--- mozilla/media/webrtc/shared_libs.mk
+++ mozilla/media/webrtc/shared_libs.mk
@@ -28,13 +28,11 @@ WEBRTC_LIBS = \
   $(call EXPAND_LIBNAME_PATH,bitrate_controller,$(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) \
@@ -45,6 +43,14 @@ WEBRTC_LIBS = \
   $(call EXPAND_LIBNAME_PATH,nrappkit,$(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:
 #
--- mozilla/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
+++ mozilla/media/webrtc/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
@@ -11,6 +11,9 @@
     {
       'target_name': 'PCM16B',
       'type': '<(library)',
+      'dependencies': [
+        '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+      ],
       'include_dirs': [
         'include',
       ],
--- mozilla/media/webrtc/trunk/src/typedefs.h
+++ mozilla/media/webrtc/trunk/src/typedefs.h
@@ -57,6 +57,14 @@
 #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