Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.

Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.

transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.

NOTRY=True

BUG=webrtc:5589, webrtc:5878, webrtc:6785

Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}

.gn

@@ -21,6 +21,7 @@ secondary_source = "//build/secondary/"
 # TODO(kjellander): Keep adding paths to this list as work in webrtc:5589 is done.
 check_targets = [
   "//webrtc/api:audio_mixer_api",
+  "//webrtc/api:transport_api",
   "//webrtc/api:rtc_stats_api",
   "//webrtc/modules/audio_coding:audio_decoder_factory_interface",
   "//webrtc/modules/audio_coding:audio_format",
@@ -37,6 +38,7 @@ check_targets = [
   "//webrtc/modules/audio_device/*",
   "//webrtc/modules/audio_mixer/*",
   "//webrtc/stats:rtc_stats",
+  "//webrtc/voice_engine",
   "//webrtc/voice_engine:level_indicator",
 ]
 

webrtc/BUILD.gn

@@ -231,6 +231,11 @@ if (!build_with_chromium) {
       "build/no_op_function.cc",
       "call.h",
       "config.h",
+
+      # TODO(aleloi): remove transport.h and the transport api
+      # dependency once clients have updated to the new transport
+      # location in webrtc/api/call/transport.h See
+      # http://bugs.webrtc.org/6785.
       "transport.h",
     ]
 
@@ -239,6 +244,7 @@ if (!build_with_chromium) {
     deps = [
       ":webrtc_common",
       "api",
+      "api:transport_api",
       "audio",
       "base",
       "call",

webrtc/api/BUILD.gn

@@ -37,6 +37,7 @@ rtc_source_set("call_api") {
   deps = [
     # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
     ":audio_mixer_api",
+    ":transport_api",
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../modules/audio_coding:audio_encoder_interface",
@@ -362,6 +363,11 @@ rtc_source_set("audio_mixer_api") {
   ]
 }
 
+rtc_source_set("transport_api") {
+  sources = [
+    "call/transport.h",
+  ]
+}
 if (rtc_include_tests) {
   config("peerconnection_unittests_config") {
     # The warnings below are enabled by default. Since GN orders compiler flags

webrtc/api/call/audio_receive_stream.h

@@ -16,12 +16,12 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
-#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/api/call/audio_send_stream.h

@@ -15,10 +15,10 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/api/call/transport.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2013 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 WEBRTC_API_CALL_TRANSPORT_H_
+#define WEBRTC_API_CALL_TRANSPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// TODO(holmer): Look into unifying this with the PacketOptions in
+// asyncpacketsocket.h.
+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;
+};
+
+class Transport {
+ public:
+  virtual bool SendRtp(const uint8_t* packet,
+                       size_t length,
+                       const PacketOptions& options) = 0;
+  virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
+
+ protected:
+  virtual ~Transport() {}
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_CALL_TRANSPORT_H_

webrtc/call/BUILD.gn

@@ -27,6 +27,7 @@ rtc_static_library("call") {
 
   deps = [
     "..:webrtc_common",
+    "../api:transport_api",
     "../audio",
     "../base:rtc_task_queue",
     "../logging:rtc_event_log_impl",

webrtc/media/BUILD.gn

@@ -226,6 +226,7 @@ rtc_static_library("rtc_media") {
     ":rtc_media_base",
     "..:webrtc_common",
     "../api:call_api",
+    "../api:transport_api",
     "../base:rtc_base_approved",
     "../call",
     "../modules/audio_mixer:audio_mixer_impl",

webrtc/media/engine/webrtcvideoengine2.h

@@ -17,6 +17,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/networkroute.h"
@@ -28,7 +29,6 @@
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/media/engine/webrtcvideodecoderfactory.h"
 #include "webrtc/media/engine/webrtcvideoencoderfactory.h"
-#include "webrtc/transport.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"

webrtc/modules/BUILD.gn

@@ -641,6 +641,7 @@ if (rtc_include_tests) {
     deps += [
       ":audio_network_adaptor_unittests",
       "..:webrtc_common",
+      "../api:transport_api",
       "../base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
       "../common_audio",
       "../common_video",

webrtc/modules/rtp_rtcp/BUILD.gn

@@ -174,6 +174,7 @@ rtc_static_library("rtp_rtcp") {
 
   deps = [
     "../..:webrtc_common",
+    "../../api:transport_api",
     "../../common_video",
     "../../system_wrappers",
     "../remote_bitrate_estimator",

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

@@ -14,6 +14,7 @@
 #include <memory>
 #include <set>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
@@ -23,7 +24,6 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/test/gtest.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

@@ -18,6 +18,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
@@ -31,7 +32,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/modules/rtp_rtcp/source/rtp_sender.h

@@ -16,6 +16,7 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/deprecation.h"
@@ -32,7 +33,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 

webrtc/modules/rtp_rtcp/test/testAPI/test_api.h

@@ -10,6 +10,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
 #define WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -18,7 +19,6 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/test/gtest.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 

webrtc/test/BUILD.gn

@@ -340,6 +340,7 @@ rtc_source_set("test_common") {
     ":rtp_test_utils",
     ":test_support",
     "..:webrtc_common",
+    "../api:transport_api",
     "../audio",
     "../base:rtc_base_approved",
     "../call",

webrtc/test/direct_transport.h

@@ -14,11 +14,11 @@
 
 #include <deque>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/test/fake_network_pipe.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 

webrtc/test/mock_transport.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_TEST_MOCK_TRANSPORT_H_
 #define WEBRTC_TEST_MOCK_TRANSPORT_H_
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/test/gmock.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 

webrtc/test/null_transport.h

@@ -10,7 +10,7 @@
 #ifndef WEBRTC_TEST_NULL_TRANSPORT_H_
 #define WEBRTC_TEST_NULL_TRANSPORT_H_
 
-#include "webrtc/transport.h"
+#include "webrtc/api/call/transport.h"
 
 namespace webrtc {
 

webrtc/transport.h

@@ -7,35 +7,12 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-
 #ifndef WEBRTC_TRANSPORT_H_
 #define WEBRTC_TRANSPORT_H_
+#include "webrtc/api/call/transport.h"
+// The contents header have moved to webrtc/api/call/transport.h. This
+// file is deprecated. See http://bugs.webrtc.org/6785.
 
-#include <stddef.h>
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// TODO(holmer): Look into unifying this with the PacketOptions in
-// asyncpacketsocket.h.
-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;
-};
-
-class Transport {
- public:
-  virtual bool SendRtp(const uint8_t* packet,
-                       size_t length,
-                       const PacketOptions& options) = 0;
-  virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
-
- protected:
-  virtual ~Transport() {}
-};
-
-}  // namespace webrtc
+// TODO(aleloi): delete this file when all dependencies are updated.
 
 #endif  // WEBRTC_TRANSPORT_H_

webrtc/video/BUILD.gn

@@ -56,6 +56,7 @@ rtc_static_library("video") {
 
   deps = [
     "..:webrtc_common",
+    "../api:transport_api",
     "../base:rtc_analytics",
     "../base:rtc_base_approved",
     "../base:rtc_task_queue",

webrtc/video/transport_adapter.h

@@ -10,9 +10,9 @@
 #ifndef WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
 #define WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/common_types.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 namespace internal {

webrtc/video_receive_stream.h

@@ -16,12 +16,12 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 

webrtc/video_send_stream.h

@@ -17,13 +17,13 @@
 #include <vector>
 #include <utility>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 

webrtc/voice_engine/BUILD.gn

@@ -89,6 +89,7 @@ rtc_static_library("voice_engine") {
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:call_api",
+    "../api:transport_api",
     "../base:rtc_base_approved",
     "../common_audio",
     "../logging:rtc_event_log_api",
@@ -177,6 +178,7 @@ if (rtc_include_tests) {
 
     deps = [
       "..:webrtc_common",
+      "../api:transport_api",
       "../system_wrappers",
       "//testing/gtest",
     ]

webrtc/voice_engine/include/voe_network.h

@@ -34,8 +34,8 @@
 #ifndef WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
 #define WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/common_types.h"
-#include "webrtc/transport.h"
 
 namespace webrtc {
 

webrtc/voice_engine/test/channel_transport/udp_transport.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_VOICE_ENGINE_TEST_CHANNEL_TRANSPORT_UDP_TRANSPORT_H_
 #define WEBRTC_VOICE_ENGINE_TEST_CHANNEL_TRANSPORT_UDP_TRANSPORT_H_
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/common_types.h"
-#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 /*