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authorAvatar Bernd Steinhauser <berniyh@exherbo.org> 2017-09-08 18:29:03 +0200
committerAvatar Bernd Steinhauser <berniyh@exherbo.org> 2017-09-08 18:29:49 +0200
commit57ab67e80161f9b517a1302899d4e64673fd6218 (patch)
treee45e48a2f17245b527f06242f0d8736007a08615
parent0e5fa79b0c18eee5dd059182818184ef3322d1f0 (diff)
downloaddesktop-57ab67e80161f9b517a1302899d4e64673fd6218.tar.gz
desktop-57ab67e80161f9b517a1302899d4e64673fd6218.tar.xz
firefox: fix build failure.
-rw-r--r--packages/net-www/firefox/files/rev-e4f5cc3fbd93.patch87
-rw-r--r--packages/net-www/firefox/firefox-55.0.3.exheres-01
2 files changed, 88 insertions, 0 deletions
diff --git a/packages/net-www/firefox/files/rev-e4f5cc3fbd93.patch b/packages/net-www/firefox/files/rev-e4f5cc3fbd93.patch
new file mode 100644
index 0000000..fcdc9cd
--- /dev/null
+++ b/packages/net-www/firefox/files/rev-e4f5cc3fbd93.patch
@@ -0,0 +1,87 @@
+Source: upstream, can be removed in ff 56
+
+# HG changeset patch
+# User Paul Adenot <paul@paul.cx>
+# Date 1502179237 -7200
+# Node ID e4f5cc3fbd9360150c9abd0698ae79a3e3758fcd
+# Parent e09c51325c7bf8c1aa69fa843f4843556bbfa74b
+Bug 1384655 - Remove obsolete latency measuring macros in downstream code we don't use anymore. r=jesup, a=lizzard
+
+MozReview-Commit-ID: ARAWGe7yFM4
+
+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)
+
diff --git a/packages/net-www/firefox/firefox-55.0.3.exheres-0 b/packages/net-www/firefox/firefox-55.0.3.exheres-0
index 4bc249d..06afabb 100644
--- a/packages/net-www/firefox/firefox-55.0.3.exheres-0
+++ b/packages/net-www/firefox/firefox-55.0.3.exheres-0
@@ -16,5 +16,6 @@ DEFAULT_SRC_PREPARE_PATCHES=(
--fuzz=2
"${FILES}"/bug1332788.patch
"${FILES}"/${PN}-55-Make-webrtc-gtest-build-work-with-system-jpeg.patch
+ "${FILES}"/rev-e4f5cc3fbd93.patch
)