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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/call/flexfec_receive_stream_impl.cc | |
parent | Initial commit. (diff) | |
download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/call/flexfec_receive_stream_impl.cc')
-rw-r--r-- | third_party/libwebrtc/call/flexfec_receive_stream_impl.cc | 223 |
1 files changed, 223 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/flexfec_receive_stream_impl.cc b/third_party/libwebrtc/call/flexfec_receive_stream_impl.cc new file mode 100644 index 0000000000..9e00078f96 --- /dev/null +++ b/third_party/libwebrtc/call/flexfec_receive_stream_impl.cc @@ -0,0 +1,223 @@ +/* + * Copyright (c) 2016 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 "call/flexfec_receive_stream_impl.h" + +#include <stddef.h> + +#include <cstdint> +#include <string> +#include <utility> + +#include "api/array_view.h" +#include "api/call/transport.h" +#include "api/rtp_parameters.h" +#include "call/rtp_stream_receiver_controller_interface.h" +#include "modules/rtp_rtcp/include/flexfec_receiver.h" +#include "modules/rtp_rtcp/include/receive_statistics.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "rtc_base/checks.h" +#include "rtc_base/location.h" +#include "rtc_base/logging.h" +#include "rtc_base/strings/string_builder.h" +#include "system_wrappers/include/clock.h" + +namespace webrtc { + +std::string FlexfecReceiveStream::Config::ToString() const { + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); + ss << "{payload_type: " << payload_type; + ss << ", remote_ssrc: " << rtp.remote_ssrc; + ss << ", local_ssrc: " << rtp.local_ssrc; + ss << ", protected_media_ssrcs: ["; + size_t i = 0; + for (; i + 1 < protected_media_ssrcs.size(); ++i) + ss << protected_media_ssrcs[i] << ", "; + if (!protected_media_ssrcs.empty()) + ss << protected_media_ssrcs[i]; + ss << "], transport_cc: " << (rtp.transport_cc ? "on" : "off"); + ss << ", rtp.extensions: ["; + i = 0; + for (; i + 1 < rtp.extensions.size(); ++i) + ss << rtp.extensions[i].ToString() << ", "; + if (!rtp.extensions.empty()) + ss << rtp.extensions[i].ToString(); + ss << "]}"; + return ss.str(); +} + +bool FlexfecReceiveStream::Config::IsCompleteAndEnabled() const { + // Check if FlexFEC is enabled. + if (payload_type < 0) + return false; + // Do we have the necessary SSRC information? + if (rtp.remote_ssrc == 0) + return false; + // TODO(brandtr): Update this check when we support multistream protection. + if (protected_media_ssrcs.size() != 1u) + return false; + return true; +} + +namespace { + +// TODO(brandtr): Update this function when we support multistream protection. +std::unique_ptr<FlexfecReceiver> MaybeCreateFlexfecReceiver( + Clock* clock, + const FlexfecReceiveStream::Config& config, + RecoveredPacketReceiver* recovered_packet_receiver) { + if (config.payload_type < 0) { + RTC_LOG(LS_WARNING) + << "Invalid FlexFEC payload type given. " + "This FlexfecReceiveStream will therefore be useless."; + return nullptr; + } + RTC_DCHECK_GE(config.payload_type, 0); + RTC_DCHECK_LE(config.payload_type, 127); + if (config.rtp.remote_ssrc == 0) { + RTC_LOG(LS_WARNING) + << "Invalid FlexFEC SSRC given. " + "This FlexfecReceiveStream will therefore be useless."; + return nullptr; + } + if (config.protected_media_ssrcs.empty()) { + RTC_LOG(LS_WARNING) + << "No protected media SSRC supplied. " + "This FlexfecReceiveStream will therefore be useless."; + return nullptr; + } + + if (config.protected_media_ssrcs.size() > 1) { + RTC_LOG(LS_WARNING) + << "The supplied FlexfecConfig contained multiple protected " + "media streams, but our implementation currently only " + "supports protecting a single media stream. " + "To avoid confusion, disabling FlexFEC completely."; + return nullptr; + } + RTC_DCHECK_EQ(1U, config.protected_media_ssrcs.size()); + return std::unique_ptr<FlexfecReceiver>(new FlexfecReceiver( + clock, config.rtp.remote_ssrc, config.protected_media_ssrcs[0], + recovered_packet_receiver)); +} + +std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule( + Clock* clock, + ReceiveStatistics* receive_statistics, + const FlexfecReceiveStreamImpl::Config& config, + RtcpRttStats* rtt_stats) { + RtpRtcpInterface::Configuration configuration; + configuration.audio = false; + configuration.receiver_only = true; + configuration.clock = clock; + configuration.receive_statistics = receive_statistics; + configuration.outgoing_transport = config.rtcp_send_transport; + configuration.rtt_stats = rtt_stats; + configuration.local_media_ssrc = config.rtp.local_ssrc; + return ModuleRtpRtcpImpl2::Create(configuration); +} + +} // namespace + +FlexfecReceiveStreamImpl::FlexfecReceiveStreamImpl( + Clock* clock, + Config config, + RecoveredPacketReceiver* recovered_packet_receiver, + RtcpRttStats* rtt_stats) + : extension_map_(std::move(config.rtp.extensions)), + remote_ssrc_(config.rtp.remote_ssrc), + transport_cc_(config.rtp.transport_cc), + payload_type_(config.payload_type), + receiver_( + MaybeCreateFlexfecReceiver(clock, config, recovered_packet_receiver)), + rtp_receive_statistics_(ReceiveStatistics::Create(clock)), + rtp_rtcp_(CreateRtpRtcpModule(clock, + rtp_receive_statistics_.get(), + config, + rtt_stats)) { + RTC_LOG(LS_INFO) << "FlexfecReceiveStreamImpl: " << config.ToString(); + RTC_DCHECK_GE(payload_type_, -1); + + packet_sequence_checker_.Detach(); + + // RTCP reporting. + rtp_rtcp_->SetRTCPStatus(config.rtcp_mode); +} + +FlexfecReceiveStreamImpl::~FlexfecReceiveStreamImpl() { + RTC_DLOG(LS_INFO) << "~FlexfecReceiveStreamImpl: ssrc: " << remote_ssrc_; +} + +void FlexfecReceiveStreamImpl::RegisterWithTransport( + RtpStreamReceiverControllerInterface* receiver_controller) { + RTC_DCHECK_RUN_ON(&packet_sequence_checker_); + RTC_DCHECK(!rtp_stream_receiver_); + + if (!receiver_) + return; + + // TODO(nisse): OnRtpPacket in this class delegates all real work to + // `receiver_`. So maybe we don't need to implement RtpPacketSinkInterface + // here at all, we'd then delete the OnRtpPacket method and instead register + // `receiver_` as the RtpPacketSinkInterface for this stream. + rtp_stream_receiver_ = + receiver_controller->CreateReceiver(remote_ssrc(), this); +} + +void FlexfecReceiveStreamImpl::UnregisterFromTransport() { + RTC_DCHECK_RUN_ON(&packet_sequence_checker_); + rtp_stream_receiver_.reset(); +} + +void FlexfecReceiveStreamImpl::OnRtpPacket(const RtpPacketReceived& packet) { + RTC_DCHECK_RUN_ON(&packet_sequence_checker_); + if (!receiver_) + return; + + receiver_->OnRtpPacket(packet); + + // Do not report media packets in the RTCP RRs generated by `rtp_rtcp_`. + if (packet.Ssrc() == remote_ssrc()) { + rtp_receive_statistics_->OnRtpPacket(packet); + } +} + +void FlexfecReceiveStreamImpl::SetPayloadType(int payload_type) { + RTC_DCHECK_RUN_ON(&packet_sequence_checker_); + RTC_DCHECK_GE(payload_type, -1); + payload_type_ = payload_type; +} + +int FlexfecReceiveStreamImpl::payload_type() const { + RTC_DCHECK_RUN_ON(&packet_sequence_checker_); + return payload_type_; +} + +void FlexfecReceiveStreamImpl::SetRtpExtensions( + std::vector<RtpExtension> extensions) { + RTC_DCHECK_RUN_ON(&packet_sequence_checker_); + extension_map_.Reset(extensions); +} + +RtpHeaderExtensionMap FlexfecReceiveStreamImpl::GetRtpExtensionMap() const { + RTC_DCHECK_RUN_ON(&packet_sequence_checker_); + return extension_map_; +} + +void FlexfecReceiveStreamImpl::SetLocalSsrc(uint32_t local_ssrc) { + RTC_DCHECK_RUN_ON(&packet_sequence_checker_); + if (local_ssrc == rtp_rtcp_->local_media_ssrc()) + return; + + rtp_rtcp_->SetLocalSsrc(local_ssrc); +} + +} // namespace webrtc |