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author | Andreas Grapentin <andreas@grapentin.org> | 2017-09-04 19:12:38 +0200 |
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committer | Andreas Grapentin <andreas@grapentin.org> | 2017-09-04 19:33:48 +0200 |
commit | fd3f2dd8bda291dafb3c546d7f9e4f4fce16a47a (patch) | |
tree | 9ae71ec907b03a0fff2a20f7c9d027a1501caa55 /libre/iceweasel/mozilla-1384655.patch | |
parent | 9285de9d87218c05dce0eaeeed9c92bf810a2b45 (diff) | |
download | abslibre-fd3f2dd8bda291dafb3c546d7f9e4f4fce16a47a.tar.gz abslibre-fd3f2dd8bda291dafb3c546d7f9e4f4fce16a47a.tar.bz2 abslibre-fd3f2dd8bda291dafb3c546d7f9e4f4fce16a47a.zip |
libre/iceweasel: updated to 55.0.3
Diffstat (limited to 'libre/iceweasel/mozilla-1384655.patch')
-rw-r--r-- | libre/iceweasel/mozilla-1384655.patch | 76 |
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) + |