Move pc/channel.h to only be used in .cc files

This is an implementation API, user classes should in principle
only use the channel_interface.h

Bug: webrtc:13931
Change-Id: I85c285217858dc087c90a50792e980f731f4439f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260185
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36674}

pc/BUILD.gn

@@ -59,7 +59,10 @@ rtc_library("proxy") {
 }
 
 rtc_source_set("channel") {
-  visibility = [ ":*" ]
+  visibility = [
+    ":*",
+    "../test/peer_scenario",
+  ]
   sources = [
     "channel.cc",
     "channel.h",
@@ -2617,6 +2620,7 @@ if (rtc_include_tests && !build_with_chromium) {
     ]
 
     deps = [
+      ":channel",
       ":channel_interface",
       ":jitter_buffer_delay",
       ":libjingle_peerconnection",

pc/channel_manager.cc

@@ -18,6 +18,7 @@
 #include "api/media_types.h"
 #include "api/sequence_checker.h"
 #include "media/base/media_constants.h"
+#include "pc/channel.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/location.h"
 #include "rtc_base/trace_event.h"

pc/channel_manager.h

@@ -26,7 +26,6 @@
 #include "media/base/media_channel.h"
 #include "media/base/media_config.h"
 #include "media/base/media_engine.h"
-#include "pc/channel.h"
 #include "pc/channel_interface.h"
 #include "pc/session_description.h"
 #include "rtc_base/system/file_wrapper.h"

pc/channel_manager_unittest.cc

@@ -17,6 +17,7 @@
 #include "media/engine/fake_webrtc_call.h"
 #include "p2p/base/fake_dtls_transport.h"
 #include "p2p/base/p2p_constants.h"
+#include "pc/channel.h"
 #include "pc/dtls_srtp_transport.h"
 #include "pc/rtp_transport_internal.h"
 #include "rtc_base/arraysize.h"

pc/jsep_transport_controller.h

@@ -46,7 +46,6 @@
 #include "p2p/base/port_allocator.h"
 #include "p2p/base/transport_description.h"
 #include "p2p/base/transport_info.h"
-#include "pc/channel.h"
 #include "pc/dtls_srtp_transport.h"
 #include "pc/dtls_transport.h"
 #include "pc/jsep_transport.h"

pc/remote_audio_source.h

@@ -21,7 +21,6 @@
 #include "api/media_stream_interface.h"
 #include "api/notifier.h"
 #include "media/base/media_channel.h"
-#include "pc/channel.h"
 #include "rtc_base/message_handler.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread.h"

pc/sdp_offer_answer.cc

@@ -39,6 +39,7 @@
 #include "p2p/base/transport_description.h"
 #include "p2p/base/transport_description_factory.h"
 #include "p2p/base/transport_info.h"
+#include "pc/channel.h"
 #include "pc/channel_interface.h"
 #include "pc/dtls_transport.h"
 #include "pc/media_stream.h"

pc/sdp_offer_answer.h

@@ -41,7 +41,6 @@
 #include "media/base/media_channel.h"
 #include "media/base/stream_params.h"
 #include "p2p/base/port_allocator.h"
-#include "pc/channel.h"
 #include "pc/channel_manager.h"
 #include "pc/connection_context.h"
 #include "pc/data_channel_controller.h"

pc/test/fake_peer_connection_for_stats.h

@@ -19,6 +19,7 @@
 #include <vector>
 
 #include "media/base/fake_media_engine.h"
+#include "pc/channel.h"
 #include "pc/stream_collection.h"
 #include "pc/test/fake_data_channel_provider.h"
 #include "pc/test/fake_peer_connection_base.h"

test/peer_scenario/BUILD.gn

@@ -43,6 +43,7 @@ if (rtc_include_tests) {
       "../../modules/audio_device:audio_device_impl",
       "../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../p2p:rtc_p2p",
+      "../../pc:channel",
       "../../pc:jsep_transport_controller",
       "../../pc:pc_test_utils",
       "../../pc:rtp_transport_internal",

test/peer_scenario/scenario_connection.cc

@@ -13,6 +13,7 @@
 #include "media/base/rtp_utils.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "p2p/client/basic_port_allocator.h"
+#include "pc/channel.h"
 #include "pc/jsep_transport_controller.h"
 #include "pc/rtp_transport_internal.h"
 #include "pc/session_description.h"