Add option to attach custom object to an rtp packet

As an alternative to attaching custom array of bytes. Bug: b/178094662 Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264 Reviewed-by: Christoffer Rodbro <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Reviewed-by: Emil Lundmark <[email protected]> Commit-Queue: Danil Chapovalov <[email protected]> Cr-Commit-Position: refs/heads/master@{#33069}
basharalbashier · Jan 25, 2021 · 5312a8f · 5312a8f
1 parent ded6636
commit 5312a8f
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Showing 8 changed files with 50 additions and 5 deletions.
diff --git a/api/BUILD.gn b/api/BUILD.gn
@@ -575,6 +575,10 @@ rtc_library("transport_api") {
     "call/transport.cc",
     "call/transport.h",
   ]
+  deps = [
+    ":refcountedbase",
+    ":scoped_refptr",
+  ]
 }
 
 rtc_source_set("bitrate_allocation") {

diff --git a/api/call/transport.h b/api/call/transport.h
@@ -16,6 +16,9 @@
 
 #include <vector>
 
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+
 namespace webrtc {
 
 // TODO(holmer): Look into unifying this with the PacketOptions in
@@ -28,9 +31,11 @@ struct PacketOptions {
   // A 16 bits positive id. Negative ids are invalid and should be interpreted
   // as packet_id not being set.
   int packet_id = -1;
+  // Deprecated: use `additional_data` instead of `application_data`.
+  std::vector<uint8_t> application_data;
   // Additional data bound to the RTP packet for use in application code,
   // outside of WebRTC.
-  std::vector<uint8_t> application_data;
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data;
   // Whether this is a retransmission of an earlier packet.
   bool is_retransmit = false;
   bool included_in_feedback = false;

diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
@@ -103,8 +103,10 @@ rtc_library("rtp_rtcp_format") {
     "..:module_api_public",
     "../../api:array_view",
     "../../api:function_view",
+    "../../api:refcountedbase",
     "../../api:rtp_headers",
     "../../api:rtp_parameters",
+    "../../api:scoped_refptr",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/transport:network_control",
     "../../api/transport/rtp:dependency_descriptor",

diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
@@ -178,6 +178,7 @@ void DEPRECATED_RtpSenderEgress::SendPacket(
 
   options.application_data.assign(packet->application_data().begin(),
                                   packet->application_data().end());
+  options.additional_data = packet->additional_data();
 
   if (packet->packet_type() != RtpPacketMediaType::kPadding &&
       packet->packet_type() != RtpPacketMediaType::kRetransmission) {

diff --git a/modules/rtp_rtcp/source/rtp_packet_received.h b/modules/rtp_rtcp/source/rtp_packet_received.h
@@ -12,14 +12,20 @@
 
 #include <stdint.h>
 
+#include <utility>
 #include <vector>
 
 #include "api/array_view.h"
+#include "api/ref_counted_base.h"
 #include "api/rtp_headers.h"
+#include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 // Class to hold rtp packet with metadata for receiver side.
+// The metadata is not parsed from the rtp packet, but may be derived from the
+// data that is parsed from the rtp packet.
 class RtpPacketReceived : public RtpPacket {
  public:
   RtpPacketReceived();
@@ -50,19 +56,29 @@ class RtpPacketReceived : public RtpPacket {
     payload_type_frequency_ = value;
   }
 
-  // Additional data bound to the RTP packet for use in application code,
-  // outside of WebRTC.
+  // An application can attach arbitrary data to an RTP packet using
+  // `application_data` or `additional_data`.
+  // The additional data does not affect WebRTC processing.
+  RTC_DEPRECATED
   rtc::ArrayView<const uint8_t> application_data() const {
     return application_data_;
   }
+  RTC_DEPRECATED
   void set_application_data(rtc::ArrayView<const uint8_t> data) {
     application_data_.assign(data.begin(), data.end());
   }
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const {
+    return additional_data_;
+  }
+  void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) {
+    additional_data_ = std::move(data);
+  }
 
  private:
   int64_t arrival_time_ms_ = 0;
   int payload_type_frequency_ = 0;
   bool recovered_ = false;
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
   std::vector<uint8_t> application_data_;
 };
 

diff --git a/modules/rtp_rtcp/source/rtp_packet_to_send.h b/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -13,17 +13,22 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <utility>
 #include <vector>
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
 #include "api/video/video_timing.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 
 namespace webrtc {
 // Class to hold rtp packet with metadata for sender side.
+// The metadata is not send over the wire, but packet sender may use it to
+// create rtp header extensions or other data that is sent over the wire.
 class RtpPacketToSend : public RtpPacket {
  public:
   // RtpPacketToSend::Type is deprecated. Use RtpPacketMediaType directly.
@@ -64,15 +69,22 @@ class RtpPacketToSend : public RtpPacket {
   }
   bool allow_retransmission() { return allow_retransmission_; }
 
-  // Additional data bound to the RTP packet for use in application code,
-  // outside of WebRTC.
+  // An application can attach arbitrary data to an RTP packet using
+  // `application_data` or `additional_data`.
+  // The additional data does not affect WebRTC processing.
   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());
   }
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const {
+    return additional_data_;
+  }
+  void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) {
+    additional_data_ = std::move(data);
+  }
 
   void set_packetization_finish_time_ms(int64_t time) {
     SetExtension<VideoTimingExtension>(
@@ -122,7 +134,10 @@ class RtpPacketToSend : public RtpPacket {
   absl::optional<RtpPacketMediaType> packet_type_;
   bool allow_retransmission_ = false;
   absl::optional<uint16_t> retransmitted_sequence_number_;
+  // TODO(danilchap): Remove applicaion_data_ when application switched to use
+  // additional_data instead.
   std::vector<uint8_t> application_data_;
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
   bool is_first_packet_of_frame_ = false;
   bool is_key_frame_ = false;
   bool fec_protect_packet_ = false;

diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -811,6 +811,7 @@ std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
 
   // Add original application data.
   rtx_packet->set_application_data(packet.application_data());
+  rtx_packet->set_additional_data(packet.additional_data());
 
   // Copy capture time so e.g. TransmissionOffset is correctly set.
   rtx_packet->set_capture_time_ms(packet.capture_time_ms());

diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -252,6 +252,7 @@ void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
 
   options.application_data.assign(packet->application_data().begin(),
                                   packet->application_data().end());
+  options.additional_data = packet->additional_data();
 
   if (packet->packet_type() != RtpPacketMediaType::kPadding &&
       packet->packet_type() != RtpPacketMediaType::kRetransmission) {