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