| Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| index b37a2f9c267da10d6c24070aaeb544bfb3729c55..9020f6a0fdf5832b77dcf495000f299a96e3df20 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
|
| @@ -15,6 +15,7 @@
|
|
|
| #include <memory>
|
| #include <vector>
|
| +#include <utility>
|
|
|
| #include "webrtc/base/checks.h"
|
| #include "webrtc/base/logging.h"
|
| @@ -25,11 +26,23 @@
|
| #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
|
| +#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
|
| +#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
|
|
|
| namespace webrtc {
|
|
|
| namespace {
|
| constexpr size_t kRedForFecHeaderLength = 1;
|
| +
|
| +void BuildRedPayload(const RtpPacketToSend& media_packet,
|
| + RtpPacketToSend* red_packet) {
|
| + uint8_t* red_payload = red_packet->AllocatePayload(
|
| + kRedForFecHeaderLength + media_packet.payload_size());
|
| + RTC_DCHECK(red_payload);
|
| + red_payload[0] = media_packet.PayloadType();
|
| + memcpy(&red_payload[kRedForFecHeaderLength], media_packet.payload(),
|
| + media_packet.payload_size());
|
| +}
|
| } // namespace
|
|
|
| RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSender* rtp_sender)
|
| @@ -72,66 +85,64 @@ RtpUtility::Payload* RTPSenderVideo::CreateVideoPayload(
|
| return payload;
|
| }
|
|
|
| -void RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer,
|
| - size_t payload_length,
|
| - size_t rtp_header_length,
|
| - uint16_t seq_num,
|
| - uint32_t rtp_timestamp,
|
| - int64_t capture_time_ms,
|
| +void RTPSenderVideo::SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
|
| StorageType storage) {
|
| - if (!rtp_sender_->SendToNetwork(data_buffer, payload_length,
|
| - rtp_header_length, capture_time_ms, storage,
|
| + // Remember some values about the packet before sending it away.
|
| + size_t packet_size = packet->size();
|
| + uint16_t seq_num = packet->SequenceNumber();
|
| + uint32_t rtp_timestamp = packet->Timestamp();
|
| + if (!rtp_sender_->SendToNetwork(std::move(packet), storage,
|
| RtpPacketSender::kLowPriority)) {
|
| LOG(LS_WARNING) << "Failed to send video packet " << seq_num;
|
| return;
|
| }
|
| rtc::CritScope cs(&stats_crit_);
|
| - video_bitrate_.Update(payload_length + rtp_header_length,
|
| - clock_->TimeInMilliseconds());
|
| + video_bitrate_.Update(packet_size, clock_->TimeInMilliseconds());
|
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
| "Video::PacketNormal", "timestamp", rtp_timestamp,
|
| "seqnum", seq_num);
|
| }
|
|
|
| -void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
|
| - size_t payload_length,
|
| - size_t rtp_header_length,
|
| - uint16_t media_seq_num,
|
| - uint32_t rtp_timestamp,
|
| - int64_t capture_time_ms,
|
| - StorageType media_packet_storage,
|
| - bool protect) {
|
| - std::unique_ptr<RedPacket> red_packet;
|
| +void RTPSenderVideo::SendVideoPacketAsRed(
|
| + std::unique_ptr<RtpPacketToSend> media_packet,
|
| + StorageType media_packet_storage,
|
| + bool protect) {
|
| + uint32_t rtp_timestamp = media_packet->Timestamp();
|
| + uint16_t media_seq_num = media_packet->SequenceNumber();
|
| +
|
| + std::unique_ptr<RtpPacketToSend> red_packet(
|
| + new RtpPacketToSend(*media_packet));
|
| + BuildRedPayload(*media_packet, red_packet.get());
|
| +
|
| std::vector<std::unique_ptr<RedPacket>> fec_packets;
|
| StorageType fec_storage = kDontRetransmit;
|
| - uint16_t next_fec_sequence_number = 0;
|
| {
|
| // Only protect while creating RED and FEC packets, not when sending.
|
| rtc::CritScope cs(&crit_);
|
| - red_packet = ProducerFec::BuildRedPacket(
|
| - data_buffer, payload_length, rtp_header_length, red_payload_type_);
|
| + red_packet->SetPayloadType(red_payload_type_);
|
| if (protect) {
|
| - producer_fec_.AddRtpPacketAndGenerateFec(data_buffer, payload_length,
|
| - rtp_header_length);
|
| + producer_fec_.AddRtpPacketAndGenerateFec(media_packet->data(),
|
| + media_packet->payload_size(),
|
| + media_packet->headers_size());
|
| }
|
| uint16_t num_fec_packets = producer_fec_.NumAvailableFecPackets();
|
| if (num_fec_packets > 0) {
|
| - next_fec_sequence_number =
|
| + uint16_t first_fec_sequence_number =
|
| rtp_sender_->AllocateSequenceNumber(num_fec_packets);
|
| fec_packets = producer_fec_.GetFecPacketsAsRed(
|
| - red_payload_type_, fec_payload_type_, next_fec_sequence_number,
|
| - rtp_header_length);
|
| + red_payload_type_, fec_payload_type_, first_fec_sequence_number,
|
| + media_packet->headers_size());
|
| RTC_DCHECK_EQ(num_fec_packets, fec_packets.size());
|
| if (retransmission_settings_ & kRetransmitFECPackets)
|
| fec_storage = kAllowRetransmission;
|
| }
|
| }
|
| - if (rtp_sender_->SendToNetwork(
|
| - red_packet->data(), red_packet->length() - rtp_header_length,
|
| - rtp_header_length, capture_time_ms, media_packet_storage,
|
| - RtpPacketSender::kLowPriority)) {
|
| + // Send |red_packet| instead of |packet| for allocated sequence number.
|
| + size_t red_packet_size = red_packet->size();
|
| + if (rtp_sender_->SendToNetwork(std::move(red_packet), media_packet_storage,
|
| + RtpPacketSender::kLowPriority)) {
|
| rtc::CritScope cs(&stats_crit_);
|
| - video_bitrate_.Update(red_packet->length(), clock_->TimeInMilliseconds());
|
| + video_bitrate_.Update(red_packet_size, clock_->TimeInMilliseconds());
|
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
| "Video::PacketRed", "timestamp", rtp_timestamp,
|
| "seqnum", media_seq_num);
|
| @@ -139,20 +150,23 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
|
| LOG(LS_WARNING) << "Failed to send RED packet " << media_seq_num;
|
| }
|
| for (const auto& fec_packet : fec_packets) {
|
| - if (rtp_sender_->SendToNetwork(
|
| - fec_packet->data(), fec_packet->length() - rtp_header_length,
|
| - rtp_header_length, capture_time_ms, fec_storage,
|
| - RtpPacketSender::kLowPriority)) {
|
| + // TODO(danilchap): Make producer_fec_ generate RtpPacketToSend to avoid
|
| + // reparsing them.
|
| + std::unique_ptr<RtpPacketToSend> rtp_packet(
|
| + new RtpPacketToSend(*media_packet));
|
| + RTC_CHECK(rtp_packet->Parse(fec_packet->data(), fec_packet->length()));
|
| + rtp_packet->set_capture_time_ms(media_packet->capture_time_ms());
|
| + uint16_t fec_sequence_number = rtp_packet->SequenceNumber();
|
| + if (rtp_sender_->SendToNetwork(std::move(rtp_packet), fec_storage,
|
| + RtpPacketSender::kLowPriority)) {
|
| rtc::CritScope cs(&stats_crit_);
|
| fec_bitrate_.Update(fec_packet->length(), clock_->TimeInMilliseconds());
|
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
| "Video::PacketFec", "timestamp", rtp_timestamp,
|
| - "seqnum", next_fec_sequence_number);
|
| + "seqnum", fec_sequence_number);
|
| } else {
|
| - LOG(LS_WARNING) << "Failed to send FEC packet "
|
| - << next_fec_sequence_number;
|
| + LOG(LS_WARNING) << "Failed to send FEC packet " << fec_sequence_number;
|
| }
|
| - ++next_fec_sequence_number;
|
| }
|
| }
|
|
|
| @@ -217,8 +231,39 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
|
| if (payload_size == 0)
|
| return false;
|
|
|
| + // Create header that will be reused in all packets.
|
| + std::unique_ptr<RtpPacketToSend> rtp_header = rtp_sender_->AllocatePacket();
|
| + rtp_header->SetPayloadType(payload_type);
|
| + rtp_header->SetTimestamp(rtp_timestamp);
|
| + rtp_header->set_capture_time_ms(capture_time_ms);
|
| + // According to
|
| + // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
|
| + // ts_126114v120700p.pdf Section 7.4.5:
|
| + // The MTSI client shall add the payload bytes as defined in this clause
|
| + // onto the last RTP packet in each group of packets which make up a key
|
| + // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
|
| + // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
|
| + // packet in each group of packets which make up another type of frame
|
| + // (e.g. a P-Frame) only if the current value is different from the previous
|
| + // value sent.
|
| + // Here we are adding it to every packet of every frame at this point.
|
| + if (video_header && video_header->rotation != kVideoRotation_0) {
|
| + // TODO(danilchap): Remove next call together with concept
|
| + // of inactive extension. Now it helps to calulate total maximum size
|
| + // or rtp header extensions that is used in FECPacketOverhead() function.
|
| + rtp_sender_->ActivateCVORtpHeaderExtension();
|
| + rtp_header->SetExtension<VideoOrientation>(video_header->rotation);
|
| + }
|
| +
|
| + size_t packet_capacity = rtp_sender_->MaxPayloadLength() -
|
| + FecPacketOverhead() -
|
| + (rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0);
|
| + RTC_DCHECK_LE(packet_capacity, rtp_header->capacity());
|
| + RTC_DCHECK_GT(packet_capacity, rtp_header->headers_size());
|
| + size_t max_data_payload_length = packet_capacity - rtp_header->headers_size();
|
| +
|
| std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
|
| - video_type, rtp_sender_->MaxDataPayloadLength(),
|
| + video_type, max_data_payload_length,
|
| video_header ? &(video_header->codecHeader) : nullptr, frame_type));
|
|
|
| StorageType storage;
|
| @@ -237,78 +282,35 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
|
| red_payload_type = red_payload_type_;
|
| }
|
|
|
| - // Register CVO rtp header extension at the first time when we receive a frame
|
| - // with pending rotation.
|
| - bool video_rotation_active = false;
|
| - if (video_header && video_header->rotation != kVideoRotation_0) {
|
| - video_rotation_active = rtp_sender_->ActivateCVORtpHeaderExtension();
|
| - }
|
| -
|
| - int rtp_header_length = rtp_sender_->RtpHeaderLength();
|
| - size_t payload_bytes_to_send = payload_size;
|
| - const uint8_t* data = payload_data;
|
| -
|
| // TODO(changbin): we currently don't support to configure the codec to
|
| // output multiple partitions for VP8. Should remove below check after the
|
| // issue is fixed.
|
| const RTPFragmentationHeader* frag =
|
| (video_type == kRtpVideoVp8) ? NULL : fragmentation;
|
|
|
| - packetizer->SetPayloadData(data, payload_bytes_to_send, frag);
|
| + packetizer->SetPayloadData(payload_data, payload_size, frag);
|
|
|
| bool first = true;
|
| bool last = false;
|
| while (!last) {
|
| - uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
|
| - size_t payload_bytes_in_packet = 0;
|
| + std::unique_ptr<RtpPacketToSend> packet(new RtpPacketToSend(*rtp_header));
|
| + uint8_t* payload = packet->AllocatePayload(max_data_payload_length);
|
| + RTC_DCHECK(payload);
|
|
|
| - if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
|
| - &payload_bytes_in_packet, &last)) {
|
| + size_t payload_bytes_in_packet = 0;
|
| + if (!packetizer->NextPacket(payload, &payload_bytes_in_packet, &last))
|
| return false;
|
| - }
|
|
|
| - // Write RTP header.
|
| - int32_t header_length = rtp_sender_->BuildRtpHeader(
|
| - dataBuffer, payload_type, last, rtp_timestamp, capture_time_ms);
|
| - if (header_length <= 0)
|
| + packet->SetPayloadSize(payload_bytes_in_packet);
|
| + packet->SetMarker(last);
|
| + if (!rtp_sender_->AssignSequenceNumber(packet.get()))
|
| return false;
|
|
|
| - // According to
|
| - // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
|
| - // ts_126114v120700p.pdf Section 7.4.5:
|
| - // The MTSI client shall add the payload bytes as defined in this clause
|
| - // onto the last RTP packet in each group of packets which make up a key
|
| - // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
|
| - // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
|
| - // packet in each group of packets which make up another type of frame
|
| - // (e.g. a P-Frame) only if the current value is different from the previous
|
| - // value sent.
|
| - // Here we are adding it to every packet of every frame at this point.
|
| - if (!video_header) {
|
| - RTC_DCHECK(!rtp_sender_->IsRtpHeaderExtensionRegistered(
|
| - kRtpExtensionVideoRotation));
|
| - } else if (video_rotation_active) {
|
| - // Checking whether CVO header extension is registered will require taking
|
| - // a lock. It'll be a no-op if it's not registered.
|
| - // TODO(guoweis): For now, all packets sent will carry the CVO such that
|
| - // the RTP header length is consistent, although the receiver side will
|
| - // only exam the packets with marker bit set.
|
| - size_t packetSize = payload_size + rtp_header_length;
|
| - RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
|
| - RTPHeader rtp_header;
|
| - rtp_parser.Parse(&rtp_header);
|
| - rtp_sender_->UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
|
| - video_header->rotation);
|
| - }
|
| if (red_payload_type != 0) {
|
| - SendVideoPacketAsRed(dataBuffer, payload_bytes_in_packet,
|
| - rtp_header_length, rtp_sender_->SequenceNumber(),
|
| - rtp_timestamp, capture_time_ms, storage,
|
| + SendVideoPacketAsRed(std::move(packet), storage,
|
| packetizer->GetProtectionType() == kProtectedPacket);
|
| } else {
|
| - SendVideoPacket(dataBuffer, payload_bytes_in_packet, rtp_header_length,
|
| - rtp_sender_->SequenceNumber(), rtp_timestamp,
|
| - capture_time_ms, storage);
|
| + SendVideoPacket(std::move(packet), storage);
|
| }
|
|
|
| if (first_frame) {
|
|
|