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firefox/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
Daniel Baumann 5e9a113729
Adding upstream version 140.0. 
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
2025-06-25 09:37:52 +02:00

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/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "VideoConduit.h"
#include <algorithm>
#include <cmath>
#include "common/browser_logging/CSFLog.h"
#include "common/YuvStamper.h"
#include "MediaConduitControl.h"
#include "nsIGfxInfo.h"
#include "nsServiceManagerUtils.h"
#include "RtpRtcpConfig.h"
#include "transport/SrtpFlow.h" // For SRTP_MAX_EXPANSION
#include "Tracing.h"
#include "VideoStreamFactory.h"
#include "WebrtcCallWrapper.h"
#include "libwebrtcglue/FrameTransformer.h"
#include "libwebrtcglue/FrameTransformerProxy.h"
#include "mozilla/StateMirroring.h"
#include "mozilla/RefPtr.h"
#include "nsThreadUtils.h"
#include "mozilla/Maybe.h"
#include "mozilla/ErrorResult.h"
#include <string>
#include <utility>
#include <vector>
// libwebrtc includes
#include "api/transport/bitrate_settings.h"
#include "api/video_codecs/h264_profile_level_id.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_codec.h"
#include "media/base/media_constants.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/ref_counted_object.h"
#include "api/call/transport.h"
#include "api/media_types.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_frame_buffer.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include <utility>
#include "call/call.h"
#include "call/rtp_config.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "CodecConfig.h"
#include "common_video/include/video_frame_buffer_pool.h"
#include "domstubs.h"
#include <iomanip>
#include <ios>
#include "jsapi/RTCStatsReport.h"
#include <limits>
#include "MainThreadUtils.h"
#include <map>
#include "MediaConduitErrors.h"
#include "MediaConduitInterface.h"
#include "MediaEventSource.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "mozilla/Assertions.h"
#include "mozilla/Atomics.h"
#include "mozilla/DataMutex.h"
#include "mozilla/dom/BindingDeclarations.h"
#include "mozilla/dom/RTCStatsReportBinding.h"
#include "mozilla/fallible.h"
#include "mozilla/mozalloc_oom.h"
#include "mozilla/MozPromise.h"
#include "mozilla/Mutex.h"
#include "mozilla/ProfilerState.h"
#include "mozilla/ReentrantMonitor.h"
#include "mozilla/ReverseIterator.h"
#include "mozilla/StateWatching.h"
#include "mozilla/glean/DomMediaWebrtcMetrics.h"
#include "mozilla/UniquePtr.h"
#include "nsCOMPtr.h"
#include "nsDebug.h"
#include "nsError.h"
#include "nsIDirectTaskDispatcher.h"
#include "nsISerialEventTarget.h"
#include "nsStringFwd.h"
#include "PerformanceRecorder.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include <sstream>
#include <stdint.h>
#include "transport/mediapacket.h"
#include "video/config/video_encoder_config.h"
#include "WebrtcVideoCodecFactory.h"
#ifdef MOZ_WIDGET_ANDROID
# include "VideoEngine.h"
#endif
// for ntohs
#ifdef _MSC_VER
# include "Winsock2.h"
#else
# include <netinet/in.h>
#endif
#define INVALID_RTP_PAYLOAD 255 // valid payload types are 0 to 127
namespace mozilla {
namespace {
const char* vcLogTag = "WebrtcVideoSessionConduit";
#ifdef LOGTAG
# undef LOGTAG
#endif
#define LOGTAG vcLogTag
using namespace cricket;
using LocalDirection = MediaSessionConduitLocalDirection;
const int kNullPayloadType = -1;
const char kRtcpFbCcmParamTmmbr[] = "tmmbr";
template <class t>
void ConstrainPreservingAspectRatioExact(uint32_t max_fs, t* width, t* height) {
// We could try to pick a better starting divisor, but it won't make any real
// performance difference.
for (size_t d = 1; d < std::min(*width, *height); ++d) {
if ((*width % d) || (*height % d)) {
continue; // Not divisible
}
if (((*width) * (*height)) / (d * d) <= max_fs) {
*width /= d;
*height /= d;
return;
}
}
*width = 0;
*height = 0;
}
/**
* Perform validation on the codecConfig to be applied
*/
MediaConduitErrorCode ValidateCodecConfig(const VideoCodecConfig& codecInfo) {
if (codecInfo.mName.empty()) {
CSFLogError(LOGTAG, "%s Empty Payload Name ", __FUNCTION__);
return kMediaConduitMalformedArgument;
}
return kMediaConduitNoError;
}
webrtc::VideoCodecType SupportedCodecType(webrtc::VideoCodecType aType) {
switch (aType) {
case webrtc::VideoCodecType::kVideoCodecVP8:
case webrtc::VideoCodecType::kVideoCodecVP9:
case webrtc::VideoCodecType::kVideoCodecH264:
case webrtc::VideoCodecType::kVideoCodecAV1:
return aType;
default:
return webrtc::VideoCodecType::kVideoCodecGeneric;
}
// NOTREACHED
}
// Call thread only.
rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
ConfigureVideoEncoderSettings(const VideoCodecConfig& aConfig,
const WebrtcVideoConduit* aConduit,
webrtc::CodecParameterMap& aParameters) {
bool is_screencast =
aConduit->CodecMode() == webrtc::VideoCodecMode::kScreensharing;
// No automatic resizing when using simulcast or screencast.
bool automatic_resize = !is_screencast && aConfig.mEncodings.size() <= 1;
bool denoising;
bool codec_default_denoising = false;
if (is_screencast) {
denoising = false;
} else {
// Use codec default if video_noise_reduction is unset.
denoising = aConduit->Denoising();
codec_default_denoising = !denoising;
}
using Av1Config = JsepVideoCodecDescription::Av1Config;
aConfig.mAv1Config.apply([&](const Av1Config& config) {
MOZ_ASSERT(aConfig.mName == kAv1CodecName);
config.mProfile.apply([&](uint8_t value) {
aParameters[kAv1FmtpProfile] = std::to_string(value);
});
config.mLevelIdx.apply([&](uint8_t value) {
aParameters[kAv1FmtpLevelIdx] = std::to_string(value);
});
config.mTier.apply([&](uint8_t value) {
aParameters[kAv1FmtpTier] = std::to_string(value);
});
});
if (aConfig.mName == kH264CodecName) {
aParameters[kH264FmtpPacketizationMode] =
std::to_string(aConfig.mPacketizationMode);
{
std::stringstream ss;
ss << std::hex << std::setfill('0');
ss << std::setw(2) << static_cast<uint32_t>(aConfig.mProfile);
ss << std::setw(2) << static_cast<uint32_t>(aConfig.mConstraints);
ss << std::setw(2) << static_cast<uint32_t>(aConfig.mLevel);
std::string profileLevelId = ss.str();
auto parsedProfileLevelId =
webrtc::ParseH264ProfileLevelId(profileLevelId.c_str());
MOZ_DIAGNOSTIC_ASSERT(parsedProfileLevelId);
if (parsedProfileLevelId) {
aParameters[kH264FmtpProfileLevelId] = profileLevelId;
}
}
aParameters[kH264FmtpSpropParameterSets] = aConfig.mSpropParameterSets;
}
if (aConfig.mName == 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::scoped_refptr<
webrtc::VideoEncoderConfig::EncoderSpecificSettings>(
new rtc::RefCountedObject<
webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(
vp8_settings));
}
if (aConfig.mName == kVp9CodecName) {
webrtc::VideoCodecVP9 vp9_settings =
webrtc::VideoEncoder::GetDefaultVp9Settings();
if (!is_screencast) {
// Always configure only 1 spatial layer for screencapture as libwebrtc
// has some special requirements when SVC is active. For non-screencapture
// the spatial layers are experimentally configurable via a pref.
vp9_settings.numberOfSpatialLayers = aConduit->SpatialLayers();
}
// VP9 denoising is disabled by default.
vp9_settings.denoisingOn = codec_default_denoising ? false : denoising;
return rtc::scoped_refptr<
webrtc::VideoEncoderConfig::EncoderSpecificSettings>(
new rtc::RefCountedObject<
webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(
vp9_settings));
}
return nullptr;
}
uint32_t GenerateRandomSSRC() {
uint32_t ssrc;
do {
SECStatus rv = PK11_GenerateRandom(reinterpret_cast<unsigned char*>(&ssrc),
sizeof(ssrc));
MOZ_RELEASE_ASSERT(rv == SECSuccess);
} while (ssrc == 0); // webrtc.org code has fits if you select an SSRC of 0
return ssrc;
}
// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
bool operator!=(
const webrtc::VideoReceiveStreamInterface::Config::Rtp& aThis,
const webrtc::VideoReceiveStreamInterface::Config::Rtp& aOther) {
return aThis.remote_ssrc != aOther.remote_ssrc ||
aThis.local_ssrc != aOther.local_ssrc ||
aThis.rtcp_mode != aOther.rtcp_mode ||
aThis.rtcp_xr.receiver_reference_time_report !=
aOther.rtcp_xr.receiver_reference_time_report ||
aThis.remb != aOther.remb || aThis.tmmbr != aOther.tmmbr ||
aThis.keyframe_method != aOther.keyframe_method ||
aThis.lntf.enabled != aOther.lntf.enabled ||
aThis.nack.rtp_history_ms != aOther.nack.rtp_history_ms ||
aThis.ulpfec_payload_type != aOther.ulpfec_payload_type ||
aThis.red_payload_type != aOther.red_payload_type ||
aThis.rtx_ssrc != aOther.rtx_ssrc ||
aThis.protected_by_flexfec != aOther.protected_by_flexfec ||
aThis.rtx_associated_payload_types !=
aOther.rtx_associated_payload_types ||
aThis.raw_payload_types != aOther.raw_payload_types;
}
#ifdef DEBUG
// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
bool operator==(
const webrtc::VideoReceiveStreamInterface::Config::Rtp& aThis,
const webrtc::VideoReceiveStreamInterface::Config::Rtp& aOther) {
return !(aThis != aOther);
}
#endif
// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
bool operator!=(const webrtc::RtpConfig& aThis,
const webrtc::RtpConfig& aOther) {
return aThis.ssrcs != aOther.ssrcs || aThis.rids != aOther.rids ||
aThis.mid != aOther.mid || aThis.rtcp_mode != aOther.rtcp_mode ||
aThis.max_packet_size != aOther.max_packet_size ||
aThis.extmap_allow_mixed != aOther.extmap_allow_mixed ||
aThis.extensions != aOther.extensions ||
aThis.payload_name != aOther.payload_name ||
aThis.payload_type != aOther.payload_type ||
aThis.raw_payload != aOther.raw_payload ||
aThis.lntf.enabled != aOther.lntf.enabled ||
aThis.nack.rtp_history_ms != aOther.nack.rtp_history_ms ||
!(aThis.ulpfec == aOther.ulpfec) ||
aThis.flexfec.payload_type != aOther.flexfec.payload_type ||
aThis.flexfec.ssrc != aOther.flexfec.ssrc ||
aThis.flexfec.protected_media_ssrcs !=
aOther.flexfec.protected_media_ssrcs ||
aThis.rtx.ssrcs != aOther.rtx.ssrcs ||
aThis.rtx.payload_type != aOther.rtx.payload_type ||
aThis.c_name != aOther.c_name;
}
#ifdef DEBUG
// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
bool operator==(const webrtc::RtpConfig& aThis,
const webrtc::RtpConfig& aOther) {
return !(aThis != aOther);
}
#endif
} // namespace
void RecvSinkProxy::OnFrame(const webrtc::VideoFrame& aFrame) {
mOwner->OnRecvFrame(aFrame);
}
void SendSinkProxy::OnFrame(const webrtc::VideoFrame& aFrame) {
MOZ_ALWAYS_SUCCEEDS(mOwner->mCallThread->Dispatch(NS_NewRunnableFunction(
__FUNCTION__,
[owner = RefPtr(mOwner), aFrame] { owner->OnSendFrame(aFrame); })));
}
/**
* Factory Method for VideoConduit
*/
RefPtr<VideoSessionConduit> VideoSessionConduit::Create(
RefPtr<WebrtcCallWrapper> aCall, nsCOMPtr<nsISerialEventTarget> aStsThread,
Options aOptions, std::string aPCHandle,
const TrackingId& aRecvTrackingId) {
MOZ_ASSERT(NS_IsMainThread());
MOZ_ASSERT(aCall, "missing required parameter: aCall");
CSFLogVerbose(LOGTAG, "%s", __FUNCTION__);
if (!aCall) {
return nullptr;
}
auto obj = MakeRefPtr<WebrtcVideoConduit>(
std::move(aCall), std::move(aStsThread), std::move(aOptions),
std::move(aPCHandle), aRecvTrackingId);
if (obj->Init() != kMediaConduitNoError) {
CSFLogError(LOGTAG, "%s VideoConduit Init Failed ", __FUNCTION__);
return nullptr;
}
CSFLogVerbose(LOGTAG, "%s Successfully created VideoConduit ", __FUNCTION__);
return obj.forget();
}
#define INIT_MIRROR(name, val) \
name(aCallThread, val, "WebrtcVideoConduit::Control::" #name " (Mirror)")
WebrtcVideoConduit::Control::Control(const RefPtr<AbstractThread>& aCallThread)
: INIT_MIRROR(mReceiving, false),
INIT_MIRROR(mTransmitting, false),
INIT_MIRROR(mLocalSsrcs, Ssrcs()),
INIT_MIRROR(mLocalRtxSsrcs, Ssrcs()),
INIT_MIRROR(mLocalCname, std::string()),
INIT_MIRROR(mMid, std::string()),
INIT_MIRROR(mRemoteSsrc, 0),
INIT_MIRROR(mRemoteRtxSsrc, 0),
INIT_MIRROR(mSyncGroup, std::string()),
INIT_MIRROR(mLocalRecvRtpExtensions, RtpExtList()),
INIT_MIRROR(mLocalSendRtpExtensions, RtpExtList()),
INIT_MIRROR(mSendCodec, Nothing()),
INIT_MIRROR(mSendRtpRtcpConfig, Nothing()),
INIT_MIRROR(mRecvCodecs, std::vector<VideoCodecConfig>()),
INIT_MIRROR(mRecvRtpRtcpConfig, Nothing()),
INIT_MIRROR(mCodecMode, webrtc::VideoCodecMode::kRealtimeVideo),
INIT_MIRROR(mFrameTransformerProxySend, nullptr),
INIT_MIRROR(mFrameTransformerProxyRecv, nullptr),
INIT_MIRROR(mVideoDegradationPreference,
webrtc::DegradationPreference::DISABLED) {}
#undef INIT_MIRROR
WebrtcVideoConduit::WebrtcVideoConduit(
RefPtr<WebrtcCallWrapper> aCall, nsCOMPtr<nsISerialEventTarget> aStsThread,
Options aOptions, std::string aPCHandle, const TrackingId& aRecvTrackingId)
: mRendererMonitor("WebrtcVideoConduit::mRendererMonitor"),
mCallThread(aCall->mCallThread),
mStsThread(std::move(aStsThread)),
mControl(aCall->mCallThread),
mWatchManager(this, aCall->mCallThread),
mMutex("WebrtcVideoConduit::mMutex"),
mDecoderFactory(MakeUnique<WebrtcVideoDecoderFactory>(
mCallThread.get(), aPCHandle, aRecvTrackingId)),
mEncoderFactory(MakeUnique<WebrtcVideoEncoderFactory>(
mCallThread.get(), std::move(aPCHandle))),
mRecvSinkProxy(this),
mSendSinkProxy(this),
mEngineTransmitting(false),
mEngineReceiving(false),
mVideoLatencyTestEnable(aOptions.mVideoLatencyTestEnable),
mMinBitrate(aOptions.mMinBitrate),
mStartBitrate(aOptions.mStartBitrate),
mPrefMaxBitrate(aOptions.mPrefMaxBitrate),
mMinBitrateEstimate(aOptions.mMinBitrateEstimate),
mDenoising(aOptions.mDenoising),
mLockScaling(aOptions.mLockScaling),
mSpatialLayers(aOptions.mSpatialLayers),
mTemporalLayers(aOptions.mTemporalLayers),
mCall(std::move(aCall)),
mSendTransport(this),
mRecvTransport(this),
mSendStreamConfig(&mSendTransport),
mVideoStreamFactory("WebrtcVideoConduit::mVideoStreamFactory"),
mRecvStreamConfig(&mRecvTransport) {
mRecvStreamConfig.rtp.rtcp_event_observer = this;
}
WebrtcVideoConduit::~WebrtcVideoConduit() {
CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
MOZ_ASSERT(!mSendStream && !mRecvStream,
"Call DeleteStreams prior to ~WebrtcVideoConduit.");
}
#define CONNECT(aCanonical, aMirror) \
do { \
/* Ensure the watchmanager is wired up before the mirror receives its \
* initial mirrored value. */ \
mCall->mCallThread->DispatchStateChange( \
NS_NewRunnableFunction(__func__, [this, self = RefPtr(this)] { \
mWatchManager.Watch(aMirror, \
&WebrtcVideoConduit::OnControlConfigChange); \
})); \
(aCanonical).ConnectMirror(&(aMirror)); \
} while (0)
void WebrtcVideoConduit::InitControl(VideoConduitControlInterface* aControl) {
MOZ_ASSERT(NS_IsMainThread());
CONNECT(aControl->CanonicalReceiving(), mControl.mReceiving);
CONNECT(aControl->CanonicalTransmitting(), mControl.mTransmitting);
CONNECT(aControl->CanonicalLocalSsrcs(), mControl.mLocalSsrcs);
CONNECT(aControl->CanonicalLocalVideoRtxSsrcs(), mControl.mLocalRtxSsrcs);
CONNECT(aControl->CanonicalLocalCname(), mControl.mLocalCname);
CONNECT(aControl->CanonicalMid(), mControl.mMid);
CONNECT(aControl->CanonicalRemoteSsrc(), mControl.mRemoteSsrc);
CONNECT(aControl->CanonicalRemoteVideoRtxSsrc(), mControl.mRemoteRtxSsrc);
CONNECT(aControl->CanonicalSyncGroup(), mControl.mSyncGroup);
CONNECT(aControl->CanonicalLocalRecvRtpExtensions(),
mControl.mLocalRecvRtpExtensions);
CONNECT(aControl->CanonicalLocalSendRtpExtensions(),
mControl.mLocalSendRtpExtensions);
CONNECT(aControl->CanonicalVideoSendCodec(), mControl.mSendCodec);
CONNECT(aControl->CanonicalVideoSendRtpRtcpConfig(),
mControl.mSendRtpRtcpConfig);
CONNECT(aControl->CanonicalVideoRecvCodecs(), mControl.mRecvCodecs);
CONNECT(aControl->CanonicalVideoRecvRtpRtcpConfig(),
mControl.mRecvRtpRtcpConfig);
CONNECT(aControl->CanonicalVideoCodecMode(), mControl.mCodecMode);
CONNECT(aControl->CanonicalFrameTransformerProxySend(),
mControl.mFrameTransformerProxySend);
CONNECT(aControl->CanonicalFrameTransformerProxyRecv(),
mControl.mFrameTransformerProxyRecv);
CONNECT(aControl->CanonicalVideoDegradationPreference(),
mControl.mVideoDegradationPreference);
}
#undef CONNECT
void WebrtcVideoConduit::OnControlConfigChange() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
bool encoderReconfigureNeeded = false;
bool remoteSsrcUpdateNeeded = false;
bool sendStreamRecreationNeeded = false;
bool sendSourceUpdateNeeded = false;
if (mControl.mRemoteSsrc.Ref() != mControl.mConfiguredRemoteSsrc) {
mControl.mConfiguredRemoteSsrc = mControl.mRemoteSsrc;
remoteSsrcUpdateNeeded = true;
}
if (mControl.mRemoteRtxSsrc.Ref() != mControl.mConfiguredRemoteRtxSsrc) {
mControl.mConfiguredRemoteRtxSsrc = mControl.mRemoteRtxSsrc;
remoteSsrcUpdateNeeded = true;
}
if (mControl.mSyncGroup.Ref() != mRecvStreamConfig.sync_group) {
mRecvStreamConfig.sync_group = mControl.mSyncGroup;
}
if (const auto [codecConfigList, rtpRtcpConfig] = std::make_pair(
mControl.mRecvCodecs.Ref(), mControl.mRecvRtpRtcpConfig.Ref());
!codecConfigList.empty() && rtpRtcpConfig.isSome() &&
(codecConfigList != mControl.mConfiguredRecvCodecs ||
rtpRtcpConfig != mControl.mConfiguredRecvRtpRtcpConfig)) {
mControl.mConfiguredRecvCodecs = codecConfigList;
mControl.mConfiguredRecvRtpRtcpConfig = rtpRtcpConfig;
webrtc::VideoReceiveStreamInterface::Config::Rtp newRtp(
mRecvStreamConfig.rtp);
MOZ_ASSERT(newRtp == mRecvStreamConfig.rtp);
newRtp.rtx_associated_payload_types.clear();
newRtp.rtcp_mode = rtpRtcpConfig->GetRtcpMode();
newRtp.nack.rtp_history_ms = 0;
newRtp.remb = false;
newRtp.tmmbr = false;
newRtp.keyframe_method = webrtc::KeyFrameReqMethod::kNone;
newRtp.ulpfec_payload_type = kNullPayloadType;
newRtp.red_payload_type = kNullPayloadType;
bool use_fec = false;
std::vector<webrtc::VideoReceiveStreamInterface::Decoder> recv_codecs;
// Try Applying the codecs in the list
// we treat as success if at least one codec was applied and reception was
// started successfully.
for (const auto& codec_config : codecConfigList) {
if (auto condError = ValidateCodecConfig(codec_config);
condError != kMediaConduitNoError) {
CSFLogError(LOGTAG, "Invalid recv codec config for %s decoder: %i",
codec_config.mName.c_str(), condError);
continue;
}
if (codec_config.mName == kUlpfecCodecName) {
newRtp.ulpfec_payload_type = codec_config.mType;
continue;
}
// Set RTX associated PT here so we can set it for RED without additional
// checks for things like preventing creating an unncessary decoder for
// RED. This assumes that any codecs created with RTX enabled
// (including those not found in SupportedCodecType) intend to use it.
if (codec_config.RtxPayloadTypeIsSet()) {
newRtp.rtx_associated_payload_types[codec_config.mRTXPayloadType] =
codec_config.mType;
}
if (codec_config.mName == kRedCodecName) {
newRtp.red_payload_type = codec_config.mType;
continue;
}
if (SupportedCodecType(
webrtc::PayloadStringToCodecType(codec_config.mName)) ==
webrtc::VideoCodecType::kVideoCodecGeneric) {
CSFLogError(LOGTAG, "%s Unknown decoder type: %s", __FUNCTION__,
codec_config.mName.c_str());
continue;
}
// Check for the keyframe request type: PLI is preferred over FIR, and FIR
// is preferred over none.
if (codec_config.RtcpFbNackIsSet(kRtcpFbNackParamPli)) {
newRtp.keyframe_method = webrtc::KeyFrameReqMethod::kPliRtcp;
} else if (newRtp.keyframe_method !=
webrtc::KeyFrameReqMethod::kPliRtcp &&
codec_config.RtcpFbCcmIsSet(kRtcpFbCcmParamFir)) {
newRtp.keyframe_method = webrtc::KeyFrameReqMethod::kFirRtcp;
}
// What if codec A has Nack and REMB, and codec B has TMMBR, and codec C
// has none? In practice, that's not a useful configuration, and
// VideoReceiveStream::Config can't represent that, so simply union the
// (boolean) settings
if (codec_config.RtcpFbNackIsSet(kParamValueEmpty)) {
newRtp.nack.rtp_history_ms = 1000;
}
newRtp.tmmbr |= codec_config.RtcpFbCcmIsSet(kRtcpFbCcmParamTmmbr);
newRtp.remb |= codec_config.RtcpFbRembIsSet();
use_fec |= codec_config.RtcpFbFECIsSet();
auto& decoder = recv_codecs.emplace_back();
decoder.video_format = webrtc::SdpVideoFormat(codec_config.mName);
decoder.payload_type = codec_config.mType;
}
if (!use_fec) {
// Reset to defaults
newRtp.ulpfec_payload_type = kNullPayloadType;
newRtp.red_payload_type = kNullPayloadType;
}
// TODO: This would be simpler, but for some reason gives
// "error: invalid operands to binary expression
// ('webrtc::VideoReceiveStreamInterface::Decoder' and
// 'webrtc::VideoReceiveStreamInterface::Decoder')"
// if (recv_codecs != mRecvStreamConfig.decoders) {
if (!std::equal(recv_codecs.begin(), recv_codecs.end(),
mRecvStreamConfig.decoders.begin(),
mRecvStreamConfig.decoders.end(),
[](const auto& aLeft, const auto& aRight) {
return aLeft == aRight;
})) {
if (recv_codecs.empty()) {
CSFLogError(LOGTAG, "%s Found no valid receive codecs", __FUNCTION__);
}
mRecvStreamConfig.decoders = std::move(recv_codecs);
}
if (mRecvStreamConfig.rtp != newRtp) {
mRecvStreamConfig.rtp = newRtp;
remoteSsrcUpdateNeeded = true;
}
}
{
// mSendStreamConfig and other members need the lock
MutexAutoLock lock(mMutex);
if (mControl.mLocalSsrcs.Ref() != mSendStreamConfig.rtp.ssrcs) {
mSendStreamConfig.rtp.ssrcs = mControl.mLocalSsrcs;
sendStreamRecreationNeeded = true;
const uint32_t localSsrc = mSendStreamConfig.rtp.ssrcs.empty()
? 0
: mSendStreamConfig.rtp.ssrcs.front();
if (localSsrc != mRecvStreamConfig.rtp.local_ssrc) {
mRecvStreamConfig.rtp.local_ssrc = localSsrc;
}
}
{
Ssrcs localRtxSsrcs = mControl.mLocalRtxSsrcs.Ref();
if (!mControl.mSendCodec.Ref()
.map([](const auto& aCodec) {
return aCodec.RtxPayloadTypeIsSet();
})
.valueOr(false)) {
localRtxSsrcs.clear();
}
if (localRtxSsrcs != mSendStreamConfig.rtp.rtx.ssrcs) {
mSendStreamConfig.rtp.rtx.ssrcs = localRtxSsrcs;
sendStreamRecreationNeeded = true;
}
}
if (mControl.mLocalCname.Ref() != mSendStreamConfig.rtp.c_name) {
mSendStreamConfig.rtp.c_name = mControl.mLocalCname;
sendStreamRecreationNeeded = true;
}
if (mControl.mMid.Ref() != mSendStreamConfig.rtp.mid) {
mSendStreamConfig.rtp.mid = mControl.mMid;
sendStreamRecreationNeeded = true;
}
if (mControl.mLocalSendRtpExtensions.Ref() !=
mSendStreamConfig.rtp.extensions) {
mSendStreamConfig.rtp.extensions = mControl.mLocalSendRtpExtensions;
sendStreamRecreationNeeded = true;
}
if (const auto [codecConfig, rtpRtcpConfig] = std::make_pair(
mControl.mSendCodec.Ref(), mControl.mSendRtpRtcpConfig.Ref());
codecConfig.isSome() && rtpRtcpConfig.isSome() &&
(codecConfig != mControl.mConfiguredSendCodec ||
rtpRtcpConfig != mControl.mConfiguredSendRtpRtcpConfig)) {
CSFLogDebug(LOGTAG, "Configuring codec %s", codecConfig->mName.c_str());
if (mControl.mConfiguredSendCodec.isSome() &&
(mControl.mConfiguredSendCodec->mName != codecConfig->mName)) {
// This tends to happen if the codec is changed mid call.
// We need to delete the stream now, if we continue to setup the new
// codec before deleting the send stream libwebrtc will throw erros.
DeleteSendStream();
}
mControl.mConfiguredSendCodec = codecConfig;
mControl.mConfiguredSendRtpRtcpConfig = rtpRtcpConfig;
if (ValidateCodecConfig(*codecConfig) == kMediaConduitNoError) {
encoderReconfigureNeeded = true;
sendSourceUpdateNeeded = true;
mCurSendCodecConfig = codecConfig;
size_t streamCount = std::min(codecConfig->mEncodings.size(),
(size_t)webrtc::kMaxSimulcastStreams);
MOZ_RELEASE_ASSERT(streamCount >= 1,
"streamCount should be at least one");
CSFLogDebug(LOGTAG,
"Updating send codec for VideoConduit:%p stream count:%zu",
this, streamCount);
// So we can comply with b=TIAS/b=AS/maxbr=X when input resolution
// changes
MOZ_ASSERT(codecConfig->mTias < INT_MAX);
mNegotiatedMaxBitrate = static_cast<int>(codecConfig->mTias);
if (!mLastSize && mMinBitrateEstimate != 0) {
// Only do this at the start; use "have we sent a frame" as a
// reasonable stand-in. min <= start <= max (but all three parameters
// are optional)
webrtc::BitrateSettings settings;
settings.min_bitrate_bps = mMinBitrateEstimate;
settings.start_bitrate_bps = mMinBitrateEstimate;
mCall->Call()->SetClientBitratePreferences(settings);
}
// XXX parse the encoded SPS/PPS data and set
// spsData/spsLen/ppsData/ppsLen
mEncoderConfig.video_format =
webrtc::SdpVideoFormat(codecConfig->mName);
mEncoderConfig.encoder_specific_settings =
ConfigureVideoEncoderSettings(
*codecConfig, this, mEncoderConfig.video_format.parameters);
mEncoderConfig.codec_type = SupportedCodecType(
webrtc::PayloadStringToCodecType(codecConfig->mName));
MOZ_RELEASE_ASSERT(mEncoderConfig.codec_type !=
webrtc::VideoCodecType::kVideoCodecGeneric);
mEncoderConfig.content_type =
mControl.mCodecMode.Ref() == webrtc::VideoCodecMode::kRealtimeVideo
? webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo
: webrtc::VideoEncoderConfig::ContentType::kScreen;
mEncoderConfig.frame_drop_enabled = true;
mEncoderConfig.min_transmit_bitrate_bps = mMinBitrate;
// Set the max bitrate, defaulting to 10Mbps, checking:
// - pref
// - b=TIAS
// - codec constraints
// - encoding parameter if there's a single stream
int maxBps = KBPS(10000);
maxBps = MinIgnoreZero(maxBps, mPrefMaxBitrate);
maxBps = MinIgnoreZero(maxBps, mNegotiatedMaxBitrate);
maxBps = MinIgnoreZero(
maxBps, static_cast<int>(codecConfig->mEncodingConstraints.maxBr));
if (codecConfig->mEncodings.size() == 1) {
maxBps = MinIgnoreZero(
maxBps,
static_cast<int>(codecConfig->mEncodings[0].constraints.maxBr));
}
mEncoderConfig.max_bitrate_bps = maxBps;
// TODO this is for webrtc-priority, but needs plumbing bits
mEncoderConfig.bitrate_priority = 1.0;
// Populate simulcast_layers with their config.
mEncoderConfig.simulcast_layers.clear();
const auto& codecConstraints = codecConfig->mEncodingConstraints;
for (size_t idx = 0; idx < streamCount; ++idx) {
webrtc::VideoStream video_stream;
const auto& encoding = codecConfig->mEncodings[idx];
const auto& encodingConstraints = encoding.constraints;
video_stream.active = encoding.active;
// Dimensions set here are used by libwebrtc to set the maximum to
// request from the source, to support scaleDownTo without another
// encoder reconfiguration on the first frame.
// We don't support scaleDownTo but use max-width and max-height if
// signaled.
video_stream.width = codecConfig->mEncodingConstraints.maxWidth;
video_stream.height = codecConfig->mEncodingConstraints.maxHeight;
// Max framerate is also used to cap the source, to avoid processing
// frames that will have to be dropped. Our signals here are both
// RTCRtpEncodingParameters.maxFramerate (per encoding) and max-fr
// for supported codecs.
video_stream.max_framerate = static_cast<int>(([&]() {
if (codecConstraints.maxFps && encodingConstraints.maxFps) {
return std::min(*codecConstraints.maxFps,
*encodingConstraints.maxFps);
}
return codecConstraints.maxFps
.orElse([&] { return encodingConstraints.maxFps; })
.valueOr(-1);
})());
// At this time, other values are not used until after
// CreateEncoderStreams(). We fill these in directly from the codec
// config in VideoStreamFactory.
mEncoderConfig.simulcast_layers.push_back(video_stream);
}
// Expected max number of encodings
mEncoderConfig.number_of_streams =
mEncoderConfig.simulcast_layers.size();
// libwebrtc disables this by default.
mSendStreamConfig.suspend_below_min_bitrate = false;
webrtc::RtpConfig newRtp = mSendStreamConfig.rtp;
MOZ_ASSERT(newRtp == mSendStreamConfig.rtp);
newRtp.payload_name = codecConfig->mName;
newRtp.payload_type = codecConfig->mType;
newRtp.rtcp_mode = rtpRtcpConfig->GetRtcpMode();
newRtp.extmap_allow_mixed = rtpRtcpConfig->GetExtmapAllowMixed();
newRtp.max_packet_size = kVideoMtu;
newRtp.rtx.payload_type = codecConfig->RtxPayloadTypeIsSet()
? codecConfig->mRTXPayloadType
: kNullPayloadType;
{
// See Bug 1297058, enabling FEC when basic NACK is to be enabled in
// H.264 is problematic
const bool useFECDefaults =
!codecConfig->RtcpFbFECIsSet() ||
(codecConfig->mName == kH264CodecName &&
codecConfig->RtcpFbNackIsSet(kParamValueEmpty));
newRtp.ulpfec.ulpfec_payload_type =
useFECDefaults ? kNullPayloadType
: codecConfig->mULPFECPayloadType;
newRtp.ulpfec.red_payload_type =
useFECDefaults ? kNullPayloadType : codecConfig->mREDPayloadType;
newRtp.ulpfec.red_rtx_payload_type =
useFECDefaults ? kNullPayloadType
: codecConfig->mREDRTXPayloadType;
}
newRtp.nack.rtp_history_ms =
codecConfig->RtcpFbNackIsSet(kParamValueEmpty) ? 1000 : 0;
newRtp.rids.clear();
if (!codecConfig->mEncodings.empty() &&
!codecConfig->mEncodings[0].rid.empty()) {
for (const auto& encoding : codecConfig->mEncodings) {
newRtp.rids.push_back(encoding.rid);
}
}
if (mSendStreamConfig.rtp != newRtp) {
mSendStreamConfig.rtp = newRtp;
sendStreamRecreationNeeded = true;
}
}
}
if (mControl.mConfiguredDegradationPreference !=
mControl.mVideoDegradationPreference) {
mControl.mConfiguredDegradationPreference =
mControl.mVideoDegradationPreference.Ref();
if (mSendStream) {
mSendStream->SetSource(mTrackSource, DegradationPreference());
}
}
{
const auto& mode = mControl.mCodecMode.Ref();
MOZ_ASSERT(mode == webrtc::VideoCodecMode::kRealtimeVideo ||
mode == webrtc::VideoCodecMode::kScreensharing);
auto contentType =
mode == webrtc::VideoCodecMode::kRealtimeVideo
? webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo
: webrtc::VideoEncoderConfig::ContentType::kScreen;
if (contentType != mEncoderConfig.content_type) {
encoderReconfigureNeeded = true;
sendSourceUpdateNeeded = true;
mEncoderConfig.content_type = contentType;
}
}
if (mControl.mConfiguredFrameTransformerProxySend.get() !=
mControl.mFrameTransformerProxySend.Ref().get()) {
mControl.mConfiguredFrameTransformerProxySend =
mControl.mFrameTransformerProxySend.Ref();
if (!mSendStreamConfig.frame_transformer) {
mSendStreamConfig.frame_transformer =
new rtc::RefCountedObject<FrameTransformer>(true);
sendStreamRecreationNeeded = true;
}
static_cast<FrameTransformer*>(mSendStreamConfig.frame_transformer.get())
->SetProxy(mControl.mConfiguredFrameTransformerProxySend);
}
if (mControl.mConfiguredFrameTransformerProxyRecv.get() !=
mControl.mFrameTransformerProxyRecv.Ref().get()) {
mControl.mConfiguredFrameTransformerProxyRecv =
mControl.mFrameTransformerProxyRecv.Ref();
if (!mRecvStreamConfig.frame_transformer) {
mRecvStreamConfig.frame_transformer =
new rtc::RefCountedObject<FrameTransformer>(true);
}
static_cast<FrameTransformer*>(mRecvStreamConfig.frame_transformer.get())
->SetProxy(mControl.mConfiguredFrameTransformerProxyRecv);
// No flag to set, we always recreate recv streams
}
if (remoteSsrcUpdateNeeded) {
SetRemoteSSRCConfig(mControl.mConfiguredRemoteSsrc,
mControl.mConfiguredRemoteRtxSsrc);
}
// Handle un-signalled SSRCs by creating random ones and then when they
// actually get set, we'll destroy and recreate.
if (mControl.mReceiving || mControl.mTransmitting) {
const auto remoteSsrc = mRecvStreamConfig.rtp.remote_ssrc;
const auto localSsrc = mRecvStreamConfig.rtp.local_ssrc;
const auto localSsrcs = mSendStreamConfig.rtp.ssrcs;
EnsureLocalSSRC();
if (mControl.mReceiving) {
EnsureRemoteSSRC();
}
if (localSsrc != mRecvStreamConfig.rtp.local_ssrc ||
remoteSsrc != mRecvStreamConfig.rtp.remote_ssrc) {
}
if (localSsrcs != mSendStreamConfig.rtp.ssrcs) {
sendStreamRecreationNeeded = true;
}
}
// Recreate receiving streams
if (mControl.mReceiving) {
DeleteRecvStream();
CreateRecvStream();
}
if (sendStreamRecreationNeeded) {
encoderReconfigureNeeded = false;
DeleteSendStream();
}
if (mControl.mTransmitting) {
CreateSendStream();
}
}
// We make sure to not hold the lock while stopping/starting/reconfiguring
// streams, so as to not cause deadlocks. These methods can cause our platform
// codecs to dispatch sync runnables to main, and main may grab the lock.
if (mSendStream) {
if (encoderReconfigureNeeded) {
MOZ_DIAGNOSTIC_ASSERT(
mSendStreamConfig.rtp.ssrcs.size() ==
mEncoderConfig.number_of_streams,
"Each video substream must have a corresponding ssrc.");
mEncoderConfig.video_stream_factory = CreateVideoStreamFactory();
mSendStream->ReconfigureVideoEncoder(mEncoderConfig.Copy());
}
if (sendSourceUpdateNeeded && mTrackSource) {
mSendStream->SetSource(mTrackSource, DegradationPreference());
}
}
if (!mControl.mReceiving) {
StopReceiving();
}
if (!mControl.mTransmitting) {
StopTransmitting();
}
if (mControl.mReceiving) {
StartReceiving();
}
if (mControl.mTransmitting) {
StartTransmitting();
}
}
std::vector<unsigned int> WebrtcVideoConduit::GetLocalSSRCs() const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
return mSendStreamConfig.rtp.ssrcs;
}
Maybe<Ssrc> WebrtcVideoConduit::GetAssociatedLocalRtxSSRC(Ssrc aSsrc) const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
for (size_t i = 0; i < mSendStreamConfig.rtp.ssrcs.size() &&
i < mSendStreamConfig.rtp.rtx.ssrcs.size();
++i) {
if (mSendStreamConfig.rtp.ssrcs[i] == aSsrc) {
return Some(mSendStreamConfig.rtp.rtx.ssrcs[i]);
}
}
return Nothing();
}
Maybe<gfx::IntSize> WebrtcVideoConduit::GetLastResolution() const {
MutexAutoLock lock(mMutex);
return mLastSize;
}
void WebrtcVideoConduit::DeleteSendStream() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertCurrentThreadOwns();
if (!mSendStream) {
return;
}
mCall->Call()->DestroyVideoSendStream(mSendStream);
mEngineTransmitting = false;
mSendStream = nullptr;
// Reset base_seqs in case ssrcs get re-used.
mRtpSendBaseSeqs.clear();
}
void WebrtcVideoConduit::CreateSendStream() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertCurrentThreadOwns();
if (mSendStream) {
return;
}
glean::webrtc_video::send_codec_used
.Get(nsDependentCString(mSendStreamConfig.rtp.payload_name.c_str()))
.Add(1);
mEncoderConfig.video_stream_factory = CreateVideoStreamFactory();
mSendStreamConfig.encoder_settings.encoder_factory = mEncoderFactory.get();
mSendStreamConfig.encoder_settings.bitrate_allocator_factory =
mCall->mVideoBitrateAllocatorFactory.get();
MOZ_DIAGNOSTIC_ASSERT(
mSendStreamConfig.rtp.ssrcs.size() == mEncoderConfig.number_of_streams,
"Each video substream must have a corresponding ssrc.");
mSendStream = mCall->Call()->CreateVideoSendStream(mSendStreamConfig.Copy(),
mEncoderConfig.Copy());
if (mTrackSource) {
mSendStream->SetSource(mTrackSource, DegradationPreference());
}
}
void WebrtcVideoConduit::DeleteRecvStream() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertCurrentThreadOwns();
if (!mRecvStream) {
return;
}
mCall->Call()->DestroyVideoReceiveStream(mRecvStream);
mEngineReceiving = false;
mRecvStream = nullptr;
}
void WebrtcVideoConduit::CreateRecvStream() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertCurrentThreadOwns();
if (mRecvStream) {
return;
}
mRecvStreamConfig.renderer = &mRecvSinkProxy;
for (auto& decoder : mRecvStreamConfig.decoders) {
glean::webrtc_video::recv_codec_used
.Get(nsDependentCString(decoder.video_format.name.c_str()))
.Add(1);
}
mRecvStreamConfig.decoder_factory = mDecoderFactory.get();
mRecvStream =
mCall->Call()->CreateVideoReceiveStream(mRecvStreamConfig.Copy());
// Ensure that we set the jitter buffer target on this stream.
mRecvStream->SetBaseMinimumPlayoutDelayMs(mJitterBufferTargetMs);
CSFLogDebug(LOGTAG, "Created VideoReceiveStream %p for SSRC %u (0x%x)",
mRecvStream, mRecvStreamConfig.rtp.remote_ssrc,
mRecvStreamConfig.rtp.remote_ssrc);
}
void WebrtcVideoConduit::NotifyUnsetCurrentRemoteSSRC() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
CSFLogDebug(LOGTAG, "%s (%p): Unsetting SSRC %u in other conduits",
__FUNCTION__, this, mRecvStreamConfig.rtp.remote_ssrc);
mCall->UnregisterConduit(this);
mCall->UnsetRemoteSSRC(mRecvStreamConfig.rtp.remote_ssrc);
mCall->RegisterConduit(this);
}
void WebrtcVideoConduit::SetRemoteSSRCConfig(uint32_t aSsrc,
uint32_t aRtxSsrc) {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
CSFLogDebug(LOGTAG, "%s: SSRC %u (0x%x)", __FUNCTION__, aSsrc, aSsrc);
if (mRecvStreamConfig.rtp.remote_ssrc != aSsrc) {
nsCOMPtr<nsIDirectTaskDispatcher> dtd = do_QueryInterface(mCallThread);
MOZ_ALWAYS_SUCCEEDS(dtd->DispatchDirectTask(NewRunnableMethod(
"WebrtcVideoConduit::NotifyUnsetCurrentRemoteSSRC", this,
&WebrtcVideoConduit::NotifyUnsetCurrentRemoteSSRC)));
}
mRecvSSRC = mRecvStreamConfig.rtp.remote_ssrc = aSsrc;
// If we have no associated PT then ensure we dont have an rtx_ssrc set.
mRecvStreamConfig.rtp.rtx_ssrc =
mRecvStreamConfig.rtp.rtx_associated_payload_types.empty() ? 0 : aRtxSsrc;
}
void WebrtcVideoConduit::SetRemoteSSRCAndRestartAsNeeded(uint32_t aSsrc,
uint32_t aRtxSsrc) {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
if (mRecvStreamConfig.rtp.remote_ssrc == aSsrc &&
mRecvStreamConfig.rtp.rtx_ssrc == aRtxSsrc) {
return;
}
SetRemoteSSRCConfig(aSsrc, aRtxSsrc);
const bool wasReceiving = mEngineReceiving;
const bool hadRecvStream = mRecvStream;
StopReceiving();
if (hadRecvStream) {
MutexAutoLock lock(mMutex);
DeleteRecvStream();
CreateRecvStream();
}
if (wasReceiving) {
StartReceiving();
}
}
void WebrtcVideoConduit::EnsureRemoteSSRC() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertCurrentThreadOwns();
const auto& ssrcs = mSendStreamConfig.rtp.ssrcs;
if (mRecvStreamConfig.rtp.remote_ssrc != 0 &&
std::find(ssrcs.begin(), ssrcs.end(),
mRecvStreamConfig.rtp.remote_ssrc) == ssrcs.end()) {
return;
}
uint32_t ssrc;
do {
ssrc = GenerateRandomSSRC();
} while (
NS_WARN_IF(std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end()));
CSFLogDebug(LOGTAG, "VideoConduit %p: Generated remote SSRC %u", this, ssrc);
SetRemoteSSRCConfig(ssrc, 0);
}
void WebrtcVideoConduit::EnsureLocalSSRC() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertCurrentThreadOwns();
auto& ssrcs = mSendStreamConfig.rtp.ssrcs;
if (ssrcs.empty()) {
ssrcs.push_back(GenerateRandomSSRC());
}
// Reverse-iterating here so that the first dupe in `ssrcs` always wins.
for (auto& ssrc : Reversed(ssrcs)) {
if (ssrc != 0 && ssrc != mRecvStreamConfig.rtp.remote_ssrc &&
std::count(ssrcs.begin(), ssrcs.end(), ssrc) == 1) {
continue;
}
do {
ssrc = GenerateRandomSSRC();
} while (NS_WARN_IF(ssrc == mRecvStreamConfig.rtp.remote_ssrc) ||
NS_WARN_IF(std::count(ssrcs.begin(), ssrcs.end(), ssrc) > 1));
CSFLogDebug(LOGTAG, "%s (%p): Generated local SSRC %u", __FUNCTION__, this,
ssrc);
}
mRecvStreamConfig.rtp.local_ssrc = ssrcs[0];
}
void WebrtcVideoConduit::UnsetRemoteSSRC(uint32_t aSsrc) {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertNotCurrentThreadOwns();
if (mRecvStreamConfig.rtp.remote_ssrc != aSsrc &&
mRecvStreamConfig.rtp.rtx_ssrc != aSsrc) {
return;
}
const auto& ssrcs = mSendStreamConfig.rtp.ssrcs;
uint32_t our_ssrc = 0;
do {
our_ssrc = GenerateRandomSSRC();
} while (NS_WARN_IF(our_ssrc == aSsrc) ||
NS_WARN_IF(std::find(ssrcs.begin(), ssrcs.end(), our_ssrc) !=
ssrcs.end()));
CSFLogDebug(LOGTAG, "%s (%p): Generated remote SSRC %u", __FUNCTION__, this,
our_ssrc);
// There is a (tiny) chance that this new random ssrc will collide with some
// other conduit's remote ssrc, in which case that conduit will choose a new
// one.
SetRemoteSSRCAndRestartAsNeeded(our_ssrc, 0);
}
/*static*/
unsigned WebrtcVideoConduit::ToLibwebrtcMaxFramerate(
const Maybe<double>& aMaxFramerate) {
Maybe<unsigned> negotiatedMaxFps;
if (aMaxFramerate.isSome()) {
// libwebrtc does not handle non-integer max framerate.
unsigned integerMaxFps = static_cast<unsigned>(std::round(*aMaxFramerate));
// libwebrtc crashes with a max framerate of 0, even though the
// spec says this is valid. For now, we treat this as no limit.
if (integerMaxFps) {
negotiatedMaxFps = Some(integerMaxFps);
}
}
// We do not use DEFAULT_VIDEO_MAX_FRAMERATE here; that is used at the very
// end in VideoStreamFactory, once codec-wide and per-encoding limits are
// known.
return negotiatedMaxFps.refOr(std::numeric_limits<unsigned int>::max());
}
Maybe<Ssrc> WebrtcVideoConduit::GetRemoteSSRC() const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
// libwebrtc uses 0 to mean a lack of SSRC. That is not to spec.
return mRecvStreamConfig.rtp.remote_ssrc == 0
? Nothing()
: Some(mRecvStreamConfig.rtp.remote_ssrc);
}
Maybe<webrtc::VideoReceiveStreamInterface::Stats>
WebrtcVideoConduit::GetReceiverStats() const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
if (!mRecvStream) {
return Nothing();
}
return Some(mRecvStream->GetStats());
}
Maybe<webrtc::VideoSendStream::Stats> WebrtcVideoConduit::GetSenderStats()
const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
if (!mSendStream) {
return Nothing();
}
return Some(mSendStream->GetStats());
}
Maybe<webrtc::Call::Stats> WebrtcVideoConduit::GetCallStats() const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
if (!mCall->Call()) {
return Nothing();
}
return Some(mCall->Call()->GetStats());
}
MediaConduitErrorCode WebrtcVideoConduit::Init() {
MOZ_ASSERT(NS_IsMainThread());
CSFLogDebug(LOGTAG, "%s this=%p", __FUNCTION__, this);
#ifdef MOZ_WIDGET_ANDROID
if (mozilla::camera::VideoEngine::SetAndroidObjects() != 0) {
CSFLogError(LOGTAG, "%s: could not set Android objects", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
#endif // MOZ_WIDGET_ANDROID
mSendPluginCreated = mEncoderFactory->CreatedGmpPluginEvent().Connect(
GetMainThreadSerialEventTarget(),
[self = detail::RawPtr(this)](uint64_t aPluginID) {
self.get()->mSendCodecPluginIDs.AppendElement(aPluginID);
});
mSendPluginReleased = mEncoderFactory->ReleasedGmpPluginEvent().Connect(
GetMainThreadSerialEventTarget(),
[self = detail::RawPtr(this)](uint64_t aPluginID) {
self.get()->mSendCodecPluginIDs.RemoveElement(aPluginID);
});
mRecvPluginCreated = mDecoderFactory->CreatedGmpPluginEvent().Connect(
GetMainThreadSerialEventTarget(),
[self = detail::RawPtr(this)](uint64_t aPluginID) {
self.get()->mRecvCodecPluginIDs.AppendElement(aPluginID);
});
mRecvPluginReleased = mDecoderFactory->ReleasedGmpPluginEvent().Connect(
GetMainThreadSerialEventTarget(),
[self = detail::RawPtr(this)](uint64_t aPluginID) {
self.get()->mRecvCodecPluginIDs.RemoveElement(aPluginID);
});
MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
__func__, [this, self = RefPtr<WebrtcVideoConduit>(this)] {
mCall->RegisterConduit(this);
})));
CSFLogDebug(LOGTAG, "%s Initialization Done", __FUNCTION__);
return kMediaConduitNoError;
}
RefPtr<GenericPromise> WebrtcVideoConduit::Shutdown() {
MOZ_ASSERT(NS_IsMainThread());
mSendPluginCreated.DisconnectIfExists();
mSendPluginReleased.DisconnectIfExists();
mRecvPluginCreated.DisconnectIfExists();
mRecvPluginReleased.DisconnectIfExists();
return InvokeAsync(
mCallThread, __func__, [this, self = RefPtr<WebrtcVideoConduit>(this)] {
if (mSendBitrate.NumDataValues() > 0) {
glean::webrtc::video_encoder_bitrate_avg_per_call_kbps
.AccumulateSingleSample(
static_cast<unsigned>(mSendBitrate.Mean() / 1000));
glean::webrtc::video_encoder_bitrate_std_dev_per_call_kbps
.AccumulateSingleSample(static_cast<unsigned>(
mSendBitrate.StandardDeviation() / 1000));
mSendBitrate.Clear();
}
if (mSendFramerate.NumDataValues() > 0) {
glean::webrtc::video_encoder_framerate_avg_per_call
.AccumulateSingleSample(
static_cast<unsigned>(mSendFramerate.Mean()));
glean::webrtc::video_encoder_framerate_10x_std_dev_per_call
.AccumulateSingleSample(static_cast<unsigned>(
mSendFramerate.StandardDeviation() * 10));
mSendFramerate.Clear();
}
if (mRecvBitrate.NumDataValues() > 0) {
glean::webrtc::video_decoder_bitrate_avg_per_call_kbps
.AccumulateSingleSample(
static_cast<unsigned>(mRecvBitrate.Mean() / 1000));
glean::webrtc::video_decoder_bitrate_std_dev_per_call_kbps
.AccumulateSingleSample(static_cast<unsigned>(
mRecvBitrate.StandardDeviation() / 1000));
mRecvBitrate.Clear();
}
if (mRecvFramerate.NumDataValues() > 0) {
glean::webrtc::video_decoder_framerate_avg_per_call
.AccumulateSingleSample(
static_cast<unsigned>(mRecvFramerate.Mean()));
glean::webrtc::video_decoder_framerate_10x_std_dev_per_call
.AccumulateSingleSample(static_cast<unsigned>(
mRecvFramerate.StandardDeviation() * 10));
mRecvFramerate.Clear();
}
mControl.mReceiving.DisconnectIfConnected();
mControl.mTransmitting.DisconnectIfConnected();
mControl.mLocalSsrcs.DisconnectIfConnected();
mControl.mLocalRtxSsrcs.DisconnectIfConnected();
mControl.mLocalCname.DisconnectIfConnected();
mControl.mMid.DisconnectIfConnected();
mControl.mRemoteSsrc.DisconnectIfConnected();
mControl.mRemoteRtxSsrc.DisconnectIfConnected();
mControl.mSyncGroup.DisconnectIfConnected();
mControl.mLocalRecvRtpExtensions.DisconnectIfConnected();
mControl.mLocalSendRtpExtensions.DisconnectIfConnected();
mControl.mSendCodec.DisconnectIfConnected();
mControl.mSendRtpRtcpConfig.DisconnectIfConnected();
mControl.mRecvCodecs.DisconnectIfConnected();
mControl.mRecvRtpRtcpConfig.DisconnectIfConnected();
mControl.mCodecMode.DisconnectIfConnected();
mControl.mFrameTransformerProxySend.DisconnectIfConnected();
mControl.mFrameTransformerProxyRecv.DisconnectIfConnected();
mControl.mVideoDegradationPreference.DisconnectIfConnected();
mWatchManager.Shutdown();
if (mTrackSource) {
mTrackSource->RemoveSink(&mSendSinkProxy);
}
mCall->UnregisterConduit(this);
mDecoderFactory->DisconnectAll();
mEncoderFactory->DisconnectAll();
{
MutexAutoLock lock(mMutex);
DeleteSendStream();
DeleteRecvStream();
}
return GenericPromise::CreateAndResolve(true, __func__);
});
}
webrtc::VideoCodecMode WebrtcVideoConduit::CodecMode() const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
return mControl.mCodecMode;
}
webrtc::DegradationPreference WebrtcVideoConduit::DegradationPreference()
const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
if (mControl.mConfiguredDegradationPreference !=
webrtc::DegradationPreference::DISABLED) {
return mControl.mConfiguredDegradationPreference;
}
if (mLockScaling || CodecMode() == webrtc::VideoCodecMode::kScreensharing) {
return webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
}
// Fall back to MAINTAIN_FRAMERATE by default. This is what libwebrtc/Chrome
// uses, because BALANCED hasn't been tuned yet. See
// https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=1939;drc=7c2b25f6a19cfeeea67f0f43ed33617840bab33d
return webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
}
MediaConduitErrorCode WebrtcVideoConduit::AttachRenderer(
RefPtr<mozilla::VideoRenderer> aVideoRenderer) {
MOZ_ASSERT(NS_IsMainThread());
CSFLogDebug(LOGTAG, "%s", __FUNCTION__);
// null renderer
if (!aVideoRenderer) {
CSFLogError(LOGTAG, "%s NULL Renderer", __FUNCTION__);
MOZ_ASSERT(false);
return kMediaConduitInvalidRenderer;
}
// This function is called only from main, so we only need to protect against
// modifying mRenderer while any webrtc.org code is trying to use it.
{
ReentrantMonitorAutoEnter enter(mRendererMonitor);
mRenderer = aVideoRenderer;
// Make sure the renderer knows the resolution
mRenderer->FrameSizeChange(mReceivingWidth, mReceivingHeight);
}
return kMediaConduitNoError;
}
void WebrtcVideoConduit::DetachRenderer() {
MOZ_ASSERT(NS_IsMainThread());
ReentrantMonitorAutoEnter enter(mRendererMonitor);
if (mRenderer) {
mRenderer = nullptr;
}
}
rtc::RefCountedObject<mozilla::VideoStreamFactory>*
WebrtcVideoConduit::CreateVideoStreamFactory() {
auto videoStreamFactory = mVideoStreamFactory.Lock();
*videoStreamFactory = new rtc::RefCountedObject<VideoStreamFactory>(
*mCurSendCodecConfig, mMinBitrate, mStartBitrate, mPrefMaxBitrate,
mNegotiatedMaxBitrate);
return videoStreamFactory->get();
}
void WebrtcVideoConduit::OnSendFrame(const webrtc::VideoFrame& aFrame) {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
// XXX Google uses a "timestamp_aligner" to translate timestamps from the
// camera via TranslateTimestamp(); we should look at doing the same. This
// avoids sampling error when capturing frames, but google had to deal with
// some broken cameras, include Logitech c920's IIRC.
const gfx::IntSize size{aFrame.width(), aFrame.height()};
CSFLogVerbose(LOGTAG, "WebrtcVideoConduit %p %s (send SSRC %u (0x%x))", this,
__FUNCTION__, mSendStreamConfig.rtp.ssrcs.front(),
mSendStreamConfig.rtp.ssrcs.front());
if (Some(size) != mLastSize) {
MOZ_ASSERT(size != gfx::IntSize(0, 0));
// Note coverity will flag this since it thinks they can be 0
MOZ_ASSERT(mCurSendCodecConfig);
{
MutexAutoLock lock(mMutex);
mLastSize = Some(size);
}
}
MOZ_ASSERT(!aFrame.color_space(), "Unexpected use of color space");
MOZ_ASSERT(!aFrame.has_update_rect(), "Unexpected use of update rect");
#ifdef MOZ_REAL_TIME_TRACING
if (profiler_is_active()) {
MutexAutoLock lock(mMutex);
nsAutoCStringN<256> ssrcsCommaSeparated;
bool first = true;
for (auto ssrc : mSendStreamConfig.rtp.ssrcs) {
if (!first) {
ssrcsCommaSeparated.AppendASCII(", ");
} else {
first = false;
}
ssrcsCommaSeparated.AppendInt(ssrc);
}
const webrtc::Timestamp currentTimestamp =
webrtc::Timestamp::Micros(aFrame.timestamp_us());
// The first frame has a delta of zero.
webrtc::TimeDelta timestampDelta =
mLastTimestampSend.isSome()
? currentTimestamp - mLastTimestampSend.value()
: webrtc::TimeDelta::Zero();
mLastTimestampSend = Some(currentTimestamp);
TRACE_COMMENT("VideoConduit::OnSendFrame (async)",
"t-delta=%.1fms, ssrcs=%s", timestampDelta.ms<double>(),
ssrcsCommaSeparated.get());
}
#endif
}
void WebrtcVideoConduit::SetTrackSource(
webrtc::VideoTrackSourceInterface* aSource) {
MOZ_ASSERT(NS_IsMainThread());
MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
__func__, [this, self = RefPtr(this), src = RefPtr(aSource)] {
if (mTrackSource) {
mTrackSource->RemoveSink(&mSendSinkProxy);
}
if (src) {
src->AddOrUpdateSink(&mSendSinkProxy, {});
}
mTrackSource = src;
if (mSendStream) {
mSendStream->SetSource(mTrackSource, DegradationPreference());
}
})));
}
// Transport Layer Callbacks
void WebrtcVideoConduit::DeliverPacket(rtc::CopyOnWriteBuffer packet,
PacketType type) {
// Currently unused.
MOZ_ASSERT(false);
}
void WebrtcVideoConduit::OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
webrtc::RTPHeader&& aHeader) {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
// We should only be handling packets on this conduit if we are set to receive them.
if (!mControl.mReceiving) {
// TODO: Create profiler marker for this and/or less noisy logging.
// CSFLogInfo(LOGTAG, "VideoConduit %p: Discarding packet SEQ# %u SSRC %u as not configured to receive.",
// this, aPacket.SequenceNumber(), aHeader.ssrc);
return;
}
mRemoteSendSSRC = aHeader.ssrc;
if (mAllowSsrcChange || mRecvStreamConfig.rtp.remote_ssrc == 0) {
bool switchRequired = mRecvStreamConfig.rtp.remote_ssrc != aHeader.ssrc;
if (switchRequired) {
// Handle the unknown ssrc (and ssrc-not-signaled case).
// We need to check that the newly received ssrc is not already
// associated with ulpfec or rtx. This is how webrtc.org handles
// things, see https://codereview.webrtc.org/1226093002.
const webrtc::VideoReceiveStreamInterface::Config::Rtp& rtp =
mRecvStreamConfig.rtp;
switchRequired =
rtp.rtx_associated_payload_types.find(aHeader.payloadType) ==
rtp.rtx_associated_payload_types.end() &&
rtp.ulpfec_payload_type != aHeader.payloadType;
}
if (switchRequired) {
CSFLogInfo(LOGTAG, "VideoConduit %p: Switching remote SSRC from %u to %u",
this, mRecvStreamConfig.rtp.remote_ssrc, aHeader.ssrc);
SetRemoteSSRCAndRestartAsNeeded(aHeader.ssrc, 0);
}
}
CSFLogVerbose(LOGTAG, "%s: seq# %u, Len %zu, SSRC %u (0x%x) ", __FUNCTION__,
aPacket.SequenceNumber(), aPacket.size(), aPacket.Ssrc(),
aPacket.Ssrc());
// Libwebrtc commit cde4b67d9d now expect calls to
// SourceTracker::GetSources() to happen on the call thread. We'll
// grab the value now while on the call thread, and dispatch to main
// to store the cached value if we have new source information.
// See Bug 1845621.
std::vector<webrtc::RtpSource> sources;
if (mRecvStream) {
sources = mRecvStream->GetSources();
}
bool needsCacheUpdate = false;
{
MutexAutoLock lock(mMutex);
needsCacheUpdate = sources != mRtpSources;
}
// only dispatch to main if we have new data
if (needsCacheUpdate) {
GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
__func__, [this, rtpSources = std::move(sources),
self = RefPtr<WebrtcVideoConduit>(this)]() {
MutexAutoLock lock(mMutex);
mRtpSources = rtpSources;
}));
}
mRtpPacketEvent.Notify();
if (mCall->Call()) {
mCall->Call()->Receiver()->DeliverRtpPacket(
webrtc::MediaType::VIDEO, std::move(aPacket),
[self = RefPtr<WebrtcVideoConduit>(this)](
const webrtc::RtpPacketReceived& packet) {
CSFLogVerbose(
LOGTAG,
"VideoConduit %p: failed demuxing packet, ssrc: %u seq: %u",
self.get(), packet.Ssrc(), packet.SequenceNumber());
return false;
});
}
}
Maybe<uint16_t> WebrtcVideoConduit::RtpSendBaseSeqFor(uint32_t aSsrc) const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
auto it = mRtpSendBaseSeqs.find(aSsrc);
if (it == mRtpSendBaseSeqs.end()) {
return Nothing();
}
return Some(it->second);
}
const dom::RTCStatsTimestampMaker& WebrtcVideoConduit::GetTimestampMaker()
const {
return mCall->GetTimestampMaker();
}
void WebrtcVideoConduit::StopTransmitting() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertNotCurrentThreadOwns();
if (!mEngineTransmitting) {
return;
}
if (mSendStream) {
CSFLogDebug(LOGTAG, "%s Stopping send stream", __FUNCTION__);
mSendStream->Stop();
}
mEngineTransmitting = false;
}
void WebrtcVideoConduit::StartTransmitting() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
MOZ_ASSERT(mSendStream);
mMutex.AssertNotCurrentThreadOwns();
if (mEngineTransmitting) {
return;
}
CSFLogDebug(LOGTAG, "%s Starting send stream", __FUNCTION__);
mSendStream->Start();
// XXX File a bug to consider hooking this up to the state of mtransport
mCall->Call()->SignalChannelNetworkState(webrtc::MediaType::VIDEO,
webrtc::kNetworkUp);
mEngineTransmitting = true;
}
void WebrtcVideoConduit::StopReceiving() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
mMutex.AssertNotCurrentThreadOwns();
// Are we receiving already? If so, stop receiving and playout
// since we can't apply new recv codec when the engine is playing.
if (!mEngineReceiving) {
return;
}
if (mRecvStream) {
CSFLogDebug(LOGTAG, "%s Stopping receive stream", __FUNCTION__);
mRecvStream->Stop();
}
mEngineReceiving = false;
}
void WebrtcVideoConduit::StartReceiving() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
MOZ_ASSERT(mRecvStream);
mMutex.AssertNotCurrentThreadOwns();
if (mEngineReceiving) {
return;
}
CSFLogDebug(LOGTAG, "%s Starting receive stream (SSRC %u (0x%x))",
__FUNCTION__, mRecvStreamConfig.rtp.remote_ssrc,
mRecvStreamConfig.rtp.remote_ssrc);
// Start Receiving on the video engine
mRecvStream->Start();
// XXX File a bug to consider hooking this up to the state of mtransport
mCall->Call()->SignalChannelNetworkState(webrtc::MediaType::VIDEO,
webrtc::kNetworkUp);
mEngineReceiving = true;
}
bool WebrtcVideoConduit::SendRtp(const uint8_t* aData, size_t aLength,
const webrtc::PacketOptions& aOptions) {
MOZ_ASSERT(aLength >= 12);
const uint16_t seqno = ntohs(*((uint16_t*)&aData[2]));
const uint32_t ssrc = ntohl(*((uint32_t*)&aData[8]));
CSFLogVerbose(
LOGTAG,
"VideoConduit %p: Sending RTP Packet seq# %u, len %zu, SSRC %u (0x%x)",
this, seqno, aLength, ssrc, ssrc);
if (!mTransportActive) {
CSFLogError(LOGTAG, "VideoConduit %p: RTP Packet Send Failed", this);
return false;
}
MediaPacket packet;
packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
packet.SetType(MediaPacket::RTP);
mSenderRtpSendEvent.Notify(std::move(packet));
// Parse the sequence number of the first rtp packet as base_seq.
const auto inserted = mRtpSendBaseSeqs_n.insert({ssrc, seqno}).second;
if (inserted || aOptions.packet_id >= 0) {
int64_t now_ms = PR_Now() / 1000;
MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
__func__, [this, self = RefPtr<WebrtcVideoConduit>(this),
packet_id = aOptions.packet_id, now_ms, ssrc, seqno] {
mRtpSendBaseSeqs.insert({ssrc, seqno});
if (packet_id >= 0) {
if (mCall->Call()) {
// TODO: This notification should ideally happen after the
// transport layer has sent the packet on the wire.
mCall->Call()->OnSentPacket({packet_id, now_ms});
}
}
})));
}
return true;
}
bool WebrtcVideoConduit::SendSenderRtcp(const uint8_t* aData, size_t aLength) {
CSFLogVerbose(
LOGTAG,
"VideoConduit %p: Sending RTCP SR Packet, len %zu, SSRC %u (0x%x)", this,
aLength, (uint32_t)ntohl(*((uint32_t*)&aData[4])),
(uint32_t)ntohl(*((uint32_t*)&aData[4])));
if (!mTransportActive) {
CSFLogError(LOGTAG, "VideoConduit %p: RTCP SR Packet Send Failed", this);
return false;
}
MediaPacket packet;
packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
packet.SetType(MediaPacket::RTCP);
mSenderRtcpSendEvent.Notify(std::move(packet));
return true;
}
bool WebrtcVideoConduit::SendReceiverRtcp(const uint8_t* aData,
size_t aLength) {
CSFLogVerbose(
LOGTAG,
"VideoConduit %p: Sending RTCP RR Packet, len %zu, SSRC %u (0x%x)", this,
aLength, (uint32_t)ntohl(*((uint32_t*)&aData[4])),
(uint32_t)ntohl(*((uint32_t*)&aData[4])));
if (!mTransportActive) {
CSFLogError(LOGTAG, "VideoConduit %p: RTCP RR Packet Send Failed", this);
return false;
}
MediaPacket packet;
packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
packet.SetType(MediaPacket::RTCP);
mReceiverRtcpSendEvent.Notify(std::move(packet));
return true;
}
void WebrtcVideoConduit::OnRecvFrame(const webrtc::VideoFrame& aFrame) {
const uint32_t localRecvSsrc = mRecvSSRC;
const uint32_t remoteSendSsrc = mRemoteSendSSRC;
CSFLogVerbose(
LOGTAG,
"VideoConduit %p: Rendering frame, Remote SSRC %u (0x%x), size %ux%u",
this, static_cast<uint32_t>(remoteSendSsrc),
static_cast<uint32_t>(remoteSendSsrc), aFrame.width(), aFrame.height());
ReentrantMonitorAutoEnter enter(mRendererMonitor);
if (!mRenderer) {
CSFLogError(LOGTAG, "VideoConduit %p: Cannot render frame, no renderer",
this);
return;
}
bool needsNewHistoryElement = mReceivedFrameHistory.mEntries.IsEmpty();
if (mReceivingWidth != aFrame.width() ||
mReceivingHeight != aFrame.height()) {
mReceivingWidth = aFrame.width();
mReceivingHeight = aFrame.height();
mRenderer->FrameSizeChange(mReceivingWidth, mReceivingHeight);
needsNewHistoryElement = true;
}
if (!needsNewHistoryElement) {
auto& currentEntry = mReceivedFrameHistory.mEntries.LastElement();
needsNewHistoryElement =
currentEntry.mRotationAngle !=
static_cast<unsigned long>(aFrame.rotation()) ||
currentEntry.mLocalSsrc != localRecvSsrc ||
currentEntry.mRemoteSsrc != remoteSendSsrc;
}
// Record frame history
const auto historyNow = mCall->GetTimestampMaker().GetNow().ToDom();
if (needsNewHistoryElement) {
dom::RTCVideoFrameHistoryEntryInternal frameHistoryElement;
frameHistoryElement.mConsecutiveFrames = 0;
frameHistoryElement.mWidth = aFrame.width();
frameHistoryElement.mHeight = aFrame.height();
frameHistoryElement.mRotationAngle =
static_cast<unsigned long>(aFrame.rotation());
frameHistoryElement.mFirstFrameTimestamp = historyNow;
frameHistoryElement.mLocalSsrc = localRecvSsrc;
frameHistoryElement.mRemoteSsrc = remoteSendSsrc;
if (!mReceivedFrameHistory.mEntries.AppendElement(frameHistoryElement,
fallible)) {
mozalloc_handle_oom(0);
}
}
auto& currentEntry = mReceivedFrameHistory.mEntries.LastElement();
currentEntry.mConsecutiveFrames++;
currentEntry.mLastFrameTimestamp = historyNow;
// Attempt to retrieve an timestamp encoded in the image pixels if enabled.
if (mVideoLatencyTestEnable && mReceivingWidth && mReceivingHeight) {
uint64_t now = PR_Now();
uint64_t timestamp = 0;
uint8_t* data =
const_cast<uint8_t*>(aFrame.video_frame_buffer()->GetI420()->DataY());
bool ok = YuvStamper::Decode(
mReceivingWidth, mReceivingHeight, mReceivingWidth, data,
reinterpret_cast<unsigned char*>(&timestamp), sizeof(timestamp), 0, 0);
if (ok) {
VideoLatencyUpdate(now - timestamp);
}
}
#ifdef MOZ_REAL_TIME_TRACING
if (profiler_is_active()) {
MutexAutoLock lock(mMutex);
// The first frame has a delta of zero.
uint32_t rtpTimestamp = aFrame.rtp_timestamp();
uint32_t timestampDelta =
mLastRTPTimestampReceive.isSome()
? rtpTimestamp - mLastRTPTimestampReceive.value()
: 0;
mLastRTPTimestampReceive = Some(rtpTimestamp);
TRACE_COMMENT("VideoConduit::OnFrame", "t-delta=%.1fms, ssrc=%u",
timestampDelta * 1000.f / webrtc::kVideoPayloadTypeFrequency,
localRecvSsrc);
}
#endif
mRenderer->RenderVideoFrame(aFrame);
}
bool WebrtcVideoConduit::AddFrameHistory(
dom::Sequence<dom::RTCVideoFrameHistoryInternal>* outHistories) const {
ReentrantMonitorAutoEnter enter(mRendererMonitor);
if (!outHistories->AppendElement(mReceivedFrameHistory, fallible)) {
mozalloc_handle_oom(0);
return false;
}
return true;
}
void WebrtcVideoConduit::SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) {
MOZ_RELEASE_ASSERT(aTargetMs <= std::numeric_limits<uint16_t>::max());
MOZ_RELEASE_ASSERT(aTargetMs >= 0);
MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
__func__,
[this, self = RefPtr<WebrtcVideoConduit>(this), targetMs = aTargetMs] {
mJitterBufferTargetMs = static_cast<uint16_t>(targetMs);
if (mRecvStream) {
mRecvStream->SetBaseMinimumPlayoutDelayMs(targetMs);
}
})));
}
void WebrtcVideoConduit::DumpCodecDB() const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
for (const auto& entry : mControl.mConfiguredRecvCodecs) {
CSFLogDebug(LOGTAG, "Payload Name: %s", entry.mName.c_str());
CSFLogDebug(LOGTAG, "Payload Type: %d", entry.mType);
CSFLogDebug(LOGTAG, "Payload Max Frame Size: %d",
entry.mEncodingConstraints.maxFs);
if (entry.mEncodingConstraints.maxFps.isSome()) {
CSFLogDebug(LOGTAG, "Payload Max Frame Rate: %f",
*entry.mEncodingConstraints.maxFps);
}
}
}
void WebrtcVideoConduit::VideoLatencyUpdate(uint64_t aNewSample) {
mRendererMonitor.AssertCurrentThreadIn();
mVideoLatencyAvg =
(sRoundingPadding * aNewSample + sAlphaNum * mVideoLatencyAvg) /
sAlphaDen;
}
uint64_t WebrtcVideoConduit::MozVideoLatencyAvg() {
mRendererMonitor.AssertCurrentThreadIn();
return mVideoLatencyAvg / sRoundingPadding;
}
void WebrtcVideoConduit::CollectTelemetryData() {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
if (mEngineTransmitting) {
webrtc::VideoSendStream::Stats stats = mSendStream->GetStats();
mSendBitrate.Push(stats.media_bitrate_bps);
mSendFramerate.Push(stats.encode_frame_rate);
}
if (mEngineReceiving) {
webrtc::VideoReceiveStreamInterface::Stats stats = mRecvStream->GetStats();
mRecvBitrate.Push(stats.total_bitrate_bps);
mRecvFramerate.Push(stats.decode_frame_rate);
}
}
void WebrtcVideoConduit::OnRtcpBye() { mRtcpByeEvent.Notify(); }
void WebrtcVideoConduit::OnRtcpTimeout() { mRtcpTimeoutEvent.Notify(); }
void WebrtcVideoConduit::SetTransportActive(bool aActive) {
MOZ_ASSERT(mStsThread->IsOnCurrentThread());
if (mTransportActive == aActive) {
return;
}
// If false, This stops us from sending
mTransportActive = aActive;
// We queue this because there might be notifications to these listeners
// pending, and we don't want to drop them by letting this jump ahead of
// those notifications. We move the listeners into the lambda in case the
// transport comes back up before we disconnect them. (The Connect calls
// happen in MediaPipeline)
// We retain a strong reference to ourself, because the listeners are holding
// a non-refcounted reference to us, and moving them into the lambda could
// conceivably allow them to outlive us.
if (!aActive) {
MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
__func__,
[self = RefPtr<WebrtcVideoConduit>(this),
recvRtpListener = std::move(mReceiverRtpEventListener)]() mutable {
recvRtpListener.DisconnectIfExists();
})));
}
}
std::vector<webrtc::RtpSource> WebrtcVideoConduit::GetUpstreamRtpSources()
const {
MOZ_ASSERT(NS_IsMainThread());
return mRtpSources;
}
void WebrtcVideoConduit::RequestKeyFrame(FrameTransformerProxy* aProxy) {
mCallThread->Dispatch(NS_NewRunnableFunction(
__func__, [this, self = RefPtr<WebrtcVideoConduit>(this),
proxy = RefPtr<FrameTransformerProxy>(aProxy)] {
bool success = false;
if (mRecvStream && mEngineReceiving) {
// This is a misnomer. This requests a keyframe from the other side.
mRecvStream->GenerateKeyFrame();
success = true;
}
proxy->KeyFrameRequestDone(success);
}));
}
void WebrtcVideoConduit::GenerateKeyFrame(const Maybe<std::string>& aRid,
FrameTransformerProxy* aProxy) {
// libwebrtc does not implement error handling in the way that
// webrtc-encoded-transform specifies. So, we'll need to do that here.
// Also, spec wants us to synchronously check whether there's an encoder, but
// that's not something that can be checked synchronously.
mCallThread->Dispatch(NS_NewRunnableFunction(
__func__, [this, self = RefPtr(this), proxy = RefPtr(aProxy), aRid] {
// If encoder is undefined, reject promise with InvalidStateError,
// abort these steps.
// If encoder is not processing video frames, reject promise with
// InvalidStateError, abort these steps.
if (!mSendStream || !mCurSendCodecConfig || !mEngineTransmitting) {
CopyableErrorResult result;
result.ThrowInvalidStateError("No encoders");
proxy->GenerateKeyFrameError(aRid, result);
return;
}
// Gather a list of video encoders, named videoEncoders from encoder,
// ordered according negotiated RIDs if any.
// NOTE: This is represented by mCurSendCodecConfig->mEncodings
// If rid is defined, remove from videoEncoders any video encoder that
// does not match rid.
// If rid is undefined, remove from videoEncoders all video encoders
// except the first one.
bool found = false;
std::vector<std::string> rids;
if (!aRid.isSome()) {
// If rid is undefined, set rid to the RID value corresponding to
// videoEncoder.
if (!mCurSendCodecConfig->mEncodings.empty()) {
if (!mCurSendCodecConfig->mEncodings[0].rid.empty()) {
rids.push_back(mCurSendCodecConfig->mEncodings[0].rid);
}
found = true;
}
} else {
for (const auto& encoding : mCurSendCodecConfig->mEncodings) {
if (encoding.rid == *aRid) {
found = true;
rids.push_back(encoding.rid);
break;
}
}
}
// If videoEncoders is empty, reject promise with NotFoundError and
// abort these steps. videoEncoders is expected to be empty if the
// corresponding RTCRtpSender is not active, or the corresponding
// RTCRtpSender track is ended.
if (!found) {
CopyableErrorResult result;
result.ThrowNotFoundError("Rid not in use");
proxy->GenerateKeyFrameError(aRid, result);
}
// NOTE: We don't do this stuff, because libwebrtc's interface is
// rid-based.
// Let videoEncoder be the first encoder in videoEncoders.
// If rid is undefined, set rid to the RID value corresponding to
// videoEncoder.
mSendStream->GenerateKeyFrame(rids);
}));
}
bool WebrtcVideoConduit::HasCodecPluginID(uint64_t aPluginID) const {
MOZ_ASSERT(NS_IsMainThread());
return mSendCodecPluginIDs.Contains(aPluginID) ||
mRecvCodecPluginIDs.Contains(aPluginID);
}
bool WebrtcVideoConduit::HasH264Hardware() {
nsCOMPtr<nsIGfxInfo> gfxInfo = do_GetService("@mozilla.org/gfx/info;1");
if (!gfxInfo) {
return false;
}
int32_t status;
nsCString discardFailureId;
return NS_SUCCEEDED(gfxInfo->GetFeatureStatus(
nsIGfxInfo::FEATURE_WEBRTC_HW_ACCELERATION_H264, discardFailureId,
&status)) &&
status == nsIGfxInfo::FEATURE_STATUS_OK;
}
bool WebrtcVideoConduit::HasAv1() {
#if defined(MOZ_AV1)
return true;
#else
return false;
#endif
}
Maybe<int> WebrtcVideoConduit::ActiveSendPayloadType() const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
if (!mSendStream) {
return Nothing();
}
if (mSendStreamConfig.rtp.payload_type == -1) {
return Nothing();
}
return Some(mSendStreamConfig.rtp.payload_type);
}
Maybe<int> WebrtcVideoConduit::ActiveRecvPayloadType() const {
MOZ_ASSERT(mCallThread->IsOnCurrentThread());
auto stats = GetReceiverStats();
if (!stats) {
return Nothing();
}
if (stats->current_payload_type == -1) {
return Nothing();
}
return Some(stats->current_payload_type);
}
} // namespace mozilla
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