forked from pool/MozillaFirefox
Accepting request 148241 from home:AndreasSchwab:ff
- Fix WebRTC to build on powerpc OBS-URL: https://build.opensuse.org/request/show/148241 OBS-URL: https://build.opensuse.org/package/show/mozilla:Factory/MozillaFirefox?expand=0&rev=314
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@ -1,3 +1,8 @@
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-------------------------------------------------------------------
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Sat Jan 12 17:25:11 UTC 2013 - schwab@linux-m68k.org
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- Fix WebRTC to build on powerpc
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-------------------------------------------------------------------
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Sun Jan 6 21:54:18 UTC 2013 - wr@rosenauer.org
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@ -101,6 +101,7 @@ Patch15: mozilla-gstreamer-760140.patch
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Patch16: mozilla-webrtc.patch
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Patch17: mozilla-libproxy-compat.patch
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Patch18: mozilla-backout-677092.patch
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Patch19: mozilla-webrtc-ppc.patch
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# Firefox/browser
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Patch30: firefox-browser-css.patch
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Patch31: firefox-kde.patch
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@ -237,6 +238,7 @@ cd $RPM_BUILD_DIR/mozilla
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%patch16 -p1
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%patch17 -p1
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%patch18 -p1
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%patch19 -p1
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#
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%patch30 -p1
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%if %suse_version >= 1110
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60
mozilla-webrtc-ppc.patch
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60
mozilla-webrtc-ppc.patch
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@ -0,0 +1,60 @@
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--- mozilla/media/webrtc/shared_libs.mk
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+++ mozilla/media/webrtc/shared_libs.mk
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@@ -28,13 +28,11 @@ WEBRTC_LIBS = \
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$(call EXPAND_LIBNAME_PATH,bitrate_controller,$(DEPTH)/media/webrtc/trunk/src/modules/modules_bitrate_controller) \
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$(call EXPAND_LIBNAME_PATH,remote_bitrate_estimator,$(DEPTH)/media/webrtc/trunk/src/modules/modules_remote_bitrate_estimator) \
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$(call EXPAND_LIBNAME_PATH,video_processing,$(DEPTH)/media/webrtc/trunk/src/modules/modules_video_processing) \
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- $(call EXPAND_LIBNAME_PATH,video_processing_sse2,$(DEPTH)/media/webrtc/trunk/src/modules/modules_video_processing_sse2) \
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$(call EXPAND_LIBNAME_PATH,voice_engine_core,$(DEPTH)/media/webrtc/trunk/src/voice_engine/voice_engine_voice_engine_core) \
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$(call EXPAND_LIBNAME_PATH,audio_conference_mixer,$(DEPTH)/media/webrtc/trunk/src/modules/modules_audio_conference_mixer) \
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$(call EXPAND_LIBNAME_PATH,audio_device,$(DEPTH)/media/webrtc/trunk/src/modules/modules_audio_device) \
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$(call EXPAND_LIBNAME_PATH,audio_processing,$(DEPTH)/media/webrtc/trunk/src/modules/modules_audio_processing) \
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$(call EXPAND_LIBNAME_PATH,aec,$(DEPTH)/media/webrtc/trunk/src/modules/modules_aec) \
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- $(call EXPAND_LIBNAME_PATH,aec_sse2,$(DEPTH)/media/webrtc/trunk/src/modules/modules_aec_sse2) \
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$(call EXPAND_LIBNAME_PATH,apm_util,$(DEPTH)/media/webrtc/trunk/src/modules/modules_apm_util) \
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$(call EXPAND_LIBNAME_PATH,aecm,$(DEPTH)/media/webrtc/trunk/src/modules/modules_aecm) \
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$(call EXPAND_LIBNAME_PATH,agc,$(DEPTH)/media/webrtc/trunk/src/modules/modules_agc) \
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@@ -45,6 +43,14 @@ WEBRTC_LIBS = \
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$(call EXPAND_LIBNAME_PATH,nrappkit,$(DEPTH)/media/mtransport/third_party/nrappkit/nrappkit_nrappkit) \
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$(NULL)
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+# if we're on an intel arch, we want SSE2 optimizations
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+ifneq (,$(INTEL_ARCHITECTURE))
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+WEBRTC_LIBS += \
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+ $(call EXPAND_LIBNAME_PATH,video_processing_sse2,$(DEPTH)/media/webrtc/trunk/src/modules/modules_video_processing_sse2) \
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+ $(call EXPAND_LIBNAME_PATH,aec_sse2,$(DEPTH)/media/webrtc/trunk/src/modules/modules_aec_sse2) \
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+ $(NULL)
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+endif
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+
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# If you enable one of these codecs in webrtc_config.gypi, you'll need to re-add the
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# relevant library from this list:
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#
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--- mozilla/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
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+++ mozilla/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
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@@ -11,6 +11,9 @@
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{
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'target_name': 'PCM16B',
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'type': '<(library)',
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+ 'dependencies': [
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+ '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
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+ ],
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'include_dirs': [
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'include',
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],
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--- mozilla/media/webrtc/trunk/src/typedefs.h
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+++ mozilla/media/webrtc/trunk/src/typedefs.h
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@@ -57,6 +57,14 @@
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#define WEBRTC_ARCH_32_BITS
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#define WEBRTC_ARCH_LITTLE_ENDIAN
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#define WEBRTC_LITTLE_ENDIAN
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+#elif defined(__powerpc__)
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+#if defined(__powerpc64__)
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+#define WEBRTC_ARCH_64_BITS
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+#else
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+#define WEBRTC_ARCH_32_BITS
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+#endif
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+#define WEBRTC_ARCH_BIG_ENDIAN
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+#define WEBRTC_BIG_ENDIAN
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#else
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#error Please add support for your architecture in typedefs.h
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#endif
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