--- a/mozilla-webrtc-ppc.patch	Sun Feb 03 16:47:11 2013 +0100
+++ /dev/null	Thu Jan 01 00:00:00 1970 +0000
@@ -1,102 +0,0 @@
-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
-