output/*: move to output/plugins/

src/output/plugins/OssOutputPlugin.cxx

@@ -0,0 +1,776 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "OssOutputPlugin.hxx"
+#include "../OutputAPI.hxx"
+#include "MixerList.hxx"
+#include "system/fd_util.h"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "util/Macros.hxx"
+#include "system/ByteOrder.hxx"
+#include "Log.hxx"
+
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <assert.h>
+
+#if defined(__OpenBSD__) || defined(__NetBSD__)
+# include <soundcard.h>
+#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+# include <sys/soundcard.h>
+#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+
+/* We got bug reports from FreeBSD users who said that the two 24 bit
+   formats generate white noise on FreeBSD, but 32 bit works.  This is
+   a workaround until we know what exactly is expected by the kernel
+   audio drivers. */
+#ifndef __linux__
+#undef AFMT_S24_PACKED
+#undef AFMT_S24_NE
+#endif
+
+#ifdef AFMT_S24_PACKED
+#include "pcm/PcmExport.hxx"
+#include "util/Manual.hxx"
+#endif
+
+struct OssOutput {
+	struct audio_output base;
+
+#ifdef AFMT_S24_PACKED
+	Manual<PcmExport> pcm_export;
+#endif
+
+	int fd;
+	const char *device;
+
+	/**
+	 * The current input audio format.  This is needed to reopen
+	 * the device after cancel().
+	 */
+	AudioFormat audio_format;
+
+	/**
+	 * The current OSS audio format.  This is needed to reopen the
+	 * device after cancel().
+	 */
+	int oss_format;
+
+	OssOutput():fd(-1), device(nullptr) {}
+
+	bool Initialize(const config_param &param, Error &error_r) {
+		return ao_base_init(&base, &oss_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
+};
+
+static constexpr Domain oss_output_domain("oss_output");
+
+enum oss_stat {
+	OSS_STAT_NO_ERROR = 0,
+	OSS_STAT_NOT_CHAR_DEV = -1,
+	OSS_STAT_NO_PERMS = -2,
+	OSS_STAT_DOESN_T_EXIST = -3,
+	OSS_STAT_OTHER = -4,
+};
+
+static enum oss_stat
+oss_stat_device(const char *device, int *errno_r)
+{
+	struct stat st;
+
+	if (0 == stat(device, &st)) {
+		if (!S_ISCHR(st.st_mode)) {
+			return OSS_STAT_NOT_CHAR_DEV;
+		}
+	} else {
+		*errno_r = errno;
+
+		switch (errno) {
+		case ENOENT:
+		case ENOTDIR:
+			return OSS_STAT_DOESN_T_EXIST;
+		case EACCES:
+			return OSS_STAT_NO_PERMS;
+		default:
+			return OSS_STAT_OTHER;
+		}
+	}
+
+	return OSS_STAT_NO_ERROR;
+}
+
+static const char *default_devices[] = { "/dev/sound/dsp", "/dev/dsp" };
+
+static bool
+oss_output_test_default_device(void)
+{
+	int fd, i;
+
+	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+		fd = open_cloexec(default_devices[i], O_WRONLY, 0);
+
+		if (fd >= 0) {
+			close(fd);
+			return true;
+		}
+
+		FormatErrno(oss_output_domain,
+			    "Error opening OSS device \"%s\"",
+			    default_devices[i]);
+	}
+
+	return false;
+}
+
+static struct audio_output *
+oss_open_default(Error &error)
+{
+	int err[ARRAY_SIZE(default_devices)];
+	enum oss_stat ret[ARRAY_SIZE(default_devices)];
+
+	const config_param empty;
+	for (int i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+		ret[i] = oss_stat_device(default_devices[i], &err[i]);
+		if (ret[i] == OSS_STAT_NO_ERROR) {
+			OssOutput *od = new OssOutput();
+			if (!od->Initialize(empty, error)) {
+				delete od;
+				return NULL;
+			}
+
+			od->device = default_devices[i];
+			return &od->base;
+		}
+	}
+
+	for (int i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+		const char *dev = default_devices[i];
+		switch(ret[i]) {
+		case OSS_STAT_NO_ERROR:
+			/* never reached */
+			break;
+		case OSS_STAT_DOESN_T_EXIST:
+			FormatWarning(oss_output_domain,
+				      "%s not found", dev);
+			break;
+		case OSS_STAT_NOT_CHAR_DEV:
+			FormatWarning(oss_output_domain,
+				      "%s is not a character device", dev);
+			break;
+		case OSS_STAT_NO_PERMS:
+			FormatWarning(oss_output_domain,
+				      "%s: permission denied", dev);
+			break;
+		case OSS_STAT_OTHER:
+			FormatErrno(oss_output_domain, err[i],
+				    "Error accessing %s", dev);
+		}
+	}
+
+	error.Set(oss_output_domain,
+		  "error trying to open default OSS device");
+	return NULL;
+}
+
+static struct audio_output *
+oss_output_init(const config_param &param, Error &error)
+{
+	const char *device = param.GetBlockValue("device");
+	if (device != NULL) {
+		OssOutput *od = new OssOutput();
+		if (!od->Initialize(param, error)) {
+			delete od;
+			return NULL;
+		}
+
+		od->device = device;
+		return &od->base;
+	}
+
+	return oss_open_default(error);
+}
+
+static void
+oss_output_finish(struct audio_output *ao)
+{
+	OssOutput *od = (OssOutput *)ao;
+
+	ao_base_finish(&od->base);
+	delete od;
+}
+
+#ifdef AFMT_S24_PACKED
+
+static bool
+oss_output_enable(struct audio_output *ao, gcc_unused Error &error)
+{
+	OssOutput *od = (OssOutput *)ao;
+
+	od->pcm_export.Construct();
+	return true;
+}
+
+static void
+oss_output_disable(struct audio_output *ao)
+{
+	OssOutput *od = (OssOutput *)ao;
+
+	od->pcm_export.Destruct();
+}
+
+#endif
+
+static void
+oss_close(OssOutput *od)
+{
+	if (od->fd >= 0)
+		close(od->fd);
+	od->fd = -1;
+}
+
+/**
+ * A tri-state type for oss_try_ioctl().
+ */
+enum oss_setup_result {
+	SUCCESS,
+	ERROR,
+	UNSUPPORTED,
+};
+
+/**
+ * Invoke an ioctl on the OSS file descriptor.  On success, SUCCESS is
+ * returned.  If the parameter is not supported, UNSUPPORTED is
+ * returned.  Any other failure returns ERROR and allocates an #Error.
+ */
+static enum oss_setup_result
+oss_try_ioctl_r(int fd, unsigned long request, int *value_r,
+		const char *msg, Error &error)
+{
+	assert(fd >= 0);
+	assert(value_r != NULL);
+	assert(msg != NULL);
+	assert(!error.IsDefined());
+
+	int ret = ioctl(fd, request, value_r);
+	if (ret >= 0)
+		return SUCCESS;
+
+	if (errno == EINVAL)
+		return UNSUPPORTED;
+
+	error.SetErrno(msg);
+	return ERROR;
+}
+
+/**
+ * Invoke an ioctl on the OSS file descriptor.  On success, SUCCESS is
+ * returned.  If the parameter is not supported, UNSUPPORTED is
+ * returned.  Any other failure returns ERROR and allocates an #Error.
+ */
+static enum oss_setup_result
+oss_try_ioctl(int fd, unsigned long request, int value,
+	      const char *msg, Error &error_r)
+{
+	return oss_try_ioctl_r(fd, request, &value, msg, error_r);
+}
+
+/**
+ * Set up the channel number, and attempts to find alternatives if the
+ * specified number is not supported.
+ */
+static bool
+oss_setup_channels(int fd, AudioFormat &audio_format, Error &error)
+{
+	const char *const msg = "Failed to set channel count";
+	int channels = audio_format.channels;
+	enum oss_setup_result result =
+		oss_try_ioctl_r(fd, SNDCTL_DSP_CHANNELS, &channels, msg, error);
+	switch (result) {
+	case SUCCESS:
+		if (!audio_valid_channel_count(channels))
+		    break;
+
+		audio_format.channels = channels;
+		return true;
+
+	case ERROR:
+		return false;
+
+	case UNSUPPORTED:
+		break;
+	}
+
+	for (unsigned i = 1; i < 2; ++i) {
+		if (i == audio_format.channels)
+			/* don't try that again */
+			continue;
+
+		channels = i;
+		result = oss_try_ioctl_r(fd, SNDCTL_DSP_CHANNELS, &channels,
+					 msg, error);
+		switch (result) {
+		case SUCCESS:
+			if (!audio_valid_channel_count(channels))
+			    break;
+
+			audio_format.channels = channels;
+			return true;
+
+		case ERROR:
+			return false;
+
+		case UNSUPPORTED:
+			break;
+		}
+	}
+
+	error.Set(oss_output_domain, msg);
+	return false;
+}
+
+/**
+ * Set up the sample rate, and attempts to find alternatives if the
+ * specified sample rate is not supported.
+ */
+static bool
+oss_setup_sample_rate(int fd, AudioFormat &audio_format,
+		      Error &error)
+{
+	const char *const msg = "Failed to set sample rate";
+	int sample_rate = audio_format.sample_rate;
+	enum oss_setup_result result =
+		oss_try_ioctl_r(fd, SNDCTL_DSP_SPEED, &sample_rate,
+				msg, error);
+	switch (result) {
+	case SUCCESS:
+		if (!audio_valid_sample_rate(sample_rate))
+			break;
+
+		audio_format.sample_rate = sample_rate;
+		return true;
+
+	case ERROR:
+		return false;
+
+	case UNSUPPORTED:
+		break;
+	}
+
+	static const int sample_rates[] = { 48000, 44100, 0 };
+	for (unsigned i = 0; sample_rates[i] != 0; ++i) {
+		sample_rate = sample_rates[i];
+		if (sample_rate == (int)audio_format.sample_rate)
+			continue;
+
+		result = oss_try_ioctl_r(fd, SNDCTL_DSP_SPEED, &sample_rate,
+					 msg, error);
+		switch (result) {
+		case SUCCESS:
+			if (!audio_valid_sample_rate(sample_rate))
+				break;
+
+			audio_format.sample_rate = sample_rate;
+			return true;
+
+		case ERROR:
+			return false;
+
+		case UNSUPPORTED:
+			break;
+		}
+	}
+
+	error.Set(oss_output_domain, msg);
+	return false;
+}
+
+/**
+ * Convert a MPD sample format to its OSS counterpart.  Returns
+ * AFMT_QUERY if there is no direct counterpart.
+ */
+static int
+sample_format_to_oss(SampleFormat format)
+{
+	switch (format) {
+	case SampleFormat::UNDEFINED:
+	case SampleFormat::FLOAT:
+	case SampleFormat::DSD:
+		return AFMT_QUERY;
+
+	case SampleFormat::S8:
+		return AFMT_S8;
+
+	case SampleFormat::S16:
+		return AFMT_S16_NE;
+
+	case SampleFormat::S24_P32:
+#ifdef AFMT_S24_NE
+		return AFMT_S24_NE;
+#else
+		return AFMT_QUERY;
+#endif
+
+	case SampleFormat::S32:
+#ifdef AFMT_S32_NE
+		return AFMT_S32_NE;
+#else
+		return AFMT_QUERY;
+#endif
+	}
+
+	return AFMT_QUERY;
+}
+
+/**
+ * Convert an OSS sample format to its MPD counterpart.  Returns
+ * SampleFormat::UNDEFINED if there is no direct counterpart.
+ */
+static SampleFormat
+sample_format_from_oss(int format)
+{
+	switch (format) {
+	case AFMT_S8:
+		return SampleFormat::S8;
+
+	case AFMT_S16_NE:
+		return SampleFormat::S16;
+
+#ifdef AFMT_S24_PACKED
+	case AFMT_S24_PACKED:
+		return SampleFormat::S24_P32;
+#endif
+
+#ifdef AFMT_S24_NE
+	case AFMT_S24_NE:
+		return SampleFormat::S24_P32;
+#endif
+
+#ifdef AFMT_S32_NE
+	case AFMT_S32_NE:
+		return SampleFormat::S32;
+#endif
+
+	default:
+		return SampleFormat::UNDEFINED;
+	}
+}
+
+/**
+ * Probe one sample format.
+ *
+ * @return the selected sample format or SampleFormat::UNDEFINED on
+ * error
+ */
+static enum oss_setup_result
+oss_probe_sample_format(int fd, SampleFormat sample_format,
+			SampleFormat *sample_format_r,
+			int *oss_format_r,
+#ifdef AFMT_S24_PACKED
+			PcmExport &pcm_export,
+#endif
+			Error &error)
+{
+	int oss_format = sample_format_to_oss(sample_format);
+	if (oss_format == AFMT_QUERY)
+		return UNSUPPORTED;
+
+	enum oss_setup_result result =
+		oss_try_ioctl_r(fd, SNDCTL_DSP_SAMPLESIZE,
+				&oss_format,
+				"Failed to set sample format", error);
+
+#ifdef AFMT_S24_PACKED
+	if (result == UNSUPPORTED && sample_format == SampleFormat::S24_P32) {
+		/* if the driver doesn't support padded 24 bit, try
+		   packed 24 bit */
+		oss_format = AFMT_S24_PACKED;
+		result = oss_try_ioctl_r(fd, SNDCTL_DSP_SAMPLESIZE,
+					 &oss_format,
+					 "Failed to set sample format", error);
+	}
+#endif
+
+	if (result != SUCCESS)
+		return result;
+
+	sample_format = sample_format_from_oss(oss_format);
+	if (sample_format == SampleFormat::UNDEFINED)
+		return UNSUPPORTED;
+
+	*sample_format_r = sample_format;
+	*oss_format_r = oss_format;
+
+#ifdef AFMT_S24_PACKED
+	pcm_export.Open(sample_format, 0, false, false,
+			oss_format == AFMT_S24_PACKED,
+			oss_format == AFMT_S24_PACKED &&
+			!IsLittleEndian());
+#endif
+
+	return SUCCESS;
+}
+
+/**
+ * Set up the sample format, and attempts to find alternatives if the
+ * specified format is not supported.
+ */
+static bool
+oss_setup_sample_format(int fd, AudioFormat &audio_format,
+			int *oss_format_r,
+#ifdef AFMT_S24_PACKED
+			PcmExport &pcm_export,
+#endif
+			Error &error)
+{
+	SampleFormat mpd_format;
+	enum oss_setup_result result =
+		oss_probe_sample_format(fd, audio_format.format,
+					&mpd_format, oss_format_r,
+#ifdef AFMT_S24_PACKED
+					pcm_export,
+#endif
+					error);
+	switch (result) {
+	case SUCCESS:
+		audio_format.format = mpd_format;
+		return true;
+
+	case ERROR:
+		return false;
+
+	case UNSUPPORTED:
+		break;
+	}
+
+	if (result != UNSUPPORTED)
+		return result == SUCCESS;
+
+	/* the requested sample format is not available - probe for
+	   other formats supported by MPD */
+
+	static const SampleFormat sample_formats[] = {
+		SampleFormat::S24_P32,
+		SampleFormat::S32,
+		SampleFormat::S16,
+		SampleFormat::S8,
+		SampleFormat::UNDEFINED /* sentinel */
+	};
+
+	for (unsigned i = 0; sample_formats[i] != SampleFormat::UNDEFINED; ++i) {
+		mpd_format = sample_formats[i];
+		if (mpd_format == audio_format.format)
+			/* don't try that again */
+			continue;
+
+		result = oss_probe_sample_format(fd, mpd_format,
+						 &mpd_format, oss_format_r,
+#ifdef AFMT_S24_PACKED
+						 pcm_export,
+#endif
+						 error);
+		switch (result) {
+		case SUCCESS:
+			audio_format.format = mpd_format;
+			return true;
+
+		case ERROR:
+			return false;
+
+		case UNSUPPORTED:
+			break;
+		}
+	}
+
+	error.Set(oss_output_domain, "Failed to set sample format");
+	return false;
+}
+
+/**
+ * Sets up the OSS device which was opened before.
+ */
+static bool
+oss_setup(OssOutput *od, AudioFormat &audio_format,
+	  Error &error)
+{
+	return oss_setup_channels(od->fd, audio_format, error) &&
+		oss_setup_sample_rate(od->fd, audio_format, error) &&
+		oss_setup_sample_format(od->fd, audio_format, &od->oss_format,
+#ifdef AFMT_S24_PACKED
+					od->pcm_export,
+#endif
+					error);
+}
+
+/**
+ * Reopen the device with the saved audio_format, without any probing.
+ */
+static bool
+oss_reopen(OssOutput *od, Error &error)
+{
+	assert(od->fd < 0);
+
+	od->fd = open_cloexec(od->device, O_WRONLY, 0);
+	if (od->fd < 0) {
+		error.FormatErrno("Error opening OSS device \"%s\"",
+				  od->device);
+		return false;
+	}
+
+	enum oss_setup_result result;
+
+	const char *const msg1 = "Failed to set channel count";
+	result = oss_try_ioctl(od->fd, SNDCTL_DSP_CHANNELS,
+			       od->audio_format.channels, msg1, error);
+	if (result != SUCCESS) {
+		oss_close(od);
+		if (result == UNSUPPORTED)
+			error.Set(oss_output_domain, msg1);
+		return false;
+	}
+
+	const char *const msg2 = "Failed to set sample rate";
+	result = oss_try_ioctl(od->fd, SNDCTL_DSP_SPEED,
+			       od->audio_format.sample_rate, msg2, error);
+	if (result != SUCCESS) {
+		oss_close(od);
+		if (result == UNSUPPORTED)
+			error.Set(oss_output_domain, msg2);
+		return false;
+	}
+
+	const char *const msg3 = "Failed to set sample format";
+	result = oss_try_ioctl(od->fd, SNDCTL_DSP_SAMPLESIZE,
+			       od->oss_format,
+			       msg3, error);
+	if (result != SUCCESS) {
+		oss_close(od);
+		if (result == UNSUPPORTED)
+			error.Set(oss_output_domain, msg3);
+		return false;
+	}
+
+	return true;
+}
+
+static bool
+oss_output_open(struct audio_output *ao, AudioFormat &audio_format,
+		Error &error)
+{
+	OssOutput *od = (OssOutput *)ao;
+
+	od->fd = open_cloexec(od->device, O_WRONLY, 0);
+	if (od->fd < 0) {
+		error.FormatErrno("Error opening OSS device \"%s\"",
+				  od->device);
+		return false;
+	}
+
+	if (!oss_setup(od, audio_format, error)) {
+		oss_close(od);
+		return false;
+	}
+
+	od->audio_format = audio_format;
+	return true;
+}
+
+static void
+oss_output_close(struct audio_output *ao)
+{
+	OssOutput *od = (OssOutput *)ao;
+
+	oss_close(od);
+}
+
+static void
+oss_output_cancel(struct audio_output *ao)
+{
+	OssOutput *od = (OssOutput *)ao;
+
+	if (od->fd >= 0) {
+		ioctl(od->fd, SNDCTL_DSP_RESET, 0);
+		oss_close(od);
+	}
+}
+
+static size_t
+oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
+		Error &error)
+{
+	OssOutput *od = (OssOutput *)ao;
+	ssize_t ret;
+
+	/* reopen the device since it was closed by dropBufferedAudio */
+	if (od->fd < 0 && !oss_reopen(od, error))
+		return 0;
+
+#ifdef AFMT_S24_PACKED
+	chunk = od->pcm_export->Export(chunk, size, size);
+#endif
+
+	while (true) {
+		ret = write(od->fd, chunk, size);
+		if (ret > 0) {
+#ifdef AFMT_S24_PACKED
+			ret = od->pcm_export->CalcSourceSize(ret);
+#endif
+			return ret;
+		}
+
+		if (ret < 0 && errno != EINTR) {
+			error.FormatErrno("Write error on %s", od->device);
+			return 0;
+		}
+	}
+}
+
+const struct audio_output_plugin oss_output_plugin = {
+	"oss",
+	oss_output_test_default_device,
+	oss_output_init,
+	oss_output_finish,
+#ifdef AFMT_S24_PACKED
+	oss_output_enable,
+	oss_output_disable,
+#else
+	nullptr,
+	nullptr,
+#endif
+	oss_output_open,
+	oss_output_close,
+	nullptr,
+	nullptr,
+	oss_output_play,
+	nullptr,
+	oss_output_cancel,
+	nullptr,
+
+	&oss_mixer_plugin,
+};