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Define GN arg owt_ffmpeg_root for the whole project. (#708)
This changes allows third_party/webrtc access owt_ffmpeg_root as well. So external headers will be included when it's defined.
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5 files changed

+16
-9
lines changed

5 files changed

+16
-9
lines changed

DEPS

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -98,7 +98,7 @@ deps = {
9898
Var('chromium_git') + '/chromium/src/third_party' + '@' + 'cd30703e732f3436f72f63c13f16ebb19803ddd6',
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# WebRTC-only dependencies (not present in Chromium).
100100
'src/third_party/webrtc':
101-
Var('deps_owt_git') + '/owt-deps-webrtc' + '@' + 'c34716479181dfaac4635098422ed3914d9555c3',
101+
Var('deps_owt_git') + '/owt-deps-webrtc' + '@' + 'f276236c868a52daa18496d619322c349e4b2b49',
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# Gradle 4.3-rc4. Used for testing Android Studio project generation for WebRTC.
103103
'src/third_party/webrtc/examples/androidtests/third_party/gradle': {
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'url': Var('chromium_git') + '/external/github.com/gradle/gradle.git' + '@' +

build_overrides/build.gni

Lines changed: 10 additions & 2 deletions
Original file line numberDiff line numberDiff line change
@@ -69,7 +69,15 @@ declare_args() {
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# Enable HEVC for WebRTC.
7070
rtc_use_h265 = false
7171

72-
# Build OWT for cloud gaming. It enables low latency features, and disables audio processing for server side. It may break normal WebRTC features, is not intended for general use.
72+
# Build OWT for cloud gaming. It enables low latency features, and disables
73+
# audio processing for server side. It may break normal WebRTC features, and
74+
# is not intended for general use.
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owt_cg_server = false
7476
owt_cg_client = false
75-
}
77+
78+
# Path to to root directory of FFmpeg, with headers in include sub-folder, and
79+
# libs in lib sub-folder. Binary libraries are not necessary for building OWT
80+
# SDK, but it is needed by your application or tests when this argument is
81+
# specified. If the value is an empty string, FFMPEG will not be used.
82+
owt_ffmpeg_root = ""
83+
}

scripts/build-win.py

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -94,7 +94,7 @@ def gngen(arch, sio_root, ffmpeg_root, ssl_root, msdk_root, quic_root, scheme, t
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# If sio_root is not specified, conference SDK is not able to build.
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gn_args.append('owt_sio_header_root="%s"' % (sio_root + r'\include'))
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if ffmpeg_root:
97-
gn_args.append('owt_ffmpeg_header_root="%s"'%(ffmpeg_root+r'\include'))
97+
gn_args.append('owt_ffmpeg_root="%s"'%(ffmpeg_root))
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if ffmpeg_root or msdk_root or cg_server:
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gn_args.append('rtc_use_h264=true')
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if msdk_root or cg_server:

scripts/build_linux.py

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -94,7 +94,7 @@ def gngen(arch, sio_root, ffmpeg_root, ssl_root, msdk_root, quic_root, scheme, t
9494
else:
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gn_args.extend(['enable_libaom=true'])
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if ffmpeg_root:
97-
gn_args.append('owt_ffmpeg_header_root="%s"'%(ffmpeg_root+'/include'))
97+
gn_args.append('owt_ffmpeg_root="%s"'%(ffmpeg_root))
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if ffmpeg_root or msdk_root or cg_server:
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gn_args.append('rtc_use_h264=true')
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if msdk_root or cg_server:

talk/owt/BUILD.gn

Lines changed: 3 additions & 4 deletions
Original file line numberDiff line numberDiff line change
@@ -12,7 +12,6 @@ declare_args() {
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owt_msdk_lib_root = ""
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owt_msdk_header_root = ""
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owt_sio_header_root = ""
15-
owt_ffmpeg_header_root = ""
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}
1716

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# Introduced for using libvpx config files. We only enable libvpx rate
@@ -236,8 +235,8 @@ static_library("owt_sdk_base") {
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defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
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}
238237

239-
if (owt_ffmpeg_header_root != "") {
240-
include_dirs += [ owt_ffmpeg_header_root ]
238+
if (owt_ffmpeg_root != "") {
239+
include_dirs += [ owt_ffmpeg_root + "/include" ]
241240
}
242241

243242
if (rtc_use_x11) {
@@ -354,7 +353,7 @@ static_library("owt_sdk_base") {
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]
355354
}
356355

357-
if (owt_ffmpeg_header_root != "") {
356+
if (owt_ffmpeg_root != "") {
358357
if (is_win) {
359358
sources += [
360359
"sdk/base/win/d3d11va_h264_decoder.cc",

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