From f515ab8c3f24befe83a548b8056708acb24b6bb7 Mon Sep 17 00:00:00 2001 From: ossu Date: Wed, 7 Dec 2016 04:52:58 -0800 Subject: [PATCH] Moved call.h and most of api/call/* into call/ BUG=webrtc:6716 Review-Url: https://codereview.webrtc.org/2550273003 Cr-Commit-Position: refs/heads/master@{#15460} --- webrtc/BUILD.gn | 3 + webrtc/DEPS | 6 + webrtc/api/BUILD.gn | 4 - webrtc/api/DEPS | 12 ++ webrtc/api/call/DEPS | 4 - webrtc/api/mediacontroller.cc | 2 +- webrtc/api/peerconnection.cc | 2 +- webrtc/api/webrtcsession.cc | 2 +- webrtc/audio/BUILD.gn | 2 +- webrtc/audio/audio_receive_stream.h | 4 +- webrtc/audio/audio_send_stream.h | 4 +- webrtc/audio/audio_state.h | 2 +- webrtc/call.h | 158 +---------------- webrtc/call/BUILD.gn | 12 ++ webrtc/{api => }/call/audio_receive_stream.h | 6 +- webrtc/{api => }/call/audio_send_stream.cc | 2 +- webrtc/{api => }/call/audio_send_stream.h | 6 +- webrtc/{api => }/call/audio_state.h | 6 +- webrtc/call/bitrate_estimator_tests.cc | 2 +- webrtc/call/call.cc | 3 +- webrtc/call/call.h | 165 ++++++++++++++++++ webrtc/call/call_perf_tests.cc | 2 +- webrtc/call/call_unittest.cc | 6 +- webrtc/call/rampup_tests.h | 2 +- webrtc/logging/BUILD.gn | 3 +- webrtc/logging/rtc_event_log/DEPS | 1 + webrtc/logging/rtc_event_log/rtc_event_log.cc | 2 +- webrtc/logging/rtc_event_log/rtc_event_log.h | 4 +- .../rtc_event_log/rtc_event_log2rtp_dump.cc | 2 +- .../rtc_event_log/rtc_event_log_parser.cc | 2 +- .../rtc_event_log/rtc_event_log_unittest.cc | 2 +- .../rtc_event_log_unittest_helper.h | 2 +- webrtc/media/base/mediaengine.h | 2 +- webrtc/media/base/videoengine_unittest.h | 2 +- webrtc/media/engine/fakewebrtccall.h | 6 +- webrtc/media/engine/webrtcmediaengine.h | 2 +- webrtc/media/engine/webrtcvideoengine2.cc | 2 +- webrtc/media/engine/webrtcvideoengine2.h | 2 +- webrtc/media/engine/webrtcvoiceengine.h | 4 +- .../engine/webrtcvoiceengine_unittest.cc | 2 +- .../neteq/tools/rtc_event_log_source.cc | 2 +- .../rtp_rtcp/include/flexfec_receiver.h | 2 +- webrtc/modules/rtp_rtcp/source/rtcp_sender.cc | 2 +- webrtc/modules/rtp_rtcp/source/rtp_sender.cc | 2 +- webrtc/test/call_test.h | 2 +- webrtc/test/direct_transport.cc | 2 +- webrtc/test/fake_network_pipe.cc | 2 +- webrtc/test/fake_network_pipe_unittest.cc | 2 +- webrtc/test/layer_filtering_transport.h | 2 +- webrtc/tools/BUILD.gn | 1 + webrtc/tools/event_log_visualizer/analyzer.cc | 10 +- webrtc/video/end_to_end_tests.cc | 2 +- webrtc/video/replay.cc | 2 +- webrtc/video/video_quality_test.cc | 2 +- webrtc/video/video_send_stream_tests.cc | 2 +- webrtc/video/vie_encoder.h | 4 +- 56 files changed, 270 insertions(+), 228 deletions(-) delete mode 100644 webrtc/api/call/DEPS rename webrtc/{api => }/call/audio_receive_stream.h (97%) rename webrtc/{api => }/call/audio_send_stream.cc (98%) rename webrtc/{api => }/call/audio_send_stream.h (96%) rename webrtc/{api => }/call/audio_state.h (93%) create mode 100644 webrtc/call/call.h diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index accd2c91d7..58300a43a0 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -229,6 +229,9 @@ if (!build_with_chromium) { # TODO(kjellander): Remove this whenever possible. GN's static_library # target type requires at least one object to avoid errors linking. "build/no_op_function.cc", + + # TODO(ossu): Keep this here until donwstream projects have updated. + # http://bugs.webrtc.org/6716 "call.h", "config.h", diff --git a/webrtc/DEPS b/webrtc/DEPS index e46a7d331d..16589362e2 100644 --- a/webrtc/DEPS +++ b/webrtc/DEPS @@ -32,6 +32,12 @@ include_rules = [ # The below rules will be removed when webrtc:4243 is fixed. specific_include_rules = { + # The call/call.h exception is here only until the peerconnection + # implementation has been moved out of api/. See: + # http://bugs.webrtc.org/5883 + "call\.h": [ + "+webrtc/call/call.h" + ], "video_frame\.h": [ "+webrtc/common_video", ], diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn index 20cf3fa2b6..e838e7ed54 100644 --- a/webrtc/api/BUILD.gn +++ b/webrtc/api/BUILD.gn @@ -20,11 +20,7 @@ group("api") { rtc_source_set("call_api") { sources = [ - "call/audio_receive_stream.h", - "call/audio_send_stream.cc", - "call/audio_send_stream.h", "call/audio_sink.h", - "call/audio_state.h", "call/flexfec_receive_stream.h", ] diff --git a/webrtc/api/DEPS b/webrtc/api/DEPS index ee97620c21..75f756de05 100644 --- a/webrtc/api/DEPS +++ b/webrtc/api/DEPS @@ -18,10 +18,22 @@ specific_include_rules = { "+base/android", # Allowed only for Android tests. "+webrtc/voice_engine", ], + # The call/call.h exceptions are here only until the peerconnection + # implementation has been moved out of api/. See: + # http://bugs.webrtc.org/5883 + "mediacontroller\.cc": [ + "+webrtc/call/call.h" + ], + "peerconnection\.cc": [ + "+webrtc/call/call.h" + ], "peerconnection_jni\.cc": [ "+webrtc/voice_engine", ], "peerconnectionfactory\.cc": [ "+webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h", ], + "webrtcsession\.cc": [ + "+webrtc/call/call.h" + ], } diff --git a/webrtc/api/call/DEPS b/webrtc/api/call/DEPS deleted file mode 100644 index d1d4309681..0000000000 --- a/webrtc/api/call/DEPS +++ /dev/null @@ -1,4 +0,0 @@ -include_rules = [ - "+webrtc/modules/audio_coding/codecs", -] - diff --git a/webrtc/api/mediacontroller.cc b/webrtc/api/mediacontroller.cc index 319dd1fb15..c64ff018b8 100644 --- a/webrtc/api/mediacontroller.cc +++ b/webrtc/api/mediacontroller.cc @@ -15,7 +15,7 @@ #include "webrtc/base/bind.h" #include "webrtc/base/checks.h" #include "webrtc/base/constructormagic.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/pc/channelmanager.h" #include "webrtc/media/base/mediachannel.h" diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc index 46bdec594a..a47b2f2d8c 100644 --- a/webrtc/api/peerconnection.cc +++ b/webrtc/api/peerconnection.cc @@ -36,7 +36,7 @@ #include "webrtc/base/stringencode.h" #include "webrtc/base/stringutils.h" #include "webrtc/base/trace_event.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/media/sctp/sctpdataengine.h" #include "webrtc/pc/channelmanager.h" diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc index 0532040394..636184b852 100644 --- a/webrtc/api/webrtcsession.cc +++ b/webrtc/api/webrtcsession.cc @@ -30,7 +30,7 @@ #include "webrtc/base/logging.h" #include "webrtc/base/stringencode.h" #include "webrtc/base/stringutils.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/media/base/mediaconstants.h" #include "webrtc/media/base/videocapturer.h" #include "webrtc/p2p/base/portallocator.h" diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn index bde074d99a..bfc2372dd6 100644 --- a/webrtc/audio/BUILD.gn +++ b/webrtc/audio/BUILD.gn @@ -30,8 +30,8 @@ rtc_static_library("audio") { deps = [ "..:webrtc_common", "../api:audio_mixer_api", - "../api:call_api", "../base:rtc_base_approved", + "../call:call_interfaces", "../common_audio", "../modules/audio_device", "../modules/audio_processing", diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h index 3bba54ebb8..7dfc5d6bb8 100644 --- a/webrtc/audio/audio_receive_stream.h +++ b/webrtc/audio/audio_receive_stream.h @@ -14,11 +14,11 @@ #include #include "webrtc/api/audio/audio_mixer.h" -#include "webrtc/api/call/audio_receive_stream.h" -#include "webrtc/api/call/audio_state.h" #include "webrtc/audio/audio_state.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/thread_checker.h" +#include "webrtc/call/audio_receive_stream.h" +#include "webrtc/call/audio_state.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" namespace webrtc { diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h index 23a7485014..de1cf82d80 100644 --- a/webrtc/audio/audio_send_stream.h +++ b/webrtc/audio/audio_send_stream.h @@ -13,10 +13,10 @@ #include -#include "webrtc/api/call/audio_send_stream.h" -#include "webrtc/api/call/audio_state.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/thread_checker.h" +#include "webrtc/call/audio_send_stream.h" +#include "webrtc/call/audio_state.h" #include "webrtc/call/bitrate_allocator.h" namespace webrtc { diff --git a/webrtc/audio/audio_state.h b/webrtc/audio/audio_state.h index 307f5cec4b..29b5a9edbf 100644 --- a/webrtc/audio/audio_state.h +++ b/webrtc/audio/audio_state.h @@ -11,12 +11,12 @@ #ifndef WEBRTC_AUDIO_AUDIO_STATE_H_ #define WEBRTC_AUDIO_AUDIO_STATE_H_ -#include "webrtc/api/call/audio_state.h" #include "webrtc/audio/audio_transport_proxy.h" #include "webrtc/audio/scoped_voe_interface.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/thread_checker.h" +#include "webrtc/call/audio_state.h" #include "webrtc/voice_engine/include/voe_base.h" namespace webrtc { diff --git a/webrtc/call.h b/webrtc/call.h index 26f8c82bd9..afea9ddd72 100644 --- a/webrtc/call.h +++ b/webrtc/call.h @@ -7,159 +7,7 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_CALL_H_ -#define WEBRTC_CALL_H_ -#include -#include - -#include "webrtc/api/call/audio_receive_stream.h" -#include "webrtc/api/call/audio_send_stream.h" -#include "webrtc/api/call/audio_state.h" -#include "webrtc/api/call/flexfec_receive_stream.h" -#include "webrtc/base/networkroute.h" -#include "webrtc/base/platform_file.h" -#include "webrtc/base/socket.h" -#include "webrtc/common_types.h" -#include "webrtc/video_receive_stream.h" -#include "webrtc/video_send_stream.h" - -namespace webrtc { - -class AudioProcessing; -class RtcEventLog; - -const char* Version(); - -enum class MediaType { - ANY, - AUDIO, - VIDEO, - DATA -}; - -class PacketReceiver { - public: - enum DeliveryStatus { - DELIVERY_OK, - DELIVERY_UNKNOWN_SSRC, - DELIVERY_PACKET_ERROR, - }; - - virtual DeliveryStatus DeliverPacket(MediaType media_type, - const uint8_t* packet, - size_t length, - const PacketTime& packet_time) = 0; - - protected: - virtual ~PacketReceiver() {} -}; - -// A Call instance can contain several send and/or receive streams. All streams -// are assumed to have the same remote endpoint and will share bitrate estimates -// etc. -class Call { - public: - struct Config { - explicit Config(RtcEventLog* event_log) : event_log(event_log) { - RTC_DCHECK(event_log); - } - - static const int kDefaultStartBitrateBps; - - // Bitrate config used until valid bitrate estimates are calculated. Also - // used to cap total bitrate used. - struct BitrateConfig { - int min_bitrate_bps = 0; - int start_bitrate_bps = kDefaultStartBitrateBps; - int max_bitrate_bps = -1; - } bitrate_config; - - // AudioState which is possibly shared between multiple calls. - // TODO(solenberg): Change this to a shared_ptr once we can use C++11. - rtc::scoped_refptr audio_state; - - // Audio Processing Module to be used in this call. - // TODO(solenberg): Change this to a shared_ptr once we can use C++11. - AudioProcessing* audio_processing = nullptr; - - // RtcEventLog to use for this call. Required. - // Use webrtc::RtcEventLog::CreateNull() for a null implementation. - RtcEventLog* event_log = nullptr; - }; - - struct Stats { - std::string ToString(int64_t time_ms) const; - - int send_bandwidth_bps = 0; // Estimated available send bandwidth. - int max_padding_bitrate_bps = 0; // Cumulative configured max padding. - int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. - int64_t pacer_delay_ms = 0; - int64_t rtt_ms = -1; - }; - - static Call* Create(const Call::Config& config); - - virtual AudioSendStream* CreateAudioSendStream( - const AudioSendStream::Config& config) = 0; - virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; - - virtual AudioReceiveStream* CreateAudioReceiveStream( - const AudioReceiveStream::Config& config) = 0; - virtual void DestroyAudioReceiveStream( - AudioReceiveStream* receive_stream) = 0; - - virtual VideoSendStream* CreateVideoSendStream( - VideoSendStream::Config config, - VideoEncoderConfig encoder_config) = 0; - virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0; - - virtual VideoReceiveStream* CreateVideoReceiveStream( - VideoReceiveStream::Config configuration) = 0; - virtual void DestroyVideoReceiveStream( - VideoReceiveStream* receive_stream) = 0; - - virtual FlexfecReceiveStream* CreateFlexfecReceiveStream( - FlexfecReceiveStream::Config configuration) = 0; - virtual void DestroyFlexfecReceiveStream( - FlexfecReceiveStream* receive_stream) = 0; - - // All received RTP and RTCP packets for the call should be inserted to this - // PacketReceiver. The PacketReceiver pointer is valid as long as the - // Call instance exists. - virtual PacketReceiver* Receiver() = 0; - - // Returns the call statistics, such as estimated send and receive bandwidth, - // pacing delay, etc. - virtual Stats GetStats() const = 0; - - // TODO(pbos): Like BitrateConfig above this is currently per-stream instead - // of maximum for entire Call. This should be fixed along with the above. - // Specifying a start bitrate (>0) will currently reset the current bitrate - // estimate. This is due to how the 'x-google-start-bitrate' flag is currently - // implemented. - virtual void SetBitrateConfig( - const Config::BitrateConfig& bitrate_config) = 0; - - // TODO(skvlad): When the unbundled case with multiple streams for the same - // media type going over different networks is supported, track the state - // for each stream separately. Right now it's global per media type. - virtual void SignalChannelNetworkState(MediaType media, - NetworkState state) = 0; - - virtual void OnTransportOverheadChanged( - MediaType media, - int transport_overhead_per_packet) = 0; - - virtual void OnNetworkRouteChanged( - const std::string& transport_name, - const rtc::NetworkRoute& network_route) = 0; - - virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; - - virtual ~Call() {} -}; - -} // namespace webrtc - -#endif // WEBRTC_CALL_H_ +// This file is deprecated. It has been moved to the location below. Please +// update your includes! See: http://bugs.webrtc.org/6716 +#include "webrtc/call/call.h" diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn index 64829e9b9d..5572941f30 100644 --- a/webrtc/call/BUILD.gn +++ b/webrtc/call/BUILD.gn @@ -8,6 +8,16 @@ import("../build/webrtc.gni") +rtc_source_set("call_interfaces") { + sources = [ + "audio_receive_stream.h", + "audio_send_stream.cc", + "audio_send_stream.h", + "audio_state.h", + "call.h", + ] +} + rtc_static_library("call") { sources = [ "bitrate_allocator.cc", @@ -22,10 +32,12 @@ rtc_static_library("call") { } public_deps = [ + ":call_interfaces", "../api:call_api", ] deps = [ + ":call_interfaces", "..:webrtc_common", "../api:transport_api", "../audio", diff --git a/webrtc/api/call/audio_receive_stream.h b/webrtc/call/audio_receive_stream.h similarity index 97% rename from webrtc/api/call/audio_receive_stream.h rename to webrtc/call/audio_receive_stream.h index ed9ff3417a..1299ded429 100644 --- a/webrtc/api/call/audio_receive_stream.h +++ b/webrtc/call/audio_receive_stream.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ -#define WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ +#ifndef WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ +#define WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ #include #include @@ -139,4 +139,4 @@ class AudioReceiveStream { }; } // namespace webrtc -#endif // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ +#endif // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/call/audio_send_stream.cc similarity index 98% rename from webrtc/api/call/audio_send_stream.cc rename to webrtc/call/audio_send_stream.cc index b6190073c1..8b6dd9e416 100644 --- a/webrtc/api/call/audio_send_stream.cc +++ b/webrtc/call/audio_send_stream.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/api/call/audio_send_stream.h" +#include "webrtc/call/audio_send_stream.h" #include diff --git a/webrtc/api/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h similarity index 96% rename from webrtc/api/call/audio_send_stream.h rename to webrtc/call/audio_send_stream.h index 487ce98721..2063589064 100644 --- a/webrtc/api/call/audio_send_stream.h +++ b/webrtc/call/audio_send_stream.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ -#define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ +#ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_ +#define WEBRTC_CALL_AUDIO_SEND_STREAM_H_ #include #include @@ -142,4 +142,4 @@ class AudioSendStream { }; } // namespace webrtc -#endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ +#endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ diff --git a/webrtc/api/call/audio_state.h b/webrtc/call/audio_state.h similarity index 93% rename from webrtc/api/call/audio_state.h rename to webrtc/call/audio_state.h index b8dca3fb4e..2c26a1749b 100644 --- a/webrtc/api/call/audio_state.h +++ b/webrtc/call/audio_state.h @@ -7,8 +7,8 @@ * 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_AUDIO_STATE_H_ -#define WEBRTC_API_CALL_AUDIO_STATE_H_ +#ifndef WEBRTC_CALL_AUDIO_STATE_H_ +#define WEBRTC_CALL_AUDIO_STATE_H_ #include "webrtc/api/audio/audio_mixer.h" #include "webrtc/base/refcount.h" @@ -46,4 +46,4 @@ class AudioState : public rtc::RefCountInterface { }; } // namespace webrtc -#endif // WEBRTC_API_CALL_AUDIO_STATE_H_ +#endif // WEBRTC_CALL_AUDIO_STATE_H_ diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc index 95d42ef117..bb4960f658 100644 --- a/webrtc/call/bitrate_estimator_tests.cc +++ b/webrtc/call/bitrate_estimator_tests.cc @@ -16,7 +16,7 @@ #include "webrtc/base/event.h" #include "webrtc/base/logging.h" #include "webrtc/base/thread_annotations.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/call_test.h" diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 3d940b619a..9a32df267b 100644 --- a/webrtc/call/call.cc +++ b/webrtc/call/call.cc @@ -12,6 +12,7 @@ #include #include #include +#include #include #include @@ -27,8 +28,8 @@ #include "webrtc/base/thread_annotations.h" #include "webrtc/base/thread_checker.h" #include "webrtc/base/trace_event.h" -#include "webrtc/call.h" #include "webrtc/call/bitrate_allocator.h" +#include "webrtc/call/call.h" #include "webrtc/call/flexfec_receive_stream.h" #include "webrtc/config.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" diff --git a/webrtc/call/call.h b/webrtc/call/call.h new file mode 100644 index 0000000000..34f68334dd --- /dev/null +++ b/webrtc/call/call.h @@ -0,0 +1,165 @@ +/* + * 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_CALL_CALL_H_ +#define WEBRTC_CALL_CALL_H_ + +#include +#include + +#include "webrtc/api/call/flexfec_receive_stream.h" +#include "webrtc/base/networkroute.h" +#include "webrtc/base/platform_file.h" +#include "webrtc/base/socket.h" +#include "webrtc/call/audio_receive_stream.h" +#include "webrtc/call/audio_send_stream.h" +#include "webrtc/call/audio_state.h" +#include "webrtc/common_types.h" +#include "webrtc/video_receive_stream.h" +#include "webrtc/video_send_stream.h" + +namespace webrtc { + +class AudioProcessing; +class RtcEventLog; + +const char* Version(); + +enum class MediaType { + ANY, + AUDIO, + VIDEO, + DATA +}; + +class PacketReceiver { + public: + enum DeliveryStatus { + DELIVERY_OK, + DELIVERY_UNKNOWN_SSRC, + DELIVERY_PACKET_ERROR, + }; + + virtual DeliveryStatus DeliverPacket(MediaType media_type, + const uint8_t* packet, + size_t length, + const PacketTime& packet_time) = 0; + + protected: + virtual ~PacketReceiver() {} +}; + +// A Call instance can contain several send and/or receive streams. All streams +// are assumed to have the same remote endpoint and will share bitrate estimates +// etc. +class Call { + public: + struct Config { + explicit Config(RtcEventLog* event_log) : event_log(event_log) { + RTC_DCHECK(event_log); + } + + static const int kDefaultStartBitrateBps; + + // Bitrate config used until valid bitrate estimates are calculated. Also + // used to cap total bitrate used. + struct BitrateConfig { + int min_bitrate_bps = 0; + int start_bitrate_bps = kDefaultStartBitrateBps; + int max_bitrate_bps = -1; + } bitrate_config; + + // AudioState which is possibly shared between multiple calls. + // TODO(solenberg): Change this to a shared_ptr once we can use C++11. + rtc::scoped_refptr audio_state; + + // Audio Processing Module to be used in this call. + // TODO(solenberg): Change this to a shared_ptr once we can use C++11. + AudioProcessing* audio_processing = nullptr; + + // RtcEventLog to use for this call. Required. + // Use webrtc::RtcEventLog::CreateNull() for a null implementation. + RtcEventLog* event_log = nullptr; + }; + + struct Stats { + std::string ToString(int64_t time_ms) const; + + int send_bandwidth_bps = 0; // Estimated available send bandwidth. + int max_padding_bitrate_bps = 0; // Cumulative configured max padding. + int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. + int64_t pacer_delay_ms = 0; + int64_t rtt_ms = -1; + }; + + static Call* Create(const Call::Config& config); + + virtual AudioSendStream* CreateAudioSendStream( + const AudioSendStream::Config& config) = 0; + virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; + + virtual AudioReceiveStream* CreateAudioReceiveStream( + const AudioReceiveStream::Config& config) = 0; + virtual void DestroyAudioReceiveStream( + AudioReceiveStream* receive_stream) = 0; + + virtual VideoSendStream* CreateVideoSendStream( + VideoSendStream::Config config, + VideoEncoderConfig encoder_config) = 0; + virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0; + + virtual VideoReceiveStream* CreateVideoReceiveStream( + VideoReceiveStream::Config configuration) = 0; + virtual void DestroyVideoReceiveStream( + VideoReceiveStream* receive_stream) = 0; + + virtual FlexfecReceiveStream* CreateFlexfecReceiveStream( + FlexfecReceiveStream::Config configuration) = 0; + virtual void DestroyFlexfecReceiveStream( + FlexfecReceiveStream* receive_stream) = 0; + + // All received RTP and RTCP packets for the call should be inserted to this + // PacketReceiver. The PacketReceiver pointer is valid as long as the + // Call instance exists. + virtual PacketReceiver* Receiver() = 0; + + // Returns the call statistics, such as estimated send and receive bandwidth, + // pacing delay, etc. + virtual Stats GetStats() const = 0; + + // TODO(pbos): Like BitrateConfig above this is currently per-stream instead + // of maximum for entire Call. This should be fixed along with the above. + // Specifying a start bitrate (>0) will currently reset the current bitrate + // estimate. This is due to how the 'x-google-start-bitrate' flag is currently + // implemented. + virtual void SetBitrateConfig( + const Config::BitrateConfig& bitrate_config) = 0; + + // TODO(skvlad): When the unbundled case with multiple streams for the same + // media type going over different networks is supported, track the state + // for each stream separately. Right now it's global per media type. + virtual void SignalChannelNetworkState(MediaType media, + NetworkState state) = 0; + + virtual void OnTransportOverheadChanged( + MediaType media, + int transport_overhead_per_packet) = 0; + + virtual void OnNetworkRouteChanged( + const std::string& transport_name, + const rtc::NetworkRoute& network_route) = 0; + + virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; + + virtual ~Call() {} +}; + +} // namespace webrtc + +#endif // WEBRTC_CALL_CALL_H_ diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc index fb5ae0d45e..36fc64fb96 100644 --- a/webrtc/call/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc @@ -16,7 +16,7 @@ #include "webrtc/base/checks.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/thread_annotations.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/config.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/modules/audio_coding/include/audio_coding_module.h" diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc index 2d75be5eda..15c642fd91 100644 --- a/webrtc/call/call_unittest.cc +++ b/webrtc/call/call_unittest.cc @@ -11,11 +11,11 @@ #include #include -#include "webrtc/api/call/audio_state.h" -#include "webrtc/call.h" +#include "webrtc/call/audio_state.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" -#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" +#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" #include "webrtc/test/gtest.h" #include "webrtc/test/mock_voice_engine.h" diff --git a/webrtc/call/rampup_tests.h b/webrtc/call/rampup_tests.h index f03b926460..2c70a95a57 100644 --- a/webrtc/call/rampup_tests.h +++ b/webrtc/call/rampup_tests.h @@ -16,7 +16,7 @@ #include #include "webrtc/base/event.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/test/call_test.h" diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn index a50d8e47dc..2daaef4ffc 100644 --- a/webrtc/logging/BUILD.gn +++ b/webrtc/logging/BUILD.gn @@ -41,7 +41,7 @@ rtc_static_library("rtc_event_log_impl") { deps = [ ":rtc_event_log_api", "..:webrtc_common", - "../api:call_api", + "../call:call_interfaces", "../modules/rtp_rtcp", ] @@ -66,6 +66,7 @@ if (rtc_include_tests) { deps = [ ":rtc_event_log_impl", ":rtc_event_log_parser", + "../call", "../modules/rtp_rtcp", "../system_wrappers:metrics_default", "//testing/gmock", diff --git a/webrtc/logging/rtc_event_log/DEPS b/webrtc/logging/rtc_event_log/DEPS index 039ebf9507..39d2020bee 100644 --- a/webrtc/logging/rtc_event_log/DEPS +++ b/webrtc/logging/rtc_event_log/DEPS @@ -1,5 +1,6 @@ include_rules = [ "+webrtc/base", + "+webrtc/call", "+webrtc/modules/rtp_rtcp", "+webrtc/system_wrappers", ] diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc index 976ff2321c..b2d3b6eaa6 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc @@ -18,7 +18,7 @@ #include "webrtc/base/event.h" #include "webrtc/base/swap_queue.h" #include "webrtc/base/thread_checker.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h index ec57b8b394..0800a43299 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log.h +++ b/webrtc/logging/rtc_event_log/rtc_event_log.h @@ -14,9 +14,9 @@ #include #include -#include "webrtc/api/call/audio_receive_stream.h" -#include "webrtc/api/call/audio_send_stream.h" #include "webrtc/base/platform_file.h" +#include "webrtc/call/audio_receive_stream.h" +#include "webrtc/call/audio_send_stream.h" #include "webrtc/video_receive_stream.h" #include "webrtc/video_send_stream.h" diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc index 337b65b80f..2336caa284 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc @@ -15,7 +15,7 @@ #include "gflags/gflags.h" #include "webrtc/base/checks.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc index 8f1b89df6b..ce55a4fb39 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc @@ -18,7 +18,7 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/file_wrapper.h" diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc index 3bff88769b..258d0fd687 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc @@ -17,7 +17,7 @@ #include "webrtc/base/buffer.h" #include "webrtc/base/checks.h" #include "webrtc/base/random.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h" #include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h" diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h index 01ade07f95..3f89e304b2 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h +++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_ #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_ -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h" namespace webrtc { diff --git a/webrtc/media/base/mediaengine.h b/webrtc/media/base/mediaengine.h index 57a2c7e6e6..5b7443b6f3 100644 --- a/webrtc/media/base/mediaengine.h +++ b/webrtc/media/base/mediaengine.h @@ -18,10 +18,10 @@ #include #include -#include "webrtc/api/call/audio_state.h" #include "webrtc/api/rtpparameters.h" #include "webrtc/base/fileutils.h" #include "webrtc/base/sigslotrepeater.h" +#include "webrtc/call/audio_state.h" #include "webrtc/media/base/codec.h" #include "webrtc/media/base/mediachannel.h" #include "webrtc/media/base/videocommon.h" diff --git a/webrtc/media/base/videoengine_unittest.h b/webrtc/media/base/videoengine_unittest.h index e307d9f703..a673d3553a 100644 --- a/webrtc/media/base/videoengine_unittest.h +++ b/webrtc/media/base/videoengine_unittest.h @@ -18,7 +18,7 @@ #include "webrtc/base/bytebuffer.h" #include "webrtc/base/gunit.h" #include "webrtc/base/timeutils.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/media/base/fakenetworkinterface.h" #include "webrtc/media/base/fakevideocapturer.h" diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h index c5e5d09303..72fba70a8c 100644 --- a/webrtc/media/engine/fakewebrtccall.h +++ b/webrtc/media/engine/fakewebrtccall.h @@ -25,10 +25,10 @@ #include #include -#include "webrtc/api/call/audio_receive_stream.h" -#include "webrtc/api/call/audio_send_stream.h" #include "webrtc/base/buffer.h" -#include "webrtc/call.h" +#include "webrtc/call/audio_receive_stream.h" +#include "webrtc/call/audio_send_stream.h" +#include "webrtc/call/call.h" #include "webrtc/video_frame.h" #include "webrtc/video_receive_stream.h" #include "webrtc/video_send_stream.h" diff --git a/webrtc/media/engine/webrtcmediaengine.h b/webrtc/media/engine/webrtcmediaengine.h index af85b2f99f..49b050c593 100644 --- a/webrtc/media/engine/webrtcmediaengine.h +++ b/webrtc/media/engine/webrtcmediaengine.h @@ -14,7 +14,7 @@ #include #include -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/config.h" #include "webrtc/media/base/mediaengine.h" diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc index 9396fccb7d..92ac698015 100644 --- a/webrtc/media/engine/webrtcvideoengine2.cc +++ b/webrtc/media/engine/webrtcvideoengine2.cc @@ -21,7 +21,7 @@ #include "webrtc/base/stringutils.h" #include "webrtc/base/timeutils.h" #include "webrtc/base/trace_event.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/common_video/h264/profile_level_id.h" #include "webrtc/media/engine/constants.h" #include "webrtc/media/engine/internalencoderfactory.h" diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h index 736d47461a..5e4233f8cc 100644 --- a/webrtc/media/engine/webrtcvideoengine2.h +++ b/webrtc/media/engine/webrtcvideoengine2.h @@ -25,7 +25,7 @@ #include "webrtc/base/thread_checker.h" #include "webrtc/media/base/videosinkinterface.h" #include "webrtc/media/base/videosourceinterface.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/media/base/mediaengine.h" #include "webrtc/media/engine/webrtcvideodecoderfactory.h" #include "webrtc/media/engine/webrtcvideoencoderfactory.h" diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h index 3bed1a0317..633e7d3e95 100644 --- a/webrtc/media/engine/webrtcvoiceengine.h +++ b/webrtc/media/engine/webrtcvoiceengine.h @@ -16,14 +16,14 @@ #include #include -#include "webrtc/api/call/audio_state.h" #include "webrtc/base/buffer.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/networkroute.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/base/stream.h" #include "webrtc/base/thread_checker.h" -#include "webrtc/call.h" +#include "webrtc/call/audio_state.h" +#include "webrtc/call/call.h" #include "webrtc/config.h" #include "webrtc/media/base/rtputils.h" #include "webrtc/media/engine/webrtccommon.h" diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc index 43142ea367..d90695de26 100644 --- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc +++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc @@ -14,7 +14,7 @@ #include "webrtc/base/arraysize.h" #include "webrtc/base/byteorder.h" #include "webrtc/base/gunit.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/p2p/base/faketransportcontroller.h" #include "webrtc/test/field_trial.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc index 6e19e93ed1..fd3c1302b4 100644 --- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc +++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc @@ -16,7 +16,7 @@ #include #include "webrtc/base/checks.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/modules/audio_coding/neteq/tools/packet.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" diff --git a/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h b/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h index 06294b4797..dc90f045d5 100644 --- a/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h +++ b/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h @@ -15,7 +15,7 @@ #include "webrtc/base/basictypes.h" #include "webrtc/base/sequenced_task_checker.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h" #include "webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc index 6d42d38fb8..29c740d134 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc @@ -18,7 +18,7 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/logging.h" #include "webrtc/base/trace_event.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/common_types.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc index 6dcff3ab37..548f8f2604 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc @@ -18,7 +18,7 @@ #include "webrtc/base/rate_limiter.h" #include "webrtc/base/trace_event.h" #include "webrtc/base/timeutils.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h index 5843f7b9ff..03bc5bb3b5 100644 --- a/webrtc/test/call_test.h +++ b/webrtc/test/call_test.h @@ -13,7 +13,7 @@ #include #include -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/test/encoder_settings.h" #include "webrtc/test/fake_audio_device.h" diff --git a/webrtc/test/direct_transport.cc b/webrtc/test/direct_transport.cc index b26aadd054..a5bd05a3c2 100644 --- a/webrtc/test/direct_transport.cc +++ b/webrtc/test/direct_transport.cc @@ -9,7 +9,7 @@ */ #include "webrtc/test/direct_transport.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/test/gtest.h" diff --git a/webrtc/test/fake_network_pipe.cc b/webrtc/test/fake_network_pipe.cc index 5a8d4fb380..88851372cb 100644 --- a/webrtc/test/fake_network_pipe.cc +++ b/webrtc/test/fake_network_pipe.cc @@ -17,7 +17,7 @@ #include #include -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/system_wrappers/include/clock.h" namespace webrtc { diff --git a/webrtc/test/fake_network_pipe_unittest.cc b/webrtc/test/fake_network_pipe_unittest.cc index 6e635e1559..aa27399594 100644 --- a/webrtc/test/fake_network_pipe_unittest.cc +++ b/webrtc/test/fake_network_pipe_unittest.cc @@ -10,7 +10,7 @@ #include -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/test/fake_network_pipe.h" #include "webrtc/test/gmock.h" diff --git a/webrtc/test/layer_filtering_transport.h b/webrtc/test/layer_filtering_transport.h index d453556235..8f8cc2e313 100644 --- a/webrtc/test/layer_filtering_transport.h +++ b/webrtc/test/layer_filtering_transport.h @@ -10,7 +10,7 @@ #ifndef WEBRTC_TEST_LAYER_FILTERING_TRANSPORT_H_ #define WEBRTC_TEST_LAYER_FILTERING_TRANSPORT_H_ -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/test/direct_transport.h" #include "webrtc/test/fake_network_pipe.h" diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn index c8604f1067..266f6690c5 100644 --- a/webrtc/tools/BUILD.gn +++ b/webrtc/tools/BUILD.gn @@ -200,6 +200,7 @@ if (rtc_enable_protobuf) { } defines = [ "ENABLE_RTC_EVENT_LOG" ] deps = [ + "../call:call_interfaces", "../logging:rtc_event_log_impl", "../logging:rtc_event_log_parser", "../modules/congestion_controller", diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc index f018b1d7a0..b57f89e73f 100644 --- a/webrtc/tools/event_log_visualizer/analyzer.cc +++ b/webrtc/tools/event_log_visualizer/analyzer.cc @@ -17,21 +17,21 @@ #include #include -#include "webrtc/api/call/audio_receive_stream.h" -#include "webrtc/api/call/audio_send_stream.h" #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/rate_statistics.h" -#include "webrtc/call.h" +#include "webrtc/call/audio_receive_stream.h" +#include "webrtc/call/audio_send_stream.h" +#include "webrtc/call/call.h" #include "webrtc/common_types.h" #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" #include "webrtc/modules/congestion_controller/include/congestion_controller.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" -#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" +#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" +#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/video_receive_stream.h" #include "webrtc/video_send_stream.h" diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc index 9d37e28f87..286da6c221 100644 --- a/webrtc/video/end_to_end_tests.cc +++ b/webrtc/video/end_to_end_tests.cc @@ -20,7 +20,7 @@ #include "webrtc/base/file.h" #include "webrtc/base/optional.h" #include "webrtc/base/rate_limiter.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/common_video/include/frame_callback.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/media/base/fakevideorenderer.h" diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc index c00fe337af..2a73917551 100644 --- a/webrtc/video/replay.cc +++ b/webrtc/video/replay.cc @@ -16,7 +16,7 @@ #include "gflags/gflags.h" #include "webrtc/base/checks.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/system_wrappers/include/clock.h" diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc index d72b60c790..38c01a7d1f 100644 --- a/webrtc/video/video_quality_test.cc +++ b/webrtc/video/video_quality_test.cc @@ -23,7 +23,7 @@ #include "webrtc/base/optional.h" #include "webrtc/base/platform_file.h" #include "webrtc/base/timeutils.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc index 9003a7fe6a..82fb6befe9 100644 --- a/webrtc/video/video_send_stream_tests.cc +++ b/webrtc/video/video_send_stream_tests.cc @@ -18,7 +18,7 @@ #include "webrtc/base/logging.h" #include "webrtc/base/platform_thread.h" #include "webrtc/base/rate_limiter.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/common_video/include/frame_callback.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" diff --git a/webrtc/video/vie_encoder.h b/webrtc/video/vie_encoder.h index 5ad1ff7be5..ca7bd9334d 100644 --- a/webrtc/video/vie_encoder.h +++ b/webrtc/video/vie_encoder.h @@ -19,7 +19,7 @@ #include "webrtc/base/event.h" #include "webrtc/base/sequenced_task_checker.h" #include "webrtc/base/task_queue.h" -#include "webrtc/call.h" +#include "webrtc/call/call.h" #include "webrtc/common_types.h" #include "webrtc/common_video/include/video_bitrate_allocator.h" #include "webrtc/common_video/rotation.h" @@ -28,10 +28,10 @@ #include "webrtc/modules/video_coding/utility/quality_scaler.h" #include "webrtc/modules/video_coding/video_coding_impl.h" #include "webrtc/system_wrappers/include/atomic32.h" +#include "webrtc/typedefs.h" #include "webrtc/video/overuse_frame_detector.h" #include "webrtc/video_encoder.h" #include "webrtc/video_send_stream.h" -#include "webrtc/typedefs.h" namespace webrtc {