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Diffstat (limited to 'src/mumble/PulseAudio.cpp')
-rw-r--r--src/mumble/PulseAudio.cpp57
1 files changed, 27 insertions, 30 deletions
diff --git a/src/mumble/PulseAudio.cpp b/src/mumble/PulseAudio.cpp
index 1527ca444..5bb0baf79 100644
--- a/src/mumble/PulseAudio.cpp
+++ b/src/mumble/PulseAudio.cpp
@@ -8,9 +8,6 @@
#include "MainWindow.h"
#include "Timer.h"
#include "User.h"
-
-// We define a global macro called 'g'. This can lead to issues when included code uses 'g' as a type or parameter name
-// (like protobuf 3.7 does). As such, for now, we have to make this our last include.
#include "Global.h"
#ifdef Q_CC_GNU
@@ -148,7 +145,7 @@ PulseAudioSystem::~PulseAudioSystem() {
}
QString PulseAudioSystem::outputDevice() const {
- QString odev = g.s.qsPulseAudioOutput;
+ QString odev = Global::get().s.qsPulseAudioOutput;
if (odev.isEmpty()) {
odev = qsDefaultOutput;
}
@@ -159,7 +156,7 @@ QString PulseAudioSystem::outputDevice() const {
}
QString PulseAudioSystem::inputDevice() const {
- QString idev = g.s.qsPulseAudioInput;
+ QString idev = Global::get().s.qsPulseAudioInput;
if (idev.isEmpty()) {
idev = qsDefaultInput;
}
@@ -192,8 +189,8 @@ void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
if (!bSourceDone || !bSinkDone || !bServerDone)
return;
- AudioInputPtr ai = g.ai;
- AudioOutputPtr ao = g.ao;
+ AudioInputPtr ai = Global::get().ai;
+ AudioOutputPtr ao = Global::get().ao;
AudioInput *raw_ai = ai.get();
AudioOutput *raw_ao = ao.get();
PulseAudioInput *pai = dynamic_cast< PulseAudioInput * >(raw_ai);
@@ -231,7 +228,7 @@ void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
do_start = true;
break;
case PA_STREAM_READY: {
- if (g.s.iOutputDelay != iDelayCache) {
+ if (Global::get().s.iOutputDelay != iDelayCache) {
do_stop = true;
} else if (odev != qsOutputCache) {
do_stop = true;
@@ -252,13 +249,13 @@ void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
const pa_sample_spec *pss = m_pulseAudio.stream_get_sample_spec(pasOutput);
const size_t sampleSize = (pss->format == PA_SAMPLE_FLOAT32NE) ? sizeof(float) : sizeof(short);
const unsigned int iBlockLen = pao->iFrameSize * pss->channels * static_cast< unsigned int >(sampleSize);
- buff.tlength = iBlockLen * (g.s.iOutputDelay + 1);
+ buff.tlength = iBlockLen * (Global::get().s.iOutputDelay + 1);
buff.minreq = iBlockLen;
buff.maxlength = -1;
buff.prebuf = -1;
buff.fragsize = iBlockLen;
- iDelayCache = g.s.iOutputDelay;
+ iDelayCache = Global::get().s.iOutputDelay;
qsOutputCache = odev;
m_pulseAudio.stream_connect_playback(pasOutput, qPrintable(odev), &buff, PA_STREAM_ADJUST_LATENCY, nullptr,
@@ -334,9 +331,9 @@ void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
bool do_stop = false;
bool do_start = false;
- if ((!pai || g.s.echoOption == EchoCancelOptionID::DISABLED) && (est == PA_STREAM_READY)) {
+ if ((!pai || Global::get().s.echoOption == EchoCancelOptionID::DISABLED) && (est == PA_STREAM_READY)) {
do_stop = true;
- } else if (pai && g.s.echoOption != EchoCancelOptionID::DISABLED) {
+ } else if (pai && Global::get().s.echoOption != EchoCancelOptionID::DISABLED) {
switch (est) {
case PA_STREAM_TERMINATED: {
if (pasSpeaker)
@@ -347,7 +344,7 @@ void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
if ((pss.format != PA_SAMPLE_FLOAT32NE) && (pss.format != PA_SAMPLE_S16NE))
pss.format = PA_SAMPLE_FLOAT32NE;
pss.rate = SAMPLE_RATE;
- if ((pss.channels == 0) || (g.s.echoOption != EchoCancelOptionID::SPEEX_MULTICHANNEL))
+ if ((pss.channels == 0) || (Global::get().s.echoOption != EchoCancelOptionID::SPEEX_MULTICHANNEL))
pss.channels = 1;
pasSpeaker =
@@ -360,7 +357,7 @@ void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
do_start = true;
break;
case PA_STREAM_READY: {
- if ((g.s.echoOption == EchoCancelOptionID::SPEEX_MULTICHANNEL) != bEchoMultiCache) {
+ if ((Global::get().s.echoOption == EchoCancelOptionID::SPEEX_MULTICHANNEL) != bEchoMultiCache) {
do_stop = true;
} else if (edev != qsEchoCache) {
do_stop = true;
@@ -386,7 +383,7 @@ void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
buff.prebuf = -1;
buff.fragsize = iBlockLen;
- bEchoMultiCache = (g.s.echoOption == EchoCancelOptionID::SPEEX_MULTICHANNEL);
+ bEchoMultiCache = (Global::get().s.echoOption == EchoCancelOptionID::SPEEX_MULTICHANNEL);
qsEchoCache = edev;
m_pulseAudio.stream_connect_record(pasSpeaker, qPrintable(edev), &buff, PA_STREAM_ADJUST_LATENCY);
@@ -483,7 +480,7 @@ void PulseAudioSystem::write_stream_callback(pa_stream *s, void *userdata) {
}
const pa_buffer_attr *bufferAttr;
if ((bufferAttr = pa.stream_get_buffer_attr(s))) {
- g.ao->setBufferSize(bufferAttr->maxlength);
+ Global::get().ao->setBufferSize(bufferAttr->maxlength);
}
pas->wakeup();
}
@@ -518,7 +515,7 @@ void PulseAudioSystem::read_callback(pa_stream *s, size_t bytes, void *userdata)
return;
}
- AudioInputPtr ai = g.ai;
+ AudioInputPtr ai = Global::get().ai;
PulseAudioInput *pai = dynamic_cast< PulseAudioInput * >(ai.get());
if (!pai) {
if (length > 0) {
@@ -569,7 +566,7 @@ void PulseAudioSystem::write_callback(pa_stream *s, size_t bytes, void *userdata
PulseAudioSystem *pas = reinterpret_cast< PulseAudioSystem * >(userdata);
Q_ASSERT(s == pas->pasOutput);
- AudioOutputPtr ao = g.ao;
+ AudioOutputPtr ao = Global::get().ao;
PulseAudioOutput *pao = dynamic_cast< PulseAudioOutput * >(ao.get());
if (!pao) {
@@ -643,13 +640,13 @@ void PulseAudioSystem::write_callback(pa_stream *s, size_t bytes, void *userdata
// do we have some mixed output?
if (pao->mix(buffer, samples)) {
// attenuate if instructed to or it's in settings
- pas->bAttenuating = (g.bAttenuateOthers || g.s.bAttenuateOthers);
+ pas->bAttenuating = (Global::get().bAttenuateOthers || Global::get().s.bAttenuateOthers);
} else {
memset(buffer, 0, bytes);
// attenuate if intructed to (self-activated)
- pas->bAttenuating = g.bAttenuateOthers;
+ pas->bAttenuating = Global::get().bAttenuateOthers;
}
// if the attenuation state has changed
@@ -668,16 +665,16 @@ void PulseAudioSystem::volume_sink_input_list_callback(pa_context *c, const pa_s
if (eol == 0) {
// If we're using the default of "enable attenuation on all ouputs" and output from an application is
// loopbacked, both the loopback and the application will be attenuated leading to double attenuation.
- if (!g.s.bOnlyAttenuateSameOutput && pas->iSinkId > -1 && !strcmp(i->driver, "module-loopback.c")) {
+ if (!Global::get().s.bOnlyAttenuateSameOutput && pas->iSinkId > -1 && !strcmp(i->driver, "module-loopback.c")) {
return;
}
// If we're not attenuating different sinks and the input is not on this sink, don't attenuate. Or,
// if the input is a loopback module and connected to Mumble's sink, also ignore it (loopbacks are used to
// connect sinks). An attenuated loopback means an indirect application attenuation.
- if (g.s.bOnlyAttenuateSameOutput && pas->iSinkId > -1) {
+ if (Global::get().s.bOnlyAttenuateSameOutput && pas->iSinkId > -1) {
if (int(i->sink) != pas->iSinkId
|| (int(i->sink) == pas->iSinkId && !strcmp(i->driver, "module-loopback.c")
- && !g.s.bAttenuateLoopbacks)) {
+ && !Global::get().s.bAttenuateLoopbacks)) {
return;
}
}
@@ -691,7 +688,7 @@ void PulseAudioSystem::volume_sink_input_list_callback(pa_context *c, const pa_s
patt.normal_volume = i->volume;
// calculate the attenuated volume
- pa_volume_t adj = static_cast< pa_volume_t >(PA_VOLUME_NORM * g.s.fOtherVolume);
+ pa_volume_t adj = static_cast< pa_volume_t >(PA_VOLUME_NORM * Global::get().s.fOtherVolume);
pa.sw_cvolume_multiply_scalar(&patt.attenuated_volume, &i->volume, adj);
// set it on the sink input
@@ -908,9 +905,9 @@ const QList< audioDevice > PulseAudioInputRegistrar::getDeviceChoices() {
QStringList qlInputDevs = pasys->qhInput.keys();
std::sort(qlInputDevs.begin(), qlInputDevs.end());
- if (qlInputDevs.contains(g.s.qsPulseAudioInput)) {
- qlInputDevs.removeAll(g.s.qsPulseAudioInput);
- qlInputDevs.prepend(g.s.qsPulseAudioInput);
+ if (qlInputDevs.contains(Global::get().s.qsPulseAudioInput)) {
+ qlInputDevs.removeAll(Global::get().s.qsPulseAudioInput);
+ qlInputDevs.prepend(Global::get().s.qsPulseAudioInput);
}
foreach (const QString &dev, qlInputDevs) { qlReturn << audioDevice(pasys->qhInput.value(dev), dev); }
@@ -940,9 +937,9 @@ const QList< audioDevice > PulseAudioOutputRegistrar::getDeviceChoices() {
QStringList qlOutputDevs = pasys->qhOutput.keys();
std::sort(qlOutputDevs.begin(), qlOutputDevs.end());
- if (qlOutputDevs.contains(g.s.qsPulseAudioOutput)) {
- qlOutputDevs.removeAll(g.s.qsPulseAudioOutput);
- qlOutputDevs.prepend(g.s.qsPulseAudioOutput);
+ if (qlOutputDevs.contains(Global::get().s.qsPulseAudioOutput)) {
+ qlOutputDevs.removeAll(Global::get().s.qsPulseAudioOutput);
+ qlOutputDevs.prepend(Global::get().s.qsPulseAudioOutput);
}
foreach (const QString &dev, qlOutputDevs) { qlReturn << audioDevice(pasys->qhOutput.value(dev), dev); }
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-05 18:32:06 +0300