libre/iceweasel: updated to 56.0

1 files changed, 0 insertions, 76 deletions

diff --git a/libre/iceweasel/mozilla-1384655.patch b/libre/iceweasel/mozilla-1384655.patch
deleted file mode 100644
index 576ae15d9..000000000
--- a/libre/iceweasel/mozilla-1384655.patch
+++ /dev/null
@@ -1,76 +0,0 @@
-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)
-