annotate src/OSS/audio.c @ 984:df7b09989aee trunk

[svn] - We got a new plugin, captain! - FileWriter is the ultimate plugin for dumping audio to files. It should be the successor of Disk Writer and Out-Lame, as it supports the same output formats as those (WAVE and MP3). The main advantage of having only one file dumping plugin for many formats is that not every plugin has to think about file handling (where to write files to, how to call them etc.) that much anymore. - FileWriter is also very extensible - adding new output formats should be very easy.

/*  BMP - Cross-platform multimedia player
 *  Copyright (C) 2003-2004  BMP development team.
 *
 *  Based on XMMS:
 *  Copyright (C) 1998-2003  XMMS development team.
 *
 *  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.
 */

extern void close_mixer_device();

#include <glib.h>
#include <audacious/util.h>
#include <string.h>

#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <sys/ioctl.h>
#include <sys/time.h>

#include "OSS.h"


#define NFRAGS		32

static gint fd = 0;
static char *buffer;
static gboolean going, prebuffer, paused, unpause, do_pause, remove_prebuffer;
static gint device_buffer_used, buffer_size, prebuffer_size, blk_size;
static gint rd_index = 0, wr_index = 0;
static gint output_time_offset = 0;
static guint64 written = 0, output_bytes = 0;
static gint flush;
static gint fragsize, device_buffer_size;
static gchar *device_name;
static GThread *buffer_thread;
static gboolean realtime, select_works;

static int (*oss_convert_func) (void **data, int length);
static int (*oss_stereo_convert_func) (void **data, int length, int fmt);

struct format_info {
    union {
        AFormat xmms;
        int oss;
    } format;
    int frequency;
    int channels;
    int bps;
};


/*
 * The format of the data from the input plugin
 * This will never change during a song. 
 */
struct format_info input;

/*
 * The format we get from the effect plugin.
 * This will be different from input if the effect plugin does
 * some kind of format conversion.
 */
struct format_info effect;

/*
 * The format of the data we actually send to the soundcard.
 * This might be different from effect if we need to resample or do
 * some other format conversion.
 */
struct format_info output;


static void
oss_calc_device_buffer_used(void)
{
    audio_buf_info buf_info;
    if (paused)
        device_buffer_used = 0;
    else if (!ioctl(fd, SNDCTL_DSP_GETOSPACE, &buf_info))
        device_buffer_used =
            (buf_info.fragstotal * buf_info.fragsize) - buf_info.bytes;
}


static gint oss_downsample(gpointer ob, guint length, guint speed,
                           guint espeed);

static int
oss_calc_bitrate(int oss_fmt, int rate, int channels)
{
    int bitrate = rate * channels;

    if (oss_fmt == AFMT_U16_BE || oss_fmt == AFMT_U16_LE ||
        oss_fmt == AFMT_S16_BE || oss_fmt == AFMT_S16_LE)
        bitrate *= 2;

    return bitrate;
}

static int
oss_get_format(AFormat fmt)
{
    int format = 0;

    switch (fmt) {
    case FMT_U8:
        format = AFMT_U8;
        break;
    case FMT_S8:
        format = AFMT_S8;
        break;
    case FMT_U16_LE:
        format = AFMT_U16_LE;
        break;
    case FMT_U16_BE:
        format = AFMT_U16_BE;
        break;
    case FMT_U16_NE:
#if (G_BYTE_ORDER == G_BIG_ENDIAN)
        format = AFMT_U16_BE;
#else
        format = AFMT_U16_LE;
#endif
        break;
    case FMT_S16_LE:
        format = AFMT_S16_LE;
        break;
    case FMT_S16_BE:
        format = AFMT_S16_BE;
        break;
    case FMT_S16_NE:
#if (G_BYTE_ORDER == G_BIG_ENDIAN)
        format = AFMT_S16_BE;
#else
        format = AFMT_S16_LE;
#endif
        break;
    }

    return format;
}

static void
oss_setup_format(AFormat fmt, int rate, int nch)
{
    effect.format.xmms = fmt;
    effect.frequency = rate;
    effect.channels = nch;
    effect.bps = oss_calc_bitrate(oss_get_format(fmt), rate, nch);

    output.format.oss = oss_get_format(fmt);
    output.frequency = rate;
    output.channels = nch;


    fragsize = 0;
    while ((1L << fragsize) < effect.bps / 25)
        fragsize++;
    fragsize--;

    device_buffer_size = ((1L << fragsize) * (NFRAGS + 1));

    oss_set_audio_params();

    output.bps = oss_calc_bitrate(output.format.oss, output.frequency,
                                  output.channels);
}


gint
oss_get_written_time(void)
{
    if (!going)
        return 0;
    return (written * 1000) / effect.bps;
}

gint
oss_get_output_time(void)
{
    guint64 bytes;

    if (!fd || !going)
        return 0;

    if (realtime)
        oss_calc_device_buffer_used();
    bytes = output_bytes < device_buffer_used ?
        0 : output_bytes - device_buffer_used;

    return output_time_offset + ((bytes * 1000) / output.bps);
}

static int
oss_used(void)
{
    if (realtime)
        return 0;
    else {
        if (wr_index >= rd_index)
            return wr_index - rd_index;
        return buffer_size - (rd_index - wr_index);
    }
}

gint
oss_playing(void)
{
    if (!going)
        return 0;
    if (realtime)
        oss_calc_device_buffer_used();
    if (!oss_used() && (device_buffer_used - (3 * blk_size)) <= 0)
        return FALSE;

    return TRUE;
}

gint
oss_free(void)
{
    if (!realtime) {
        if (remove_prebuffer && prebuffer) {
            prebuffer = FALSE;
            remove_prebuffer = FALSE;
        }
        if (prebuffer)
            remove_prebuffer = TRUE;

        if (rd_index > wr_index)
            return (rd_index - wr_index) - device_buffer_size - 1;
        return (buffer_size - (wr_index - rd_index)) - device_buffer_size - 1;
    }
    else if (paused)
        return 0;
    else
        return 1000000;
}

static inline ssize_t
write_all(int fd, const void *buf, size_t count)
{
    size_t done = 0;
    do {
        ssize_t n = write(fd, (gchar *) buf + done, count - done);
        if (n == -1) {
            if (errno == EINTR)
                continue;
            else
                break;
        }
        done += n;
    } while (count > done);

    return done;
}

static void
oss_write_audio(gpointer data, int length)
{

    audio_buf_info abuf_info;
    AFormat new_format;
    int new_frequency, new_channels;
    EffectPlugin *ep;

    new_format = input.format.xmms;
    new_frequency = input.frequency;
    new_channels = input.channels;


    ep = get_current_effect_plugin();
    if (effects_enabled() && ep && ep->query_format) {
        ep->query_format(&new_format, &new_frequency, &new_channels);
    }

    if (new_format != effect.format.xmms ||
        new_frequency != effect.frequency ||
        new_channels != effect.channels) {
        output_time_offset += (output_bytes * 1000) / output.bps;
        output_bytes = 0;
        close(fd);
        fd = open(device_name, O_WRONLY);
        oss_setup_format(new_format, new_frequency, new_channels);
    }
    if (effects_enabled() && ep && ep->mod_samples)
        length = ep->mod_samples(&data, length,
                                 input.format.xmms,
                                 input.frequency, input.channels);
    if (realtime && !ioctl(fd, SNDCTL_DSP_GETOSPACE, &abuf_info)) {
        while (abuf_info.bytes < length) {
            xmms_usleep(10000);
            if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &abuf_info))
                break;
        }
    }

    if (oss_convert_func != NULL)
        length = oss_convert_func(&data, length);

    if (oss_stereo_convert_func != NULL)
        length = oss_stereo_convert_func(&data, length, output.format.oss);

    if (effect.frequency == output.frequency)
        output_bytes += write_all(fd, data, length);
    else
        output_bytes += oss_downsample(data, length,
                                       effect.frequency, output.frequency);
}

static void
swap_endian(guint16 * data, int length)
{
    int i;
    for (i = 0; i < length; i += 2, data++)
        *data = GUINT16_SWAP_LE_BE(*data);
}

#define NOT_NATIVE_ENDIAN ((IS_BIG_ENDIAN &&				\
			   (output.format.oss == AFMT_S16_LE ||		\
			    output.format.oss == AFMT_U16_LE)) ||	\
			  (!IS_BIG_ENDIAN &&				\
			   (output.format.oss == AFMT_S16_BE ||		\
			    output.format.oss == AFMT_U16_BE)))


#define RESAMPLE_STEREO(sample_type)				\
do {								\
	const gint shift = sizeof (sample_type);		\
        gint i, in_samples, out_samples, x, delta;		\
	sample_type *inptr = (sample_type *)ob, *outptr;	\
	guint nlen = (((length >> shift) * espeed) / speed);	\
	if (nlen == 0)						\
		break;						\
	nlen <<= shift;						\
	if (NOT_NATIVE_ENDIAN)					\
		swap_endian(ob, length);			\
	if(nlen > nbuffer_size)					\
	{							\
		nbuffer = g_realloc(nbuffer, nlen);		\
		nbuffer_size = nlen;				\
	}							\
	outptr = (sample_type *)nbuffer;			\
	in_samples = length >> shift;				\
        out_samples = nlen >> shift;				\
	delta = (in_samples << 12) / out_samples;		\
	for (x = 0, i = 0; i < out_samples; i++)		\
	{							\
		gint x1, frac;					\
		x1 = (x >> 12) << 12;				\
		frac = x - x1;					\
		*outptr++ =					\
			(sample_type)				\
			((inptr[(x1 >> 12) << 1] *		\
			  ((1<<12) - frac) +			\
			  inptr[((x1 >> 12) + 1) << 1] *	\
			  frac) >> 12);				\
		*outptr++ =					\
			(sample_type)				\
			((inptr[((x1 >> 12) << 1) + 1] *	\
			  ((1<<12) - frac) +			\
			  inptr[(((x1 >> 12) + 1) << 1) + 1] *	\
			  frac) >> 12);				\
		x += delta;					\
	}							\
	if (NOT_NATIVE_ENDIAN)					\
		swap_endian(nbuffer, nlen);			\
	w = write_all(fd, nbuffer, nlen);			\
} while (0)


#define RESAMPLE_MONO(sample_type)				\
do {								\
	const gint shift = sizeof (sample_type) - 1;		\
        gint i, x, delta, in_samples, out_samples;		\
	sample_type *inptr = (sample_type *)ob, *outptr;	\
	guint nlen = (((length >> shift) * espeed) / speed);	\
	if (nlen == 0)						\
		break;						\
	nlen <<= shift;						\
	if (NOT_NATIVE_ENDIAN)					\
		swap_endian(ob, length);			\
	if(nlen > nbuffer_size)					\
	{							\
		nbuffer = g_realloc(nbuffer, nlen);		\
		nbuffer_size = nlen;				\
	}							\
	outptr = (sample_type *)nbuffer;			\
	in_samples = length >> shift;				\
        out_samples = nlen >> shift;				\
	delta = ((length >> shift) << 12) / out_samples;	\
	for (x = 0, i = 0; i < out_samples; i++)		\
	{							\
		gint x1, frac;					\
		x1 = (x >> 12) << 12;				\
		frac = x - x1;					\
		*outptr++ =					\
			(sample_type)				\
			((inptr[x1 >> 12] * ((1<<12) - frac) +	\
			  inptr[(x1 >> 12) + 1] * frac) >> 12);	\
		x += delta;					\
	}							\
	if (NOT_NATIVE_ENDIAN)					\
		swap_endian(nbuffer, nlen);			\
	w = write_all(fd, nbuffer, nlen);			\
} while (0)


static gint
oss_downsample(gpointer ob, guint length, guint speed, guint espeed)
{
    guint w = 0;
    static gpointer nbuffer = NULL;
    static guint nbuffer_size = 0;

    switch (output.format.oss) {
    case AFMT_S16_BE:
    case AFMT_S16_LE:
        if (output.channels == 2)
            RESAMPLE_STEREO(gint16);
        else
            RESAMPLE_MONO(gint16);
        break;
    case AFMT_U16_BE:
    case AFMT_U16_LE:
        if (output.channels == 2)
            RESAMPLE_STEREO(guint16);
        else
            RESAMPLE_MONO(guint16);
        break;
    case AFMT_S8:
        if (output.channels == 2)
            RESAMPLE_STEREO(gint8);
        else
            RESAMPLE_MONO(gint8);
        break;
    case AFMT_U8:
        if (output.channels == 2)
            RESAMPLE_STEREO(guint8);
        else
            RESAMPLE_MONO(guint8);
        break;
    }
    return w;
}

void
oss_write(gpointer ptr, int length)
{
    int cnt, off = 0;

    if (!realtime) {
        remove_prebuffer = FALSE;

        written += length;
        while (length > 0) {
            cnt = MIN(length, buffer_size - wr_index);
            memcpy(buffer + wr_index, (char *) ptr + off, cnt);
            wr_index = (wr_index + cnt) % buffer_size;
            length -= cnt;
            off += cnt;
        }
    }
    else {
        if (paused)
            return;
        oss_write_audio(ptr, length);
        written += length;
    }
}

void
oss_close(void)
{
    if (!going)
        return;
    going = 0;
    if (!realtime)
        g_thread_join(buffer_thread);
    else {
        ioctl(fd, SNDCTL_DSP_RESET, 0);
        close(fd);
    }
    g_free(device_name);
    oss_free_convert_buffer();
    wr_index = 0;
    rd_index = 0;

    close_mixer_device();
}

void
oss_flush(gint time)
{
    if (!realtime) {
        flush = time;
        while (flush != -1)
            xmms_usleep(10000);
    }
    else {
        ioctl(fd, SNDCTL_DSP_RESET, 0);
        close(fd);
        fd = open(device_name, O_WRONLY);
        oss_set_audio_params();
        output_time_offset = time;
        written = ((guint64) time * input.bps) / 1000;
        output_bytes = 0;
    }
}

void
oss_pause(short p)
{
    if (!realtime) {
        if (p == TRUE)
            do_pause = TRUE;
        else
            unpause = TRUE;
    }
    else
        paused = p;

}

gpointer
oss_loop(gpointer arg)
{
    gint length, cnt;
    fd_set set;
    struct timeval tv;

    while (going) {
        if (oss_used() > prebuffer_size)
            prebuffer = FALSE;
        if (oss_used() > 0 && !paused && !prebuffer) {
            tv.tv_sec = 0;
            tv.tv_usec = 10000;
            FD_ZERO(&set);
            FD_SET(fd, &set);
            if (!select_works || (select(fd + 1, NULL, &set, NULL, &tv) > 0)) {
                length = MIN(blk_size, oss_used());
                while (length > 0) {
                    cnt = MIN(length, buffer_size - rd_index);
                    oss_write_audio(buffer + rd_index, cnt);
                    rd_index = (rd_index + cnt) % buffer_size;
                    length -= cnt;
                }
                if (!oss_used())
                    ioctl(fd, SNDCTL_DSP_POST, 0);
            }
        }
        else
            xmms_usleep(10000);
        oss_calc_device_buffer_used();
        if (do_pause && !paused) {
            do_pause = FALSE;
            paused = TRUE;
            /*
             * We lose some data here that is sent to the
             * soundcard, but not yet played.  I don't
             * think this is worth fixing.
             */
            ioctl(fd, SNDCTL_DSP_RESET, 0);
        }
        else if (unpause && paused) {
            unpause = FALSE;
            close(fd);
            fd = open(device_name, O_WRONLY);
            oss_set_audio_params();
            paused = FALSE;
        }

        if (flush != -1) {
            /*
             * This close and open is a work around of a
             * bug that exists in some drivers which cause
             * the driver to get fucked up by a reset
             */

            ioctl(fd, SNDCTL_DSP_RESET, 0);
            close(fd);
            fd = open(device_name, O_WRONLY);
            oss_set_audio_params();
            output_time_offset = flush;
            written = ((guint64) flush * input.bps) / 1000;
            rd_index = wr_index = output_bytes = 0;
            flush = -1;
            prebuffer = TRUE;
        }

    }

    ioctl(fd, SNDCTL_DSP_RESET, 0);
    close(fd);
    g_free(buffer);
    return NULL;
}

void
oss_set_audio_params(void)
{
    int frag, stereo, ret;
    struct timeval tv;
    fd_set set;

    ioctl(fd, SNDCTL_DSP_RESET, 0);
    frag = (NFRAGS << 16) | fragsize;
    ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &frag);
    /*
     * Set the stream format.  This ioctl() might fail, but should
     * return a format that works if it does.
     */
    ioctl(fd, SNDCTL_DSP_SETFMT, &output.format.oss);
    if (ioctl(fd, SNDCTL_DSP_SETFMT, &output.format.oss) == -1)
        g_warning("SNDCTL_DSP_SETFMT ioctl failed: %s", strerror(errno));

    stereo = output.channels - 1;
    ioctl(fd, SNDCTL_DSP_STEREO, &stereo);
    output.channels = stereo + 1;

    oss_stereo_convert_func = oss_get_stereo_convert_func(output.channels,
                                                          effect.channels);

    if (ioctl(fd, SNDCTL_DSP_SPEED, &output.frequency) == -1)
        g_warning("SNDCTL_DSP_SPEED ioctl failed: %s", strerror(errno));

    if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &blk_size) == -1)
        blk_size = 1L << fragsize;

    oss_convert_func =
        oss_get_convert_func(output.format.oss,
                             oss_get_format(effect.format.xmms));

    /*
     * Stupid hack to find out if the driver support selects, some
     * drivers won't work properly without a select and some won't
     * work with a select :/
     */

    tv.tv_sec = 0;
    tv.tv_usec = 50000;
    FD_ZERO(&set);
    FD_SET(fd, &set);
    ret = select(fd + 1, NULL, &set, NULL, &tv);
    if (ret > 0)
        select_works = TRUE;
    else
        select_works = FALSE;
}

gint
oss_open(AFormat fmt, gint rate, gint nch)
{

    if (oss_cfg.use_alt_audio_device && oss_cfg.alt_audio_device)
        device_name = g_strdup(oss_cfg.alt_audio_device);
    else {
        if (oss_cfg.audio_device > 0)
            device_name =
                g_strdup_printf("%s%d", DEV_DSP, oss_cfg.audio_device);
        else
            device_name = g_strdup(DEV_DSP);
    }

    fd = open(device_name, O_WRONLY);

    if (fd == -1) {
        g_warning("oss_open(): Failed to open audio device (%s): %s",
                  device_name, strerror(errno));
        g_free(device_name);
        return 0;
    }

    input.format.xmms = fmt;
    input.frequency = rate;
    input.channels = nch;
    input.bps = oss_calc_bitrate(oss_get_format(fmt), rate, nch);

    oss_setup_format(fmt, rate, nch);

    realtime = xmms_check_realtime_priority();

    if (!realtime) {
        buffer_size = (oss_cfg.buffer_size * input.bps) / 1000;
        if (buffer_size < 8192)
            buffer_size = 8192;
        prebuffer_size = (buffer_size * oss_cfg.prebuffer) / 100;
        if (buffer_size - prebuffer_size < 4096)
            prebuffer_size = buffer_size - 4096;

        buffer_size += device_buffer_size;
        buffer = g_malloc0(buffer_size);
    }
    flush = -1;
    prebuffer = TRUE;
    wr_index = rd_index = output_time_offset = written = output_bytes = 0;
    paused = FALSE;
    do_pause = FALSE;
    unpause = FALSE;
    remove_prebuffer = FALSE;

    going = 1;
    if (!realtime)
        buffer_thread = g_thread_create(oss_loop, NULL, TRUE, NULL);
    return 1;
}

void oss_tell(AFormat * fmt, gint * rate, gint * nch)
{
	(*fmt) = input.format.xmms;
	(*rate) = input.frequency;
	(*nch) = input.channels;
}