Bug 694814: Patch 2: modifications to webrtc.org single_rw_fifo r=glandium,ted

media/webrtc/moz.build

@@ -14,8 +14,11 @@ webrtc_non_unified_sources = [
     'trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c',   # Because of name clash in the kDampFilter variable
     'trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c', # Because of name clash in the kDampFilter variable
     'trunk/webrtc/modules/audio_coding/neteq4/audio_vector.cc',                  # Because of explicit template specializations
+    'trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc',       # Because of LATE()
+    'trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc',# Because of LATE()
     'trunk/webrtc/modules/audio_device/opensl/opensles_input.cc',                # Because of name clash in the kOption variable
     'trunk/webrtc/modules/audio_device/opensl/opensles_output.cc',               # Because of name clash in the kOption variable
+    'trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.cc',                # Because of name clash with #define FF
     'trunk/webrtc/modules/audio_device/win/audio_device_core_win.cc',            # Because of ordering assumptions in strsafe.h
     'trunk/webrtc/modules/audio_processing/aec/aec_core.c',                      # Because of name clash in the ComfortNoise function
     'trunk/webrtc/modules/audio_processing/aecm/aecm_core.c',                    # Because of name clash in the ComfortNoise function

media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo.cc

@@ -9,6 +9,9 @@
  */
 
 #include "webrtc/modules/audio_device/android/single_rw_fifo.h"
+#if defined(_MSC_VER)
+#include <windows.h>
+#endif
 
 static int UpdatePos(int pos, int capacity) {
   return (pos + 1) % capacity;
@@ -19,7 +22,11 @@ namespace webrtc {
 namespace subtle {
 
 inline void MemoryBarrier() {
+#if defined(_MSC_VER)
+  ::MemoryBarrier();
+#else
   __sync_synchronize();
+#endif
 }
 
 }  // namespace subtle

media/webrtc/trunk/webrtc/modules/audio_device/audio_device.gypi

@@ -123,6 +123,9 @@
             'win/audio_device_utility_win.h',
             'win/audio_mixer_manager_win.cc',
             'win/audio_mixer_manager_win.h',
+            # used externally for getUserMedia
+            'opensl/single_rw_fifo.cc',
+            'opensl/single_rw_fifo.h',
           ],
           'conditions': [
             ['OS=="android"', {
@@ -154,8 +157,6 @@
                     'opensl/opensles_input.cc',
                     'opensl/opensles_input.h',
                     'opensl/opensles_output.h',
-                    'opensl/single_rw_fifo.cc',
-                    'opensl/single_rw_fifo.h',
                     'shared/audio_device_utility_shared.cc',
                     'shared/audio_device_utility_shared.h',
                   ],