Bug 1654112 - Update include paths for moved upstream code. r=ng

Differential Revision: https://phabricator.services.mozilla.com/D130059
2020-10-23 14:02:54 -04:00 · 2020-10-23 14:02:54 -04:00 · 38d5d300d9
--- a/dom/bindings/moz.build
+++ b/dom/bindings/moz.build
@ -99,7 +99,6 @@ LOCAL_INCLUDES += [
    "/media/webrtc/",
    "/netwerk/base/",
    "/third_party/libwebrtc",
-    "/third_party/libwebrtc/webrtc",
 ]

 LOCAL_INCLUDES += ["/third_party/msgpack/include"]
--- a/dom/media/MediaManager.cpp
+++ b/dom/media/MediaManager.cpp
@ -68,7 +68,7 @@
 #  include "MediaEngineWebRTC.h"
 #  include "MediaEngineWebRTCAudio.h"
 #  include "browser_logging/WebRtcLog.h"
-#  include "webrtc/modules/audio_processing/include/audio_processing.h"
+#  include "modules/audio_processing/include/audio_processing.h"
 #endif

 #if defined(XP_WIN)
--- a/dom/media/VideoFrameConverter.h
+++ b/dom/media/VideoFrameConverter.h
@ -16,11 +16,11 @@
 #include "mozilla/TaskQueue.h"
 #include "mozilla/dom/ImageBitmapBinding.h"
 #include "mozilla/dom/ImageUtils.h"
-#include "webrtc/api/video/video_frame.h"
-#include "webrtc/common_video/include/i420_buffer_pool.h"
-#include "webrtc/common_video/include/video_frame_buffer.h"
-#include "webrtc/rtc_base/keep_ref_until_done.h"
-#include "webrtc/system_wrappers/include/clock.h"
+#include "api/video/video_frame.h"
+#include "common_video/include/i420_buffer_pool.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "rtc_base/keep_ref_until_done.h"
+#include "system_wrappers/include/clock.h"

 // The number of frame buffers VideoFrameConverter may create before returning
 // errors.
--- a/dom/media/bridge/moz.build
+++ b/dom/media/bridge/moz.build
@ -24,7 +24,6 @@ LOCAL_INCLUDES += [
    "/ipc/chromium/src",
    "/media/webrtc/",
    "/third_party/libwebrtc",
-    "/third_party/libwebrtc/webrtc",
 ]

 if CONFIG["MOZ_WEBRTC"]:
--- a/dom/media/gtest/moz.build
+++ b/dom/media/gtest/moz.build
@ -12,7 +12,6 @@ LOCAL_INCLUDES += [
    "/dom/media/mediasink",
    "/dom/media/webrtc/common/",
    "/third_party/libwebrtc",
-    "/third_party/libwebrtc/webrtc",
 ]

 UNIFIED_SOURCES += [
--- a/dom/media/moz.build
+++ b/dom/media/moz.build
@ -344,7 +344,6 @@ if CONFIG["MOZ_WEBRTC"]:
    LOCAL_INCLUDES += [
        "/dom/media/webrtc/common",
        "/third_party/libwebrtc",
-        "/third_party/libwebrtc/webrtc",
    ]

 DEFINES["MOZILLA_INTERNAL_API"] = True
--- a/dom/media/systemservices/CamerasChild.h
+++ b/dom/media/systemservices/CamerasChild.h
@ -17,7 +17,7 @@

 // conflicts with #include of scoped_ptr.h
 #undef FF
-#include "webrtc/modules/video_capture/video_capture_defines.h"
+#include "modules/video_capture/video_capture_defines.h"

 namespace mozilla {

--- a/dom/media/systemservices/CamerasParent.cpp
+++ b/dom/media/systemservices/CamerasParent.cpp
@ -24,7 +24,7 @@
 #include "nsThreadUtils.h"
 #include "nsNetUtil.h"

-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"

 #if defined(_WIN32)
 #  include <process.h>
--- a/dom/media/systemservices/CamerasParent.h
+++ b/dom/media/systemservices/CamerasParent.h
@ -12,14 +12,10 @@
 #include "mozilla/ipc/Shmem.h"
 #include "mozilla/ShmemPool.h"
 #include "mozilla/Atomics.h"
-#include "webrtc/modules/video_capture/video_capture.h"
-#include "webrtc/modules/video_capture/video_capture_defines.h"
-#include "webrtc/common_video/include/incoming_video_stream.h"
-#include "webrtc/media/base/videosinkinterface.h"
-
-// conflicts with #include of scoped_ptr.h
-#undef FF
-#include "webrtc/common_types.h"
+#include "api/video/video_sink_interface.h"
+#include "common_video/include/incoming_video_stream.h"
+#include "modules/video_capture/video_capture.h"
+#include "modules/video_capture/video_capture_defines.h"

 #include "CamerasChild.h"

--- a/dom/media/systemservices/VideoEngine.cpp
+++ b/dom/media/systemservices/VideoEngine.cpp
@ -6,9 +6,9 @@

 #include "VideoEngine.h"
 #include "video_engine/desktop_capture_impl.h"
-#include "webrtc/system_wrappers/include/clock.h"
+#include "system_wrappers/include/clock.h"
 #ifdef WEBRTC_ANDROID
-#  include "webrtc/modules/video_capture/video_capture.h"
+#  include "modules/video_capture/video_capture.h"
 #endif

 #ifdef MOZ_WIDGET_ANDROID
--- a/dom/media/systemservices/VideoEngine.h
+++ b/dom/media/systemservices/VideoEngine.h
@ -10,9 +10,9 @@
 #include "MediaEngine.h"
 #include "VideoFrameUtils.h"
 #include "mozilla/media/MediaUtils.h"
-#include "webrtc/modules/video_capture/video_capture_impl.h"
-#include "webrtc/modules/video_capture/video_capture_defines.h"
-#include "webrtc/modules/video_capture/video_capture_factory.h"
+#include "modules/video_capture/video_capture_impl.h"
+#include "modules/video_capture/video_capture_defines.h"
+#include "modules/video_capture/video_capture_factory.h"
 #include <memory>
 #include <functional>

--- a/dom/media/systemservices/VideoFrameUtils.cpp
+++ b/dom/media/systemservices/VideoFrameUtils.cpp
@ -5,7 +5,7 @@
 * You can obtain one at http://mozilla.org/MPL/2.0/. */

 #include "VideoFrameUtils.h"
-#include "webrtc/api/video/video_frame.h"
+#include "api/video/video_frame.h"
 #include "mozilla/ShmemPool.h"

 namespace mozilla {
--- a/dom/media/systemservices/moz.build
+++ b/dom/media/systemservices/moz.build
@ -22,7 +22,6 @@ if CONFIG["MOZ_WEBRTC"]:
        "/dom/media/webrtc",
        "/media/libyuv/libyuv/include",
        "/third_party/libwebrtc",
-        "/third_party/libwebrtc/webrtc",
        "/tools/profiler/public",
    ]

--- a/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
+++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
@ -16,8 +16,8 @@
 #include "VideoFrameUtils.h"
 #include "VideoUtils.h"
 #include "ImageContainer.h"
-#include "webrtc/common_video/include/video_frame_buffer.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"

 namespace mozilla {

--- a/dom/media/webrtc/MediaEngineRemoteVideoSource.h
+++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.h
@ -34,8 +34,8 @@
 #include "NullTransport.h"

 // WebRTC includes
-#include "webrtc/common_video/include/i420_buffer_pool.h"
-#include "webrtc/modules/video_capture/video_capture_defines.h"
+#include "common_video/include/i420_buffer_pool.h"
+#include "modules/video_capture/video_capture_defines.h"

 namespace webrtc {
 using CaptureCapability = VideoCaptureCapability;
--- a/dom/media/webrtc/MediaEngineWebRTC.h
+++ b/dom/media/webrtc/MediaEngineWebRTC.h
@ -36,10 +36,7 @@
 #include "prthread.h"

 // WebRTC library includes follow
-// Video Engine
-// conflicts with #include of scoped_ptr.h
-#undef FF
-#include "webrtc/modules/video_capture/video_capture_defines.h"
+#include "modules/video_capture/video_capture_defines.h"

 namespace mozilla {

--- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
@ -19,13 +19,8 @@
 #include "transport/runnable_utils.h"
 #include "Tracing.h"

-// scoped_ptr.h uses FF
-#ifdef FF
-#  undef FF
-#endif
-#include "webrtc/voice_engine/voice_engine_defines.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/common_audio/include/audio_util.h"
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/include/audio_processing.h"

 using namespace webrtc;

--- a/dom/media/webrtc/MediaEngineWebRTCAudio.h
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.h
@ -11,7 +11,7 @@
 #include "AudioDeviceInfo.h"
 #include "MediaEngineWebRTC.h"
 #include "MediaTrackListener.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/include/audio_processing.h"

 namespace mozilla {

--- a/dom/media/webrtc/common/NullTransport.h
+++ b/dom/media/webrtc/common/NullTransport.h
@ -7,7 +7,7 @@

 #include "mozilla/Attributes.h"

-#include "webrtc/api/call/transport.h"
+#include "api/call/transport.h"

 namespace mozilla {

--- a/dom/media/webrtc/common/browser_logging/CSFLog.cpp
+++ b/dom/media/webrtc/common/browser_logging/CSFLog.cpp
@ -7,7 +7,6 @@
 #include <stdarg.h>

 #include "CSFLog.h"
-#include "rtc_base/basictypes.h"

 #include <map>
 #include "prrwlock.h"
--- a/dom/media/webrtc/common/browser_logging/WebRtcLog.cpp
+++ b/dom/media/webrtc/common/browser_logging/WebRtcLog.cpp
@ -7,7 +7,6 @@
 #include "mozilla/Logging.h"
 #include "mozilla/StaticPtr.h"
 #include "prenv.h"
-#include "common_types.h"
 #include "rtc_base/logging.h"

 #include "nscore.h"
--- a/dom/media/webrtc/common/moz.build
+++ b/dom/media/webrtc/common/moz.build
@ -9,7 +9,7 @@ EXPORTS.mozilla.dom += ["CandidateInfo.h"]

 LOCAL_INCLUDES += [
    "/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr",
-    "/third_party/libwebrtc/webrtc",
+    "/third_party/libwebrtc",
 ]

 UNIFIED_SOURCES += [
--- a/dom/media/webrtc/jsapi/moz.build
+++ b/dom/media/webrtc/jsapi/moz.build
@ -15,7 +15,6 @@ LOCAL_INCLUDES += [
    "/netwerk/dns",  # For nsDNSService2.h
    "/third_party/libsrtp/src/include",
    "/third_party/libwebrtc",
-    "/third_party/libwebrtc/webrtc",
 ]

 UNIFIED_SOURCES += [
--- a/dom/media/webrtc/jsep/JsepSessionImpl.cpp
+++ b/dom/media/webrtc/jsep/JsepSessionImpl.cpp
@ -22,7 +22,7 @@
 #include "nss.h"
 #include "pk11pub.h"

-#include "webrtc/api/rtpparameters.h"
+#include "api/rtp_parameters.h"

 #include "jsep/JsepTrack.h"
 #include "jsep/JsepTransport.h"
--- a/dom/media/webrtc/jsep/moz.build
+++ b/dom/media/webrtc/jsep/moz.build
@ -9,7 +9,6 @@ LOCAL_INCLUDES += [
    "/dom/media/webrtc",
    "/media/webrtc",
    "/third_party/libwebrtc",
-    "/third_party/libwebrtc/webrtc",
    "/third_party/sipcc",
 ]

--- a/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
+++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
@ -21,15 +21,13 @@

 #include "pk11pub.h"

-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h"
-
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "webrtc/voice_engine/include/voe_errors.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
-#include "webrtc/system_wrappers/include/clock.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/voice_engine/include/voe_errors.h"
+#include "modules/voice_engine/voice_engine_impl.h"
+#include "system_wrappers/include/clock.h"

 #ifdef MOZ_WIDGET_ANDROID
 #  include "AndroidBridge.h"
--- a/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h
+++ b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h
@ -20,16 +20,12 @@

 #include "ImageContainer.h"

-#include "webrtc/call/call.h"
-#include "webrtc/common_types.h"
-#include "webrtc/common_types.h"
-#include "webrtc/api/video/video_frame_buffer.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_device/include/fake_audio_device.h"
-#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/voice_engine/include/voe_base.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/video/video_frame_buffer.h"
+#include "call/call.h"
+#include "modules/audio_device/include/fake_audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"

 #include <vector>
 #include <set>
--- a/dom/media/webrtc/libwebrtcglue/MediaDataCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/MediaDataCodec.h
@ -6,7 +6,6 @@
 #define MEDIA_DATA_CODEC_H_

 #include "MediaConduitInterface.h"
-#include "webrtc/common_types.h"

 namespace mozilla {

--- a/dom/media/webrtc/libwebrtcglue/RtpRtcpConfig.h
+++ b/dom/media/webrtc/libwebrtcglue/RtpRtcpConfig.h
@ -4,7 +4,7 @@

 #ifndef __RTPRTCP_CONFIG_H__
 #define __RTPRTCP_CONFIG_H__
-#include "webrtc/common_types.h"
+#include "api/rtp_headers.h"

 namespace mozilla {
 class RtpRtcpConfig {
--- a/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.cpp
+++ b/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.cpp
@ -6,8 +6,9 @@

 #include "RtpSourceObserver.h"
 #include "nsThreadUtils.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/modules/include/module_common_types.h"
+
+#include "modules/include/module_common_types.h"
+#include "system_wrappers/include/clock.h"

 namespace mozilla {

--- a/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.h
+++ b/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.h
@ -12,9 +12,11 @@

 #include "nsISupportsImpl.h"
 #include "mozilla/dom/RTCRtpSourcesBinding.h"
-#include "webrtc/common_types.h"
 #include "jsapi/RTCStatsReport.h"

+#include "api/rtp_headers.h"
+#include "modules/include/module_common_types.h"
+
 // Unit Test class
 namespace test {
 class RtpSourcesTest;
--- a/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
+++ b/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
@ -27,14 +27,13 @@
 #include "pk11pub.h"

 #include "api/video_codecs/sdp_video_format.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "media/base/media_constants.h"
 #include "media/engine/encoder_simulcast_proxy.h"
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/media/base/mediaconstants.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
-#include "webrtc/common_video/include/video_frame_buffer.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"

 #include "mozilla/Unused.h"

--- a/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h
+++ b/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h
@ -10,8 +10,8 @@
 #include "CodecConfig.h"
 #include "mozilla/Atomics.h"
 #include "mozilla/UniquePtr.h"
-#include "webrtc/media/base/videoadapter.h"
-#include "call/video_config.h"
+#include "api/video_codecs/video_encoder_config.h"
+#include "media/base/video_adapter.h"

 namespace mozilla {

--- a/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
@ -44,7 +44,7 @@
 #include "MediaConduitInterface.h"
 #include "AudioConduit.h"
 #include "VideoConduit.h"
-#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_codec_interface.h"

 #include "gmp-video-host.h"
 #include "GMPVideoDecoderProxy.h"
--- a/dom/media/webrtc/libwebrtcglue/WebrtcImageBuffer.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcImageBuffer.h
@ -7,8 +7,8 @@
 #ifndef WebrtcImageBuffer_h__
 #define WebrtcImageBuffer_h__

-#include "webrtc/common_video/include/video_frame_buffer.h"
-#include "webrtc/rtc_base/keep_ref_until_done.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "rtc_base/keep_ref_until_done.h"

 namespace mozilla {
 namespace layers {
--- a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.h
@ -11,8 +11,8 @@
 #include "PlatformDecoderModule.h"
 #include "VideoConduit.h"
 #include "WebrtcImageBuffer.h"
-#include "webrtc/common_video/include/video_frame_buffer.h"
-#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "modules/video_coding/include/video_codec_interface.h"

 namespace webrtc {
 class DecodedImageCallback;
--- a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
@ -11,7 +11,7 @@
 #include "PlatformEncoderModule.h"
 #include "WebrtcGmpVideoCodec.h"
 #include "common_video/include/bitrate_adjuster.h"
-#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_codec_interface.h"

 namespace mozilla {

--- a/dom/media/webrtc/libwebrtcglue/moz.build
+++ b/dom/media/webrtc/libwebrtcglue/moz.build
@ -13,7 +13,6 @@ LOCAL_INCLUDES += [
    "/media/libyuv/libyuv/include",
    "/media/webrtc",
    "/third_party/libwebrtc",
-    "/third_party/libwebrtc/webrtc",
 ]

 UNIFIED_SOURCES += [
--- a/dom/media/webrtc/moz.build
+++ b/dom/media/webrtc/moz.build
@ -63,7 +63,6 @@ if CONFIG["MOZ_WEBRTC"]:
        "/dom/media/webrtc/common/browser_logging",
        "/media/libyuv/libyuv/include",
        "/third_party/libwebrtc",
-        "/third_party/libwebrtc/webrtc",
    ]

 if CONFIG["MOZ_WEBRTC_SIGNALING"]:
--- a/dom/media/webrtc/third_party_build/moz.build
+++ b/dom/media/webrtc/third_party_build/moz.build
@ -10,50 +10,38 @@ with Files("**"):
 include("/build/gn.mozbuild")

 webrtc_non_unified_sources = [
-    "../../../../third_party/libwebrtc/webrtc/common_audio/vad/vad_core.c",  # Because of name clash in the kInitCheck variable
-    "../../../../third_party/libwebrtc/webrtc/common_audio/vad/webrtc_vad.c",  # Because of name clash in the kInitCheck variable
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_coding/acm2/codec_manager.cc",  # Because of duplicate IsCodecRED/etc
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_coding/codecs/g722/g722_decode.c",  # Because of name clash in the saturate function
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_coding/codecs/g722/g722_encode.c",  # Because of name clash in the saturate function
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c",  # Because of name clash in the exp2_Q10_T function
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c",  # Because of name clash in the exp2_Q10_T function
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c",  # Because of name clash in the kDampFilter variable
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c",  # Because of name clash in the kDampFilter variable
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_coding/neteq/audio_vector.cc",  # Because of explicit template specializations
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_device/android/audio_manager.cc",  # Because of TAG redefinition
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_device/android/audio_record_jni.cc",  # Becuse of commonly named module static vars
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_device/android/audio_track_jni.cc",  # Becuse of commonly named module static vars
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_device/android/opensles_player.cc",  # Because of TAG redefinition
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc",  # Because of LATE()
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc",  # Because of LATE()
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_device/win/audio_device_core_win.cc",  # Because of ordering assumptions in strsafe.h
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/aec/echo_cancellation.cc",  # Because of conflicts over 'near' on windows
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/aecm/aecm_core.cc",  # Because of the PART_LEN2 define
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/aecm/aecm_core_c.cc",  # Because of the PART_LEN2 define
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc",  # Because of the PART_LEN2 define
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc",  # Because of the PART_LEN2 define
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc",  # Because of the PART_LEN2 define
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/agc/legacy/analog_agc.c",  # Because of name clash in the kInitCheck variable
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc",  # Because of needing to define _USE_MATH_DEFINES before including <cmath>
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc",  # Because of needing to define _USE_MATH_DEFINES before including <cmath>
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc",  # Because of needing to define _USE_MATH_DEFINES before including <cmath>
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc",  # Because of name clash in the MapError function
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/echo_control_mobile_impl.cc",  # Because of name clash in the MapError function
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc",  # Because of kAlpha
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/gain_control_impl.cc",  # Because of name clash in the Handle typedef
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc",  # Because of name clash in the Handle typedef
-    "../../../../third_party/libwebrtc/webrtc/modules/audio_processing/rms_level.cc",  # Because of name clash in the kMinLevel variable
-    "../../../../third_party/libwebrtc/webrtc/modules/congestion_controller/trendline_estimator.cc",  # Because of name clash in kDeltaCounterMax
-    "../../../../third_party/libwebrtc/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc",  # Because base/logging.h uses #ifndef LOG before defining anything
-    "../../../../third_party/libwebrtc/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc",  # Because of duplicate definitions of static consts against remote_bitrate_estimator_abs_send_time.cc
-    "../../../../third_party/libwebrtc/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc",  # Because of identically named functions and vars between flexfec_receiver.cc and flexfec_sender.cc in an anonymous namespaces
-    "../../../../third_party/libwebrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc",  # Because of identically named functions and vars between tmmbr.cc and tmmbn.cc in an anonymous namespaces
-    "../../../../third_party/libwebrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc",  # Because of identically named functions and vars between tmmbr.cc and tmmbn.cc in an anonymous namespaces
-    "../../../../third_party/libwebrtc/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc",  # Because of identically named constant kRedForFecHeaderLength in an anonymous namespace
-    "../../../../third_party/libwebrtc/webrtc/modules/video_capture/windows/device_info_ds.cc",  # Because of the MEDIASUBTYPE_HDYC variable
-    "../../../../third_party/libwebrtc/webrtc/modules/video_capture/windows/help_functions_ds.cc",  # Because of initguid.h
-    "../../../../third_party/libwebrtc/webrtc/modules/video_capture/windows/sink_filter_ds.cc",  # Because of the MEDIASUBTYPE_HDYC variable and initguid.h
-    "../../../../third_party/libwebrtc/webrtc/video/overuse_frame_detector.cc",  # Because of name clash with call_stats.cc on kWeightFactor
+    "../../../../third_party/libwebrtc/common_audio/vad/vad_core.c",                                  # Because of name clash in the kInitCheck variable
+    "../../../../third_party/libwebrtc/common_audio/vad/webrtc_vad.c",                                # Because of name clash in the kInitCheck variable
+    "../../../../third_party/libwebrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c",   # Because of name clash in the exp2_Q10_T function
+    "../../../../third_party/libwebrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c",   # Because of name clash in the exp2_Q10_T function
+    "../../../../third_party/libwebrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c",   # Because of name clash in the kDampFilter variable
+    "../../../../third_party/libwebrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c", # Because of name clash in the kDampFilter variable
+    "../../../../third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc",                   # Because of explicit template specializations
+    "../../../../third_party/libwebrtc/modules/audio_device/android/audio_manager.cc",                # Because of TAG redefinition
+    "../../../../third_party/libwebrtc/modules/audio_device/android/audio_record_jni.cc",             # Becuse of commonly named module static vars
+    "../../../../third_party/libwebrtc/modules/audio_device/android/audio_track_jni.cc",              # Becuse of commonly named module static vars
+    "../../../../third_party/libwebrtc/modules/audio_device/android/opensles_player.cc",              # Because of TAG redefinition
+    "../../../../third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.cc",       # Because of LATE()
+    "../../../../third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc",# Because of LATE()
+    "../../../../third_party/libwebrtc/modules/audio_device/win/audio_device_core_win.cc",            # Because of ordering assumptions in strsafe.h
+    "../../../../third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.cc",                   # Because of the PART_LEN2 define
+    "../../../../third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc",                 # Because of the PART_LEN2 define
+    "../../../../third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_mips.cc",              # Because of the PART_LEN2 define
+    "../../../../third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_neon.cc",              # Because of the PART_LEN2 define
+    "../../../../third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.cc",         # Because of the PART_LEN2 define
+    "../../../../third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.cc",         # Because of name clash in the MapError function
+    "../../../../third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc", # Because of kAlpha
+    "../../../../third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc",                # Because of name clash in the Handle typedef
+    "../../../../third_party/libwebrtc/modules/audio_processing/rms_level.cc",                        # Because of name clash in the kMinLevel variable
+    "../../../../third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc",       # Because base/logging.h uses #ifndef LOG before defining anything
+    "../../../../third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc", # Because of duplicate definitions of static consts against remote_bitrate_estimator_abs_send_time.cc
+    "../../../../third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver.cc",                  # Because of identically named functions and vars between flexfec_receiver.cc and flexfec_sender.cc in an anonymous namespaces
+    "../../../../third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc",                 # Because of identically named functions and vars between tmmbr.cc and tmmbn.cc in an anonymous namespaces
+    "../../../../third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc",                 # Because of identically named functions and vars between tmmbr.cc and tmmbn.cc in an anonymous namespaces
+    "../../../../third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.cc",                  # Because of identically named constant kRedForFecHeaderLength in an anonymous namespace
+    "../../../../third_party/libwebrtc/modules/video_capture/windows/device_info_ds.cc",              # Because of the MEDIASUBTYPE_HDYC variable
+    "../../../../third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.cc",           # Because of initguid.h
+    "../../../../third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.cc",              # Because of the MEDIASUBTYPE_HDYC variable and initguid.h
 ]

 if CONFIG["MOZ_WIDGET_TOOLKIT"] == "gtk":
--- a/dom/media/webrtc/transportbridge/MediaPipeline.cpp
+++ b/dom/media/webrtc/transportbridge/MediaPipeline.cpp
@ -46,8 +46,8 @@
 #include "jsapi/MediaTransportHandler.h"
 #include "Tracing.h"
 #include "libwebrtcglue/WebrtcImageBuffer.h"
-#include "webrtc/common_video/include/video_frame_buffer.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"

 // Max size given stereo is 480*2*2 = 1920 (10ms of 16-bits stereo audio at
 // 48KHz)
--- a/dom/media/webrtc/transportbridge/MediaPipeline.h
+++ b/dom/media/webrtc/transportbridge/MediaPipeline.h
@ -24,7 +24,7 @@
 #include "MediaSegment.h"
 #include "jsapi/PacketDumper.h"

-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "test/rtp_header_parser.h"

 // Should come from MediaEngine.h, but that's a pain to include here
 // because of the MOZILLA_EXTERNAL_LINKAGE stuff.
--- a/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp
+++ b/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp
@ -9,8 +9,7 @@

 #include "MediaPipelineFilter.h"

-#include "webrtc/common_types.h"
-#include "webrtc/api/rtpparameters.h"
+#include "api/rtpparameters.h"
 #include "mozilla/Logging.h"

 // defined in MediaPipeline.cpp
--- a/dom/media/webrtc/transportbridge/RtpLogger.h
+++ b/dom/media/webrtc/transportbridge/RtpLogger.h
@ -7,7 +7,7 @@
 #ifndef rtplogger_h__
 #define rtplogger_h__

-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "test/rtp_header_parser.h"
 #include "transport/mediapacket.h"

 namespace mozilla {
--- a/dom/media/webrtc/transportbridge/moz.build
+++ b/dom/media/webrtc/transportbridge/moz.build
@ -15,7 +15,6 @@ LOCAL_INCLUDES += [
    "/third_party/libsrtp/src/crypto/include",
    "/third_party/libsrtp/src/include",
    "/third_party/libwebrtc",
-    "/third_party/libwebrtc/webrtc",
 ]

 UNIFIED_SOURCES += [
--- a/ipc/glue/moz.build
+++ b/ipc/glue/moz.build
@ -217,7 +217,6 @@ LOCAL_INCLUDES += [
    "/dom/storage",
    "/netwerk/base",
    "/third_party/libwebrtc",
-    "/third_party/libwebrtc/webrtc",
    "/xpcom/build",
 ]

--- a/media/webrtc/signaling/gtest/MockCall.h
+++ b/media/webrtc/signaling/gtest/MockCall.h
@ -6,8 +6,8 @@
 #define MOCK_CALL_H_

 #include "mozilla/Assertions.h"
-#include <webrtc/api/call/audio_sink.h>
-#include <webrtc/call/call.h>
+#include <api/call/audio_sink.h>
+#include <call/call.h>

 using namespace webrtc;

--- a/media/webrtc/signaling/gtest/moz.build
+++ b/media/webrtc/signaling/gtest/moz.build
@ -34,7 +34,6 @@ if (
        "/media/webrtc/",
        "/third_party/libsrtp/src/include",
        "/third_party/libwebrtc",
-        "/third_party/libwebrtc/webrtc",
        "/third_party/sipcc",
    ]

--- a/media/webrtc/signaling/gtest/rtpsources_unittests.cpp
+++ b/media/webrtc/signaling/gtest/rtpsources_unittests.cpp
@ -1,6 +1,6 @@
 #include <RtpSourceObserver.h>
 #include "RTCStatsReport.h"
-#include "webrtc/modules/include/module_common_types.h"
+#include "modules/include/module_common_types.h"
 #define GTEST_HAS_RTTI 0
 #include "gtest/gtest.h"

--- a/media/webrtc/signaling/gtest/videoconduit_unittests.cpp
+++ b/media/webrtc/signaling/gtest/videoconduit_unittests.cpp
@ -14,8 +14,8 @@
 #include "RtpRtcpConfig.h"
 #include "WebrtcGmpVideoCodec.h"

-#include "webrtc/media/base/videoadapter.h"
-#include "webrtc/media/base/videosinkinterface.h"
+#include "api/video/video_sink_interface.h"
+#include "media/base/video_adapter.h"

 #include "MockCall.h"