From 25d9c357ba0a6e5a76c099d3fc7615efd40a20c8 Mon Sep 17 00:00:00 2001 From: Andreas Grapentin Date: Wed, 4 Oct 2017 12:52:42 +0200 Subject: libre/iceweasel: updated to 56.0 --- libre/iceweasel/mozilla-1384655.patch | 76 ----------------------------------- 1 file changed, 76 deletions(-) delete mode 100644 libre/iceweasel/mozilla-1384655.patch (limited to 'libre/iceweasel/mozilla-1384655.patch') 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(x), 0) --#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \ -- reinterpret_cast(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(x), 0) --#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \ -- reinterpret_cast(x), frames) - extern "C" - { - static void playOnmove(void *arg, int delta) - { - static_cast(arg)->_playDelay -= delta; - } - - static void recOnmove(void *arg, int delta) - -- cgit v1.2.3