1 files changed, 76 insertions, 0 deletions
diff --git a/libre/iceweasel/mozilla-1384655.patch b/libre/iceweasel/mozilla-1384655.patch
new file mode 100644
index 000000000..576ae15d9
--- /dev/null
+++ b/libre/iceweasel/mozilla-1384655.patch
@@ -0,0 +1,76 @@
+diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
++++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+@@ -12,24 +12,16 @@
+ 
+ #include "webrtc/base/logging.h"
+ #include "webrtc/modules/audio_device/audio_device_config.h"
+ #include "webrtc/modules/audio_device/linux/audio_device_alsa_linux.h"
+ 
+ #include "webrtc/system_wrappers/include/event_wrapper.h"
+ #include "webrtc/system_wrappers/include/sleep.h"
+ #include "webrtc/system_wrappers/include/trace.h"
+- 
+-#include "Latency.h"
+-
+-#define LOG_FIRST_CAPTURE(x) LogTime(AsyncLatencyLogger::AudioCaptureBase, \
+-                                     reinterpret_cast<uint64_t>(x), 0)
+-#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \
+-                                                 reinterpret_cast<uint64_t>(x), frames)
+-
+ 
+ webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
+ 
+ // Accesses ALSA functions through our late-binding symbol table instead of
+ // directly. This way we don't have to link to libasound, which means our binary
+ // will work on systems that don't have it.
+ #define LATE(sym) \
+   LATESYM_GET(webrtc_adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym)
+@@ -2138,20 +2130,18 @@ bool AudioDeviceLinuxALSA::RecThreadProc
+                buffer, size);
+         _recordingFramesLeft -= frames;
+ 
+         if (!_recordingFramesLeft)
+         { // buf is full
+             _recordingFramesLeft = _recordingFramesIn10MS;
+ 
+             if (_firstRecord) {
+-              LOG_FIRST_CAPTURE(this);
+               _firstRecord = false;
+             }
+-            LOG_CAPTURE_FRAMES(this, _recordingFramesIn10MS);
+             // store the recorded buffer (no action will be taken if the
+             // #recorded samples is not a full buffer)
+             _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+                                                _recordingFramesIn10MS);
+ 
+             uint32_t currentMicLevel = 0;
+             uint32_t newMicLevel = 0;
+ 
+diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
+--- a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
++++ b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
+@@ -13,22 +13,16 @@
+ 
+ #include "webrtc/modules/audio_device/audio_device_config.h"
+ #include "webrtc/modules/audio_device/sndio/audio_device_sndio.h"
+ 
+ #include "webrtc/system_wrappers/include/event_wrapper.h"
+ #include "webrtc/system_wrappers/include/sleep.h"
+ #include "webrtc/system_wrappers/include/trace.h"
+ 
+-#include "Latency.h"
+-
+-#define LOG_FIRST_CAPTURE(x) LogTime(AsyncLatencyLogger::AudioCaptureBase, \
+-                                     reinterpret_cast<uint64_t>(x), 0)
+-#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \
+-                                                 reinterpret_cast<uint64_t>(x), frames)
+ extern "C"
+ {
+     static void playOnmove(void *arg, int delta)
+     {
+         static_cast<webrtc::AudioDeviceSndio *>(arg)->_playDelay -= delta;
+     }
+ 
+     static void recOnmove(void *arg, int delta)
+