Add application_data field(s) to RtpPacketToSend and PacketOptions.

Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions.

Bug: webrtc:8906
Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d
Reviewed-on: https://webrtc-review.googlesource.com/55600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22174}

api/BUILD.gn

@@ -201,6 +201,7 @@ rtc_source_set("audio_options_api") {
 rtc_source_set("transport_api") {
   visibility = [ "*" ]
   sources = [
+    "call/transport.cc",
     "call/transport.h",
   ]
 }

api/call/transport.cc

@@ -0,0 +1,19 @@
+/*
+ *  Copyright (c) 2018 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 "api/call/transport.h"
+
+namespace webrtc {
+
+PacketOptions::PacketOptions() = default;
+
+PacketOptions::~PacketOptions() = default;
+
+}  // namespace webrtc

api/call/transport.h

@@ -13,15 +13,22 @@
 
 #include <stddef.h>
 #include <stdint.h>
+#include <vector>
 
 namespace webrtc {
 
 // TODO(holmer): Look into unifying this with the PacketOptions in
 // asyncpacketsocket.h.
 struct PacketOptions {
+  PacketOptions();
+  ~PacketOptions();
+
   // A 16 bits positive id. Negative ids are invalid and should be interpreted
   // as packet_id not being set.
   int packet_id = -1;
+  // Additional data bound to the RTP packet for use in application code,
+  // outside of WebRTC.
+  std::vector<uint8_t> application_data;
 };
 
 class Transport {

modules/rtp_rtcp/BUILD.gn

@@ -77,6 +77,7 @@ rtc_source_set("rtp_rtcp_format") {
     "source/rtp_header_extensions.cc",
     "source/rtp_packet.cc",
     "source/rtp_packet_received.cc",
+    "source/rtp_packet_to_send.cc",
   ]
 
   deps = [

modules/rtp_rtcp/source/rtp_packet_to_send.cc

@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2018 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 "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions)
+    : RtpPacket(extensions) {}
+RtpPacketToSend::RtpPacketToSend(const RtpPacketToSend& packet) = default;
+RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions,
+                                 size_t capacity)
+    : RtpPacket(extensions, capacity) {}
+
+RtpPacketToSend& RtpPacketToSend::operator=(const RtpPacketToSend& packet) =
+    default;
+
+RtpPacketToSend::~RtpPacketToSend() = default;
+
+}  // namespace webrtc

modules/rtp_rtcp/source/rtp_packet_to_send.h

@@ -10,6 +10,9 @@
 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
 #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
 
+#include <vector>
+
+#include "api/array_view.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 
@@ -17,19 +20,29 @@ namespace webrtc {
 // Class to hold rtp packet with metadata for sender side.
 class RtpPacketToSend : public RtpPacket {
  public:
-  explicit RtpPacketToSend(const ExtensionManager* extensions)
-      : RtpPacket(extensions) {}
-  RtpPacketToSend(const RtpPacketToSend& packet) = default;
-  RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
-      : RtpPacket(extensions, capacity) {}
+  explicit RtpPacketToSend(const ExtensionManager* extensions);
+  RtpPacketToSend(const RtpPacketToSend& packet);
+  RtpPacketToSend(const ExtensionManager* extensions, size_t capacity);
 
-  RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default;
+  RtpPacketToSend& operator=(const RtpPacketToSend& packet);
+
+  ~RtpPacketToSend();
 
   // Time in local time base as close as it can to frame capture time.
   int64_t capture_time_ms() const { return capture_time_ms_; }
 
   void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
 
+  // Additional data bound to the RTP packet for use in application code,
+  // outside of WebRTC.
+  rtc::ArrayView<const uint8_t> application_data() const {
+    return application_data_;
+  }
+
+  void set_application_data(rtc::ArrayView<const uint8_t> data) {
+    application_data_.assign(data.begin(), data.end());
+  }
+
   void set_packetization_finish_time_ms(int64_t time) {
     SetExtension<VideoTimingExtension>(
         VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
@@ -56,6 +69,7 @@ class RtpPacketToSend : public RtpPacket {
 
  private:
   int64_t capture_time_ms_ = 0;
+  std::vector<uint8_t> application_data_;
 };
 
 }  // namespace webrtc

modules/rtp_rtcp/source/rtp_sender.cc

@@ -782,6 +782,8 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
     AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
                                  pacing_info);
   }
+  options.application_data.assign(packet_to_send->application_data().begin(),
+                                  packet_to_send->application_data().end());
 
   if (!is_retransmit && !send_over_rtx) {
     UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
@@ -914,6 +916,8 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
     AddPacketToTransportFeedback(options.packet_id, *packet.get(),
                                  PacedPacketInfo());
   }
+  options.application_data.assign(packet->application_data().begin(),
+                                  packet->application_data().end());
 
   UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
   UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
@@ -1211,6 +1215,9 @@ std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
   auto payload = packet.payload();
   memcpy(rtx_payload + kRtxHeaderSize, payload.data(), payload.size());
 
+  // Add original application data.
+  rtx_packet->set_application_data(packet.application_data());
+
   return rtx_packet;
 }