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}
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 66b4bb1..c427844 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 @@
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 3387f27..1c18a8b 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 @@
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 b9ce7ac..455c43c 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 @@
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 39f100a..d698b33 100644
--- a/audio/voip/test/BUILD.gn
+++ b/audio/voip/test/BUILD.gn
@@ -36,6 +36,7 @@
"../../../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 @@
"../../../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 @@
"../../../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 3391265..ebb1772 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 @@
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 bedb82e..91d114c 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 @@
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);
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index 40005ef..2689195 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -22,8 +22,8 @@
#include "call/rtp_stream_receiver_controller_interface.h"
#include "modules/rtp_rtcp/include/flexfec_receiver.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.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/location.h"
@@ -132,7 +132,7 @@
configuration.outgoing_transport = config.rtcp_send_transport;
configuration.rtt_stats = rtt_stats;
configuration.local_media_ssrc = config.local_ssrc;
- return RtpRtcp::Create(configuration);
+ return ModuleRtpRtcpImpl2::Create(configuration);
}
} // namespace
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index fd35876..e7dbb20 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -22,8 +22,8 @@
#include "api/video_codecs/video_codec.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/pacing/packet_router.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/include/video_codec_interface.h"
@@ -253,7 +253,7 @@
configuration.need_rtp_packet_infos = rtp_config.lntf.enabled;
- auto rtp_rtcp = RtpRtcp::Create(configuration);
+ auto rtp_rtcp = ModuleRtpRtcpImpl2::Create(configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index 2db523c..be36ec8 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -50,6 +50,7 @@
class TransportFeedback;
}
+// TODO(tommi): See if we can remove Module.
class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
public:
struct Configuration {
@@ -158,8 +159,15 @@
RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
};
- // Creates an RTP/RTCP module object using provided |configuration|.
- static std::unique_ptr<RtpRtcp> Create(const Configuration& configuration);
+ // DEPRECATED. Do not use. Currently instantiates a deprecated version of the
+ // RtpRtcp module.
+ static std::unique_ptr<RtpRtcp> RTC_DEPRECATED
+ Create(const Configuration& configuration) {
+ return DEPRECATED_Create(configuration);
+ }
+
+ static std::unique_ptr<RtpRtcp> DEPRECATED_Create(
+ const Configuration& configuration);
// **************************************************************************
// Receiver functions
diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 55e1e44..0bfd18e 100644
--- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -19,9 +19,9 @@
#include "call/rtp_stream_receiver_controller.h"
#include "call/rtx_receive_stream.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/rate_limiter.h"
#include "test/gtest.h"
@@ -134,7 +134,7 @@
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
configuration.local_media_ssrc = kTestSsrc;
configuration.rtx_send_ssrc = kTestRtxSsrc;
- rtp_rtcp_module_ = RtpRtcp::Create(configuration);
+ rtp_rtcp_module_ = ModuleRtpRtcpImpl2::Create(configuration);
FieldTrialBasedConfig field_trials;
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 0bd37eb..795ac29 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -48,6 +48,14 @@
&packet_history,
config.paced_sender ? config.paced_sender : &non_paced_sender) {}
+std::unique_ptr<RtpRtcp> RtpRtcp::DEPRECATED_Create(
+ const Configuration& configuration) {
+ RTC_DCHECK(configuration.clock);
+ RTC_LOG(LS_ERROR)
+ << "*********** USING WebRTC INTERNAL IMPLEMENTATION DETAILS ***********";
+ return std::make_unique<ModuleRtpRtcpImpl>(configuration);
+}
+
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: rtcp_sender_(configuration),
rtcp_receiver_(configuration, this),
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index c8f10ac..76335f7 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -48,11 +48,6 @@
&packet_history,
config.paced_sender ? config.paced_sender : &non_paced_sender) {}
-std::unique_ptr<RtpRtcp> RtpRtcp::Create(const Configuration& configuration) {
- RTC_DCHECK(configuration.clock);
- return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
-}
-
ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
: rtcp_sender_(configuration),
rtcp_receiver_(configuration, this),
@@ -86,6 +81,14 @@
RTC_DCHECK_RUN_ON(&construction_thread_checker_);
}
+// static
+std::unique_ptr<RtpRtcp> ModuleRtpRtcpImpl2::Create(
+ const Configuration& configuration) {
+ RTC_DCHECK(configuration.clock);
+ RTC_DCHECK(TaskQueueBase::Current());
+ return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
+}
+
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
int64_t ModuleRtpRtcpImpl2::TimeUntilNextProcess() {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index 67409c0..87a8107 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -24,13 +24,13 @@
#include "api/video/video_bitrate_allocation.h"
#include "modules/include/module_fec_types.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType
#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
#include "modules/rtp_rtcp/source/rtcp_receiver.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
#include "rtc_base/critical_section.h"
@@ -49,6 +49,14 @@
explicit ModuleRtpRtcpImpl2(const RtpRtcp::Configuration& configuration);
~ModuleRtpRtcpImpl2() override;
+ // This method is provided to easy with migrating away from the
+ // RtpRtcp::Create factory method. Since this is an internal implementation
+ // detail though, creating an instance of ModuleRtpRtcpImpl2 directly should
+ // be fine.
+ static std::unique_ptr<RtpRtcp> Create(const Configuration& configuration);
+
+ // TODO(tommi): Make implementation private?
+
// Returns the number of milliseconds until the module want a worker thread to
// call Process.
int64_t TimeUntilNextProcess() override;
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index 3e35f42..e406b53 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -18,6 +18,7 @@
#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_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -67,7 +68,7 @@
public:
RtpSenderAudioTest()
: fake_clock_(kStartTime),
- rtp_module_(RtpRtcp::Create([&] {
+ rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
RtpRtcp::Configuration config;
config.audio = true;
config.clock = &fake_clock_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 80481dc..2dbb2e7 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -24,7 +24,6 @@
#include "common_video/generic_frame_descriptor/generic_frame_info.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_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
@@ -33,6 +32,7 @@
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.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_impl2.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/rate_limiter.h"
@@ -169,7 +169,7 @@
: field_trials_(GetParam()),
fake_clock_(kStartTime),
retransmission_rate_limiter_(&fake_clock_, 1000),
- rtp_module_(RtpRtcp::Create([&] {
+ rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
@@ -920,7 +920,7 @@
RtpSenderVideoWithFrameTransformerTest()
: fake_clock_(kStartTime),
retransmission_rate_limiter_(&fake_clock_, 1000),
- rtp_module_(RtpRtcp::Create([&] {
+ rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport_;
diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index 6e8e11d..d0c2c27 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -16,7 +16,7 @@
#include "api/video/video_bitrate_allocation.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/task_utils/to_queued_task.h"
@@ -244,7 +244,7 @@
config.outgoing_transport = receive_transport_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
config.local_media_ssrc = (*receive_configs)[0].rtp.local_ssrc;
- rtp_rtcp_ = RtpRtcp::Create(config);
+ rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config);
rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
}
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index ad8b038..a4c102e 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -25,7 +25,6 @@
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
@@ -35,6 +34,7 @@
#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/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
#include "modules/utility/include/process_thread.h"
@@ -97,7 +97,7 @@
configuration.rtcp_cname_callback = rtcp_cname_callback;
configuration.local_media_ssrc = local_ssrc;
- std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
+ std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::DEPRECATED_Create(configuration);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
return rtp_rtcp;
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 2c7bd4b..54a25d2 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -25,7 +25,6 @@
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
@@ -35,6 +34,7 @@
#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/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
#include "modules/utility/include/process_thread.h"
@@ -97,7 +97,7 @@
configuration.rtcp_cname_callback = rtcp_cname_callback;
configuration.local_media_ssrc = local_ssrc;
- std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
+ std::unique_ptr<RtpRtcp> rtp_rtcp = ModuleRtpRtcpImpl2::Create(configuration);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
return rtp_rtcp;
diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc
index 57fba8f..22ca595 100644
--- a/video/rtp_video_stream_receiver2_unittest.cc
+++ b/video/rtp_video_stream_receiver2_unittest.cc
@@ -37,6 +37,7 @@
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_frame_transformer.h"
+#include "test/time_controller/simulated_task_queue.h"
using ::testing::_;
using ::testing::ElementsAre;
@@ -237,6 +238,9 @@
return config;
}
+ TokenTaskQueue task_queue_;
+ TokenTaskQueue::CurrentTaskQueueSetter task_queue_setter_{&task_queue_};
+
const webrtc::test::ScopedFieldTrials override_field_trials_;
VideoReceiveStream::Config config_;
MockNackSender mock_nack_sender_;
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 449e194..cb77f13 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -25,10 +25,10 @@
#include "call/simulated_network.h"
#include "call/video_send_stream.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "modules/video_coding/codecs/vp9/include/vp9.h"
@@ -1677,7 +1677,7 @@
config.clock = Clock::GetRealTimeClock();
config.outgoing_transport = feedback_transport_.get();
config.retransmission_rate_limiter = &retranmission_rate_limiter_;
- rtp_rtcp_ = RtpRtcp::Create(config);
+ rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
}
