
case UnsignalledSsrcHandler::kDeliverPacket:
break;
}
if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet,
packet_time_us) !=
webrtc::PacketReceiver::DELIVERY_OK) {
RTC_LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
}
last_unsignalled_ssrc_creation_time_ms_ = rtc::TimeMillis();
}));
}
void WebRtcVideoChannel::OnPacketSent(const rtc::SentPacket& sent_packet) {
RTC_DCHECK_RUN_ON(&network_thread_checker_);
// TODO(tommi): We shouldn't need to go through call_ to deliver this
// notification. We should already have direct access to
// video_send_delay_stats_ and transport_send_ptr_ via `stream_`.
// So we should be able to remove OnSentPacket from Call and handle this per
// channel instead. At the moment Call::OnSentPacket calls OnSentPacket for
// the video stats, for all sent packets, including audio, which causes
// unnecessary lookups.
call_->OnSentPacket(sent_packet);
}
void WebRtcVideoChannel::BackfillBufferedPackets(
rtc::ArrayView<const uint32_t> ssrcs) {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!unknown_ssrc_packet_buffer_) {
return;
}
int delivery_ok_cnt = 0;
int delivery_unknown_ssrc_cnt = 0;
int delivery_packet_error_cnt = 0;
webrtc::PacketReceiver* receiver = this->call_->Receiver();
unknown_ssrc_packet_buffer_->BackfillPackets(
ssrcs, [&](uint32_t /*ssrc*/, int64_t packet_time_us,
rtc::CopyOnWriteBuffer packet) {
switch (receiver->DeliverPacket(webrtc::MediaType::VIDEO, packet,
packet_time_us)) {
case webrtc::PacketReceiver::DELIVERY_OK:
delivery_ok_cnt++;
break;
case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC:
delivery_unknown_ssrc_cnt++;
break;
case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR:
delivery_packet_error_cnt++;
break;
}
});
rtc::StringBuilder out;
out << "[ ";
for (uint32_t ssrc : ssrcs) {
out << std::to_string(ssrc) << " ";
}
out << "]";
auto level = rtc::LS_INFO;
if (delivery_unknown_ssrc_cnt > 0 || delivery_packet_error_cnt > 0) {
level = rtc::LS_ERROR;
}
int total =
delivery_ok_cnt + delivery_unknown_ssrc_cnt + delivery_packet_error_cnt;
RTC_LOG_V(level) << "Backfilled " << total
<< " packets for ssrcs: " << out.Release()
<< " ok: " << delivery_ok_cnt
<< " error: " << delivery_packet_error_cnt
<< " unknown: " << delivery_unknown_ssrc_cnt;
}
void WebRtcVideoChannel::OnReadyToSend(bool ready) {
RTC_DCHECK_RUN_ON(&network_thread_checker_);
RTC_LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
call_->SignalChannelNetworkState(
webrtc::MediaType::VIDEO,
ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
}
void WebRtcVideoChannel::OnNetworkRouteChanged(
absl::string_view transport_name,
const rtc::NetworkRoute& network_route) {
RTC_DCHECK_RUN_ON(&network_thread_checker_);
worker_thread_->PostTask(ToQueuedTask(
task_safety_,
[this, name = std::string(transport_name), route = network_route] {
RTC_DCHECK_RUN_ON(&thread_checker_);
webrtc::RtpTransportControllerSendInterface* transport =
call_->GetTransportControllerSend();
transport->OnNetworkRouteChanged(name, route);
transport->OnTransportOverheadChanged(route.packet_overhead);
}));
}
void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {
RTC_DCHECK_RUN_ON(&network_thread_checker_);
MediaChannel::SetInterface(iface);
// Set the RTP recv/send buffer to a bigger size.
// The group should be a positive integer with an explicit size, in
// which case that is used as UDP recevie buffer size. All other values shall
// result in the default value being used.
const std::string group_name_recv_buf_size =
call_->trials().Lookup("WebRTC-IncreasedReceivebuffers");
int recv_buffer_size = kVideoRtpRecvBufferSize;
if (!group_name_recv_buf_size.empty() &&
(sscanf(group_name_recv_buf_size.c_str(), "%d", &recv_buffer_size) != 1 ||
recv_buffer_size <= 0)) {
RTC_LOG(LS_WARNING) << "Invalid receive buffer size: "
<< group_name_recv_buf_size;
recv_buffer_size = kVideoRtpRecvBufferSize;
}
MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_RCVBUF,
recv_buffer_size);
// 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.
const std::string group_name_send_buf_size =
call_->trials().Lookup("WebRTC-SendBufferSizeBytes");
int send_buffer_size = kVideoRtpSendBufferSize;
if (!group_name_send_buf_size.empty() &&
(sscanf(group_name_send_buf_size.c_str(), "%d", &send_buffer_size) != 1 ||
send_buffer_size <= 0)) {
RTC_LOG(LS_WARNING) << "Invalid send buffer size: "
<< group_name_send_buf_size;
send_buffer_size = kVideoRtpSendBufferSize;
}
MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_SNDBUF,
send_buffer_size);
}
void WebRtcVideoChannel::SetFrameDecryptor(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto matching_stream = receive_streams_.find(ssrc);
if (matching_stream != receive_streams_.end()) {
matching_stream->second->SetFrameDecryptor(frame_decryptor);
}
}
void WebRtcVideoChannel::SetFrameEncryptor(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto matching_stream = send_streams_.find(ssrc);
if (matching_stream != send_streams_.end()) {
matching_stream->second->SetFrameEncryptor(frame_encryptor);
} else {
RTC_LOG(LS_ERROR) << "No stream found to attach frame encryptor";
}
}
void WebRtcVideoChannel::SetEncoderSelector(
uint32_t ssrc,
webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto matching_stream = send_streams_.find(ssrc);
if (matching_stream != send_streams_.end()) {
matching_stream->second->SetEncoderSelector(encoder_selector);
} else {
RTC_LOG(LS_ERROR) << "No stream found to attach encoder selector";
}
}
void WebRtcVideoChannel::SetVideoCodecSwitchingEnabled(bool enabled) {
RTC_DCHECK_RUN_ON(&thread_checker_);
allow_codec_switching_ = enabled;
if (allow_codec_switching_) {
RTC_LOG(LS_INFO) << "Encoder switching enabled.";
}
}
bool WebRtcVideoChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc,
int delay_ms) {
RTC_DCHECK_RUN_ON(&thread_checker_);
absl::optional<uint32_t> default_ssrc = GetDefaultReceiveStreamSsrc();
// SSRC of 0 represents the default receive stream.
if (ssrc == 0) {
default_recv_base_minimum_delay_ms_ = delay_ms;
}
if (ssrc == 0 && !default_ssrc) {
return true;
}
if (ssrc == 0 && default_ssrc) {
ssrc = default_ssrc.value();
}
auto stream = receive_streams_.find(ssrc);
if (stream != receive_streams_.end()) {
stream->second->SetBaseMinimumPlayoutDelayMs(delay_ms);
return true;
} else {
RTC_LOG(LS_ERROR) << "No stream found to set base minimum playout delay";
return false;
}
}
absl::optional<int> WebRtcVideoChannel::GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
// SSRC of 0 represents the default receive stream.
if (ssrc == 0) {
return default_recv_base_minimum_delay_ms_;
}
auto stream = receive_streams_.find(ssrc);
if (stream != receive_streams_.end()) {
return stream->second->GetBaseMinimumPlayoutDelayMs();
} else {
RTC_LOG(LS_ERROR) << "No stream found to get base minimum playout delay";
return absl::nullopt;
}
}
absl::optional<uint32_t> WebRtcVideoChannel::GetDefaultReceiveStreamSsrc() {
RTC_DCHECK_RUN_ON(&thread_checker_);
absl::optional<uint32_t> ssrc;
for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) {
if (it->second->IsDefaultStream()) {
ssrc.emplace(it->first);
break;
}
}
return ssrc;
}
std::vector<webrtc::RtpSource> WebRtcVideoChannel::GetSources(
uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto it = receive_streams_.find(ssrc);
if (it == receive_streams_.end()) {
// TODO(bugs.webrtc.org/9781): Investigate standard compliance
// with sources for streams that has been removed.
RTC_LOG(LS_ERROR) << "Attempting to get contributing sources for SSRC:"
<< ssrc << " which doesn't exist.";
return {};
}
return it->second->GetSources();
}
bool WebRtcVideoChannel::SendRtp(const uint8_t* data,
size_t len,
const webrtc::PacketOptions& options) {
MediaChannel::SendRtp(data, len, options);
return true;
}
bool WebRtcVideoChannel::SendRtcp(const uint8_t* data, size_t len) {
MediaChannel::SendRtcp(data, len);
return true;
}
WebRtcVideoChannel::WebRtcVideoSendStream::VideoSendStreamParameters::
VideoSendStreamParameters(
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
int max_bitrate_bps,
const absl::optional<VideoCodecSettings>& codec_settings)
: config(std::move(config)),
options(options),
max_bitrate_bps(max_bitrate_bps),
conference_mode(false),
codec_settings(codec_settings) {}
WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
bool enable_cpu_overuse_detection,
int max_bitrate_bps,
const absl::optional<VideoCodecSettings>& codec_settings,
const absl::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_(call->worker_thread()),
ssrcs_(sp.ssrcs),
ssrc_groups_(sp.ssrc_groups),
call_(call),
enable_cpu_overuse_detection_(enable_cpu_overuse_detection),
source_(nullptr),
stream_(nullptr),
parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
sending_(false),
disable_automatic_resize_(
IsEnabled(call->trials(), "WebRTC-Video-DisableAutomaticResize")) {
// Maximum packet size may come in RtpConfig from external transport, for
// example from QuicTransportInterface implementation, so do not exceed
// given max_packet_size.
parameters_.config.rtp.max_packet_size =
std::min<size_t>(parameters_.config.rtp.max_packet_size, kVideoMtu);
parameters_.conference_mode = send_params.conference_mode;
sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
// ValidateStreamParams should prevent this from happening.
RTC_CHECK(!parameters_.config.rtp.ssrcs.empty());
rtp_parameters_.encodings[0].ssrc = parameters_.config.rtp.ssrcs[0];
// RTX.
sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
&parameters_.config.rtp.rtx.ssrcs);
// FlexFEC SSRCs.
// TODO(brandtr): This code needs to be generalized when we add support for
// multistream protection.
if (IsEnabled(call_->trials(), "WebRTC-FlexFEC-03")) {
uint32_t flexfec_ssrc;
bool flexfec_enabled = false;
for (uint32_t primary_ssrc : parameters_.config.rtp.ssrcs) {
if (sp.GetFecFrSsrc(primary_ssrc, &flexfec_ssrc)) {
if (flexfec_enabled) {
RTC_LOG(LS_INFO)
<< "Multiple FlexFEC streams in local SDP, but "
"our implementation only supports a single FlexFEC "
"stream. Will not enable FlexFEC for proposed "
"stream with SSRC: "
<< flexfec_ssrc << ".";
continue;
}
flexfec_enabled = true;
parameters_.config.rtp.flexfec.ssrc = flexfec_ssrc;
parameters_.config.rtp.flexfec.protected_media_ssrcs = {primary_ssrc};
}
}
}
parameters_.config.rtp.c_name = sp.cname;
if (rtp_extensions) {
parameters_.config.rtp.extensions = *rtp_extensions;
rtp_parameters_.header_extensions = *rtp_extensions;
}
parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
? webrtc::RtcpMode::kReducedSize
: webrtc::RtcpMode::kCompound;
parameters_.config.rtp.mid = send_params.mid;
rtp_parameters_.rtcp.reduced_size = send_params.rtcp.reduced_size;
if (codec_settings) {
SetCodec(*codec_settings);
}
}
WebRtcVideoChannel::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
if (stream_ != NULL) {
call_->DestroyVideoSendStream(stream_);
}
}
bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend(
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetVideoSend");
RTC_DCHECK_RUN_ON(&thread_checker_);
if (options) {
VideoOptions old_options = parameters_.options;
parameters_.options.SetAll(*options);
if (parameters_.options.is_screencast.value_or(false) !=
old_options.is_screencast.value_or(false) &&
parameters_.codec_settings) {
// If screen content settings change, we may need to recreate the codec
// instance so that the correct type is used.
SetCodec(*parameters_.codec_settings);
// Mark screenshare parameter as being updated, then test for any other
// changes that may require codec reconfiguration.
old_options.is_screencast = options->is_screencast;
}
if (parameters_.options != old_options) {
ReconfigureEncoder();
}
}
if (source_ && stream_) {
stream_->SetSource(nullptr, webrtc::DegradationPreference::DISABLED);
}
// Switch to the new source.
source_ = source;
if (source && stream_) {
stream_->SetSource(source_, GetDegradationPreference());
}
return true;
}
webrtc::DegradationPreference
WebRtcVideoChannel::WebRtcVideoSendStream::GetDegradationPreference() const {
// Do not adapt resolution for screen content as this will likely
// result in blurry and unreadable text.
// `this` acts like a VideoSource to make sure SinkWants are handled on the
// correct thread.
if (!enable_cpu_overuse_detection_) {
return webrtc::DegradationPreference::DISABLED;
}
webrtc::DegradationPreference degradation_preference;
if (rtp_parameters_.degradation_preference.has_value()) {
degradation_preference = *rtp_parameters_.degradation_preference;
} else {
if (parameters_.options.content_hint ==
webrtc::VideoTrackInterface::ContentHint::kFluid) {
degradation_preference =
webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
} else if (parameters_.options.is_screencast.value_or(false) ||
parameters_.options.content_hint ==
webrtc::VideoTrackInterface::ContentHint::kDetailed ||
parameters_.options.content_hint ==
webrtc::VideoTrackInterface::ContentHint::kText) {
degradation_preference =
webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
} else if (IsEnabled(call_->trials(), "WebRTC-Video-BalancedDegradation")) {
// Standard wants balanced by default, but it needs to be tuned first.
degradation_preference = webrtc::DegradationPreference::BALANCED;
} else {
// Keep MAINTAIN_FRAMERATE by default until BALANCED has been tuned for
// all codecs and launched.
degradation_preference =
webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
}
}
return degradation_preference;
}
const std::vector<uint32_t>&
WebRtcVideoChannel::WebRtcVideoSendStream::GetSsrcs() const {
return ssrcs_;
}
void WebRtcVideoChannel::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, 0);
parameters_.config.rtp.payload_name = codec_settings.codec.name;
parameters_.config.rtp.payload_type = codec_settings.codec.id;
parameters_.config.rtp.raw_payload =
codec_settings.codec.packetization == kPacketizationParamRaw;
parameters_.config.rtp.ulpfec = codec_settings.ulpfec;
parameters_.config.rtp.flexfec.payload_type =
codec_settings.flexfec_payload_type;
// Set RTX payload type if RTX is enabled.
if (!parameters_.config.rtp.rtx.ssrcs.empty()) {
if (codec_settings.rtx_payload_type == -1) {
RTC_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;
}
}
const bool has_lntf = HasLntf(codec_settings.codec);
parameters_.config.rtp.lntf.enabled = has_lntf;
parameters_.config.encoder_settings.capabilities.loss_notification = has_lntf;
parameters_.config.rtp.nack.rtp_history_ms =
HasNack(codec_settings.codec) ? kNackHistoryMs : 0;
parameters_.codec_settings = codec_settings;
// TODO(nisse): Avoid recreation, it should be enough to call
// ReconfigureEncoder.
RTC_LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetCodec.";
RecreateWebRtcStream();
}
void WebRtcVideoChannel::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;
rtp_parameters_.rtcp.reduced_size =
parameters_.config.rtp.rtcp_mode == webrtc::RtcpMode::kReducedSize;
recreate_stream = true;
}
if (params.extmap_allow_mixed) {
parameters_.config.rtp.extmap_allow_mixed = *params.extmap_allow_mixed;
recreate_stream = true;
}
if (params.rtp_header_extensions) {
parameters_.config.rtp.extensions = *params.rtp_header_extensions;
rtp_parameters_.header_extensions = *params.rtp_header_extensions;
recreate_stream = true;
}
if (params.mid) {
parameters_.config.rtp.mid = *params.mid;
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.send_codec) {
SetCodec(*params.send_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) {
RTC_LOG(LS_INFO)
<< "RecreateWebRtcStream (send) because of SetSendParameters";
RecreateWebRtcStream();
}
}
webrtc::RTCError WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters(
const webrtc::RtpParameters& new_parameters) {
RTC_DCHECK_RUN_ON(&thread_checker_);
webrtc::RTCError error = CheckRtpParametersInvalidModificationAndValues(
rtp_parameters_, new_parameters);
if (!error.ok()) {
return error;
}
bool new_param = false;
for (size_t i = 0; i < rtp_parameters_.encodings.size(); ++i) {
if ((new_parameters.encodings[i].min_bitrate_bps !=
rtp_parameters_.encodings[i].min_bitrate_bps) ||
(new_parameters.encodings[i].max_bitrate_bps !=
rtp_parameters_.encodings[i].max_bitrate_bps) ||
(new_parameters.encodings[i].max_framerate !=
rtp_parameters_.encodings[i].max_framerate) ||
(new_parameters.encodings[i].scale_resolution_down_by !=
rtp_parameters_.encodings[i].scale_resolution_down_by) ||
(new_parameters.encodings[i].num_temporal_layers !=
rtp_parameters_.encodings[i].num_temporal_layers)) {
new_param = true;
break;
}
}
bool new_degradation_preference = false;
if (new_parameters.degradation_preference !=
rtp_parameters_.degradation_preference) {
new_degradation_preference = true;
}
// Some fields (e.g. bitrate priority) only need to update the bitrate
// allocator which is updated via ReconfigureEncoder (however, note that the
// actual encoder should only be reconfigured if needed).
bool reconfigure_encoder = new_param ||
(new_parameters.encodings[0].bitrate_priority !=
rtp_parameters_.encodings[0].bitrate_priority) ||
new_parameters.encodings[0].scalability_mode !=
rtp_parameters_.encodings[0].scalability_mode;
// Note that the simulcast encoder adapter relies on the fact that layers
// de/activation triggers encoder reinitialization.
bool new_send_state = false;
for (size_t i = 0; i < rtp_parameters_.encodings.size(); ++i) {
bool new_active = IsLayerActive(new_parameters.encodings[i]);
bool old_active = IsLayerActive(rtp_parameters_.encodings[i]);
if (new_active != old_active) {
new_send_state = true;
}
}
rtp_parameters_ = new_parameters;
// Codecs are currently handled at the WebRtcVideoChannel level.
rtp_parameters_.codecs.clear();
if (reconfigure_encoder || new_send_state) {
ReconfigureEncoder();
}
if (new_send_state) {
UpdateSendState();
}
if (new_degradation_preference) {
if (source_ && stream_) {
stream_->SetSource(source_, GetDegradationPreference());
}
}
return webrtc::RTCError::OK();
}
webrtc::RtpParameters
WebRtcVideoChannel::WebRtcVideoSendStream::GetRtpParameters() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
return rtp_parameters_;
}
void WebRtcVideoChannel::WebRtcVideoSendStream::SetFrameEncryptor(
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor) {
RTC_DCHECK_RUN_ON(&thread_checker_);
parameters_.config.frame_encryptor = frame_encryptor;
if (stream_) {
RTC_LOG(LS_INFO)
<< "RecreateWebRtcStream (send) because of SetFrameEncryptor, ssrc="
<< parameters_.config.rtp.ssrcs[0];
RecreateWebRtcStream();
}
}
void WebRtcVideoChannel::WebRtcVideoSendStream::SetEncoderSelector(
webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
RTC_DCHECK_RUN_ON(&thread_checker_);
parameters_.config.encoder_selector = encoder_selector;
if (stream_) {
RTC_LOG(LS_INFO)
<< "RecreateWebRtcStream (send) because of SetEncoderSelector, ssrc="
<< parameters_.config.rtp.ssrcs[0];
RecreateWebRtcStream();
}
}
void WebRtcVideoChannel::WebRtcVideoSendStream::UpdateSendState() {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (sending_) {
RTC_DCHECK(stream_ != nullptr);
size_t num_layers = rtp_parameters_.encodings.size();
if (parameters_.encoder_config.number_of_streams == 1) {
// SVC is used. Only one simulcast layer is present.
num_layers = 1;
}
std::vector<bool> active_layers(num_layers);
for (size_t i = 0; i < num_layers; ++i) {
active_layers[i] = IsLayerActive(rtp_parameters_.encodings[i]);
}
if (parameters_.encoder_config.number_of_streams == 1 &&
rtp_parameters_.encodings.size() > 1) {
// SVC is used.
// The only present simulcast layer should be active if any of the
// configured SVC layers is active.
active_layers[0] =
absl::c_any_of(rtp_parameters_.encodings,
[](const auto& encoding) { return encoding.active; });
}
// This updates what simulcast layers are sending, and possibly starts
// or stops the VideoSendStream.
stream_->UpdateActiveSimulcastLayers(active_layers);
} else {
if (stream_ != nullptr) {
stream_->Stop();
}
}
}
webrtc::VideoEncoderConfig
WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(
const VideoCodec& codec) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
webrtc::VideoEncoderConfig encoder_config;
encoder_config.codec_type = webrtc::PayloadStringToCodecType(codec.name);
encoder_config.video_format =
webrtc::SdpVideoFormat(codec.name, codec.params);
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 disabled for simulcast
// or a screencast (and not in simulcast screenshare experiment), only
// configure a single stream.
encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size();
if (IsCodecDisabledForSimulcast(codec.name, call_->trials())) {
encoder_config.number_of_streams = 1;
}
// parameters_.max_bitrate comes from the max bitrate set at the SDP
// (m-section) level with the attribute "b=AS." Note that we override this
// value below if the RtpParameters max bitrate set with
// RtpSender::SetParameters has a lower value.
int stream_max_bitrate = parameters_.max_bitrate_bps;
// When simulcast is enabled (when there are multiple encodings),
// encodings[i].max_bitrate_bps will be enforced by
// encoder_config.simulcast_layers[i].max_bitrate_bps. Otherwise, it's
// enforced by stream_max_bitrate, taking the minimum of the two maximums
// (one coming from SDP, the other coming from RtpParameters).
if (rtp_parameters_.encodings[0].max_bitrate_bps &&
rtp_parameters_.encodings.size() == 1) {
stream_max_bitrate =
MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps),
parameters_.max_bitrate_bps);
}
// The codec max bitrate comes from the "x-google-max-bitrate" parameter
// attribute set in the SDP for a specific codec. As done in
// WebRtcVideoChannel::SetSendParameters, this value does not override the
// stream max_bitrate set above.
int codec_max_bitrate_kbps;
if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps) &&
stream_max_bitrate == -1) {
stream_max_bitrate = codec_max_bitrate_kbps * 1000;
}
encoder_config.max_bitrate_bps = stream_max_bitrate;
// The encoder config's default bitrate priority is set to 1.0,
// unless it is set through the sender's encoding parameters.
// The bitrate priority, which is used in the bitrate allocation, is done
// on a per sender basis, so we use the first encoding's value.
encoder_config.bitrate_priority =
rtp_parameters_.encodings[0].bitrate_priority;
// Application-controlled state is held in the encoder_config's
// simulcast_layers. Currently this is used to control which simulcast layers
// are active and for configuring the min/max bitrate and max framerate.
// The encoder_config's simulcast_layers is also used for non-simulcast (when
// there is a single layer).
RTC_DCHECK_GE(rtp_parameters_.encodings.size(),
encoder_config.number_of_streams);
RTC_DCHECK_GT(encoder_config.number_of_streams, 0);
// Copy all provided constraints.
encoder_config.simulcast_layers.resize(rtp_parameters_.encodings.size());
for (size_t i = 0; i < encoder_config.simulcast_layers.size(); ++i) {
encoder_config.simulcast_layers[i].active =
rtp_parameters_.encodings[i].active;
encoder_config.simulcast_layers[i].scalability_mode =
webrtc::ScalabilityModeFromString(
rtp_parameters_.encodings[i].scalability_mode.value_or(""));
if (rtp_parameters_.encodings[i].min_bitrate_bps) {
encoder_config.simulcast_layers[i].min_bitrate_bps =
*rtp_parameters_.encodings[i].min_bitrate_bps;
}
if (rtp_parameters_.encodings[i].max_bitrate_bps) {
encoder_config.simulcast_layers[i].max_bitrate_bps =
*rtp_parameters_.encodings[i].max_bitrate_bps;
}
if (rtp_parameters_.encodings[i].max_framerate) {
encoder_config.simulcast_layers[i].max_framerate =
*rtp_parameters_.encodings[i].max_framerate;
}
if (rtp_parameters_.encodings[i].scale_resolution_down_by) {
encoder_config.simulcast_layers[i].scale_resolution_down_by =
*rtp_parameters_.encodings[i].scale_resolution_down_by;
}
if (rtp_parameters_.encodings[i].num_temporal_layers) {
encoder_config.simulcast_layers[i].num_temporal_layers =
*rtp_parameters_.encodings[i].num_temporal_layers;
}
}
encoder_config.legacy_conference_mode = parameters_.conference_mode;
encoder_config.is_quality_scaling_allowed =
!disable_automatic_resize_ && !is_screencast &&
(parameters_.config.rtp.ssrcs.size() == 1 ||
NumActiveStreams(rtp_parameters_) == 1);
// Ensure frame dropping is always enabled.
encoder_config.frame_drop_enabled = true;
int max_qp = kDefaultQpMax;
codec.GetParam(kCodecParamMaxQuantization, &max_qp);
encoder_config.video_stream_factory =
rtc::make_ref_counted<EncoderStreamFactory>(
codec.name, max_qp, is_screencast, parameters_.conference_mode);
return encoder_config;
}
void WebRtcVideoChannel::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, 0);
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 WebRtcVideoChannel::WebRtcVideoSendStream::SetSend(bool send) {
RTC_DCHECK_RUN_ON(&thread_checker_);
sending_ = send;
UpdateSendState();
}
std::vector<VideoSenderInfo>
WebRtcVideoChannel::WebRtcVideoSendStream::GetPerLayerVideoSenderInfos(
bool log_stats) {
RTC_DCHECK_RUN_ON(&thread_checker_);
VideoSenderInfo common_info;
if (parameters_.codec_settings) {
common_info.codec_name = parameters_.codec_settings->codec.name;
common_info.codec_payload_type = parameters_.codec_settings->codec.id;
}
std::vector<VideoSenderInfo> infos;
webrtc::VideoSendStream::Stats stats;
if (stream_ == nullptr) {
for (uint32_t ssrc : parameters_.config.rtp.ssrcs) {
common_info.add_ssrc(ssrc);
}
infos.push_back(common_info);
return infos;
} else {
stats = stream_->GetStats();
if (log_stats)
RTC_LOG(LS_INFO) << stats.ToString(rtc::TimeMillis());
// Metrics that are in common for all substreams.
common_info.adapt_changes = stats.number_of_cpu_adapt_changes;
common_info.adapt_reason =
stats.cpu_limited_resolution ? ADAPTREASON_CPU : ADAPTREASON_NONE;
common_info.has_entered_low_resolution = stats.has_entered_low_resolution;
// 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)
common_info.adapt_reason |= ADAPTREASON_BANDWIDTH;
common_info.quality_limitation_reason = stats.quality_limitation_reason;
common_info.quality_limitation_durations_ms =
stats.quality_limitation_durations_ms;
common_info.quality_limitation_resolution_changes =
stats.quality_limitation_resolution_changes;
common_info.encoder_implementation_name = stats.encoder_implementation_name;
common_info.ssrc_groups = ssrc_groups_;
common_info.frames = stats.frames;
common_info.framerate_input = stats.input_frame_rate;
common_info.avg_encode_ms = stats.avg_encode_time_ms;
common_info.encode_usage_percent = stats.encode_usage_percent;
common_info.nominal_bitrate = stats.media_bitrate_bps;
common_info.content_type = stats.content_type;
common_info.aggregated_framerate_sent = stats.encode_frame_rate;
common_info.aggregated_huge_frames_sent = stats.huge_frames_sent;
// If we don't have any substreams, get the remaining metrics from `stats`.
// Otherwise, these values are obtained from `sub_stream` below.
if (stats.substreams.empty()) {
for (uint32_t ssrc : parameters_.config.rtp.ssrcs) {
common_info.add_ssrc(ssrc);
}
common_info.framerate_sent = stats.encode_frame_rate;
common_info.frames_encoded = stats.frames_encoded;
common_info.total_encode_time_ms = stats.total_encode_time_ms;
common_info.total_encoded_bytes_target = stats.total_encoded_bytes_target;
common_info.frames_sent = stats.frames_encoded;
common_info.huge_frames_sent = stats.huge_frames_sent;
infos.push_back(common_info);
return infos;
}
}
auto outbound_rtp_substreams =
MergeInfoAboutOutboundRtpSubstreams(stats.substreams);
for (const auto& pair : outbound_rtp_substreams) {
auto info = common_info;
info.add_ssrc(pair.first);
info.rid = parameters_.config.rtp.GetRidForSsrc(pair.first);
auto stream_stats = pair.second;
RTC_DCHECK_EQ(stream_stats.type,
webrtc::VideoSendStream::StreamStats::StreamType::kMedia);
info.payload_bytes_sent = stream_stats.rtp_stats.transmitted.payload_bytes;
info.header_and_padding_bytes_sent =
stream_stats.rtp_stats.transmitted.header_bytes +
stream_stats.rtp_stats.transmitted.padding_bytes;
info.packets_sent = stream_stats.rtp_stats.transmitted.packets;
info.total_packet_send_delay_ms += stream_stats.total_packet_send_delay_ms;
info.send_frame_width = stream_stats.width;
info.send_frame_height = stream_stats.height;
info.key_frames_encoded = stream_stats.frame_counts.key_frames;
info.framerate_sent = stream_stats.encode_frame_rate;
info.frames_encoded = stream_stats.frames_encoded;
info.frames_sent = stream_stats.frames_encoded;
info.retransmitted_bytes_sent =
stream_stats.rtp_stats.retransmitted.payload_bytes;
info.retransmitted_packets_sent =
stream_stats.rtp_stats.retransmitted.packets;
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 (stream_stats.report_block_data.has_value()) {
info.packets_lost =
stream_stats.report_block_data->report_block().packets_lost;
info.fraction_lost =
static_cast<float>(
stream_stats.report_block_data->report_block().fraction_lost) /
(1 << 8);
info.report_block_datas.push_back(*stream_stats.report_block_data);
}
info.qp_sum = stream_stats.qp_sum;
info.total_encode_time_ms = stream_stats.total_encode_time_ms;
info.total_encoded_bytes_target = stream_stats.total_encoded_bytes_target;
info.huge_frames_sent = stream_stats.huge_frames_sent;
infos.push_back(info);
}
return infos;
}
VideoSenderInfo
WebRtcVideoChannel::WebRtcVideoSendStream::GetAggregatedVideoSenderInfo(
const std::vector<VideoSenderInfo>& infos) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_CHECK(!infos.empty());
if (infos.size() == 1) {
return infos[0];
}
VideoSenderInfo info = infos[0];
info.local_stats.clear();
for (uint32_t ssrc : parameters_.config.rtp.ssrcs) {
info.add_ssrc(ssrc);
}
info.framerate_sent = info.aggregated_framerate_sent;
info.huge_frames_sent = info.aggregated_huge_frames_sent;
for (size_t i = 1; i < infos.size(); i++) {
info.key_frames_encoded += infos[i].key_frames_encoded;
info.payload_bytes_sent += infos[i].payload_bytes_sent;
info.header_and_padding_bytes_sent +=
infos[i].header_and_padding_bytes_sent;
info.packets_sent += infos[i].packets_sent;
info.total_packet_send_delay_ms += infos[i].total_packet_send_delay_ms;
info.retransmitted_bytes_sent += infos[i].retransmitted_bytes_sent;
info.retransmitted_packets_sent += infos[i].retransmitted_packets_sent;
info.packets_lost += infos[i].packets_lost;
if (infos[i].send_frame_width > info.send_frame_width)
info.send_frame_width = infos[i].send_frame_width;
if (infos[i].send_frame_height > info.send_frame_height)
info.send_frame_height = infos[i].send_frame_height;
info.firs_rcvd += infos[i].firs_rcvd;
info.nacks_rcvd += infos[i].nacks_rcvd;
info.plis_rcvd += infos[i].plis_rcvd;
if (infos[i].report_block_datas.size())
info.report_block_datas.push_back(infos[i].report_block_datas[0]);
if (infos[i].qp_sum) {
if (!info.qp_sum) {
info.qp_sum = 0;
}
info.qp_sum = *info.qp_sum + *infos[i].qp_sum;
}
info.frames_encoded += infos[i].frames_encoded;
info.frames_sent += infos[i].frames_sent;
info.total_encode_time_ms += infos[i].total_encode_time_ms;
info.total_encoded_bytes_target += infos[i].total_encoded_bytes_target;
}
return info;
}
void WebRtcVideoChannel::WebRtcVideoSendStream::FillBitrateInfo(
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 WebRtcVideoChannel::WebRtcVideoSendStream::
SetEncoderToPacketizerFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer) {
RTC_DCHECK_RUN_ON(&thread_checker_);
parameters_.config.frame_transformer = std::move(frame_transformer);
if (stream_)
RecreateWebRtcStream();
}
void WebRtcVideoChannel::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) {
RTC_LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
"payload type the set codec. Ignoring RTX.";
config.rtp.rtx.ssrcs.clear();
}
if (parameters_.encoder_config.number_of_streams == 1) {
// SVC is used instead of simulcast. Remove unnecessary SSRCs.
if (config.rtp.ssrcs.size() > 1) {
config.rtp.ssrcs.resize(1);
if (config.rtp.rtx.ssrcs.size() > 1) {
config.rtp.rtx.ssrcs.resize(1);
}
}
}
stream_ = call_->CreateVideoSendStream(std::move(config),
parameters_.encoder_config.Copy());
parameters_.encoder_config.encoder_specific_settings = NULL;
if (source_) {
stream_->SetSource(source_, GetDegradationPreference());
}
// Call stream_->Start() if necessary conditions are met.
UpdateSendState();
}
WebRtcVideoChannel::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
WebRtcVideoChannel* channel,
webrtc::Call* call,
const StreamParams& sp,
webrtc::VideoReceiveStreamInterface::Config config,
bool default_stream,
const std::vector<VideoCodecSettings>& recv_codecs,
const webrtc::FlexfecReceiveStream::Config& flexfec_config)
: channel_(channel),
call_(call),
stream_params_(sp),
stream_(NULL),
default_stream_(default_stream),
config_(std::move(config)),
flexfec_config_(flexfec_config),
flexfec_stream_(nullptr),
sink_(NULL),
first_frame_timestamp_(-1),
estimated_remote_start_ntp_time_ms_(0) {
RTC_DCHECK(config_.decoder_factory);
config_.renderer = this;
ConfigureCodecs(recv_codecs);
flexfec_config_.payload_type = flexfec_config.payload_type;
RecreateReceiveStream();
}
WebRtcVideoChannel::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() {
call_->DestroyVideoReceiveStream(stream_);
if (flexfec_stream_)
call_->DestroyFlexfecReceiveStream(flexfec_stream_);
}
webrtc::VideoReceiveStreamInterface&
WebRtcVideoChannel::WebRtcVideoReceiveStream::stream() {
RTC_DCHECK(stream_);
return *stream_;
}
webrtc::FlexfecReceiveStream*
WebRtcVideoChannel::WebRtcVideoReceiveStream::flexfec_stream() {
return flexfec_stream_;
}
const std::vector<uint32_t>&
WebRtcVideoChannel::WebRtcVideoReceiveStream::GetSsrcs() const {
return stream_params_.ssrcs;
}
std::vector<webrtc::RtpSource>
WebRtcVideoChannel::WebRtcVideoReceiveStream::GetSources() {
RTC_DCHECK(stream_);
return stream_->GetSources();
}
webrtc::RtpParameters
WebRtcVideoChannel::WebRtcVideoReceiveStream::GetRtpParameters() const {
webrtc::RtpParameters rtp_parameters;
std::vector<uint32_t> primary_ssrcs;
stream_params_.GetPrimarySsrcs(&primary_ssrcs);
for (uint32_t ssrc : primary_ssrcs) {
rtp_parameters.encodings.emplace_back();
rtp_parameters.encodings.back().ssrc = ssrc;
}
rtp_parameters.header_extensions = config_.rtp.extensions;
rtp_parameters.rtcp.reduced_size =
config_.rtp.rtcp_mode == webrtc::RtcpMode::kReducedSize;
return rtp_parameters;
}
bool WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureCodecs(
const std::vector<VideoCodecSettings>& recv_codecs) {
RTC_DCHECK(!recv_codecs.empty());
std::map<int, int> rtx_associated_payload_types;
std::set<int> raw_payload_types;
std::vector<webrtc::VideoReceiveStreamInterface::Decoder> decoders;
for (const auto& recv_codec : recv_codecs) {
decoders.emplace_back(
webrtc::SdpVideoFormat(recv_codec.codec.name, recv_codec.codec.params),
recv_codec.codec.id);
rtx_associated_payload_types.insert(
{recv_codec.rtx_payload_type, recv_codec.codec.id});
if (recv_codec.codec.packetization == kPacketizationParamRaw) {
raw_payload_types.insert(recv_codec.codec.id);
}
}
bool recreate_needed = (stream_ == nullptr);
const auto& codec = recv_codecs.front();
if (config_.rtp.ulpfec_payload_type != codec.ulpfec.ulpfec_payload_type) {
config_.rtp.ulpfec_payload_type = codec.ulpfec.ulpfec_payload_type;
recreate_needed = true;
}
if (config_.rtp.red_payload_type != codec.ulpfec.red_payload_type) {
config_.rtp.red_payload_type = codec.ulpfec.red_payload_type;
recreate_needed = true;
}
const bool has_lntf = HasLntf(codec.codec);
if (config_.rtp.lntf.enabled != has_lntf) {
config_.rtp.lntf.enabled = has_lntf;
recreate_needed = true;
}
const int rtp_history_ms = HasNack(codec.codec) ? kNackHistoryMs : 0;
if (rtp_history_ms != config_.rtp.nack.rtp_history_ms) {
config_.rtp.nack.rtp_history_ms = rtp_history_ms;
recreate_needed = true;
}
// The rtx-time parameter can be used to override the hardcoded default for
// the NACK buffer length.
if (codec.rtx_time != -1 && config_.rtp.nack.rtp_history_ms != 0) {
config_.rtp.nack.rtp_history_ms = codec.rtx_time;
recreate_needed = true;
}
const bool has_rtr = HasRrtr(codec.codec);
if (has_rtr != config_.rtp.rtcp_xr.receiver_reference_time_report) {
config_.rtp.rtcp_xr.receiver_reference_time_report = has_rtr;
recreate_needed = true;
}
if (codec.ulpfec.red_rtx_payload_type != -1) {
rtx_associated_payload_types[codec.ulpfec.red_rtx_payload_type] =
codec.ulpfec.red_payload_type;
}
if (config_.rtp.rtx_associated_payload_types !=
rtx_associated_payload_types) {
rtx_associated_payload_types.swap(config_.rtp.rtx_associated_payload_types);
recreate_needed = true;
}
if (raw_payload_types != config_.rtp.raw_payload_types) {
raw_payload_types.swap(config_.rtp.raw_payload_types);
recreate_needed = true;
}
if (decoders != config_.decoders) {
decoders.swap(config_.decoders);
recreate_needed = true;
}
return recreate_needed;
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetFeedbackParameters(
bool lntf_enabled,
bool nack_enabled,
bool transport_cc_enabled,
webrtc::RtcpMode rtcp_mode,
int rtx_time) {
int nack_history_ms =
nack_enabled ? rtx_time != -1 ? rtx_time : kNackHistoryMs : 0;
if (config_.rtp.lntf.enabled == lntf_enabled &&
config_.rtp.nack.rtp_history_ms == nack_history_ms &&
config_.rtp.transport_cc == transport_cc_enabled &&
config_.rtp.rtcp_mode == rtcp_mode) {
RTC_LOG(LS_INFO)
<< "Ignoring call to SetFeedbackParameters because parameters are "
"unchanged; lntf="
<< lntf_enabled << ", nack=" << nack_enabled
<< ", transport_cc=" << transport_cc_enabled
<< ", rtx_time=" << rtx_time;
return;
}
config_.rtp.lntf.enabled = lntf_enabled;
config_.rtp.nack.rtp_history_ms = nack_history_ms;
config_.rtp.transport_cc = transport_cc_enabled;
config_.rtp.rtcp_mode = rtcp_mode;
// TODO(brandtr): We should be spec-compliant and set `transport_cc` here
// based on the rtcp-fb for the FlexFEC codec, not the media codec.
flexfec_config_.rtp.transport_cc = config_.rtp.transport_cc;
flexfec_config_.rtcp_mode = config_.rtp.rtcp_mode;
RTC_LOG(LS_INFO) << "RecreateReceiveStream (recv) because of "
"SetFeedbackParameters; nack="
<< nack_enabled << ", transport_cc=" << transport_cc_enabled;
RecreateReceiveStream();
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetRecvParameters(
const ChangedRecvParameters& params) {
bool video_needs_recreation = false;
if (params.codec_settings) {
video_needs_recreation = ConfigureCodecs(*params.codec_settings);
}
if (params.rtp_header_extensions) {
if (config_.rtp.extensions != *params.rtp_header_extensions) {
config_.rtp.extensions = *params.rtp_header_extensions;
if (stream_) {
stream_->SetRtpExtensions(config_.rtp.extensions);
} else {
video_needs_recreation = true;
}
}
if (flexfec_config_.rtp.extensions != *params.rtp_header_extensions) {
flexfec_config_.rtp.extensions = *params.rtp_header_extensions;
if (flexfec_stream_) {
flexfec_stream_->SetRtpExtensions(flexfec_config_.rtp.extensions);
} else if (flexfec_config_.IsCompleteAndEnabled()) {
video_needs_recreation = true;
}
}
}
if (params.flexfec_payload_type) {
flexfec_config_.payload_type = *params.flexfec_payload_type;
// TODO(tommi): See if it is better to always have a flexfec stream object
// configured and instead of recreating the video stream, reconfigure the
// flexfec object from within the rtp callback (soon to be on the network
// thread).
if (flexfec_stream_ || flexfec_config_.IsCompleteAndEnabled())
video_needs_recreation = true;
}
if (video_needs_recreation) {
RecreateReceiveStream();
}
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::RecreateReceiveStream() {
absl::optional<int> base_minimum_playout_delay_ms;
absl::optional<webrtc::VideoReceiveStreamInterface::RecordingState>
recording_state;
if (stream_) {
base_minimum_playout_delay_ms = stream_->GetBaseMinimumPlayoutDelayMs();
recording_state = stream_->SetAndGetRecordingState(
webrtc::VideoReceiveStreamInterface::RecordingState(),
/*generate_key_frame=*/false);
call_->DestroyVideoReceiveStream(stream_);
stream_ = nullptr;
}
if (flexfec_stream_) {
call_->DestroyFlexfecReceiveStream(flexfec_stream_);
flexfec_stream_ = nullptr;
}
if (flexfec_config_.IsCompleteAndEnabled()) {
flexfec_stream_ = call_->CreateFlexfecReceiveStream(flexfec_config_);
}
webrtc::VideoReceiveStreamInterface::Config config = config_.Copy();
config.rtp.protected_by_flexfec = (flexfec_stream_ != nullptr);
config.rtp.packet_sink_ = flexfec_stream_;
stream_ = call_->CreateVideoReceiveStream(std::move(config));
if (base_minimum_playout_delay_ms) {
stream_->SetBaseMinimumPlayoutDelayMs(
base_minimum_playout_delay_ms.value());
}
if (recording_state) {
stream_->SetAndGetRecordingState(std::move(*recording_state),
/*generate_key_frame=*/false);
}
stream_->Start();
if (IsEnabled(call_->trials(), "WebRTC-Video-BufferPacketsWithUnknownSsrc")) {
channel_->BackfillBufferedPackets(stream_params_.ssrcs);
}
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::OnFrame(
const webrtc::VideoFrame& frame) {
webrtc::MutexLock lock(&sink_lock_);
int64_t time_now_ms = rtc::TimeMillis();
if (first_frame_timestamp_ < 0)
first_frame_timestamp_ = time_now_ms;
int64_t elapsed_time_ms = time_now_ms - first_frame_timestamp_;
if (frame.ntp_time_ms() > 0)
estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms;
if (sink_ == NULL) {
RTC_LOG(LS_WARNING)
<< "VideoReceiveStreamInterface not connected to a VideoSink.";
return;
}
sink_->OnFrame(frame);
}
bool WebRtcVideoChannel::WebRtcVideoReceiveStream::IsDefaultStream() const {
return default_stream_;
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
config_.frame_decryptor = frame_decryptor;
if (stream_) {
RTC_LOG(LS_INFO)
<< "Setting FrameDecryptor (recv) because of SetFrameDecryptor, "
"remote_ssrc="
<< config_.rtp.remote_ssrc;
stream_->SetFrameDecryptor(frame_decryptor);
}
}
bool WebRtcVideoChannel::WebRtcVideoReceiveStream::SetBaseMinimumPlayoutDelayMs(
int delay_ms) {
return stream_ ? stream_->SetBaseMinimumPlayoutDelayMs(delay_ms) : false;
}
int WebRtcVideoChannel::WebRtcVideoReceiveStream::GetBaseMinimumPlayoutDelayMs()
const {
return stream_ ? stream_->GetBaseMinimumPlayoutDelayMs() : 0;
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetSink(
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
webrtc::MutexLock lock(&sink_lock_);
sink_ = sink;
}
std::string
WebRtcVideoChannel::WebRtcVideoReceiveStream::GetCodecNameFromPayloadType(
int payload_type) {
for (const webrtc::VideoReceiveStreamInterface::Decoder& decoder :
config_.decoders) {
if (decoder.payload_type == payload_type) {
return decoder.video_format.name;
}
}
return "";
}
VideoReceiverInfo
WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
bool log_stats) {
VideoReceiverInfo info;
info.ssrc_groups = stream_params_.ssrc_groups;
info.add_ssrc(config_.rtp.remote_ssrc);
webrtc::VideoReceiveStreamInterface::Stats stats = stream_->GetStats();
info.decoder_implementation_name = stats.decoder_implementation_name;
if (stats.current_payload_type != -1) {
info.codec_payload_type = stats.current_payload_type;
}
info.payload_bytes_rcvd = stats.rtp_stats.packet_counter.payload_bytes;
info.header_and_padding_bytes_rcvd =
stats.rtp_stats.packet_counter.header_bytes +
stats.rtp_stats.packet_counter.padding_bytes;
info.packets_rcvd = stats.rtp_stats.packet_counter.packets;
info.packets_lost = stats.rtp_stats.packets_lost;
info.jitter_ms = stats.rtp_stats.jitter / (kVideoCodecClockrate / 1000);
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;
info.frame_height = stats.height;
{
webrtc::MutexLock frame_cs(&sink_lock_);
info.capture_start_ntp_time_ms = estimated_remote_start_ntp_time_ms_;
}
info.decode_ms = stats.decode_ms;
info.max_decode_ms = stats.max_decode_ms;
info.current_delay_ms = stats.current_delay_ms;
info.target_delay_ms = stats.target_delay_ms;
info.jitter_buffer_ms = stats.jitter_buffer_ms;
info.jitter_buffer_delay_seconds = stats.jitter_buffer_delay_seconds;
info.jitter_buffer_emitted_count = stats.jitter_buffer_emitted_count;
info.min_playout_delay_ms = stats.min_playout_delay_ms;
info.render_delay_ms = stats.render_delay_ms;
info.frames_received =
stats.frame_counts.key_frames + stats.frame_counts.delta_frames;
info.frames_dropped = stats.frames_dropped;
info.frames_decoded = stats.frames_decoded;
info.key_frames_decoded = stats.frame_counts.key_frames;
info.frames_rendered = stats.frames_rendered;
info.qp_sum = stats.qp_sum;
info.total_decode_time_ms = stats.total_decode_time_ms;
info.total_processing_delay = stats.total_processing_delay;
info.total_assembly_time = stats.total_assembly_time;
info.frames_assembled_from_multiple_packets =
stats.frames_assembled_from_multiple_packets;
info.last_packet_received_timestamp_ms =
stats.rtp_stats.last_packet_received_timestamp_ms;
info.estimated_playout_ntp_timestamp_ms =
stats.estimated_playout_ntp_timestamp_ms;
info.first_frame_received_to_decoded_ms =
stats.first_frame_received_to_decoded_ms;
info.total_inter_frame_delay = stats.total_inter_frame_delay;
info.total_squared_inter_frame_delay = stats.total_squared_inter_frame_delay;
info.interframe_delay_max_ms = stats.interframe_delay_max_ms;
info.freeze_count = stats.freeze_count;
info.pause_count = stats.pause_count;
info.total_freezes_duration_ms = stats.total_freezes_duration_ms;
info.total_pauses_duration_ms = stats.total_pauses_duration_ms;
info.total_frames_duration_ms = stats.total_frames_duration_ms;
info.sum_squared_frame_durations = stats.sum_squared_frame_durations;
info.content_type = stats.content_type;
info.codec_name = GetCodecNameFromPayloadType(stats.current_payload_type);
info.firs_sent = stats.rtcp_packet_type_counts.fir_packets;
info.plis_sent = stats.rtcp_packet_type_counts.pli_packets;
info.nacks_sent = stats.rtcp_packet_type_counts.nack_packets;
// TODO(bugs.webrtc.org/10662): Add stats for LNTF.
info.timing_frame_info = stats.timing_frame_info;
if (log_stats)
RTC_LOG(LS_INFO) << stats.ToString(rtc::TimeMillis());
return info;
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::
SetRecordableEncodedFrameCallback(
std::function<void(const webrtc::RecordableEncodedFrame&)> callback) {
if (stream_) {
stream_->SetAndGetRecordingState(
webrtc::VideoReceiveStreamInterface::RecordingState(
std::move(callback)),
/*generate_key_frame=*/true);
} else {
RTC_LOG(LS_ERROR) << "Absent receive stream; ignoring setting encoded "
"frame sink";
}
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::
ClearRecordableEncodedFrameCallback() {
if (stream_) {
stream_->SetAndGetRecordingState(
webrtc::VideoReceiveStreamInterface::RecordingState(),
/*generate_key_frame=*/false);
} else {
RTC_LOG(LS_ERROR) << "Absent receive stream; ignoring clearing encoded "
"frame sink";
}
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::GenerateKeyFrame() {
if (stream_) {
stream_->GenerateKeyFrame();
} else {
RTC_LOG(LS_ERROR)
<< "Absent receive stream; ignoring key frame generation request.";
}
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::
SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer) {
config_.frame_transformer = frame_transformer;
if (stream_)
stream_->SetDepacketizerToDecoderFrameTransformer(frame_transformer);
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetLocalSsrc(uint32_t ssrc) {
config_.rtp.local_ssrc = ssrc;
call_->OnLocalSsrcUpdated(stream(), ssrc);
if (flexfec_stream_)
call_->OnLocalSsrcUpdated(*flexfec_stream_, ssrc);
}
WebRtcVideoChannel::VideoCodecSettings::VideoCodecSettings()
: flexfec_payload_type(-1), rtx_payload_type(-1), rtx_time(-1) {}
bool WebRtcVideoChannel::VideoCodecSettings::operator==(
const WebRtcVideoChannel::VideoCodecSettings& other) const {
return codec == other.codec && ulpfec == other.ulpfec &&
flexfec_payload_type == other.flexfec_payload_type &&
rtx_payload_type == other.rtx_payload_type &&
rtx_time == other.rtx_time;
}
bool WebRtcVideoChannel::VideoCodecSettings::EqualsDisregardingFlexfec(
const WebRtcVideoChannel::VideoCodecSettings& a,
const WebRtcVideoChannel::VideoCodecSettings& b) {
return a.codec == b.codec && a.ulpfec == b.ulpfec &&
a.rtx_payload_type == b.rtx_payload_type && a.rtx_time == b.rtx_time;
}
bool WebRtcVideoChannel::VideoCodecSettings::operator!=(
const WebRtcVideoChannel::VideoCodecSettings& other) const {
return !(*this == other);
}
std::vector<WebRtcVideoChannel::VideoCodecSettings>
WebRtcVideoChannel::MapCodecs(const std::vector<VideoCodec>& codecs) {
if (codecs.empty()) {
return {};
}
std::vector<VideoCodecSettings> video_codecs;
std::map<int, VideoCodec::CodecType> payload_codec_type;
// `rtx_mapping` maps video payload type to rtx payload type.
std::map<int, int> rtx_mapping;
std::map<int, int> rtx_time_mapping;
webrtc::UlpfecConfig ulpfec_config;
absl::optional<int> flexfec_payload_type;
for (const VideoCodec& in_codec : codecs) {
const int payload_type = in_codec.id;
if (payload_codec_type.find(payload_type) != payload_codec_type.end()) {
RTC_LOG(LS_ERROR) << "Payload type already registered: "
<< in_codec.ToString();
return {};
}
payload_codec_type[payload_type] = in_codec.GetCodecType();
switch (in_codec.GetCodecType()) {
case VideoCodec::CODEC_RED: {
if (ulpfec_config.red_payload_type != -1) {
RTC_LOG(LS_ERROR)
<< "Duplicate RED codec: ignoring PT=" << payload_type
<< " in favor of PT=" << ulpfec_config.red_payload_type
<< " which was specified first.";
break;
}
ulpfec_config.red_payload_type = payload_type;
break;
}
case VideoCodec::CODEC_ULPFEC: {
if (ulpfec_config.ulpfec_payload_type != -1) {
RTC_LOG(LS_ERROR)
<< "Duplicate ULPFEC codec: ignoring PT=" << payload_type
<< " in favor of PT=" << ulpfec_config.ulpfec_payload_type
<< " which was specified first.";
break;
}
ulpfec_config.ulpfec_payload_type = payload_type;
break;
}
case VideoCodec::CODEC_FLEXFEC: {
if (flexfec_payload_type) {
RTC_LOG(LS_ERROR)
<< "Duplicate FLEXFEC codec: ignoring PT=" << payload_type
<< " in favor of PT=" << *flexfec_payload_type
<< " which was specified first.";
break;
}
flexfec_payload_type = payload_type;
break;
}
case VideoCodec::CODEC_RTX: {
int associated_payload_type;
if (!in_codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_payload_type) ||
!IsValidRtpPayloadType(associated_payload_type)) {
RTC_LOG(LS_ERROR)
<< "RTX codec with invalid or no associated payload type: "
<< in_codec.ToString();
return {};
}
int rtx_time;
if (in_codec.GetParam(kCodecParamRtxTime, &rtx_time) && rtx_time > 0) {
rtx_time_mapping[associated_payload_type] = rtx_time;
}
rtx_mapping[associated_payload_type] = payload_type;
break;
}
case VideoCodec::CODEC_VIDEO: {
video_codecs.emplace_back();
video_codecs.back().codec = in_codec;
break;
}
}
}
// One of these codecs should have been a video codec. Only having FEC
// parameters into this code is a logic error.
RTC_DCHECK(!video_codecs.empty());
for (const auto& entry : rtx_mapping) {
const int associated_payload_type = entry.first;
const int rtx_payload_type = entry.second;
auto it = payload_codec_type.find(associated_payload_type);
if (it == payload_codec_type.end()) {
RTC_LOG(LS_ERROR) << "RTX codec (PT=" << rtx_payload_type
<< ") mapped to PT=" << associated_payload_type
<< " which is not in the codec list.";
return {};
}
const VideoCodec::CodecType associated_codec_type = it->second;
if (associated_codec_type != VideoCodec::CODEC_VIDEO &&
associated_codec_type != VideoCodec::CODEC_RED) {
RTC_LOG(LS_ERROR)
<< "RTX PT=" << rtx_payload_type
<< " not mapped to regular video codec or RED codec (PT="
<< associated_payload_type << ").";
return {};
}
if (associated_payload_type == ulpfec_config.red_payload_type) {
ulpfec_config.red_rtx_payload_type = rtx_payload_type;
}
}
for (VideoCodecSettings& codec_settings : video_codecs) {
const int payload_type = codec_settings.codec.id;
codec_settings.ulpfec = ulpfec_config;
codec_settings.flexfec_payload_type = flexfec_payload_type.value_or(-1);
auto it = rtx_mapping.find(payload_type);
if (it != rtx_mapping.end()) {
const int rtx_payload_type = it->second;
codec_settings.rtx_payload_type = rtx_payload_type;
auto rtx_time_it = rtx_time_mapping.find(payload_type);
if (rtx_time_it != rtx_time_mapping.end()) {
const int rtx_time = rtx_time_it->second;
if (rtx_time < kNackHistoryMs) {
codec_settings.rtx_time = rtx_time;
} else {
codec_settings.rtx_time = kNackHistoryMs;
}
}
}
}
return video_codecs;
}
WebRtcVideoChannel::WebRtcVideoReceiveStream*
WebRtcVideoChannel::FindReceiveStream(uint32_t ssrc) {
if (ssrc == 0) {
absl::optional<uint32_t> default_ssrc = GetDefaultReceiveStreamSsrc();
if (!default_ssrc) {
return nullptr;
}
ssrc = *default_ssrc;
}
auto it = receive_streams_.find(ssrc);
if (it != receive_streams_.end()) {
return it->second;
}
return nullptr;
}
void WebRtcVideoChannel::SetRecordableEncodedFrameCallback(
uint32_t ssrc,
std::function<void(const webrtc::RecordableEncodedFrame&)> callback) {
RTC_DCHECK_RUN_ON(&thread_checker_);
WebRtcVideoReceiveStream* stream = FindReceiveStream(ssrc);
if (stream) {
stream->SetRecordableEncodedFrameCallback(std::move(callback));
} else {
RTC_LOG(LS_ERROR) << "Absent receive stream; ignoring setting encoded "
"frame sink for ssrc "
<< ssrc;
}
}
void WebRtcVideoChannel::ClearRecordableEncodedFrameCallback(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&thread_checker_);
WebRtcVideoReceiveStream* stream = FindReceiveStream(ssrc);
if (stream) {
stream->ClearRecordableEncodedFrameCallback();
} else {
RTC_LOG(LS_ERROR) << "Absent receive stream; ignoring clearing encoded "
"frame sink for ssrc "
<< ssrc;
}
}
void WebRtcVideoChannel::GenerateKeyFrame(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&thread_checker_);
WebRtcVideoReceiveStream* stream = FindReceiveStream(ssrc);
if (stream) {
stream->GenerateKeyFrame();
} else {
RTC_LOG(LS_ERROR)
<< "Absent receive stream; ignoring key frame generation for ssrc "
<< ssrc;
}
}
void WebRtcVideoChannel::SetEncoderToPacketizerFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto matching_stream = send_streams_.find(ssrc);
if (matching_stream != send_streams_.end()) {
matching_stream->second->SetEncoderToPacketizerFrameTransformer(
std::move(frame_transformer));
}
}
void WebRtcVideoChannel::SetDepacketizerToDecoderFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK(frame_transformer);
RTC_DCHECK_RUN_ON(&thread_checker_);
if (ssrc == 0) {
// If the receiver is unsignaled, save the frame transformer and set it when
// the stream is associated with an ssrc.
unsignaled_frame_transformer_ = std::move(frame_transformer);
return;
}
auto matching_stream = receive_streams_.find(ssrc);
if (matching_stream != receive_streams_.end()) {
matching_stream->second->SetDepacketizerToDecoderFrameTransformer(
std::move(frame_transformer));
}
}
// TODO(bugs.webrtc.org/8785): Consider removing max_qp as member of
// EncoderStreamFactory and instead set this value individually for each stream
// in the VideoEncoderConfig.simulcast_layers.
EncoderStreamFactory::EncoderStreamFactory(
std::string codec_name,
int max_qp,
bool is_screenshare,
bool conference_mode,
const webrtc::FieldTrialsView* trials)
: codec_name_(codec_name),
max_qp_(max_qp),
is_screenshare_(is_screenshare),
conference_mode_(conference_mode),
trials_(trials ? *trials : fallback_trials_) {}
std::vector<webrtc::VideoStream> EncoderStreamFactory::CreateEncoderStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config) {
RTC_DCHECK_GT(encoder_config.number_of_streams, 0);
RTC_DCHECK_GE(encoder_config.simulcast_layers.size(),
encoder_config.number_of_streams);
const absl::optional<webrtc::DataRate> experimental_min_bitrate =
GetExperimentalMinVideoBitrate(encoder_config.codec_type);
if (encoder_config.number_of_streams > 1 ||
((absl::EqualsIgnoreCase(codec_name_, kVp8CodecName) ||
absl::EqualsIgnoreCase(codec_name_, kH264CodecName)) &&
is_screenshare_ && conference_mode_)) {
return CreateSimulcastOrConferenceModeScreenshareStreams(
width, height, encoder_config, experimental_min_bitrate);
}
return CreateDefaultVideoStreams(width, height, encoder_config,
experimental_min_bitrate);
}
std::vector<webrtc::VideoStream>
EncoderStreamFactory::CreateDefaultVideoStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config,
const absl::optional<webrtc::DataRate>& experimental_min_bitrate) const {
std::vector<webrtc::VideoStream> layers;
// 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, is_screenshare_) *
1000;
int min_bitrate_bps =
experimental_min_bitrate
? rtc::saturated_cast<int>(experimental_min_bitrate->bps())
: webrtc::kDefaultMinVideoBitrateBps;
if (encoder_config.simulcast_layers[0].min_bitrate_bps > 0) {
// Use set min bitrate.
min_bitrate_bps = encoder_config.simulcast_layers[0].min_bitrate_bps;
// If only min bitrate is configured, make sure max is above min.
if (encoder_config.max_bitrate_bps <= 0)
max_bitrate_bps = std::max(min_bitrate_bps, max_bitrate_bps);
}
int max_framerate = (encoder_config.simulcast_layers[0].max_framerate > 0)
? encoder_config.simulcast_layers[0].max_framerate
: kDefaultVideoMaxFramerate;
webrtc::VideoStream layer;
layer.width = width;
layer.height = height;
layer.max_framerate = max_framerate;
if (encoder_config.simulcast_layers[0].scale_resolution_down_by > 1.) {
layer.width = ScaleDownResolution(
layer.width,
encoder_config.simulcast_layers[0].scale_resolution_down_by,
kMinLayerSize);
layer.height = ScaleDownResolution(
layer.height,
encoder_config.simulcast_layers[0].scale_resolution_down_by,
kMinLayerSize);
}
// In the case that the application sets a max bitrate that's lower than the
// min bitrate, we adjust it down (see bugs.webrtc.org/9141).
layer.min_bitrate_bps = std::min(min_bitrate_bps, max_bitrate_bps);
if (encoder_config.simulcast_layers[0].target_bitrate_bps <= 0) {
layer.target_bitrate_bps = max_bitrate_bps;
} else {
layer.target_bitrate_bps =
encoder_config.simulcast_layers[0].target_bitrate_bps;
}
layer.max_bitrate_bps = max_bitrate_bps;
layer.max_qp = max_qp_;
layer.bitrate_priority = encoder_config.bitrate_priority;
if (absl::EqualsIgnoreCase(codec_name_, kVp9CodecName)) {
RTC_DCHECK(encoder_config.encoder_specific_settings);
// Use VP9 SVC layering from codec settings which might be initialized
// though field trial in ConfigureVideoEncoderSettings.
webrtc::VideoCodecVP9 vp9_settings;
encoder_config.encoder_specific_settings->FillVideoCodecVp9(&vp9_settings);
layer.num_temporal_layers = vp9_settings.numberOfTemporalLayers;
}
if (IsTemporalLayersSupported(codec_name_)) {
// Use configured number of temporal layers if set.
if (encoder_config.simulcast_layers[0].num_temporal_layers) {
layer.num_temporal_layers =
*encoder_config.simulcast_layers[0].num_temporal_layers;
}
}
layer.scalability_mode = encoder_config.simulcast_layers[0].scalability_mode;
layers.push_back(layer);
return layers;
}
std::vector<webrtc::VideoStream>
EncoderStreamFactory::CreateSimulcastOrConferenceModeScreenshareStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config,
const absl::optional<webrtc::DataRate>& experimental_min_bitrate) const {
std::vector<webrtc::VideoStream> layers;
const bool temporal_layers_supported =
absl::EqualsIgnoreCase(codec_name_, kVp8CodecName) ||
absl::EqualsIgnoreCase(codec_name_, kH264CodecName);
// Use legacy simulcast screenshare if conference mode is explicitly enabled
// or use the regular simulcast configuration path which is generic.
layers = GetSimulcastConfig(FindRequiredActiveLayers(encoder_config),
encoder_config.number_of_streams, width, height,
encoder_config.bitrate_priority, max_qp_,
is_screenshare_ && conference_mode_,
temporal_layers_supported, trials_);
// Allow an experiment to override the minimum bitrate for the lowest
// spatial layer. The experiment's configuration has the lowest priority.
if (experimental_min_bitrate) {
layers[0].min_bitrate_bps =
rtc::saturated_cast<int>(experimental_min_bitrate->bps());
}
// Update the active simulcast layers and configured bitrates.
bool is_highest_layer_max_bitrate_configured = false;
const bool has_scale_resolution_down_by = absl::c_any_of(
encoder_config.simulcast_layers, [](const webrtc::VideoStream& layer) {
return layer.scale_resolution_down_by != -1.;
});
bool default_scale_factors_used = true;
if (has_scale_resolution_down_by) {
default_scale_factors_used = IsScaleFactorsPowerOfTwo(encoder_config);
}
const bool norm_size_configured =
webrtc::NormalizeSimulcastSizeExperiment::GetBase2Exponent().has_value();
const int normalized_width =
(default_scale_factors_used || norm_size_configured) &&
(width >= kMinLayerSize)
? NormalizeSimulcastSize(width, encoder_config.number_of_streams)
: width;
const int normalized_height =
(default_scale_factors_used || norm_size_configured) &&
(height >= kMinLayerSize)
? NormalizeSimulcastSize(height, encoder_config.number_of_streams)
: height;
for (size_t i = 0; i < layers.size(); ++i) {
layers[i].active = encoder_config.simulcast_layers[i].active;
layers[i].scalability_mode =
encoder_config.simulcast_layers[i].scalability_mode;
// Update with configured num temporal layers if supported by codec.
if (encoder_config.simulcast_layers[i].num_temporal_layers &&
IsTemporalLayersSupported(codec_name_)) {
layers[i].num_temporal_layers =
*encoder_config.simulcast_layers[i].num_temporal_layers;
}
if (encoder_config.simulcast_layers[i].max_framerate > 0) {
layers[i].max_framerate =
encoder_config.simulcast_layers[i].max_framerate;
}
if (has_scale_resolution_down_by) {
const double scale_resolution_down_by = std::max(
encoder_config.simulcast_layers[i].scale_resolution_down_by, 1.0);
layers[i].width = ScaleDownResolution(
normalized_width, scale_resolution_down_by, kMinLayerSize);
layers[i].height = ScaleDownResolution(
normalized_height, scale_resolution_down_by, kMinLayerSize);
}
// Update simulcast bitrates with configured min and max bitrate.
if (encoder_config.simulcast_layers[i].min_bitrate_bps > 0) {
layers[i].min_bitrate_bps =
encoder_config.simulcast_layers[i].min_bitrate_bps;
}
if (encoder_config.simulcast_layers[i].max_bitrate_bps > 0) {
layers[i].max_bitrate_bps =
encoder_config.simulcast_layers[i].max_bitrate_bps;
}
if (encoder_config.simulcast_layers[i].target_bitrate_bps > 0) {
layers[i].target_bitrate_bps =
encoder_config.simulcast_layers[i].target_bitrate_bps;
}
if (encoder_config.simulcast_layers[i].min_bitrate_bps > 0 &&
encoder_config.simulcast_layers[i].max_bitrate_bps > 0) {
// Min and max bitrate are configured.
// Set target to 3/4 of the max bitrate (or to max if below min).
if (encoder_config.simulcast_layers[i].target_bitrate_bps <= 0)
layers[i].target_bitrate_bps = layers[i].max_bitrate_bps * 3 / 4;
if (layers[i].target_bitrate_bps < layers[i].min_bitrate_bps)
layers[i].target_bitrate_bps = layers[i].max_bitrate_bps;
} else if (encoder_config.simulcast_layers[i].min_bitrate_bps > 0) {
// Only min bitrate is configured, make sure target/max are above min.
layers[i].target_bitrate_bps =
std::max(layers[i].target_bitrate_bps, layers[i].min_bitrate_bps);
layers[i].max_bitrate_bps =
std::max(layers[i].max_bitrate_bps, layers[i].min_bitrate_bps);
} else if (encoder_config.simulcast_layers[i].max_bitrate_bps > 0) {
// Only max bitrate is configured, make sure min/target are below max.
// Keep target bitrate if it is set explicitly in encoding config.
// Otherwise set target bitrate to 3/4 of the max bitrate
// or the one calculated from GetSimulcastConfig() which is larger.
layers[i].min_bitrate_bps =
std::min(layers[i].min_bitrate_bps, layers[i].max_bitrate_bps);
if (encoder_config.simulcast_layers[i].target_bitrate_bps <= 0) {
layers[i].target_bitrate_bps = std::max(
layers[i].target_bitrate_bps, layers[i].max_bitrate_bps * 3 / 4);
}
layers[i].target_bitrate_bps = std::max(
std::min(layers[i].target_bitrate_bps, layers[i].max_bitrate_bps),
layers[i].min_bitrate_bps);
}
if (i == layers.size() - 1) {
is_highest_layer_max_bitrate_configured =
encoder_config.simulcast_layers[i].max_bitrate_bps > 0;
}
}
if (!is_screenshare_ && !is_highest_layer_max_bitrate_configured &&
encoder_config.max_bitrate_bps > 0) {
// No application-configured maximum for the largest layer.
// If there is bitrate leftover, give it to the largest layer.
BoostMaxSimulcastLayer(
webrtc::DataRate::BitsPerSec(encoder_config.max_bitrate_bps), &layers);
}
// Sort the layers by max_bitrate_bps, they might not always be from
// smallest to biggest
std::vector<size_t> index(layers.size());
std::iota(index.begin(), index.end(), 0);
std::stable_sort(index.begin(), index.end(), [&layers](size_t a, size_t b) {
return layers[a].max_bitrate_bps < layers[b].max_bitrate_bps;
});
if (!layers[index[0]].active) {
// Adjust min bitrate of the first active layer to allow it to go as low as
// the lowest (now inactive) layer could.
// Otherwise, if e.g. a single HD stream is active, it would have 600kbps
// min bitrate, which would always be allocated to the stream.
// This would lead to congested network, dropped frames and overall bad
// experience.
const int min_configured_bitrate = layers[index[0]].min_bitrate_bps;
for (size_t i = 0; i < layers.size(); ++i) {
if (layers[index[i]].active) {
layers[index[i]].min_bitrate_bps = min_configured_bitrate;
break;
}
}
}
return layers;
}
} // namespace cricket