diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index 40368597b2..5c3d9bc0cb 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -178,8 +178,10 @@ source_set("webrtc") { public_configs = [ ":common_inherited_config" ] deps = [ + "audio", ":webrtc_common", "base:rtc_base", + "call", "common_audio", "common_video", "modules/audio_coding", @@ -247,16 +249,16 @@ source_set("gtest_prod") { if (rtc_enable_protobuf) { proto_library("rtc_event_log_proto") { sources = [ - "video/rtc_event_log.proto", + "call/rtc_event_log.proto", ] - proto_out_dir = "webrtc/video" + proto_out_dir = "webrtc/call" } } source_set("rtc_event_log") { sources = [ - "video/rtc_event_log.cc", - "video/rtc_event_log.h", + "call/rtc_event_log.cc", + "call/rtc_event_log.h", ] defines = [] diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn new file mode 100644 index 0000000000..db0791fc97 --- /dev/null +++ b/webrtc/audio/BUILD.gn @@ -0,0 +1,31 @@ +# Copyright (c) 2015 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. + +import("../build/webrtc.gni") + +source_set("audio") { + sources = [ + "audio_receive_stream.cc", + "audio_receive_stream.h", + ] + + configs += [ "..:common_config" ] + public_configs = [ "..:common_inherited_config" ] + + if (is_clang) { + # Suppress warnings from Chrome's Clang plugins. + # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. + configs -= [ "//build/config/clang:find_bad_constructs" ] + } + + deps = [ + "..:webrtc_common", + "../voice_engine", + "../system_wrappers", + ] +} diff --git a/webrtc/audio/OWNERS b/webrtc/audio/OWNERS new file mode 100644 index 0000000000..f0cc72a885 --- /dev/null +++ b/webrtc/audio/OWNERS @@ -0,0 +1,9 @@ +solenberg@webrtc.org +tina.legrand@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* + +per-file BUILD.gn=kjellander@webrtc.org diff --git a/webrtc/video/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc similarity index 98% rename from webrtc/video/audio_receive_stream.cc rename to webrtc/audio/audio_receive_stream.cc index a1cf2ca33e..064749c61e 100644 --- a/webrtc/video/audio_receive_stream.cc +++ b/webrtc/audio/audio_receive_stream.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/video/audio_receive_stream.h" +#include "webrtc/audio/audio_receive_stream.h" #include diff --git a/webrtc/video/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h similarity index 92% rename from webrtc/video/audio_receive_stream.h rename to webrtc/audio/audio_receive_stream.h index c9ac04af39..70ad4fcf2b 100644 --- a/webrtc/video/audio_receive_stream.h +++ b/webrtc/audio/audio_receive_stream.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ -#define WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ +#ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ +#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ #include "webrtc/audio_receive_stream.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" @@ -50,4 +50,4 @@ class AudioReceiveStream : public webrtc::AudioReceiveStream { } // namespace internal } // namespace webrtc -#endif // WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_ +#endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ diff --git a/webrtc/video/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc similarity index 98% rename from webrtc/video/audio_receive_stream_unittest.cc rename to webrtc/audio/audio_receive_stream_unittest.cc index cf5314cea1..d6cce69dbf 100644 --- a/webrtc/video/audio_receive_stream_unittest.cc +++ b/webrtc/audio/audio_receive_stream_unittest.cc @@ -10,9 +10,9 @@ #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/audio/audio_receive_stream.h" #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" -#include "webrtc/video/audio_receive_stream.h" namespace webrtc { diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi new file mode 100644 index 0000000000..42f91aca16 --- /dev/null +++ b/webrtc/audio/webrtc_audio.gypi @@ -0,0 +1,21 @@ +# 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. +{ + 'variables': { + 'webrtc_audio_dependencies': [ + '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine', + '<(webrtc_root)/webrtc.gyp:rtc_event_log', + ], + 'webrtc_audio_sources': [ + 'audio/audio_receive_stream.cc', + 'audio/audio_receive_stream.h', + ], + }, +} diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn new file mode 100644 index 0000000000..ebd30db97f --- /dev/null +++ b/webrtc/call/BUILD.gn @@ -0,0 +1,33 @@ +# Copyright (c) 2015 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. + +import("../build/webrtc.gni") + +source_set("call") { + sources = [ + "call.cc", + "transport_adapter.cc", + "transport_adapter.h", + ] + + configs += [ "..:common_config" ] + public_configs = [ "..:common_inherited_config" ] + + if (is_clang) { + # Suppress warnings from Chrome's Clang plugins. + # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. + configs -= [ "//build/config/clang:find_bad_constructs" ] + } + + deps = [ + "..:rtc_event_log", + "..:webrtc_common", + "../modules/rtp_rtcp", + "../system_wrappers", + ] +} diff --git a/webrtc/call/OWNERS b/webrtc/call/OWNERS new file mode 100644 index 0000000000..792de19042 --- /dev/null +++ b/webrtc/call/OWNERS @@ -0,0 +1,11 @@ +mflodman@webrtc.org +pbos@webrtc.org +solenberg@webrtc.org +stefan@webrtc.org + +# These are for the common case of adding or renaming files. If you're doing +# structural changes, please get a review from a reviewer in this file. +per-file *.gyp=* +per-file *.gypi=* + +per-file BUILD.gn=kjellander@webrtc.org diff --git a/webrtc/video/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc similarity index 100% rename from webrtc/video/bitrate_estimator_tests.cc rename to webrtc/call/bitrate_estimator_tests.cc diff --git a/webrtc/video/call.cc b/webrtc/call/call.cc similarity index 98% rename from webrtc/video/call.cc rename to webrtc/call/call.cc index 2b2d596855..21109c21f5 100644 --- a/webrtc/video/call.cc +++ b/webrtc/call/call.cc @@ -13,26 +13,23 @@ #include #include +#include "webrtc/audio/audio_receive_stream.h" #include "webrtc/base/checks.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/call.h" +#include "webrtc/call/rtc_event_log.h" #include "webrtc/common.h" #include "webrtc/config.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/utility/interface/process_thread.h" -#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" -#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" -#include "webrtc/modules/video_render/include/video_render.h" #include "webrtc/system_wrappers/interface/cpu_info.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" #include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace_event.h" -#include "webrtc/video/audio_receive_stream.h" -#include "webrtc/video/rtc_event_log.h" #include "webrtc/video/video_receive_stream.h" #include "webrtc/video/video_send_stream.h" #include "webrtc/voice_engine/include/voe_codec.h" diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc similarity index 99% rename from webrtc/video/call_perf_tests.cc rename to webrtc/call/call_perf_tests.cc index bbf4caaebd..a7714b19c6 100644 --- a/webrtc/video/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc @@ -17,6 +17,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/call.h" +#include "webrtc/call/transport_adapter.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" @@ -33,7 +34,6 @@ #include "webrtc/test/rtp_rtcp_observer.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/perf_test.h" -#include "webrtc/video/transport_adapter.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_codec.h" #include "webrtc/voice_engine/include/voe_network.h" diff --git a/webrtc/video/packet_injection_tests.cc b/webrtc/call/packet_injection_tests.cc similarity index 100% rename from webrtc/video/packet_injection_tests.cc rename to webrtc/call/packet_injection_tests.cc diff --git a/webrtc/video/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc similarity index 99% rename from webrtc/video/rtc_event_log.cc rename to webrtc/call/rtc_event_log.cc index 8e6d1b9f22..23b120cb8e 100644 --- a/webrtc/video/rtc_event_log.cc +++ b/webrtc/call/rtc_event_log.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/video/rtc_event_log.h" +#include "webrtc/call/rtc_event_log.h" #include @@ -23,9 +23,9 @@ #ifdef ENABLE_RTC_EVENT_LOG // Files generated at build-time by the protobuf compiler. #ifdef WEBRTC_ANDROID_PLATFORM_BUILD -#include "external/webrtc/webrtc/video/rtc_event_log.pb.h" +#include "external/webrtc/webrtc/call/rtc_event_log.pb.h" #else -#include "webrtc/video/rtc_event_log.pb.h" +#include "webrtc/call/rtc_event_log.pb.h" #endif #endif diff --git a/webrtc/video/rtc_event_log.h b/webrtc/call/rtc_event_log.h similarity index 95% rename from webrtc/video/rtc_event_log.h rename to webrtc/call/rtc_event_log.h index daab3c116f..a4b7cadadb 100644 --- a/webrtc/video/rtc_event_log.h +++ b/webrtc/call/rtc_event_log.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_RTC_EVENT_LOG_H_ -#define WEBRTC_VIDEO_RTC_EVENT_LOG_H_ +#ifndef WEBRTC_CALL_RTC_EVENT_LOG_H_ +#define WEBRTC_CALL_RTC_EVENT_LOG_H_ #include @@ -75,4 +75,4 @@ class RtcEventLog { } // namespace webrtc -#endif // WEBRTC_VIDEO_RTC_EVENT_LOG_H_ +#endif // WEBRTC_CALL_RTC_EVENT_LOG_H_ diff --git a/webrtc/video/rtc_event_log.proto b/webrtc/call/rtc_event_log.proto similarity index 100% rename from webrtc/video/rtc_event_log.proto rename to webrtc/call/rtc_event_log.proto diff --git a/webrtc/video/rtc_event_log2rtp_dump.cc b/webrtc/call/rtc_event_log2rtp_dump.cc similarity index 98% rename from webrtc/video/rtc_event_log2rtp_dump.cc rename to webrtc/call/rtc_event_log2rtp_dump.cc index 4f1d93bbe4..8357d4856a 100644 --- a/webrtc/video/rtc_event_log2rtp_dump.cc +++ b/webrtc/call/rtc_event_log2rtp_dump.cc @@ -15,15 +15,15 @@ #include "gflags/gflags.h" #include "webrtc/base/checks.h" #include "webrtc/base/scoped_ptr.h" +#include "webrtc/call/rtc_event_log.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/test/rtp_file_writer.h" -#include "webrtc/video/rtc_event_log.h" // Files generated at build-time by the protobuf compiler. #ifdef WEBRTC_ANDROID_PLATFORM_BUILD -#include "external/webrtc/webrtc/video/rtc_event_log.pb.h" +#include "external/webrtc/webrtc/call/rtc_event_log.pb.h" #else -#include "webrtc/video/rtc_event_log.pb.h" +#include "webrtc/call/rtc_event_log.pb.h" #endif namespace { diff --git a/webrtc/video/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc similarity index 99% rename from webrtc/video/rtc_event_log_unittest.cc rename to webrtc/call/rtc_event_log_unittest.cc index cc8a8d2fdf..a916a2d816 100644 --- a/webrtc/video/rtc_event_log_unittest.cc +++ b/webrtc/call/rtc_event_log_unittest.cc @@ -19,18 +19,18 @@ #include "webrtc/base/checks.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" +#include "webrtc/call/rtc_event_log.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" #include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/test/test_suite.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" -#include "webrtc/video/rtc_event_log.h" // Files generated at build-time by the protobuf compiler. #ifdef WEBRTC_ANDROID_PLATFORM_BUILD -#include "external/webrtc/webrtc/video/rtc_event_log.pb.h" +#include "external/webrtc/webrtc/call/rtc_event_log.pb.h" #else -#include "webrtc/video/rtc_event_log.pb.h" +#include "webrtc/call/rtc_event_log.pb.h" #endif namespace webrtc { @@ -50,7 +50,7 @@ const char* kExtensionNames[] = {RtpExtension::kTOffset, RtpExtension::kTransportSequenceNumber}; const size_t kNumExtensions = 5; -} // namepsace +} // namespace // TODO(terelius): Place this definition with other parsing functions? MediaType GetRuntimeMediaType(rtclog::MediaType media_type) { diff --git a/webrtc/video/transport_adapter.cc b/webrtc/call/transport_adapter.cc similarity index 97% rename from webrtc/video/transport_adapter.cc rename to webrtc/call/transport_adapter.cc index 653e55caa4..733f3dba8f 100644 --- a/webrtc/video/transport_adapter.cc +++ b/webrtc/call/transport_adapter.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/video/transport_adapter.h" +#include "webrtc/call/transport_adapter.h" #include "webrtc/base/checks.h" diff --git a/webrtc/video/transport_adapter.h b/webrtc/call/transport_adapter.h similarity index 88% rename from webrtc/video/transport_adapter.h rename to webrtc/call/transport_adapter.h index c9a7938546..c14c2625ce 100644 --- a/webrtc/video/transport_adapter.h +++ b/webrtc/call/transport_adapter.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_VIDEO_TRANSPORT_ADAPTER_H_ -#define WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_ +#ifndef WEBRTC_CALL_TRANSPORT_ADAPTER_H_ +#define WEBRTC_CALL_TRANSPORT_ADAPTER_H_ #include "webrtc/common_types.h" #include "webrtc/system_wrappers/interface/atomic32.h" @@ -34,4 +34,4 @@ class TransportAdapter : public webrtc::Transport { } // namespace internal } // namespace webrtc -#endif // WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_ +#endif // WEBRTC_CALL_TRANSPORT_ADAPTER_H_ diff --git a/webrtc/call/webrtc_call.gypi b/webrtc/call/webrtc_call.gypi new file mode 100644 index 0000000000..4a606f884e --- /dev/null +++ b/webrtc/call/webrtc_call.gypi @@ -0,0 +1,22 @@ +# 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. +{ + 'variables': { + 'webrtc_call_dependencies': [ + '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/webrtc.gyp:rtc_event_log', + ], + 'webrtc_call_sources': [ + 'call/call.cc', + 'call/transport_adapter.cc', + 'call/transport_adapter.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 14e105123e..bfd49d3cec 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 @@ -16,15 +16,15 @@ #include #include "webrtc/base/checks.h" +#include "webrtc/call/rtc_event_log.h" #include "webrtc/modules/audio_coding/neteq/tools/packet.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/video/rtc_event_log.h" // Files generated at build-time by the protobuf compiler. #ifdef WEBRTC_ANDROID_PLATFORM_BUILD -#include "external/webrtc/webrtc/video/rtc_event_log.pb.h" +#include "external/webrtc/webrtc/call/rtc_event_log.pb.h" #else -#include "webrtc/video/rtc_event_log.pb.h" +#include "webrtc/call/rtc_event_log.pb.h" #endif namespace webrtc { diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn index 04d95ceaa6..a2233c2b57 100644 --- a/webrtc/video/BUILD.gn +++ b/webrtc/video/BUILD.gn @@ -35,17 +35,12 @@ source_set("video") { "../video_engine/vie_remb.h", "../video_engine/vie_sync_module.cc", "../video_engine/vie_sync_module.h", - "audio_receive_stream.cc", - "audio_receive_stream.h", - "call.cc", "encoded_frame_callback_adapter.cc", "encoded_frame_callback_adapter.h", "receive_statistics_proxy.cc", "receive_statistics_proxy.h", "send_statistics_proxy.cc", "send_statistics_proxy.h", - "transport_adapter.cc", - "transport_adapter.h", "video_capture_input.cc", "video_capture_input.h", "video_decoder.cc", diff --git a/webrtc/video/OWNERS b/webrtc/video/OWNERS index b5f9aeba7e..3f5e1653ec 100644 --- a/webrtc/video/OWNERS +++ b/webrtc/video/OWNERS @@ -1,6 +1,6 @@ mflodman@webrtc.org -stefan@webrtc.org pbos@webrtc.org +stefan@webrtc.org # These are for the common case of adding or renaming files. If you're doing # structural changes, please get a review from a reviewer in this file. diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc index 7224478e14..1248f6cebc 100644 --- a/webrtc/video/end_to_end_tests.cc +++ b/webrtc/video/end_to_end_tests.cc @@ -18,6 +18,7 @@ #include "webrtc/base/event.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" +#include "webrtc/call/transport_adapter.h" #include "webrtc/frame_callback.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" @@ -42,7 +43,6 @@ #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" #include "webrtc/test/testsupport/perf_test.h" -#include "webrtc/video/transport_adapter.h" #include "webrtc/video_encoder.h" namespace webrtc { diff --git a/webrtc/video/rampup_tests.h b/webrtc/video/rampup_tests.h index 8c18a52eb9..62edda3406 100644 --- a/webrtc/video/rampup_tests.h +++ b/webrtc/video/rampup_tests.h @@ -17,10 +17,10 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" +#include "webrtc/call/transport_adapter.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/system_wrappers/interface/event_wrapper.h" #include "webrtc/test/call_test.h" -#include "webrtc/video/transport_adapter.h" namespace webrtc { diff --git a/webrtc/video/video_receive_stream.h b/webrtc/video/video_receive_stream.h index 15742383d0..ef2e135a8b 100644 --- a/webrtc/video/video_receive_stream.h +++ b/webrtc/video/video_receive_stream.h @@ -15,13 +15,13 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" +#include "webrtc/call/transport_adapter.h" #include "webrtc/common_video/interface/incoming_video_stream.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/modules/video_render/include/video_render_defines.h" #include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/video/encoded_frame_callback_adapter.h" #include "webrtc/video/receive_statistics_proxy.h" -#include "webrtc/video/transport_adapter.h" #include "webrtc/video_engine/vie_channel.h" #include "webrtc/video_engine/vie_channel_group.h" #include "webrtc/video_engine/vie_encoder.h" diff --git a/webrtc/video/video_send_stream.h b/webrtc/video/video_send_stream.h index 58593146d4..94f5f5a5b8 100644 --- a/webrtc/video/video_send_stream.h +++ b/webrtc/video/video_send_stream.h @@ -15,15 +15,15 @@ #include #include "webrtc/call.h" +#include "webrtc/call/transport_adapter.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/video/encoded_frame_callback_adapter.h" #include "webrtc/video/send_statistics_proxy.h" -#include "webrtc/video/transport_adapter.h" #include "webrtc/video/video_capture_input.h" #include "webrtc/video_receive_stream.h" #include "webrtc/video_send_stream.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" namespace webrtc { diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc index b396cd6836..688278eadc 100644 --- a/webrtc/video/video_send_stream_tests.cc +++ b/webrtc/video/video_send_stream_tests.cc @@ -17,6 +17,7 @@ #include "webrtc/base/criticalsection.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" +#include "webrtc/call/transport_adapter.h" #include "webrtc/frame_callback.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" @@ -36,7 +37,6 @@ #include "webrtc/test/null_transport.h" #include "webrtc/test/testsupport/perf_test.h" #include "webrtc/video/send_statistics_proxy.h" -#include "webrtc/video/transport_adapter.h" #include "webrtc/video_frame.h" #include "webrtc/video_send_stream.h" diff --git a/webrtc/video/webrtc_video.gypi b/webrtc/video/webrtc_video.gypi index 198fe3e06b..66ab577ae1 100644 --- a/webrtc/video/webrtc_video.gypi +++ b/webrtc/video/webrtc_video.gypi @@ -24,17 +24,12 @@ '<(webrtc_root)/webrtc.gyp:rtc_event_log', ], 'webrtc_video_sources': [ - 'video/audio_receive_stream.cc', - 'video/audio_receive_stream.h', - 'video/call.cc', 'video/encoded_frame_callback_adapter.cc', 'video/encoded_frame_callback_adapter.h', 'video/receive_statistics_proxy.cc', 'video/receive_statistics_proxy.h', 'video/send_statistics_proxy.cc', 'video/send_statistics_proxy.h', - 'video/transport_adapter.cc', - 'video/transport_adapter.h', 'video/video_capture_input.cc', 'video/video_capture_input.h', 'video/video_decoder.cc', diff --git a/webrtc/voice_engine/channel_manager.h b/webrtc/voice_engine/channel_manager.h index d01fc68d0f..81768e010f 100644 --- a/webrtc/voice_engine/channel_manager.h +++ b/webrtc/voice_engine/channel_manager.h @@ -15,10 +15,10 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" +#include "webrtc/call/rtc_event_log.h" #include "webrtc/system_wrappers/interface/atomic32.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/typedefs.h" -#include "webrtc/video/rtc_event_log.h" namespace webrtc { diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc index e7c9fbf906..eeb12aba04 100644 --- a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc +++ b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc @@ -11,11 +11,11 @@ #include #include +#include "webrtc/call/rtc_event_log.h" #include "webrtc/test/test_suite.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" #include "webrtc/voice_engine/voice_engine_defines.h" -#include "webrtc/video/rtc_event_log.h" class CodecTest : public AfterStreamingFixture { protected: diff --git a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc index bb524639ed..312ac7ca31 100644 --- a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc +++ b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc @@ -20,12 +20,12 @@ #include "gflags/gflags.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" +#include "webrtc/call/rtc_event_log.h" #include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/test/channel_transport/include/channel_transport.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/trace_to_stderr.h" -#include "webrtc/video/rtc_event_log.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_codec.h" diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp index 283aa46208..13d88b3a14 100644 --- a/webrtc/webrtc.gyp +++ b/webrtc/webrtc.gyp @@ -22,10 +22,10 @@ # This target should only be built if enable_protobuf is defined 'target_name': 'rtc_event_log_proto', 'type': 'static_library', - 'sources': ['video/rtc_event_log.proto',], + 'sources': ['call/rtc_event_log.proto',], 'variables': { - 'proto_in_dir': 'video', - 'proto_out_dir': 'webrtc/video', + 'proto_in_dir': 'call', + 'proto_out_dir': 'webrtc/call', }, 'includes': ['build/protoc.gypi'], }, @@ -36,7 +36,7 @@ { 'target_name': 'rtc_event_log2rtp_dump', 'type': 'executable', - 'sources': ['video/rtc_event_log2rtp_dump.cc',], + 'sources': ['call/rtc_event_log2rtp_dump.cc',], 'dependencies': [ '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', 'rtc_event_log', @@ -49,6 +49,8 @@ ], 'includes': [ 'build/common.gypi', + 'audio/webrtc_audio.gypi', + 'call/webrtc_call.gypi', 'video/webrtc_video.gypi', ], 'variables': { @@ -105,10 +107,14 @@ 'video_renderer.h', 'video_send_stream.h', + '<@(webrtc_audio_sources)', + '<@(webrtc_call_sources)', '<@(webrtc_video_sources)', ], 'dependencies': [ 'common.gyp:*', + '<@(webrtc_audio_dependencies)', + '<@(webrtc_call_dependencies)', '<@(webrtc_video_dependencies)', 'rtc_event_log', ], @@ -127,8 +133,8 @@ 'target_name': 'rtc_event_log', 'type': 'static_library', 'sources': [ - 'video/rtc_event_log.cc', - 'video/rtc_event_log.h', + 'call/rtc_event_log.cc', + 'call/rtc_event_log.h', ], 'conditions': [ # If enable_protobuf is defined, we want to compile the protobuf diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi index 06bb25fdde..3e0fcbf432 100644 --- a/webrtc/webrtc_tests.gypi +++ b/webrtc/webrtc_tests.gypi @@ -146,18 +146,17 @@ ], }, { - # TODO(pbos): Rename target to webrtc_tests or rtc_tests, this target is - # not meant to only include video. + # TODO(pbos): Add separate target webrtc_audio_tests and move files there. 'target_name': 'video_engine_tests', 'type': '<(gtest_target_type)', 'sources': [ + 'audio/audio_receive_stream_unittest.cc', + 'call/bitrate_estimator_tests.cc', + 'call/packet_injection_tests.cc', 'test/common_unittest.cc', 'test/testsupport/metrics/video_metrics_unittest.cc', 'tools/agc/agc_manager_unittest.cc', - 'video/audio_receive_stream_unittest.cc', - 'video/bitrate_estimator_tests.cc', 'video/end_to_end_tests.cc', - 'video/packet_injection_tests.cc', 'video/send_statistics_proxy_unittest.cc', 'video/video_capture_input_unittest.cc', 'video/video_decoder_unittest.cc', @@ -194,7 +193,7 @@ 'webrtc.gyp:rtc_event_log_proto', ], 'sources': [ - 'video/rtc_event_log_unittest.cc', + 'call/rtc_event_log_unittest.cc', ], }], ], @@ -203,11 +202,10 @@ 'target_name': 'webrtc_perf_tests', 'type': '<(gtest_target_type)', 'sources': [ + 'call/call_perf_tests.cc', 'modules/audio_coding/neteq/test/neteq_performance_unittest.cc', 'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc', - 'tools/agc/agc_manager_integrationtest.cc', - 'video/call_perf_tests.cc', 'video/full_stack.cc', 'video/rampup_tests.cc', 'video/rampup_tests.h',