Add FlexfecSender.

webrtc/modules/rtp_rtcp/source/flexfec_sender_impl.cc

+/*
+ *  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 "webrtc/modules/rtp_rtcp/source/flexfec_sender_impl.h"
+
+#include <utility>
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+
+namespace webrtc {
+
+namespace {
+
+// Let first sequence number be in the first half of the interval.
+constexpr uint16_t kMaxInitRtpSeqNumber = 0x7fff;
+
+// See breakdown in flexfec_header_reader_writer.cc.
+constexpr size_t kFlexfecMaxHeaderSize = 32;
+
+// Since we will mainly use FlexFEC to protect video streams, we use a 90 kHz
+// clock for the RTP timestamps. (This is according to the RFC, which states
+// that it is RECOMMENDED to use the same clock frequency for FlexFEC as for
+// the protected media stream.)
+// The constant converts from clock millisecond timestamps to the 90 kHz
+// RTP timestamp.
+const int kMsToRtpTimestamp = kVideoPayloadTypeFrequency / 1000;
+
+// How often to log the generated FEC packets to the text log.
+constexpr int64_t kPacketLogIntervalMs = 10000;
+
+}  // namespace
+
+std::unique_ptr<FlexfecSender> FlexfecSender::Create(
+    int flexfec_payload_type,
+    uint32_t flexfec_ssrc,
+    uint32_t protected_media_ssrc,
+    const std::vector<RtpExtension>& rtp_header_extensions,
+    Clock* clock) {
+  // This object should not have been instantiated if FlexFEC is disabled.
+  RTC_DCHECK_GE(flexfec_payload_type, 0);
+  RTC_DCHECK_LE(flexfec_payload_type, 127);
+  return std::unique_ptr<FlexfecSender>(new FlexfecSenderImpl(
+      flexfec_payload_type, flexfec_ssrc, protected_media_ssrc,
+      rtp_header_extensions, clock));
+}
+
+FlexfecSenderImpl::FlexfecSenderImpl(
+    int flexfec_payload_type,
+    uint32_t flexfec_ssrc,
+    uint32_t protected_media_ssrc,
+    const std::vector<RtpExtension>& rtp_header_extensions,
+    Clock* clock)
+    : clock_(clock),
+      random_(clock_->TimeInMicroseconds()),
+      last_generated_packet_ms_(-1),
+      flexfec_payload_type_(flexfec_payload_type),
+      // Initialize the timestamp offset and RTP sequence numbers and randomly.

[stefan-webrtc, 2016/11/02 13:02:42]

-and

[brandtr, 2016/11/02 14:04:37]

Done.

+      // (This is not intended to be cryptographically strong.)
+      timestamp_offset_(random_.Rand<uint32_t>()),
+      flexfec_ssrc_(flexfec_ssrc),
+      protected_media_ssrc_(protected_media_ssrc),
+      seq_num_(random_.Rand(1, kMaxInitRtpSeqNumber)),
+      ulpfec_generator_(ForwardErrorCorrection::CreateFlexfec()),
+      rtp_header_extension_map_() {
+  // It's OK to create this object on a different thread/task queue than
+  // the one used during main operation.
+  sequence_checker_.Detach();
+
+  // Register RTP header extensions for BWE.
+  for (const auto& extension : rtp_header_extensions) {
+    if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
+      rtp_header_extension_map_.Register(kRtpExtensionTransportSequenceNumber,
+                                         extension.id);
+    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
+      rtp_header_extension_map_.Register(kRtpExtensionAbsoluteSendTime,
+                                         extension.id);
+    } else if (extension.uri == RtpExtension::kTimestampOffsetUri) {
+      rtp_header_extension_map_.Register(kRtpExtensionTransmissionTimeOffset,
+                                         extension.id);
+    } else {
+      LOG(LS_WARNING) << "RTP header extension with id: " << extension.id
+                      << ", uri: " << extension.uri
+                      << ", is unsupported by FlexfecSender.";
+    }
+  }
+}
+
+FlexfecSenderImpl::~FlexfecSenderImpl() = default;
+
+// We are reusing the implementation from UlpfecGenerator for SetFecParameters,
+// AddRtpPacketAndGenerateFec, and FecAvailable.
+void FlexfecSenderImpl::SetFecParameters(const FecProtectionParams& params) {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
+  ulpfec_generator_.SetFecParameters(&params);
+}
+
+bool FlexfecSenderImpl::AddRtpPacketAndGenerateFec(
+    const RtpPacketToSend& packet) {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
+  // TODO(brandtr): Generalize this SSRC check when we support multistream
+  // protection.
+  RTC_DCHECK_EQ(packet.Ssrc(), protected_media_ssrc_);
+  return ulpfec_generator_.AddRtpPacketAndGenerateFec(
+             packet.data(), packet.payload_size(), packet.headers_size()) == 0;
+}
+
+bool FlexfecSenderImpl::FecAvailable() const {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
+  return ulpfec_generator_.FecAvailable();
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+FlexfecSenderImpl::GetFecPackets() {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets_to_send;
+  fec_packets_to_send.reserve(ulpfec_generator_.generated_fec_packets_.size());
+  for (const auto& fec_packet : ulpfec_generator_.generated_fec_packets_) {
+    std::unique_ptr<RtpPacketToSend> fec_packet_to_send(
+        new RtpPacketToSend(&rtp_header_extension_map_));
+
+    // RTP header.
+    fec_packet_to_send->SetMarker(false);
+    fec_packet_to_send->SetPayloadType(flexfec_payload_type_);
+    fec_packet_to_send->SetSequenceNumber(seq_num_++);
+    fec_packet_to_send->SetTimestamp(
+        timestamp_offset_ +
+        static_cast<uint32_t>(kMsToRtpTimestamp *
+                              clock_->TimeInMilliseconds()));
+    // Set "capture time" so that the TransmissionOffset header extension
+    // can be set by the RTPSender.
+    fec_packet_to_send->set_capture_time_ms(clock_->TimeInMilliseconds());
+    fec_packet_to_send->SetSsrc(flexfec_ssrc_);
+    // Reserve extensions, if registered. These will be set by the RTPSender.
+    fec_packet_to_send->ReserveExtension<AbsoluteSendTime>();
+    fec_packet_to_send->ReserveExtension<TransmissionOffset>();
+    fec_packet_to_send->ReserveExtension<TransportSequenceNumber>();
+
+    // RTP payload.
+    uint8_t* payload = fec_packet_to_send->AllocatePayload(fec_packet->length);
+    memcpy(payload, fec_packet->data, fec_packet->length);
+
+    fec_packets_to_send.push_back(std::move(fec_packet_to_send));
+  }
+  ulpfec_generator_.ResetState();
+
+  // Periodically log generated FEC packets.
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (!fec_packets_to_send.empty() &&
+      now_ms - last_generated_packet_ms_ > kPacketLogIntervalMs) {
+    LOG(LS_INFO) << "Generated " << fec_packets_to_send.size()
+                 << " FlexFEC packets with payload type: "
+                 << flexfec_payload_type_ << " and SSRC: " << flexfec_ssrc_
+                 << ".";
+    last_generated_packet_ms_ = now_ms;
+  }

[stefan-webrtc, 2016/11/02 13:02:42]

Will this be valuable, or mostly confusing when we don't do the same thing for
audio/video?

[brandtr, 2016/11/02 14:04:37]

It's valueable for now, when I'm still wiring things up, because then it's easy
to see whenever the FlexfecSender is active. Long-term there's probably no need
for it, and as you say, it may be confusing the the media streams do not output
logs like this. I'll add a TODO to remove it, when things are properly wired up.

==

Second thought; what is our opinion about logging things like this to the
VERBOSE level?

[stefan-webrtc, 2016/11/03 13:42:40]

I don't have an opinion. If it's used for things like this in other places it's
fine. I just want to keep the number of lines of code down, and if it's not very
useful it should probably not be there. :) 

Keep for now if you use it.

+
+  return fec_packets_to_send;
+}
+
+size_t FlexfecSenderImpl::MaxPacketOverhead() const {
+  return kFlexfecMaxHeaderSize;
+}
+
+}  // namespace webrtc