--- a/dom/bindings/moz.build
+++ b/dom/bindings/moz.build
@@ -90,17 +90,16 @@ LOCAL_INCLUDES += [
"/js/xpconnect/src",
"/js/xpconnect/wrappers",
"/layout/generic",
"/layout/style",
"/layout/xul/tree",
"/media/webrtc/",
"/netwerk/base/",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
]
LOCAL_INCLUDES += ["/third_party/msgpack/include"]
DEFINES["GOOGLE_PROTOBUF_NO_RTTI"] = True
DEFINES["GOOGLE_PROTOBUF_NO_STATIC_INITIALIZER"] = True
UNIFIED_SOURCES += [
--- a/dom/media/MediaManager.cpp
+++ b/dom/media/MediaManager.cpp
@@ -63,17 +63,17 @@
#include "pk11pub.h"
/* Using WebRTC backend on Desktops (Mac, Windows, Linux), otherwise default */
#include "MediaEngineDefault.h"
#if defined(MOZ_WEBRTC)
# 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)
# include <iphlpapi.h>
# include <objbase.h>
# include <tchar.h>
# include <winsock2.h>
--- a/dom/media/VideoFrameConverter.h
+++ b/dom/media/VideoFrameConverter.h
@@ -11,21 +11,21 @@
#include "MediaTimer.h"
#include "VideoSegment.h"
#include "VideoUtils.h"
#include "nsISupportsImpl.h"
#include "nsThreadUtils.h"
#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.
// Sometimes these are released synchronously but they can be forwarded all the
// way to the encoder for asynchronous encoding. With a pool size of 5,
// we allow 1 buffer for the current conversion, and 4 buffers to be queued at
// the encoder.
#define CONVERTER_BUFFER_POOL_SIZE 5
--- a/dom/media/bridge/moz.build
+++ b/dom/media/bridge/moz.build
@@ -19,17 +19,16 @@ LOCAL_INCLUDES += [
"/dom/media/webrtc/common/time_profiling",
"/dom/media/webrtc/jsapi",
"/dom/media/webrtc/libwebrtcglue",
"/dom/media/webrtc/transport",
"/dom/media/webrtc/transportbridge",
"/ipc/chromium/src",
"/media/webrtc/",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
]
if CONFIG["MOZ_WEBRTC"]:
XPCOM_MANIFESTS += [
"components.conf",
]
FINAL_LIBRARY = "xul"
--- a/dom/media/gtest/moz.build
+++ b/dom/media/gtest/moz.build
@@ -7,17 +7,16 @@
include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
DEFINES["ENABLE_SET_CUBEB_BACKEND"] = True
LOCAL_INCLUDES += [
"/dom/media/mediasink",
"/dom/media/webrtc/common/",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
]
UNIFIED_SOURCES += [
"MockCubeb.cpp",
"MockMediaResource.cpp",
"TestAudioBuffers.cpp",
"TestAudioCallbackDriver.cpp",
"TestAudioCompactor.cpp",
--- a/dom/media/moz.build
+++ b/dom/media/moz.build
@@ -339,17 +339,16 @@ LOCAL_INCLUDES += [
"/netwerk/base",
"/toolkit/content/tests/browser/",
]
if CONFIG["MOZ_WEBRTC"]:
LOCAL_INCLUDES += [
"/dom/media/webrtc/common",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
]
DEFINES["MOZILLA_INTERNAL_API"] = True
DEFINES["TRACING"] = True
if CONFIG["MOZ_ANDROID_HLS_SUPPORT"]:
DEFINES["MOZ_ANDROID_HLS_SUPPORT"] = True
--- a/dom/media/systemservices/CamerasChild.h
+++ b/dom/media/systemservices/CamerasChild.h
@@ -12,17 +12,17 @@
#include "MediaEventSource.h"
#include "mozilla/Mutex.h"
#include "mozilla/camera/PCamerasChild.h"
#include "mozilla/camera/PCamerasParent.h"
#include "nsCOMPtr.h"
// 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 {
namespace ipc {
class BackgroundChildImpl;
} // namespace ipc
namespace camera {
--- a/dom/media/systemservices/CamerasParent.cpp
+++ b/dom/media/systemservices/CamerasParent.cpp
@@ -19,17 +19,17 @@
#include "mozilla/dom/CanonicalBrowsingContext.h"
#include "mozilla/dom/WindowGlobalParent.h"
#include "mozilla/Preferences.h"
#include "mozilla/StaticPrefs_permissions.h"
#include "nsIPermissionManager.h"
#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>
# define getpid() _getpid()
#endif
#undef LOG
#undef LOG_VERBOSE
--- a/dom/media/systemservices/CamerasParent.h
+++ b/dom/media/systemservices/CamerasParent.h
@@ -7,24 +7,20 @@
#ifndef mozilla_CamerasParent_h
#define mozilla_CamerasParent_h
#include "VideoEngine.h"
#include "mozilla/camera/PCamerasParent.h"
#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"
#include "base/thread.h"
namespace mozilla {
namespace camera {
--- a/dom/media/systemservices/VideoEngine.cpp
+++ b/dom/media/systemservices/VideoEngine.cpp
@@ -1,19 +1,19 @@
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim: set sw=2 ts=8 et ft=cpp : */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#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
# include "mozilla/jni/Utils.h"
#endif
namespace mozilla::camera {
--- a/dom/media/systemservices/VideoEngine.h
+++ b/dom/media/systemservices/VideoEngine.h
@@ -5,19 +5,19 @@
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef mozilla_VideoEngine_h
#define mozilla_VideoEngine_h
#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>
namespace mozilla {
namespace camera {
// Historically the video engine was part of webrtc
// it was removed (and reimplemented in Talk)
--- a/dom/media/systemservices/VideoFrameUtils.cpp
+++ b/dom/media/systemservices/VideoFrameUtils.cpp
@@ -1,16 +1,16 @@
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim: set sw=2 ts=8 et ft=cpp : */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* 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 {
uint32_t VideoFrameUtils::TotalRequiredBufferSize(
const webrtc::VideoFrame& aVideoFrame) {
auto i420 = aVideoFrame.video_frame_buffer()->ToI420();
auto height = i420->height();
--- a/dom/media/systemservices/moz.build
+++ b/dom/media/systemservices/moz.build
@@ -17,17 +17,16 @@ if CONFIG["MOZ_WEBRTC"]:
"CamerasParent.cpp",
"VideoEngine.cpp",
"VideoFrameUtils.cpp",
]
LOCAL_INCLUDES += [
"/dom/media/webrtc",
"/media/libyuv/libyuv/include",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
"/tools/profiler/public",
]
if CONFIG["OS_TARGET"] == "Android":
UNIFIED_SOURCES += [
"android_video_capture/device_info_android.cc",
"android_video_capture/video_capture_android.cc",
]
--- a/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
+++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
@@ -10,18 +10,18 @@
#include "MediaManager.h"
#include "MediaTrackConstraints.h"
#include "mozilla/ErrorNames.h"
#include "mozilla/gfx/Point.h"
#include "mozilla/RefPtr.h"
#include "Tracing.h"
#include "VideoFrameUtils.h"
#include "VideoUtils.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 {
extern LazyLogModule gMediaManagerLog;
#define LOG(...) MOZ_LOG(gMediaManagerLog, LogLevel::Debug, (__VA_ARGS__))
#define LOG_FRAME(...) \
MOZ_LOG(gMediaManagerLog, LogLevel::Verbose, (__VA_ARGS__))
--- a/dom/media/webrtc/MediaEngineRemoteVideoSource.h
+++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.h
@@ -29,18 +29,18 @@
#include "mozilla/dom/MediaStreamTrackBinding.h"
// Camera Access via IPC
#include "CamerasChild.h"
#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;
}
namespace mozilla {
// Fitness distance is defined in
--- a/dom/media/webrtc/MediaEngineWebRTC.h
+++ b/dom/media/webrtc/MediaEngineWebRTC.h
@@ -31,20 +31,17 @@
#include "nsComponentManagerUtils.h"
#include "nsDirectoryServiceDefs.h"
#include "nsRefPtrHashtable.h"
#include "nsThreadUtils.h"
#include "prcvar.h"
#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 {
class MediaEngineWebRTC : public MediaEngine {
typedef MediaEngine Super;
public:
explicit MediaEngineWebRTC(MediaEnginePrefs& aPrefs);
--- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
@@ -14,23 +14,18 @@
#include "MediaTrackConstraints.h"
#include "mozilla/Assertions.h"
#include "mozilla/ErrorNames.h"
#include "nsContentUtils.h"
#include "nsIDUtils.h"
#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;
// These are restrictions from the webrtc.org code
#define MAX_CHANNELS 2
#define MONO 1
#define MAX_SAMPLING_FREQ 48000 // Hz - multiple of 100
--- a/dom/media/webrtc/MediaEngineWebRTCAudio.h
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.h
@@ -6,17 +6,17 @@
#ifndef MediaEngineWebRTCAudio_h
#define MediaEngineWebRTCAudio_h
#include "AudioPacketizer.h"
#include "AudioSegment.h"
#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 {
class AudioInputProcessing;
class AudioInputTrack;
// This class is created and used exclusively on the Media Manager thread, with
// exactly two exceptions:
--- a/dom/media/webrtc/common/NullTransport.h
+++ b/dom/media/webrtc/common/NullTransport.h
@@ -2,17 +2,17 @@
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef NULL_TRANSPORT_H_
#define NULL_TRANSPORT_H_
#include "mozilla/Attributes.h"
-#include "webrtc/api/call/transport.h"
+#include "api/call/transport.h"
namespace mozilla {
/**
* NullTransport is registered as ExternalTransport to throw away data
*/
class NullTransport : public webrtc::Transport {
public:
--- a/dom/media/webrtc/common/browser_logging/CSFLog.cpp
+++ b/dom/media/webrtc/common/browser_logging/CSFLog.cpp
@@ -2,17 +2,16 @@
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#include <stdio.h>
#include <string.h>
#include <stdarg.h>
#include "CSFLog.h"
-#include "rtc_base/basictypes.h"
#include <map>
#include "prrwlock.h"
#include "prthread.h"
#include "nsThreadUtils.h"
#include "mozilla/Logging.h"
#include "mozilla/Sprintf.h"
--- a/dom/media/webrtc/common/browser_logging/WebRtcLog.cpp
+++ b/dom/media/webrtc/common/browser_logging/WebRtcLog.cpp
@@ -2,17 +2,16 @@
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "WebRtcLog.h"
#include "mozilla/Logging.h"
#include "mozilla/StaticPtr.h"
#include "prenv.h"
-#include "common_types.h"
#include "rtc_base/logging.h"
#include "nscore.h"
#include "nsString.h"
#include "nsXULAppAPI.h"
#include "mozilla/Preferences.h"
#include "nsIFile.h"
--- a/dom/media/webrtc/common/moz.build
+++ b/dom/media/webrtc/common/moz.build
@@ -4,17 +4,17 @@
# License, v. 2.0. If a copy of the MPL was not distributed with this
# file, You can obtain one at http://mozilla.org/MPL/2.0/.
include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
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 += [
"browser_logging/CSFLog.cpp",
"browser_logging/WebRtcLog.cpp",
"time_profiling/timecard.c",
"YuvStamper.cpp",
]
--- a/dom/media/webrtc/jsapi/moz.build
+++ b/dom/media/webrtc/jsapi/moz.build
@@ -10,17 +10,16 @@ LOCAL_INCLUDES += [
"/dom/base",
"/dom/media",
"/dom/media/webrtc",
"/ipc/chromium/src",
"/media/webrtc",
"/netwerk/dns", # For nsDNSService2.h
"/third_party/libsrtp/src/include",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
]
UNIFIED_SOURCES += [
"MediaTransportHandler.cpp",
"MediaTransportHandlerIPC.cpp",
"MediaTransportParent.cpp",
"PacketDumper.cpp",
"PeerConnectionCtx.cpp",
--- a/dom/media/webrtc/jsep/JsepSessionImpl.cpp
+++ b/dom/media/webrtc/jsep/JsepSessionImpl.cpp
@@ -17,17 +17,17 @@
#include "mozilla/Telemetry.h"
#include "mozilla/UniquePtr.h"
#include "mozilla/net/DataChannelProtocol.h"
#include "nsDebug.h"
#include "nspr.h"
#include "nss.h"
#include "pk11pub.h"
-#include "webrtc/api/rtpparameters.h"
+#include "api/rtp_parameters.h"
#include "jsep/JsepTrack.h"
#include "jsep/JsepTransport.h"
#include "sdp/HybridSdpParser.h"
#include "sdp/SipccSdp.h"
namespace mozilla {
--- a/dom/media/webrtc/jsep/moz.build
+++ b/dom/media/webrtc/jsep/moz.build
@@ -4,15 +4,14 @@
# License, v. 2.0. If a copy of the MPL was not distributed with this
# file, You can obtain one at http://mozilla.org/MPL/2.0/.
include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
LOCAL_INCLUDES += [
"/dom/media/webrtc",
"/media/webrtc",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
"/third_party/sipcc",
]
UNIFIED_SOURCES += ["JsepSessionImpl.cpp", "JsepTrack.cpp", "SsrcGenerator.cpp"]
FINAL_LIBRARY = "xul"
--- a/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
+++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
@@ -16,25 +16,23 @@
#include "mozilla/media/MediaUtils.h"
#include "nsServiceManagerUtils.h"
#include "nsThreadUtils.h"
#include "mozilla/Telemetry.h"
#include "transport/runnable_utils.h"
#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"
#endif
namespace mozilla {
static const char* acLogTag = "WebrtcAudioSessionConduit";
--- a/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h
+++ b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h
@@ -15,26 +15,22 @@
#include "RtcpEventObserver.h"
#include "CodecConfig.h"
#include "VideoTypes.h"
#include "MediaConduitErrors.h"
#include "jsapi/RTCStatsReport.h"
#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>
namespace webrtc {
class VideoFrame;
}
--- a/dom/media/webrtc/libwebrtcglue/MediaDataCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/MediaDataCodec.h
@@ -1,17 +1,16 @@
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef MEDIA_DATA_CODEC_H_
#define MEDIA_DATA_CODEC_H_
#include "MediaConduitInterface.h"
-#include "webrtc/common_types.h"
namespace mozilla {
class WebrtcVideoDecoder;
class WebrtcVideoEncoder;
class MediaDataCodec {
public:
/**
--- a/dom/media/webrtc/libwebrtcglue/RtpRtcpConfig.h
+++ b/dom/media/webrtc/libwebrtcglue/RtpRtcpConfig.h
@@ -1,15 +1,15 @@
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef __RTPRTCP_CONFIG_H__
#define __RTPRTCP_CONFIG_H__
-#include "webrtc/common_types.h"
+#include "api/rtp_headers.h"
namespace mozilla {
class RtpRtcpConfig {
public:
RtpRtcpConfig() = delete;
explicit RtpRtcpConfig(const webrtc::RtcpMode aMode) : mRtcpMode(aMode) {}
webrtc::RtcpMode GetRtcpMode() const { return mRtcpMode; }
--- a/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.cpp
+++ b/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.cpp
@@ -1,18 +1,19 @@
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#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 {
using EntryType = dom::RTCRtpSourceEntryType;
double RtpSourceObserver::RtpSourceEntry::ToLinearAudioLevel() const {
// Spec indicates that a value of 127 should be set to 0
if (audioLevel == 127) {
--- a/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.h
+++ b/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.h
@@ -7,19 +7,21 @@
#ifndef AUDIOLEVELOBSERVER_H
#define AUDIOLEVELOBSERVER_H
#include <vector>
#include <map>
#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;
}
namespace mozilla {
/* Observes reception of RTP packets and tabulates data about the
--- a/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
+++ b/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
@@ -22,24 +22,23 @@
#include "nsIPrefService.h"
#include "nsServiceManagerUtils.h"
#include "nsThreadUtils.h"
#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"
#if defined(MOZ_WIDGET_ANDROID)
# include "VideoEngine.h"
#endif
#include "GmpVideoCodec.h"
--- a/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h
+++ b/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h
@@ -5,18 +5,18 @@
* file, You can obtain one at https://mozilla.org/MPL/2.0/. */
#ifndef VideoStreamFactory_h
#define VideoStreamFactory_h
#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 {
// Factory class for VideoStreams... vie_encoder.cc will call this to
// reconfigure.
class VideoStreamFactory
: public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
public:
--- a/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
@@ -39,17 +39,17 @@
#include "nsThreadUtils.h"
#include "mozilla/Monitor.h"
#include "mozilla/Mutex.h"
#include "mozIGeckoMediaPluginService.h"
#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"
#include "GMPVideoEncoderProxy.h"
#include "jsapi/PeerConnectionImpl.h"
namespace mozilla {
--- a/dom/media/webrtc/libwebrtcglue/WebrtcImageBuffer.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcImageBuffer.h
@@ -2,18 +2,18 @@
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#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 {
class Image;
}
class ImageBuffer : public webrtc::VideoFrameBuffer {
public:
--- a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.h
@@ -6,18 +6,18 @@
#define WebrtcMediaDataDecoderCodec_h__
#include "MediaConduitInterface.h"
#include "MediaInfo.h"
#include "MediaResult.h"
#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;
}
namespace mozilla {
namespace layers {
class Image;
class ImageContainer;
--- a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
@@ -6,17 +6,17 @@
#define WebrtcMediaDataEncoderCodec_h__
#include "MediaConduitInterface.h"
#include "MediaInfo.h"
#include "MediaResult.h"
#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 {
class MediaData;
class PEMFactory;
class SharedThreadPool;
class TaskQueue;
--- a/dom/media/webrtc/libwebrtcglue/moz.build
+++ b/dom/media/webrtc/libwebrtcglue/moz.build
@@ -8,17 +8,16 @@ include("/dom/media/webrtc/third_party_b
LOCAL_INCLUDES += [
"!/ipc/ipdl/_ipdlheaders",
"/dom/media/gmp", # for GMPLog.h,
"/dom/media/webrtc",
"/ipc/chromium/src",
"/media/libyuv/libyuv/include",
"/media/webrtc",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
]
UNIFIED_SOURCES += [
"AudioConduit.cpp",
"GmpVideoCodec.cpp",
"MediaDataCodec.cpp",
"RtpSourceObserver.cpp",
"VideoConduit.cpp",
--- a/dom/media/webrtc/moz.build
+++ b/dom/media/webrtc/moz.build
@@ -58,17 +58,16 @@ if CONFIG["MOZ_WEBRTC"]:
LOCAL_INCLUDES += [
"..",
"/dom/base",
"/dom/media",
"/dom/media/webrtc/common",
"/dom/media/webrtc/common/browser_logging",
"/media/libyuv/libyuv/include",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
]
if CONFIG["MOZ_WEBRTC_SIGNALING"]:
DIRS += [
"common",
"jsapi",
"jsep",
"libwebrtcglue",
--- a/dom/media/webrtc/third_party_build/moz.build
+++ b/dom/media/webrtc/third_party_build/moz.build
@@ -5,60 +5,48 @@
# file, You can obtain one at http://mozilla.org/MPL/2.0/.
with Files("**"):
BUG_COMPONENT = ("Core", "WebRTC")
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":
DIRS += ["../../../../third_party/pipewire/libpipewire"]
GN_DIRS += ["../../../../third_party/libwebrtc/webrtc"]
gn_vars_copy = gn_vars.copy()
--- a/dom/media/webrtc/transportbridge/MediaPipeline.cpp
+++ b/dom/media/webrtc/transportbridge/MediaPipeline.cpp
@@ -41,18 +41,18 @@
#include "mozilla/gfx/Point.h"
#include "mozilla/gfx/Types.h"
#include "nsError.h"
#include "nsThreadUtils.h"
#include "transport/runnable_utils.h"
#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)
#define AUDIO_SAMPLE_BUFFER_MAX_BYTES (480 * 2 * 2)
static_assert((WEBRTC_MAX_SAMPLE_RATE / 100) * sizeof(uint16_t) * 2 <=
AUDIO_SAMPLE_BUFFER_MAX_BYTES,
"AUDIO_SAMPLE_BUFFER_MAX_BYTES is not large enough");
--- a/dom/media/webrtc/transportbridge/MediaPipeline.h
+++ b/dom/media/webrtc/transportbridge/MediaPipeline.h
@@ -19,17 +19,17 @@
#include "transport/SrtpFlow.h" // For SRTP_MAX_EXPANSION
#include "transport/mediapacket.h"
#include "transport/runnable_utils.h"
#include "AudioPacketizer.h"
#include "MediaPipelineFilter.h"
#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.
#define WEBRTC_MAX_SAMPLE_RATE 48000
class nsIPrincipal;
namespace mozilla {
--- a/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp
+++ b/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp
@@ -4,18 +4,17 @@
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
// Original author: bcampen@mozilla.com
#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
extern mozilla::LazyLogModule gMediaPipelineLog;
#define DEBUG_LOG(x) MOZ_LOG(gMediaPipelineLog, LogLevel::Debug, x)
namespace mozilla {
--- a/dom/media/webrtc/transportbridge/RtpLogger.h
+++ b/dom/media/webrtc/transportbridge/RtpLogger.h
@@ -2,17 +2,17 @@
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
// Original author: nohlmeier@mozilla.com
#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 {
/* This class logs RTP and RTCP packets in hex in a format compatible to
* text2pcap.
* Example to convert the MOZ log file into a PCAP file:
* egrep '(RTP_PACKET|RTCP_PACKET)' moz.log | \
--- a/dom/media/webrtc/transportbridge/moz.build
+++ b/dom/media/webrtc/transportbridge/moz.build
@@ -10,17 +10,16 @@ LOCAL_INCLUDES += [
"/dom/media",
"/dom/media/webrtc",
"/ipc/chromium/src",
"/media/libyuv/libyuv/include",
"/media/webrtc",
"/third_party/libsrtp/src/crypto/include",
"/third_party/libsrtp/src/include",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
]
UNIFIED_SOURCES += [
"MediaPipeline.cpp",
"MediaPipelineFilter.cpp",
"RtpLogger.cpp",
]
--- a/ipc/glue/moz.build
+++ b/ipc/glue/moz.build
@@ -211,17 +211,16 @@ else:
LOCAL_INCLUDES += [
"/caps",
"/dom/broadcastchannel",
"/dom/indexedDB",
"/dom/storage",
"/netwerk/base",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
"/xpcom/build",
]
IPDL_SOURCES = [
"InputStreamParams.ipdlh",
"IPCStream.ipdlh",
"PBackground.ipdl",
"PBackgroundSharedTypes.ipdlh",
--- a/media/webrtc/signaling/gtest/MockCall.h
+++ b/media/webrtc/signaling/gtest/MockCall.h
@@ -1,18 +1,18 @@
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef MOCK_CALL_H_
#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;
namespace test {
class MockAudioSendStream : public webrtc::AudioSendStream {
public:
MockAudioSendStream() : mConfig(nullptr) {}
--- a/media/webrtc/signaling/gtest/moz.build
+++ b/media/webrtc/signaling/gtest/moz.build
@@ -29,17 +29,16 @@ if (
"/dom/media/webrtc/transport",
"/dom/media/webrtc/transport/test",
"/dom/media/webrtc/transport/third_party/nrappkit/src/registry",
"/dom/media/webrtc/transportbridge",
"/ipc/chromium/src",
"/media/webrtc/",
"/third_party/libsrtp/src/include",
"/third_party/libwebrtc",
- "/third_party/libwebrtc/webrtc",
"/third_party/sipcc",
]
SOURCES += [
"audioconduit_unittests.cpp",
"jsep_session_unittest.cpp",
"jsep_track_unittest.cpp",
"mediapipeline_unittest.cpp",
--- a/media/webrtc/signaling/gtest/rtpsources_unittests.cpp
+++ b/media/webrtc/signaling/gtest/rtpsources_unittests.cpp
@@ -1,11 +1,11 @@
#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"
using namespace mozilla;
namespace test {
class RtpSourcesTest : public ::testing::Test {
using RtpSourceHistory = RtpSourceObserver::RtpSourceHistory;
--- a/media/webrtc/signaling/gtest/videoconduit_unittests.cpp
+++ b/media/webrtc/signaling/gtest/videoconduit_unittests.cpp
@@ -9,18 +9,18 @@
#include "nspr.h"
#include "nss.h"
#include "ssl.h"
#include "VideoConduit.h"
#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"
using namespace mozilla;
namespace test {
class MockVideoSink : public rtc::VideoSinkInterface<webrtc::VideoFrame> {