Split webrtc/video into webrtc/{audio,call,video}.

Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}

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",
+    "call/rtc_event_log.cc",
-    "video/rtc_event_log.h",
+    "call/rtc_event_log.h",
   ]
 
   defines = []

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",
+  ]
+}

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

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 <string>
 

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_
+#ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
-#define WEBRTC_VIDEO_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_

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 {
 

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',
+    ],
+  },
+}

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",
+  ]
+}

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

webrtc/call/call.cc

@@ -13,26 +13,23 @@
 #include <map>
 #include <vector>
 
+#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"

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"

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 <deque>
 
@@ -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
 

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_
+#ifndef WEBRTC_CALL_RTC_EVENT_LOG_H_
-#define WEBRTC_VIDEO_RTC_EVENT_LOG_H_
+#define WEBRTC_CALL_RTC_EVENT_LOG_H_
 
 #include <string>
 
@@ -75,4 +75,4 @@ class RtcEventLog {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_RTC_EVENT_LOG_H_
+#endif  // WEBRTC_CALL_RTC_EVENT_LOG_H_

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 {

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) {

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"
 

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_
+#ifndef WEBRTC_CALL_TRANSPORT_ADAPTER_H_
-#define WEBRTC_VIDEO_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_

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',
+    ],
+  },
+}

webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc

@@ -16,15 +16,15 @@
 #include <limits>
 
 #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 {

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",

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.

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 {

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 {
 

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"

webrtc/video/video_send_stream.h

@@ -15,15 +15,15 @@
 #include <vector>
 
 #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 {
 

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"
 

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',

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 {
 

webrtc/voice_engine/test/auto_test/standard/codec_test.cc

@@ -11,11 +11,11 @@
 #include <stdio.h>
 #include <string>
 
+#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:

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"

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_in_dir': 'call',
-            'proto_out_dir': 'webrtc/video',
+            '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',
+        'call/rtc_event_log.cc',
-        'video/rtc_event_log.h',
+        'call/rtc_event_log.h',
       ],
       'conditions': [
         # If enable_protobuf is defined, we want to compile the protobuf

webrtc/webrtc_tests.gypi

@@ -146,18 +146,17 @@
       ],
     },
     {
-      # TODO(pbos): Rename target to webrtc_tests or rtc_tests, this target is
+      # TODO(pbos): Add separate target webrtc_audio_tests and move files there.
-      # not meant to only include video.
       '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',