Move rtp_packet_sender.h to api/

Old copy of the header and some previous usage is kept around
for compatibility with downstream projects for now.

Bug: chromium:345101934
Change-Id: Icbe42fb8450d3a4115799438d209da4eda127bab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357441
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42681}

api/BUILD.gn

@@ -558,6 +558,11 @@ rtc_source_set("audio_quality_analyzer_api") {
   ]
 }
 
+rtc_library("rtp_packet_sender") {
+  visibility = [ "*" ]
+  sources = [ "rtp_packet_sender.h" ]
+}
+
 rtc_source_set("stats_observer_interface") {
   visibility = [ "*" ]
   testonly = true

api/rtp_packet_sender.h

@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2019 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.
+ */
+
+#ifndef API_RTP_PACKET_SENDER_H_
+#define API_RTP_PACKET_SENDER_H_
+
+#include <memory>
+#include <vector>
+
+namespace webrtc {
+
+class RtpPacketToSend;
+
+class RtpPacketSender {
+ public:
+  virtual ~RtpPacketSender() = default;
+
+  // Insert a set of packets into queue, for eventual transmission. Based on the
+  // type of packets, they will be prioritized and scheduled relative to other
+  // packets and the current target send rate.
+  virtual void EnqueuePackets(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets) = 0;
+
+  // Clear any pending packets with the given SSRC from the queue.
+  // TODO(crbug.com/1395081): Make pure virtual when downstream code has been
+  // updated.
+  virtual void RemovePacketsForSsrc(uint32_t ssrc) {}
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTP_PACKET_SENDER_H_

modules/rtp_rtcp/BUILD.gn

@@ -21,7 +21,8 @@ rtc_library("rtp_rtcp_format") {
     "include/rtcp_statistics.h",
     "include/rtp_cvo.h",
     "include/rtp_header_extension_map.h",
-    "include/rtp_packet_sender.h",
+    "include/rtp_packet_sender.h",  # Kept for compatibility with downstream
+                                    # projects
     "include/rtp_rtcp_defines.h",
     "source/byte_io.h",
     "source/rtcp_packet.h",
@@ -114,6 +115,7 @@ rtc_library("rtp_rtcp_format") {
     "../../api:function_view",
     "../../api:refcountedbase",
     "../../api:rtp_headers",
+    "../../api:rtp_packet_sender",  # For compatibility with downstream projects
     "../../api:rtp_parameters",
     "../../api:scoped_refptr",
     "../../api/audio_codecs:audio_codecs_api",
@@ -281,6 +283,7 @@ rtc_library("rtp_rtcp") {
     "../../api:function_view",
     "../../api:rtp_headers",
     "../../api:rtp_packet_info",
+    "../../api:rtp_packet_sender",
     "../../api:rtp_parameters",
     "../../api:scoped_refptr",
     "../../api:sequence_checker",
@@ -671,6 +674,7 @@ if (rtc_include_tests) {
       "../../api:mock_transformable_video_frame",
       "../../api:rtp_headers",
       "../../api:rtp_packet_info",
+      "../../api:rtp_packet_sender",
       "../../api:rtp_parameters",
       "../../api:scoped_refptr",
       "../../api:time_controller",

modules/rtp_rtcp/include/rtp_packet_sender.h

@@ -17,24 +17,7 @@
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 
-namespace webrtc {
+// Kept for compatibility with the newer location of the header
-
+#include "api/rtp_packet_sender.h"
-class RtpPacketSender {
- public:
-  virtual ~RtpPacketSender() = default;
-
-  // Insert a set of packets into queue, for eventual transmission. Based on the
-  // type of packets, they will be prioritized and scheduled relative to other
-  // packets and the current target send rate.
-  virtual void EnqueuePackets(
-      std::vector<std::unique_ptr<RtpPacketToSend>> packets) = 0;
-
-  // Clear any pending packets with the given SSRC from the queue.
-  // TODO(crbug.com/1395081): Make pure virtual when downstream code has been
-  // updated.
-  virtual void RemovePacketsForSsrc(uint32_t ssrc) {}
-};
-
-}  // namespace webrtc
 
 #endif  // MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_

modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -15,6 +15,7 @@
 
 #include "absl/strings/string_view.h"
 #include "api/rtc_event_log/rtc_event.h"
+#include "api/rtp_packet_sender.h"
 #include "api/units/frequency.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -23,7 +24,6 @@
 #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "modules/rtp_rtcp/include/rtp_cvo.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/packet_sequencer.h"
 #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"

test/peer_scenario/tests/bwe_ramp_up_test.cc

@@ -16,6 +16,7 @@
 #include "api/units/time_delta.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.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_util.h"
 #include "pc/media_session.h"
 #include "pc/test/mock_peer_connection_observers.h"

test/peer_scenario/tests/unsignaled_stream_test.cc

@@ -10,11 +10,9 @@
 
 #include "media/base/stream_params.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
-#include "pc/media_session.h"
 #include "pc/session_description.h"
-#include "test/field_trial.h"
-#include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/peer_scenario/peer_scenario.h"