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openSUSE:12.3
MozillaFirefox
mozilla-webrtc-ppc.patch
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File mozilla-webrtc-ppc.patch of Package MozillaFirefox
Submitted-by: schwab@@linux-m68k.org Subject: fix PPC build References: (not delivered with the patch but apparently mix of:) Bug 750869 - Support WebRTC for Android in our build system (TM:20) Bug 814693 - Build failure on Debian powerpc (TM:20) diff --git a/media/webrtc/shared_libs.mk b/media/webrtc/shared_libs.mk --- a/media/webrtc/shared_libs.mk +++ b/media/webrtc/shared_libs.mk @@ -23,33 +23,39 @@ WEBRTC_LIBS = \ $(call EXPAND_LIBNAME_PATH,video_render_module,$(DEPTH)/media/webrtc/trunk/src/modules/modules_video_render_module) \ $(call EXPAND_LIBNAME_PATH,video_engine_core,$(DEPTH)/media/webrtc/trunk/src/video_engine/video_engine_video_engine_core) \ $(call EXPAND_LIBNAME_PATH,media_file,$(DEPTH)/media/webrtc/trunk/src/modules/modules_media_file) \ $(call EXPAND_LIBNAME_PATH,rtp_rtcp,$(DEPTH)/media/webrtc/trunk/src/modules/modules_rtp_rtcp) \ $(call EXPAND_LIBNAME_PATH,udp_transport,$(DEPTH)/media/webrtc/trunk/src/modules/modules_udp_transport) \ $(call EXPAND_LIBNAME_PATH,bitrate_controller,$(DEPTH)/media/webrtc/trunk/src/modules/modules_bitrate_controller) \ $(call EXPAND_LIBNAME_PATH,remote_bitrate_estimator,$(DEPTH)/media/webrtc/trunk/src/modules/modules_remote_bitrate_estimator) \ $(call EXPAND_LIBNAME_PATH,video_processing,$(DEPTH)/media/webrtc/trunk/src/modules/modules_video_processing) \ - $(call EXPAND_LIBNAME_PATH,video_processing_sse2,$(DEPTH)/media/webrtc/trunk/src/modules/modules_video_processing_sse2) \ $(call EXPAND_LIBNAME_PATH,voice_engine_core,$(DEPTH)/media/webrtc/trunk/src/voice_engine/voice_engine_voice_engine_core) \ $(call EXPAND_LIBNAME_PATH,audio_conference_mixer,$(DEPTH)/media/webrtc/trunk/src/modules/modules_audio_conference_mixer) \ $(call EXPAND_LIBNAME_PATH,audio_device,$(DEPTH)/media/webrtc/trunk/src/modules/modules_audio_device) \ $(call EXPAND_LIBNAME_PATH,audio_processing,$(DEPTH)/media/webrtc/trunk/src/modules/modules_audio_processing) \ $(call EXPAND_LIBNAME_PATH,aec,$(DEPTH)/media/webrtc/trunk/src/modules/modules_aec) \ - $(call EXPAND_LIBNAME_PATH,aec_sse2,$(DEPTH)/media/webrtc/trunk/src/modules/modules_aec_sse2) \ $(call EXPAND_LIBNAME_PATH,apm_util,$(DEPTH)/media/webrtc/trunk/src/modules/modules_apm_util) \ $(call EXPAND_LIBNAME_PATH,aecm,$(DEPTH)/media/webrtc/trunk/src/modules/modules_aecm) \ $(call EXPAND_LIBNAME_PATH,agc,$(DEPTH)/media/webrtc/trunk/src/modules/modules_agc) \ $(call EXPAND_LIBNAME_PATH,ns,$(DEPTH)/media/webrtc/trunk/src/modules/modules_ns) \ $(call EXPAND_LIBNAME_PATH,yuv,$(DEPTH)/media/webrtc/trunk/third_party/libyuv/libyuv_libyuv) \ $(call EXPAND_LIBNAME_PATH,webrtc_jpeg,$(DEPTH)/media/webrtc/trunk/src/common_video/common_video_webrtc_jpeg) \ $(call EXPAND_LIBNAME_PATH,nicer,$(DEPTH)/media/mtransport/third_party/nICEr/nicer_nicer) \ $(call EXPAND_LIBNAME_PATH,nrappkit,$(DEPTH)/media/mtransport/third_party/nrappkit/nrappkit_nrappkit) \ $(NULL) +# if we're on an intel arch, we want SSE2 optimizations +ifneq (,$(INTEL_ARCHITECTURE)) +WEBRTC_LIBS += \ + $(call EXPAND_LIBNAME_PATH,video_processing_sse2,$(DEPTH)/media/webrtc/trunk/src/modules/modules_video_processing_sse2) \ + $(call EXPAND_LIBNAME_PATH,aec_sse2,$(DEPTH)/media/webrtc/trunk/src/modules/modules_aec_sse2) \ + $(NULL) +endif + # If you enable one of these codecs in webrtc_config.gypi, you'll need to re-add the # relevant library from this list: # # $(call EXPAND_LIBNAME_PATH,G722,$(DEPTH)/media/webrtc/trunk/src/modules/modules_G722) \ # $(call EXPAND_LIBNAME_PATH,iLBC,$(DEPTH)/media/webrtc/trunk/src/modules/modules_iLBC) \ # $(call EXPAND_LIBNAME_PATH,iSAC,$(DEPTH)/media/webrtc/trunk/src/modules/modules_iSAC) \ # $(call EXPAND_LIBNAME_PATH,iSACFix,$(DEPTH)/media/webrtc/trunk/src/modules/modules_iSACFix) \ # diff --git a/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi b/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi --- a/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi +++ b/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi @@ -6,16 +6,19 @@ # in the file PATENTS. All contributing project authors may # be found in the AUTHORS file in the root of the source tree. { 'targets': [ { 'target_name': 'PCM16B', 'type': '<(library)', + 'dependencies': [ + '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing', + ], 'include_dirs': [ 'include', ], 'direct_dependent_settings': { 'include_dirs': [ 'include', ], }, diff --git a/media/webrtc/trunk/src/typedefs.h b/media/webrtc/trunk/src/typedefs.h --- a/media/webrtc/trunk/src/typedefs.h +++ b/media/webrtc/trunk/src/typedefs.h @@ -52,16 +52,24 @@ //#define WEBRTC_ARCH_ARMEL #define WEBRTC_ARCH_32_BITS #define WEBRTC_ARCH_LITTLE_ENDIAN #define WEBRTC_LITTLE_ENDIAN #elif defined(__MIPSEL__) #define WEBRTC_ARCH_32_BITS #define WEBRTC_ARCH_LITTLE_ENDIAN #define WEBRTC_LITTLE_ENDIAN +#elif defined(__powerpc__) +#if defined(__powerpc64__) +#define WEBRTC_ARCH_64_BITS +#else +#define WEBRTC_ARCH_32_BITS +#endif +#define WEBRTC_ARCH_BIG_ENDIAN +#define WEBRTC_BIG_ENDIAN #else #error Please add support for your architecture in typedefs.h #endif #if defined(__SSE2__) || defined(_MSC_VER) #define WEBRTC_USE_SSE2 #endif
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