Deprecate the static RtpRtcp::Create() method.

The method is being used externally to create instances
of the deprecated internal implementation.

Instead, I'm moving how we instantiate the internal implementation into
the implementation itself and move towards keeping the interface
separate from a single implementation.

Change-Id: I743aa86dc4c812b545699c546c253c104719260e
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176404
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31420}

6 files changed, 13 insertions, 6 deletions

diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 66b4bb11f5..c4278444ab 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -33,11 +33,11 @@
 #include "modules/pacing/packet_router.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/absolute_capture_time_receiver.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/critical_section.h"
@@ -507,7 +507,7 @@ ChannelReceive::ChannelReceive(
   if (frame_transformer)
     InitFrameTransformerDelegate(std::move(frame_transformer));
 
-  _rtpRtcpModule = RtpRtcp::Create(configuration);
+  _rtpRtcpModule = ModuleRtpRtcpImpl2::Create(configuration);
   _rtpRtcpModule->SetSendingMediaStatus(false);
   _rtpRtcpModule->SetRemoteSSRC(remote_ssrc_);
 
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 3387f271ba..1c18a8b9b7 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -29,6 +29,7 @@
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_processing/rms_level.h"
 #include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
@@ -530,7 +531,7 @@ ChannelSend::ChannelSend(
 
   configuration.local_media_ssrc = ssrc;
 
-  _rtpRtcpModule = RtpRtcp::Create(configuration);
+  _rtpRtcpModule = ModuleRtpRtcpImpl2::Create(configuration);
   _rtpRtcpModule->SetSendingMediaStatus(false);
 
   rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(
diff --git a/audio/voip/audio_channel.cc b/audio/voip/audio_channel.cc
index b9ce7accd1..455c43c48b 100644
--- a/audio/voip/audio_channel.cc
+++ b/audio/voip/audio_channel.cc
@@ -16,6 +16,7 @@
 #include "api/audio_codecs/audio_format.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/critical_section.h"
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
@@ -51,7 +52,7 @@ AudioChannel::AudioChannel(
   rtp_config.outgoing_transport = transport;
   rtp_config.local_media_ssrc = local_ssrc;
 
-  rtp_rtcp_ = RtpRtcp::Create(rtp_config);
+  rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
 
   rtp_rtcp_->SetSendingMediaStatus(false);
   rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
diff --git a/audio/voip/test/BUILD.gn b/audio/voip/test/BUILD.gn
index 39f100a3aa..d698b3321d 100644
--- a/audio/voip/test/BUILD.gn
+++ b/audio/voip/test/BUILD.gn
@@ -36,6 +36,7 @@ if (rtc_include_tests) {
       "../../../api/task_queue:default_task_queue_factory",
       "../../../modules/audio_mixer:audio_mixer_impl",
       "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../../modules/utility",
       "../../../rtc_base:logging",
@@ -56,6 +57,7 @@ if (rtc_include_tests) {
       "../../../api/audio_codecs:builtin_audio_encoder_factory",
       "../../../api/task_queue:default_task_queue_factory",
       "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../rtc_base:logging",
       "../../../rtc_base:rtc_event",
       "../../../test:mock_transport",
@@ -72,6 +74,7 @@ if (rtc_include_tests) {
       "../../../api/audio_codecs:builtin_audio_encoder_factory",
       "../../../api/task_queue:default_task_queue_factory",
       "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../../rtc_base:logging",
       "../../../rtc_base:rtc_event",
diff --git a/audio/voip/test/audio_egress_unittest.cc b/audio/voip/test/audio_egress_unittest.cc
index 3391265880..ebb1772b30 100644
--- a/audio/voip/test/audio_egress_unittest.cc
+++ b/audio/voip/test/audio_egress_unittest.cc
@@ -14,6 +14,7 @@
 #include "api/task_queue/default_task_queue_factory.h"
 #include "modules/audio_mixer/sine_wave_generator.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/event.h"
 #include "rtc_base/logging.h"
 #include "test/gmock.h"
@@ -36,7 +37,7 @@ std::unique_ptr<RtpRtcp> CreateRtpStack(Clock* clock,
   rtp_config.rtcp_report_interval_ms = 5000;
   rtp_config.outgoing_transport = transport;
   rtp_config.local_media_ssrc = remote_ssrc;
-  auto rtp_rtcp = RtpRtcp::Create(rtp_config);
+  auto rtp_rtcp = ModuleRtpRtcpImpl2::Create(rtp_config);
   rtp_rtcp->SetSendingMediaStatus(false);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
   return rtp_rtcp;
diff --git a/audio/voip/test/audio_ingress_unittest.cc b/audio/voip/test/audio_ingress_unittest.cc
index bedb82e211..91d114c52d 100644
--- a/audio/voip/test/audio_ingress_unittest.cc
+++ b/audio/voip/test/audio_ingress_unittest.cc
@@ -15,6 +15,7 @@
 #include "api/task_queue/default_task_queue_factory.h"
 #include "audio/voip/audio_egress.h"
 #include "modules/audio_mixer/sine_wave_generator.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/event.h"
 #include "rtc_base/logging.h"
 #include "test/gmock.h"
@@ -45,7 +46,7 @@ class AudioIngressTest : public ::testing::Test {
     rtp_config.rtcp_report_interval_ms = 5000;
     rtp_config.outgoing_transport = &transport_;
     rtp_config.local_media_ssrc = 0xdeadc0de;
-    rtp_rtcp_ = RtpRtcp::Create(rtp_config);
+    rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
 
     rtp_rtcp_->SetSendingMediaStatus(false);
     rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);