Revert "Refactor RtpVideoStreamReceiver without RtpReceiver."
This reverts commit 0b9e01d605a174a52635626c885738a222abff46. Reason for revert: Appears to breaks AV sync, failing perftests: CallPerfTest.PlaysOutAudioAndVideoInSyncWithVideoNtpDrift CallPerfTest.PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift CallPerfTest.PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift Original change's description: > Refactor RtpVideoStreamReceiver without RtpReceiver. > > Bug: webrtc:7135 > Change-Id: Iabf3330e579b892efc160683f9f90efbf6ff9a40 > Reviewed-on: https://webrtc-review.googlesource.com/92398 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24232} TBR=danilchap@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,sprang@webrtc.org Change-Id: I70a1dcb519c51937e35e04ac82b2ab495bec0a3d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7135 Reviewed-on: https://webrtc-review.googlesource.com/93260 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24235}
This commit is contained in:
Niels Moller
@ -1277,7 +1277,6 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
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return DELIVERY_OK;
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}
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} else if (media_type == MediaType::VIDEO) {
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parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
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if (video_receiver_controller_.OnRtpPacket(parsed_packet)) {
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received_bytes_per_second_counter_.Add(length);
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received_video_bytes_per_second_counter_.Add(length);
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@ -1328,7 +1327,6 @@ void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
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parsed_packet.IdentifyExtensions(it->second.extensions);
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// TODO(brandtr): Update here when we support protecting audio packets too.
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parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
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video_receiver_controller_.OnRtpPacket(parsed_packet);
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}
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@ -54,8 +54,11 @@ RtpDepacketizer* RtpDepacketizer::Create(VideoCodecType type) {
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return new RtpDepacketizerVp8();
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case kVideoCodecVP9:
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return new RtpDepacketizerVp9();
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default:
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case kVideoCodecGeneric:
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return new RtpDepacketizerGeneric();
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default:
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RTC_NOTREACHED();
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}
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return nullptr;
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}
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} // namespace webrtc
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@ -14,20 +14,17 @@
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#include <utility>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "media/base/mediaconstants.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_cvo.h"
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#include "modules/rtp_rtcp/include/rtp_receiver.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
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#include "modules/rtp_rtcp/source/rtp_format.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/video_coding/h264_sprop_parameter_sets.h"
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#include "modules/video_coding/h264_sps_pps_tracker.h"
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@ -100,6 +97,9 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
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process_thread_(process_thread),
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ntp_estimator_(clock_),
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rtp_header_extensions_(config_.rtp.extensions),
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rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
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this,
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&rtp_payload_registry_)),
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rtp_receive_statistics_(rtp_receive_statistics),
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ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc, this)),
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receiving_(false),
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@ -168,25 +168,26 @@ RtpVideoStreamReceiver::~RtpVideoStreamReceiver() {
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UpdateHistograms();
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}
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void RtpVideoStreamReceiver::AddReceiveCodec(
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bool RtpVideoStreamReceiver::AddReceiveCodec(
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const VideoCodec& video_codec,
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const std::map<std::string, std::string>& codec_params) {
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pt_codec_type_.emplace(video_codec.plType, video_codec.codecType);
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pt_codec_params_.emplace(video_codec.plType, codec_params);
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pt_codec_params_.insert(make_pair(video_codec.plType, codec_params));
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return rtp_payload_registry_.RegisterReceivePayload(video_codec) == 0;
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}
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absl::optional<Syncable::Info> RtpVideoStreamReceiver::GetSyncInfo() const {
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if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
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Syncable::Info info;
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if (!rtp_receiver_->GetLatestTimestamps(
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&info.latest_received_capture_timestamp,
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&info.latest_receive_time_ms)) {
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return absl::nullopt;
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}
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Syncable::Info info;
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if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs,
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&info.capture_time_ntp_frac, nullptr, nullptr,
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&info.capture_time_source_clock) != 0) {
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return absl::nullopt;
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}
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info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
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info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
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// Leaves info.current_delay_ms uninitialized.
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return info;
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@ -243,20 +244,12 @@ void RtpVideoStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
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RtpPacketReceived packet;
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if (!packet.Parse(rtp_packet, rtp_packet_length))
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return;
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if (packet.PayloadType() == config_.rtp.red_payload_type) {
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RTC_LOG(LS_WARNING) << "Discarding recovered packet with RED encapsulation";
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return;
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}
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packet.IdentifyExtensions(rtp_header_extensions_);
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packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
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// TODO(nisse): UlpfecReceiverImpl::ProcessReceivedFec passes both
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// original (decapsulated) media packets and recovered packets to
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// this callback. We need a way to distinguish, for setting
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// packet.recovered() correctly. Ideally, move RED decapsulation out
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// of the Ulpfec implementation.
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ReceivePacket(packet);
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RTPHeader header;
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packet.GetHeader(&header);
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ReceivePacket(rtp_packet, rtp_packet_length, header);
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}
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// This method handles both regular RTP packets and packets recovered
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@ -270,9 +263,6 @@ void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
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if (!packet.recovered()) {
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int64_t now_ms = clock_->TimeInMilliseconds();
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// TODO(nisse): Exclude out-of-order packets?
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last_received_rtp_timestamp_ = packet.Timestamp();
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last_received_rtp_system_time_ms_ = now_ms;
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// Periodically log the RTP header of incoming packets.
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if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
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@ -295,14 +285,18 @@ void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
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}
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}
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ReceivePacket(packet);
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// TODO(nisse): Delete use of GetHeader, but needs refactoring of
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// ReceivePacket and IncomingPacket methods below.
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RTPHeader header;
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packet.GetHeader(&header);
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header.payload_type_frequency = kVideoPayloadTypeFrequency;
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ReceivePacket(packet.data(), packet.size(), header);
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// Update receive statistics after ReceivePacket.
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// Receive statistics will be reset if the payload type changes (make sure
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// that the first packet is included in the stats).
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if (!packet.recovered()) {
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RTPHeader header;
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packet.GetHeader(&header);
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// TODO(nisse): We should pass a recovered flag to stats, to aid
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// fixing bug bugs.webrtc.org/6339.
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rtp_receive_statistics_->IncomingPacket(header, packet.size(),
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@ -394,55 +388,22 @@ void RtpVideoStreamReceiver::RemoveSecondarySink(
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secondary_sinks_.erase(it);
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}
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void RtpVideoStreamReceiver::ReceivePacket(const RtpPacketReceived& packet) {
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if (packet.PayloadType() == config_.rtp.red_payload_type) {
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RTPHeader header;
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packet.GetHeader(&header);
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ParseAndHandleEncapsulatingHeader(packet.data(), packet.size(), header);
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void RtpVideoStreamReceiver::ReceivePacket(const uint8_t* packet,
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size_t packet_length,
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const RTPHeader& header) {
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if (header.payloadType == config_.rtp.red_payload_type) {
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ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
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return;
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}
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const auto codec_type_it = pt_codec_type_.find(packet.PayloadType());
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if (codec_type_it == pt_codec_type_.end()) {
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return;
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const uint8_t* payload = packet + header.headerLength;
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assert(packet_length >= header.headerLength);
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size_t payload_length = packet_length - header.headerLength;
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const auto pl =
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rtp_payload_registry_.PayloadTypeToPayload(header.payloadType);
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if (pl) {
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rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
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pl->typeSpecific);
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}
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auto depacketizer =
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absl::WrapUnique(RtpDepacketizer::Create(codec_type_it->second));
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if (!depacketizer) {
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RTC_LOG(LS_ERROR) << "Failed to create depacketizer.";
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return;
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}
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RtpDepacketizer::ParsedPayload parsed_payload;
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if (!depacketizer->Parse(&parsed_payload, packet.payload().data(),
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packet.payload().size())) {
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RTC_LOG(LS_WARNING) << "Failed parsing payload.";
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return;
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}
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WebRtcRTPHeader webrtc_rtp_header = {};
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packet.GetHeader(&webrtc_rtp_header.header);
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webrtc_rtp_header.frameType = parsed_payload.frame_type;
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webrtc_rtp_header.video_header() = parsed_payload.video_header();
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webrtc_rtp_header.video_header().rotation = kVideoRotation_0;
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webrtc_rtp_header.video_header().content_type = VideoContentType::UNSPECIFIED;
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webrtc_rtp_header.video_header().video_timing.flags =
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VideoSendTiming::kInvalid;
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// Retrieve the video rotation information.
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packet.GetExtension<VideoOrientation>(
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&webrtc_rtp_header.video_header().rotation);
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packet.GetExtension<VideoContentTypeExtension>(
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&webrtc_rtp_header.video_header().content_type);
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packet.GetExtension<VideoTimingExtension>(
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&webrtc_rtp_header.video_header().video_timing);
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packet.GetExtension<PlayoutDelayLimits>(
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&webrtc_rtp_header.video_header().playout_delay);
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OnReceivedPayloadData(parsed_payload.payload, parsed_payload.payload_length,
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&webrtc_rtp_header);
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}
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void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
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@ -495,7 +456,7 @@ bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
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rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
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int64_t rtt = 0;
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rtp_rtcp_->RTT(config_.rtp.remote_ssrc, &rtt, nullptr, nullptr, nullptr);
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rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
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if (rtt == 0) {
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// Waiting for valid rtt.
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return true;
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@ -17,9 +17,6 @@
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/video_codecs/video_codec.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "call/syncable.h"
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#include "call/video_receive_stream.h"
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@ -27,6 +24,7 @@
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/video_coding/h264_sps_pps_tracker.h"
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@ -76,7 +74,7 @@ class RtpVideoStreamReceiver : public RtpData,
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video_coding::OnCompleteFrameCallback* complete_frame_callback);
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~RtpVideoStreamReceiver();
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void AddReceiveCodec(const VideoCodec& video_codec,
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bool AddReceiveCodec(const VideoCodec& video_codec,
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const std::map<std::string, std::string>& codec_params);
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void StartReceive();
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@ -140,10 +138,10 @@ class RtpVideoStreamReceiver : public RtpData,
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void RemoveSecondarySink(const RtpPacketSinkInterface* sink);
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private:
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// Entry point doing non-stats work for a received packet. Called
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// for the same packet both before and after RED decapsulation.
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void ReceivePacket(const RtpPacketReceived& packet);
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// Parses and handles RED headers.
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void ReceivePacket(const uint8_t* packet,
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size_t packet_length,
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const RTPHeader& header);
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// Parses and handles for instance RTX and RED headers.
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// This function assumes that it's being called from only one thread.
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void ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
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size_t packet_length,
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@ -162,8 +160,10 @@ class RtpVideoStreamReceiver : public RtpData,
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ProcessThread* const process_thread_;
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RemoteNtpTimeEstimator ntp_estimator_;
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RTPPayloadRegistry rtp_payload_registry_;
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RtpHeaderExtensionMap rtp_header_extensions_;
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const std::unique_ptr<RtpReceiver> rtp_receiver_;
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ReceiveStatistics* const rtp_receive_statistics_;
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std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
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@ -183,10 +183,6 @@ class RtpVideoStreamReceiver : public RtpData,
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std::map<int64_t, uint16_t> last_seq_num_for_pic_id_
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RTC_GUARDED_BY(last_seq_num_cs_);
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video_coding::H264SpsPpsTracker tracker_;
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absl::optional<uint32_t> last_received_rtp_timestamp_;
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absl::optional<int64_t> last_received_rtp_system_time_ms_;
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std::map<uint8_t, VideoCodecType> pt_codec_type_;
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// TODO(johan): Remove pt_codec_params_ once
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// https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
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// Maps a payload type to a map of out-of-band supplied codec parameters.
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@ -24,6 +24,7 @@
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#include "common_video/h264/profile_level_id.h"
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#include "common_video/include/incoming_video_stream.h"
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#include "common_video/libyuv/include/webrtc_libyuv.h"
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#include "modules/rtp_rtcp/include/rtp_receiver.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_coding/frame_object.h"
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@ -202,7 +203,8 @@ void VideoReceiveStream::Start() {
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video_receiver_.RegisterExternalDecoder(decoder.decoder,
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decoder.payload_type);
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VideoCodec codec = CreateDecoderVideoCodec(decoder);
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rtp_video_stream_receiver_.AddReceiveCodec(codec, decoder.codec_params);
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RTC_CHECK(rtp_video_stream_receiver_.AddReceiveCodec(codec,
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decoder.codec_params));
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RTC_CHECK_EQ(VCM_OK, video_receiver_.RegisterReceiveCodec(
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&codec, num_cpu_cores_, false));
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}
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