forked from pool/MozillaFirefox
61 lines
3.0 KiB
Diff
61 lines
3.0 KiB
Diff
|
--- 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
|