output: rename plugin source files

1 files changed, 684 insertions, 0 deletions

diff --git a/src/output/oss_output_plugin.c b/src/output/oss_output_plugin.c
new file mode 100644
index 000000000..d7df594d3
--- /dev/null
+++ b/src/output/oss_output_plugin.c
@@ -0,0 +1,684 @@
+/*
+ * Copyright (C) 2003-2011 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 "output_api.h"
+#include "mixer_list.h"
+#include "fd_util.h"
+
+#include <glib.h>
+
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <assert.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "oss"
+
+#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
+
+struct oss_data {
+	int fd;
+	const char *device;
+
+	/**
+	 * The current input audio format.  This is needed to reopen
+	 * the device after cancel().
+	 */
+	struct audio_format audio_format;
+};
+
+/**
+ * The quark used for GError.domain.
+ */
+static inline GQuark
+oss_output_quark(void)
+{
+	return g_quark_from_static_string("oss_output");
+}
+
+static struct oss_data *
+oss_data_new(void)
+{
+	struct oss_data *ret = g_new(struct oss_data, 1);
+
+	ret->device = NULL;
+	ret->fd = -1;
+
+	return ret;
+}
+
+static void
+oss_data_free(struct oss_data *od)
+{
+	g_free(od);
+}
+
+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 = G_N_ELEMENTS(default_devices); --i >= 0; ) {
+		fd = open_cloexec(default_devices[i], O_WRONLY, 0);
+
+		if (fd >= 0) {
+			close(fd);
+			return true;
+		}
+		g_warning("Error opening OSS device \"%s\": %s\n",
+			  default_devices[i], strerror(errno));
+	}
+
+	return false;
+}
+
+static void *
+oss_open_default(GError **error)
+{
+	int i;
+	int err[G_N_ELEMENTS(default_devices)];
+	enum oss_stat ret[G_N_ELEMENTS(default_devices)];
+
+	for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
+		ret[i] = oss_stat_device(default_devices[i], &err[i]);
+		if (ret[i] == OSS_STAT_NO_ERROR) {
+			struct oss_data *od = oss_data_new();
+			od->device = default_devices[i];
+			return od;
+		}
+	}
+
+	for (i = G_N_ELEMENTS(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:
+			g_warning("%s not found\n", dev);
+			break;
+		case OSS_STAT_NOT_CHAR_DEV:
+			g_warning("%s is not a character device\n", dev);
+			break;
+		case OSS_STAT_NO_PERMS:
+			g_warning("%s: permission denied\n", dev);
+			break;
+		case OSS_STAT_OTHER:
+			g_warning("Error accessing %s: %s\n",
+				  dev, strerror(err[i]));
+		}
+	}
+
+	g_set_error(error, oss_output_quark(), 0,
+		    "error trying to open default OSS device");
+	return NULL;
+}
+
+static void *
+oss_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
+		const struct config_param *param,
+		GError **error)
+{
+	const char *device = config_get_block_string(param, "device", NULL);
+	if (device != NULL) {
+		struct oss_data *od = oss_data_new();
+		od->device = device;
+		return od;
+	}
+
+	return oss_open_default(error);
+}
+
+static void
+oss_output_finish(void *data)
+{
+	struct oss_data *od = data;
+
+	oss_data_free(od);
+}
+
+static void
+oss_close(struct oss_data *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 a GError.
+ */
+static enum oss_setup_result
+oss_try_ioctl_r(int fd, unsigned long request, int *value_r,
+		const char *msg, GError **error_r)
+{
+	assert(fd >= 0);
+	assert(value_r != NULL);
+	assert(msg != NULL);
+	assert(error_r == NULL || *error_r == NULL);
+
+	int ret = ioctl(fd, request, value_r);
+	if (ret >= 0)
+		return SUCCESS;
+
+	if (errno == EINVAL)
+		return UNSUPPORTED;
+
+	g_set_error(error_r, oss_output_quark(), errno,
+		    "%s: %s", msg, g_strerror(errno));
+	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 a GError.
+ */
+static enum oss_setup_result
+oss_try_ioctl(int fd, unsigned long request, int value,
+	      const char *msg, GError **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, struct audio_format *audio_format, GError **error_r)
+{
+	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_r);
+	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_r);
+		switch (result) {
+		case SUCCESS:
+			if (!audio_valid_channel_count(channels))
+			    break;
+
+			audio_format->channels = channels;
+			return true;
+
+		case ERROR:
+			return false;
+
+		case UNSUPPORTED:
+			break;
+		}
+	}
+
+	g_set_error(error_r, oss_output_quark(), EINVAL, "%s", 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, struct audio_format *audio_format,
+		      GError **error_r)
+{
+	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_r);
+	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_r);
+		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;
+		}
+	}
+
+	g_set_error(error_r, oss_output_quark(), EINVAL, "%s", 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(enum sample_format format)
+{
+	switch (format) {
+	case SAMPLE_FORMAT_UNDEFINED:
+		return AFMT_QUERY;
+
+	case SAMPLE_FORMAT_S8:
+		return AFMT_S8;
+
+	case SAMPLE_FORMAT_S16:
+		return AFMT_S16_NE;
+
+	case SAMPLE_FORMAT_S24:
+#ifdef AFMT_S24_PACKED
+		return AFMT_S24_PACKED;
+#else
+		return AFMT_QUERY;
+#endif
+
+	case SAMPLE_FORMAT_S24_P32:
+#ifdef AFMT_S24_NE
+		return AFMT_S24_NE;
+#else
+		return AFMT_QUERY;
+#endif
+
+	case SAMPLE_FORMAT_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
+ * SAMPLE_FORMAT_UNDEFINED if there is no direct counterpart.
+ */
+static enum sample_format
+sample_format_from_oss(int format)
+{
+	switch (format) {
+	case AFMT_S8:
+		return SAMPLE_FORMAT_S8;
+
+	case AFMT_S16_NE:
+		return SAMPLE_FORMAT_S16;
+
+#ifdef AFMT_S24_PACKED
+	case AFMT_S24_PACKED:
+		return SAMPLE_FORMAT_S24;
+#endif
+
+#ifdef AFMT_S24_NE
+	case AFMT_S24_NE:
+		return SAMPLE_FORMAT_S24_P32;
+#endif
+
+#ifdef AFMT_S32_NE
+	case AFMT_S32_NE:
+		return SAMPLE_FORMAT_S32;
+#endif
+
+	default:
+		return SAMPLE_FORMAT_UNDEFINED;
+	}
+}
+
+/**
+ * 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, struct audio_format *audio_format,
+			GError **error_r)
+{
+	const char *const msg = "Failed to set sample format";
+	int oss_format = sample_format_to_oss(audio_format->format);
+	enum oss_setup_result result = oss_format != AFMT_QUERY
+		? oss_try_ioctl_r(fd, SNDCTL_DSP_SAMPLESIZE,
+				  &oss_format, msg, error_r)
+		: UNSUPPORTED;
+	enum sample_format mpd_format;
+	switch (result) {
+	case SUCCESS:
+		mpd_format = sample_format_from_oss(oss_format);
+		if (mpd_format == SAMPLE_FORMAT_UNDEFINED)
+			break;
+
+		audio_format->format = mpd_format;
+
+#ifdef AFMT_S24_PACKED
+		if (oss_format == AFMT_S24_PACKED)
+			audio_format->reverse_endian =
+				G_BYTE_ORDER != G_LITTLE_ENDIAN;
+#endif
+		return true;
+
+	case ERROR:
+		return false;
+
+	case UNSUPPORTED:
+		break;
+	}
+
+	/* the requested sample format is not available - probe for
+	   other formats supported by MPD */
+
+	static const enum sample_format sample_formats[] = {
+		SAMPLE_FORMAT_S24_P32,
+		SAMPLE_FORMAT_S32,
+		SAMPLE_FORMAT_S24,
+		SAMPLE_FORMAT_S16,
+		SAMPLE_FORMAT_S8,
+		SAMPLE_FORMAT_UNDEFINED /* sentinel */
+	};
+
+	for (unsigned i = 0; sample_formats[i] != SAMPLE_FORMAT_UNDEFINED; ++i) {
+		mpd_format = sample_formats[i];
+		if (mpd_format == audio_format->format)
+			/* don't try that again */
+			continue;
+
+		oss_format = sample_format_to_oss(mpd_format);
+		if (oss_format == AFMT_QUERY)
+			/* not supported by this OSS version */
+			continue;
+
+		result = oss_try_ioctl_r(fd, SNDCTL_DSP_SAMPLESIZE,
+					 &oss_format, msg, error_r);
+		switch (result) {
+		case SUCCESS:
+			mpd_format = sample_format_from_oss(oss_format);
+			if (mpd_format == SAMPLE_FORMAT_UNDEFINED)
+				break;
+
+			audio_format->format = mpd_format;
+
+#ifdef AFMT_S24_PACKED
+			if (oss_format == AFMT_S24_PACKED)
+				audio_format->reverse_endian =
+					G_BYTE_ORDER != G_LITTLE_ENDIAN;
+#endif
+			return true;
+
+		case ERROR:
+			return false;
+
+		case UNSUPPORTED:
+			break;
+		}
+	}
+
+	g_set_error(error_r, oss_output_quark(), EINVAL, "%s", msg);
+	return false;
+}
+
+/**
+ * Sets up the OSS device which was opened before.
+ */
+static bool
+oss_setup(struct oss_data *od, struct audio_format *audio_format,
+	  GError **error_r)
+{
+	return oss_setup_channels(od->fd, audio_format, error_r) &&
+		oss_setup_sample_rate(od->fd, audio_format, error_r) &&
+		oss_setup_sample_format(od->fd, audio_format, error_r);
+}
+
+/**
+ * Reopen the device with the saved audio_format, without any probing.
+ */
+static bool
+oss_reopen(struct oss_data *od, GError **error_r)
+{
+	assert(od->fd < 0);
+
+	od->fd = open_cloexec(od->device, O_WRONLY, 0);
+	if (od->fd < 0) {
+		g_set_error(error_r, oss_output_quark(), errno,
+			    "Error opening OSS device \"%s\": %s",
+			    od->device, strerror(errno));
+		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_r);
+	if (result != SUCCESS) {
+		oss_close(od);
+		if (result == UNSUPPORTED)
+			g_set_error(error_r, oss_output_quark(), EINVAL,
+				    "%s", 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_r);
+	if (result != SUCCESS) {
+		oss_close(od);
+		if (result == UNSUPPORTED)
+			g_set_error(error_r, oss_output_quark(), EINVAL,
+				    "%s", msg2);
+		return false;
+	}
+
+	const char *const msg3 = "Failed to set sample format";
+	assert(sample_format_to_oss(od->audio_format.format) != AFMT_QUERY);
+	result = oss_try_ioctl(od->fd, SNDCTL_DSP_SAMPLESIZE,
+			       sample_format_to_oss(od->audio_format.format),
+			       msg3, error_r);
+	if (result != SUCCESS) {
+		oss_close(od);
+		if (result == UNSUPPORTED)
+			g_set_error(error_r, oss_output_quark(), EINVAL,
+				    "%s", msg3);
+		return false;
+	}
+
+	return true;
+}
+
+static bool
+oss_output_open(void *data, struct audio_format *audio_format, GError **error)
+{
+	struct oss_data *od = data;
+
+	od->fd = open_cloexec(od->device, O_WRONLY, 0);
+	if (od->fd < 0) {
+		g_set_error(error, oss_output_quark(), errno,
+			    "Error opening OSS device \"%s\": %s",
+			    od->device, strerror(errno));
+		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(void *data)
+{
+	struct oss_data *od = data;
+
+	oss_close(od);
+}
+
+static void
+oss_output_cancel(void *data)
+{
+	struct oss_data *od = data;
+
+	if (od->fd >= 0) {
+		ioctl(od->fd, SNDCTL_DSP_RESET, 0);
+		oss_close(od);
+	}
+}
+
+static size_t
+oss_output_play(void *data, const void *chunk, size_t size, GError **error)
+{
+	struct oss_data *od = data;
+	ssize_t ret;
+
+	/* reopen the device since it was closed by dropBufferedAudio */
+	if (od->fd < 0 && !oss_reopen(od, error))
+		return 0;
+
+	while (true) {
+		ret = write(od->fd, chunk, size);
+		if (ret > 0)
+			return (size_t)ret;
+
+		if (ret < 0 && errno != EINTR) {
+			g_set_error(error, oss_output_quark(), errno,
+				    "Write error on %s: %s",
+				    od->device, strerror(errno));
+			return 0;
+		}
+	}
+}
+
+const struct audio_output_plugin oss_output_plugin = {
+	.name = "oss",
+	.test_default_device = oss_output_test_default_device,
+	.init = oss_output_init,
+	.finish = oss_output_finish,
+	.open = oss_output_open,
+	.close = oss_output_close,
+	.play = oss_output_play,
+	.cancel = oss_output_cancel,
+
+	.mixer_plugin = &oss_mixer_plugin,
+};