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-rw-r--r--configure.ac41
-rw-r--r--drumgizmo/Makefile.am6
-rw-r--r--drumgizmo/drumgizmoc.cc1
-rw-r--r--drumgizmo/enginefactory.cc9
-rw-r--r--drumgizmo/enginefactory.h5
-rw-r--r--drumgizmo/output/oss.cc151
-rw-r--r--drumgizmo/output/oss.h56
-rw-r--r--man/drumgizmo.16
8 files changed, 274 insertions, 1 deletions
diff --git a/configure.ac b/configure.ac
index 070f380..d437b0f 100644
--- a/configure.ac
+++ b/configure.ac
@@ -288,6 +288,7 @@ AC_ARG_ENABLE([cli],
AS_HELP_STRING([--enable-cli], [Compile the command line interface [default=yes]]),,
[enable_cli="yes"])
+
AS_IF(
[test "x$enable_cli" = "xyes"],
[enable_cli=yes
@@ -422,7 +423,44 @@ AS_IF(
have_output_wavfile=no]
)
- OUTPUT_PLUGINS="dummy alsa wavfile jackaudio"
+ dnl *** oss
+ case $host_os in
+ freebsd*)
+ enable_oss_value=yes
+ ;;
+ *)
+ enable_oss_value=no
+ ;;
+ esac
+ AC_ARG_ENABLE([output_oss],
+ AS_HELP_STRING(
+ [--disable-output-oss],
+ [Disable output oss plugin [enabled by default on FreeBSD, disabled otherwise]]),,
+ [enable_output_oss=$enable_oss_value]
+ )
+
+ AS_IF(
+ [test "x$enable_output_oss" = "xyes"],
+ [AC_MSG_CHECKING(for OSS)
+ AC_COMPILE_IFELSE(
+ [AC_LANG_SOURCE([[
+ #include <sys/soundcard.h>
+ #ifndef AFMT_S32_NE
+ # error no oss
+ #endif
+ ]])],
+ [
+ have_output_oss=yes
+ AC_MSG_RESULT(yes)
+ ],
+ [AC_MSG_FAILURE([no OSS headers found])]
+ )],
+
+ [AC_MSG_RESULT([*** output oss plugin disabled per user request ***])
+ have_output_oss=no]
+ )
+
+ OUTPUT_PLUGINS="dummy alsa wavfile jackaudio oss"
AC_SUBST(OUTPUT_PLUGINS)
dnl
@@ -448,6 +486,7 @@ AM_CONDITIONAL([HAVE_OUTPUT_DUMMY], [test "x$have_output_dummy" = "xyes"])
AM_CONDITIONAL([HAVE_OUTPUT_ALSA], [test "x$have_output_alsa" = "xyes"])
AM_CONDITIONAL([HAVE_OUTPUT_WAVFILE], [test "x$have_output_wavfile" = "xyes"])
AM_CONDITIONAL([HAVE_OUTPUT_JACKAUDIO], [test "x$have_output_jackaudio" = "xyes"])
+AM_CONDITIONAL([HAVE_OUTPUT_OSS], [test "x$have_output_oss" = "xyes"])
dnl ======================
dnl Check for sndfile
diff --git a/drumgizmo/Makefile.am b/drumgizmo/Makefile.am
index ee51e66..6fe9428 100644
--- a/drumgizmo/Makefile.am
+++ b/drumgizmo/Makefile.am
@@ -67,6 +67,12 @@ drumgizmo_SOURCES += output/jackaudio.cc
drumgizmo_CXXFLAGS += -DHAVE_OUTPUT_JACKAUDIO
endif # HAVE_OUTPUT_JACKAUDIO
+if HAVE_OUTPUT_OSS
+drumgizmo_SOURCES += output/oss.cc
+drumgizmo_CXXFLAGS += -DHAVE_OUTPUT_OSS
+endif # HAVE_OUTPUT_OSS
+
+
# Only compile jackclient.cc if at least one of the jack modules are included.
if HAVE_OUTPUT_JACKAUDIO
drumgizmo_SOURCES += jackclient.cc
diff --git a/drumgizmo/drumgizmoc.cc b/drumgizmo/drumgizmoc.cc
index 56b2367..01025e7 100644
--- a/drumgizmo/drumgizmoc.cc
+++ b/drumgizmo/drumgizmoc.cc
@@ -88,6 +88,7 @@ static const char usage_str[] =
" alsa: dev=<device> (default 'default'), frames=<frames> (default "
"32)\n"
" srate=<samplerate> (default 441000)\n"
+ " oss: dev=<device> (default '/dev/dsp'), srate=<samplerate>\n"
" wavfile: file=<filename> (default 'output'), srate=<samplerate> "
"(default 44100)\n"
" jackaudio:\n"
diff --git a/drumgizmo/enginefactory.cc b/drumgizmo/enginefactory.cc
index d013ef9..a1b8a0b 100644
--- a/drumgizmo/enginefactory.cc
+++ b/drumgizmo/enginefactory.cc
@@ -63,6 +63,9 @@ EngineFactory::EngineFactory()
#ifdef HAVE_OUTPUT_JACKAUDIO
output.push_back("jackaudio");
#endif
+#ifdef HAVE_OUTPUT_OSS
+ output.push_back("oss");
+#endif
}
#ifdef USE_JACK
@@ -146,6 +149,12 @@ std::unique_ptr<AudioOutputEngine> EngineFactory::createOutput(const std::string
return std::make_unique<JackAudioOutputEngine>(*jack);
}
#endif
+#ifdef HAVE_OUTPUT_OSS
+ if(name == "oss")
+ {
+ return std::make_unique<OSSOutputEngine>();
+ }
+#endif
// todo: add more engines
diff --git a/drumgizmo/enginefactory.h b/drumgizmo/enginefactory.h
index b2c0428..2a1457e 100644
--- a/drumgizmo/enginefactory.h
+++ b/drumgizmo/enginefactory.h
@@ -69,6 +69,11 @@
#include "output/jackaudio.h"
#endif
+#ifdef HAVE_OUTPUT_OSS
+#include "output/oss.h"
+#endif
+
+
//! Factory for various input- and output engines
class EngineFactory
{
diff --git a/drumgizmo/output/oss.cc b/drumgizmo/output/oss.cc
new file mode 100644
index 0000000..4e02e97
--- /dev/null
+++ b/drumgizmo/output/oss.cc
@@ -0,0 +1,151 @@
+/* -*- Mode: c++ -*- */
+/***************************************************************************
+ * oss.cc
+ *
+ * Tue Apr 11 19:42:45 CET 2017
+ * Copyright 2017 Goran Mekić
+ * meka@tilda.center
+ ****************************************************************************/
+
+/*
+ * This file is part of DrumGizmo.
+ *
+ * DrumGizmo is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * DrumGizmo is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with DrumGizmo; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ */
+#include "oss.h"
+#include <sys/soundcard.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <iostream>
+
+
+OSSOutputEngine::OSSOutputEngine()
+ : dev{"/dev/dsp"}
+ , num_channels{16}
+ , srate{44100}
+ , format{AFMT_S32_NE}
+ , data{}
+{
+ data.clear();
+ data.resize(1024 * num_channels);
+}
+
+bool OSSOutputEngine::init(const Channels& channels)
+{
+ int tmp, mode = O_WRONLY;
+ num_channels = channels.size();
+
+ if((fd = open(dev.data(), mode, 0)) == -1)
+ {
+ std::cerr << dev.data() << ' ' << std::strerror(errno) << std::endl;
+ return false;
+ }
+
+ tmp = format;
+ if(ioctl(fd, SNDCTL_DSP_SETFMT, &tmp) == -1 || tmp != format)
+ {
+ std::cerr << "Setting audio format failed " << std::strerror(errno);
+ std::cerr << std::endl;
+ return false;
+ }
+
+ tmp = num_channels;
+ if(ioctl(fd, SNDCTL_DSP_CHANNELS, &tmp) == -1 || tmp != num_channels)
+ {
+ std::cerr << "Can not set number of channels to " << num_channels;
+ std::cerr << ": " << std::strerror(errno) << std::endl;
+ return false;
+ }
+
+ tmp = srate;
+ if(ioctl(fd, SNDCTL_DSP_SPEED, &tmp) == -1 || tmp != srate)
+ {
+ std::cerr << "Can not set sampling frequency to " << srate << ": ";
+ std::cerr << std::strerror(errno) << std::endl;
+ return false;
+ }
+
+ return true;
+}
+
+void OSSOutputEngine::setParm(const std::string& parm, const std::string& value)
+{
+ if(parm == "dev")
+ {
+ dev = value;
+ }
+ else if(parm == "srate")
+ {
+ try
+ {
+ srate = std::stoi(value);
+ }
+ catch(...)
+ {
+ std::cerr << "[OSSOutputEngine] Invalid samplerate ";
+ std::cerr << value << std::endl;
+ }
+ }
+ else
+ {
+ std::cerr << "[OSSOutputEngine] Unsupported parameter '";
+ std::cerr << parm << std::endl;
+ }
+}
+
+void OSSOutputEngine::run(int ch, sample_t* samples, size_t nsamples)
+{
+ // Convert to int format and
+ // write channel data in interleaved buffer
+ std::int32_t sample;
+ for(size_t i = 0; i < nsamples; ++i)
+ {
+ // Hard clip if needed
+ if(samples[i] > 1)
+ {
+ sample = INT32_MAX;
+ }
+ else if(samples[i] < -1)
+ {
+ sample = -INT32_MAX;
+ }
+ else
+ {
+ sample = samples[i] * INT32_MAX;
+ }
+ data[i * num_channels + ch] = sample;
+ }
+}
+
+void OSSOutputEngine::post(size_t nsamples)
+{
+ auto data_size = data.size() * sizeof(*data.data());
+ auto size_written = write(fd, data.data(), data_size);
+ if(size_written != data_size)
+ {
+ std::cerr << "Audio write: " << std::strerror(errno) << std::endl;
+ }
+}
+
+std::size_t OSSOutputEngine::getBufferSize() const
+{
+ return data.size() / num_channels;
+}
+
+std::size_t OSSOutputEngine::getSamplerate() const
+{
+ return srate;
+}
diff --git a/drumgizmo/output/oss.h b/drumgizmo/output/oss.h
new file mode 100644
index 0000000..d44cd1c
--- /dev/null
+++ b/drumgizmo/output/oss.h
@@ -0,0 +1,56 @@
+/* -*- Mode: c++ -*- */
+/***************************************************************************
+ * oss.h
+ *
+ * Tue Apr 11 19:42:45 CET 2017
+ * Copyright 2017 Goran Mekić
+ * meka@tilda.center
+ ****************************************************************************/
+
+/*
+ * This file is part of DrumGizmo.
+ *
+ * DrumGizmo is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * DrumGizmo is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with DrumGizmo; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ */
+#pragma once
+#include "audiooutputengine.h"
+
+class OSSOutputEngine
+ : public AudioOutputEngine
+{
+public:
+ OSSOutputEngine();
+ ~OSSOutputEngine() {};
+
+ // based on AudioOutputEngine
+ bool init(const Channels& chan) override;
+ void setParm(const std::string& parm, const std::string& value) override;
+ bool start() override { return true; };
+ void stop() override {};
+ void pre(size_t nsamples) override { data.resize(nsamples * num_channels); };
+ void run(int ch, sample_t* samples, size_t nsamples) override;
+ void post(size_t nsamples) override;
+ bool isFreewheeling() const override { return false; };
+ std::size_t getBufferSize() const override;
+ std::size_t getSamplerate() const override;
+
+private:
+ std::string dev;
+ int fd;
+ std::size_t num_channels;
+ unsigned int srate;
+ unsigned int format;
+ std::vector<std::int32_t> data;
+};
diff --git a/man/drumgizmo.1 b/man/drumgizmo.1
index f3c1b58..4aab17c 100644
--- a/man/drumgizmo.1
+++ b/man/drumgizmo.1
@@ -75,6 +75,12 @@ file=<filename> (default 'output')
.P
srate=<samplerate> (default 44100)
+\fBoss:\fR
+.P
+dev=<device> (default /dev/dsp)
+.P
+srate=<samplerate> (default 44100)
+
\fBjackaudio:\fR
\fBdummy:\fR
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-01 08:27:53 +0000