Add OSS output support

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