forked from pool/MozillaFirefox
1ad53d1168
* blocklist updates * backed out bmo#677092 (removed patch) * fixed problems involving HTTP proxy transactions OBS-URL: https://build.opensuse.org/package/show/mozilla:Factory/MozillaFirefox?expand=0&rev=315
103 lines
5.2 KiB
Diff
103 lines
5.2 KiB
Diff
Submitted-by: schwab@@linux-m68k.org
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Subject: fix PPC build
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References: (not delivered with the patch but apparently mix of:)
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Bug 750869 - Support WebRTC for Android in our build system (TM:20)
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Bug 814693 - Build failure on Debian powerpc (TM:20)
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diff --git a/media/webrtc/shared_libs.mk b/media/webrtc/shared_libs.mk
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--- a/media/webrtc/shared_libs.mk
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+++ b/media/webrtc/shared_libs.mk
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@@ -23,33 +23,39 @@ WEBRTC_LIBS = \
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$(call EXPAND_LIBNAME_PATH,video_render_module,$(DEPTH)/media/webrtc/trunk/src/modules/modules_video_render_module) \
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$(call EXPAND_LIBNAME_PATH,video_engine_core,$(DEPTH)/media/webrtc/trunk/src/video_engine/video_engine_video_engine_core) \
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$(call EXPAND_LIBNAME_PATH,media_file,$(DEPTH)/media/webrtc/trunk/src/modules/modules_media_file) \
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$(call EXPAND_LIBNAME_PATH,rtp_rtcp,$(DEPTH)/media/webrtc/trunk/src/modules/modules_rtp_rtcp) \
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$(call EXPAND_LIBNAME_PATH,udp_transport,$(DEPTH)/media/webrtc/trunk/src/modules/modules_udp_transport) \
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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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$(call EXPAND_LIBNAME_PATH,ns,$(DEPTH)/media/webrtc/trunk/src/modules/modules_ns) \
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$(call EXPAND_LIBNAME_PATH,yuv,$(DEPTH)/media/webrtc/trunk/third_party/libyuv/libyuv_libyuv) \
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$(call EXPAND_LIBNAME_PATH,webrtc_jpeg,$(DEPTH)/media/webrtc/trunk/src/common_video/common_video_webrtc_jpeg) \
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$(call EXPAND_LIBNAME_PATH,nicer,$(DEPTH)/media/mtransport/third_party/nICEr/nicer_nicer) \
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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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# $(call EXPAND_LIBNAME_PATH,G722,$(DEPTH)/media/webrtc/trunk/src/modules/modules_G722) \
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# $(call EXPAND_LIBNAME_PATH,iLBC,$(DEPTH)/media/webrtc/trunk/src/modules/modules_iLBC) \
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# $(call EXPAND_LIBNAME_PATH,iSAC,$(DEPTH)/media/webrtc/trunk/src/modules/modules_iSAC) \
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# $(call EXPAND_LIBNAME_PATH,iSACFix,$(DEPTH)/media/webrtc/trunk/src/modules/modules_iSACFix) \
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#
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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
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--- a/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
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+++ b/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
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@@ -6,16 +6,19 @@
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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{
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'targets': [
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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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'direct_dependent_settings': {
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'include_dirs': [
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'include',
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],
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},
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diff --git a/media/webrtc/trunk/src/typedefs.h b/media/webrtc/trunk/src/typedefs.h
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--- a/media/webrtc/trunk/src/typedefs.h
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+++ b/media/webrtc/trunk/src/typedefs.h
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@@ -52,16 +52,24 @@
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//#define WEBRTC_ARCH_ARMEL
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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(__MIPSEL__)
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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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#if defined(__SSE2__) || defined(_MSC_VER)
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#define WEBRTC_USE_SSE2
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#endif
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