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mpv/audio/out/ao_oss.c
wm4 037c37519b audio/out: require AO drivers to report period size and correct buffer 
Before this change, AOs could have internal alignment, and play() would
not consume the trailing data if the size passed to it is not aligned.
Change this to require AOs to report their alignment (via period_size),
and make sure to always send aligned data.

The buffer reported by get_space() now always has to be correct and
reliable. If play() does not consume all data provided (which is bounded
by get_space()), an error is printed.

This is preparation for potential further AO changes.

I casually checked alsa/lavc/null/pcm, the other AOs might or might not
work.
2017-06-25 15:57:43 +02:00

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/*
* OSS audio output driver
*
* Original author: A'rpi
* Support for >2 output channels added 2001-11-25
* - Steve Davies <steve@daviesfam.org>
*
* This file is part of mpv.
*
* mpv 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.
*
* mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdio.h>
#include <stdlib.h>
#include <sys/ioctl.h>
#include <unistd.h>
#include <sys/time.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <poll.h>
#include <errno.h>
#include <string.h>
#include <strings.h>
#include "config.h"
#include "options/options.h"
#include "common/common.h"
#include "common/msg.h"
#include "osdep/timer.h"
#include "osdep/endian.h"
#include <sys/soundcard.h>
#include "audio/format.h"
#include "ao.h"
#include "internal.h"
// Define to 0 if the device must be reopened to reset it (stop all playback,
// clear the buffer), and the device should be closed when unused.
// Define to 1 if SNDCTL_DSP_RESET should be used to reset without close.
#define KEEP_DEVICE (defined(SNDCTL_DSP_RESET) && !defined(__NetBSD__))
#define PATH_DEV_DSP "/dev/dsp"
#define PATH_DEV_MIXER "/dev/mixer"
struct priv {
int audio_fd;
int prepause_samples;
int oss_mixer_channel;
int audio_delay_method;
int buffersize;
int outburst;
bool device_failed;
double audio_end;
char *oss_mixer_device;
char *cfg_oss_mixer_channel;
};
static const char *const mixer_channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
/* like alsa except for 6.1 and 7.1, from pcm/matrix_map.h */
static const struct mp_chmap oss_layouts[MP_NUM_CHANNELS + 1] = {
{0}, // empty
MP_CHMAP_INIT_MONO, // mono
MP_CHMAP2(FL, FR), // stereo
MP_CHMAP3(FL, FR, LFE), // 2.1
MP_CHMAP4(FL, FR, BL, BR), // 4.0
MP_CHMAP5(FL, FR, BL, BR, FC), // 5.0
MP_CHMAP6(FL, FR, BL, BR, FC, LFE), // 5.1
MP_CHMAP7(FL, FR, BL, BR, FC, LFE, BC), // 6.1
MP_CHMAP8(FL, FR, BL, BR, FC, LFE, SL, SR), // 7.1
};
#if !defined(AFMT_S16_NE) && defined(AFMT_S16_LE) && defined(AFMT_S16_BE)
#define AFMT_S16_NE MP_SELECT_LE_BE(AFMT_S16_LE, AFMT_S16_BE)
#endif
#if !defined(AFMT_S24_NE) && defined(AFMT_S24_LE) && defined(AFMT_S24_BE)
#define AFMT_S24_NE MP_SELECT_LE_BE(AFMT_S24_LE, AFMT_S24_BE)
#endif
#if !defined(AFMT_S32_NE) && defined(AFMT_S32_LE) && defined(AFMT_S32_BE)
#define AFMT_S32_NE AFMT_S32MP_SELECT_LE_BE(AFMT_S32_LE, AFMT_S32_BE)
#endif
static const int format_table[][2] = {
{AFMT_U8, AF_FORMAT_U8},
{AFMT_S16_NE, AF_FORMAT_S16},
#ifdef AFMT_S24_NE
{AFMT_S24_NE, AF_FORMAT_S24},
#endif
#ifdef AFMT_S32_NE
{AFMT_S32_NE, AF_FORMAT_S32},
#endif
#ifdef AFMT_FLOAT
{AFMT_FLOAT, AF_FORMAT_FLOAT},
#endif
#ifdef AFMT_MPEG
{AFMT_MPEG, AF_FORMAT_S_MP3},
#endif
{-1, -1}
};
#ifndef AFMT_AC3
#define AFMT_AC3 -1
#endif
static int format2oss(int format)
{
for (int n = 0; format_table[n][0] != -1; n++) {
if (format_table[n][1] == format)
return format_table[n][0];
}
return -1;
}
static int oss2format(int format)
{
for (int n = 0; format_table[n][0] != -1; n++) {
if (format_table[n][0] == format)
return format_table[n][1];
}
return 0;
}
#ifdef SNDCTL_DSP_GETPLAYVOL
static int volume_oss4(struct ao *ao, ao_control_vol_t *vol, int cmd)
{
struct priv *p = ao->priv;
int v;
if (p->audio_fd < 0)
return CONTROL_ERROR;
if (cmd == AOCONTROL_GET_VOLUME) {
if (ioctl(p->audio_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1)
return CONTROL_ERROR;
vol->right = (v & 0xff00) >> 8;
vol->left = v & 0x00ff;
return CONTROL_OK;
} else if (cmd == AOCONTROL_SET_VOLUME) {
v = ((int) vol->right << 8) | (int) vol->left;
if (ioctl(p->audio_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1)
return CONTROL_ERROR;
return CONTROL_OK;
} else
return CONTROL_UNKNOWN;
}
#endif
// to set/get/query special features/parameters
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct priv *p = ao->priv;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME: {
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int fd, v, devs;
#ifdef SNDCTL_DSP_GETPLAYVOL
// Try OSS4 first
if (volume_oss4(ao, vol, cmd) == CONTROL_OK)
return CONTROL_OK;
#endif
if (!af_fmt_is_pcm(ao->format))
return CONTROL_TRUE;
if ((fd = open(p->oss_mixer_device, O_RDONLY)) != -1) {
ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
if (devs & (1 << p->oss_mixer_channel)) {
if (cmd == AOCONTROL_GET_VOLUME) {
ioctl(fd, MIXER_READ(p->oss_mixer_channel), &v);
vol->right = (v & 0xFF00) >> 8;
vol->left = v & 0x00FF;
} else {
v = ((int)vol->right << 8) | (int)vol->left;
ioctl(fd, MIXER_WRITE(p->oss_mixer_channel), &v);
}
} else {
close(fd);
return CONTROL_ERROR;
}
close(fd);
return CONTROL_OK;
}
return CONTROL_ERROR;
}
#ifdef SNDCTL_DSP_GETPLAYVOL
case AOCONTROL_HAS_SOFT_VOLUME:
return CONTROL_TRUE;
#endif
}
return CONTROL_UNKNOWN;
}
// 1: ok, 0: not writable, -1: error
static int device_writable(struct ao *ao)
{
struct priv *p = ao->priv;
struct pollfd fd = {.fd = p->audio_fd, .events = POLLOUT};
return poll(&fd, 1, 0);
}
static void close_device(struct ao *ao)
{
struct priv *p = ao->priv;
p->device_failed = false;
if (p->audio_fd == -1)
return;
#if defined(SNDCTL_DSP_RESET)
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
#endif
close(p->audio_fd);
p->audio_fd = -1;
}
// close audio device
static void uninit(struct ao *ao)
{
close_device(ao);
}
static bool try_format(struct ao *ao, int *format)
{
struct priv *p = ao->priv;
int oss_format = format2oss(*format);
if (oss_format == -1 && af_fmt_is_spdif(*format))
oss_format = AFMT_AC3;
if (oss_format == -1) {
MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n",
af_fmt_to_str(*format));
*format = 0;
return false;
}
int actual_format = oss_format;
if (ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &actual_format) < 0)
actual_format = -1;
if (actual_format == oss_format)
return true;
MP_WARN(ao, "Can't set audio device to %s output.\n", af_fmt_to_str(*format));
*format = oss2format(actual_format);
if (actual_format != -1 && !*format)
MP_ERR(ao, "Unknown/Unsupported OSS format: 0x%x.\n", actual_format);
return false;
}
static int reopen_device(struct ao *ao, bool allow_format_changes)
{
struct priv *p = ao->priv;
int samplerate = ao->samplerate;
int format = ao->format;
struct mp_chmap channels = ao->channels;
const char *device = PATH_DEV_DSP;
if (ao->device)
device = ao->device;
MP_VERBOSE(ao, "using '%s' dsp device\n", device);
#ifdef __linux__
p->audio_fd = open(device, O_WRONLY | O_NONBLOCK);
#else
p->audio_fd = open(device, O_WRONLY);
#endif
if (p->audio_fd < 0) {
MP_ERR(ao, "Can't open audio device %s: %s\n",
device, mp_strerror(errno));
goto fail;
}
#ifdef __linux__
/* Remove the non-blocking flag */
if (fcntl(p->audio_fd, F_SETFL, 0) < 0) {
MP_ERR(ao, "Can't make file descriptor blocking: %s\n",
mp_strerror(errno));
goto fail;
}
#endif
#if defined(FD_CLOEXEC) && defined(F_SETFD)
fcntl(p->audio_fd, F_SETFD, FD_CLOEXEC);
#endif
if (af_fmt_is_spdif(format)) {
if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
goto fail;
// Probably could be fixed by setting number of channels; needs testing.
if (channels.num != 2) {
MP_ERR(ao, "Format %s not implemented.\n", af_fmt_to_str(format));
goto fail;
}
}
int try_formats[AF_FORMAT_COUNT];
af_get_best_sample_formats(format, try_formats);
for (int n = 0; try_formats[n]; n++) {
format = try_formats[n];
if (try_format(ao, &format))
break;
}
if (!format) {
MP_ERR(ao, "Can't set sample format.\n");
goto fail;
}
MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(format));
if (!af_fmt_is_spdif(format)) {
struct mp_chmap_sel sel = {0};
for (int n = 0; n < MP_NUM_CHANNELS + 1; n++)
mp_chmap_sel_add_map(&sel, &oss_layouts[n]);
if (!ao_chmap_sel_adjust(ao, &sel, &channels))
goto fail;
int reqchannels = channels.num;
// We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
if (reqchannels > 2) {
int nchannels = reqchannels;
if (ioctl(p->audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1 ||
nchannels != reqchannels)
{
MP_ERR(ao, "Failed to set audio device to %d channels.\n",
reqchannels);
goto fail;
}
} else {
int c = reqchannels - 1;
if (ioctl(p->audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
MP_ERR(ao, "Failed to set audio device to %d channels.\n",
reqchannels);
goto fail;
}
if (!ao_chmap_sel_get_def(ao, &sel, &channels, c + 1))
goto fail;
}
MP_VERBOSE(ao, "using %d channels (requested: %d)\n",
channels.num, reqchannels);
// set rate
if (ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
goto fail;
MP_VERBOSE(ao, "using %d Hz samplerate\n", samplerate);
}
audio_buf_info zz = {0};
if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &zz) == -1) {
int r = 0;
MP_WARN(ao, "driver doesn't support SNDCTL_DSP_GETOSPACE\n");
if (ioctl(p->audio_fd, SNDCTL_DSP_GETBLKSIZE, &r) == -1)
MP_VERBOSE(ao, "%d bytes/frag (config.h)\n", p->outburst);
else {
p->outburst = r;
MP_VERBOSE(ao, "%d bytes/frag (GETBLKSIZE)\n", p->outburst);
}
} else {
MP_VERBOSE(ao, "frags: %3d/%d (%d bytes/frag) free: %6d\n",
zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
p->buffersize = zz.bytes;
p->outburst = zz.fragsize;
}
if (allow_format_changes) {
ao->format = format;
ao->samplerate = samplerate;
ao->channels = channels;
} else {
if (format != ao->format || samplerate != ao->samplerate ||
!mp_chmap_equals(&channels, &ao->channels))
{
MP_ERR(ao, "Could not reselect previous audio format.\n");
goto fail;
}
}
ao->period_size = channels.num * af_fmt_to_bytes(format);
p->outburst -= p->outburst % ao->period_size; // round down
return 0;
fail:
close_device(ao);
return -1;
}
// open & setup audio device
// return: 0=success -1=fail
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
const char *mchan = NULL;
if (p->cfg_oss_mixer_channel && p->cfg_oss_mixer_channel[0])
mchan = p->cfg_oss_mixer_channel;
if (mchan) {
int fd, devs, i;
if ((fd = open(p->oss_mixer_device, O_RDONLY)) == -1) {
MP_ERR(ao, "Can't open mixer device %s: %s\n",
p->oss_mixer_device, mp_strerror(errno));
} else {
ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
close(fd);
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (!strcasecmp(mixer_channels[i], mchan)) {
if (!(devs & (1 << i))) {
MP_ERR(ao, "Audio card mixer does not have "
"channel '%s', using default.\n", mchan);
i = SOUND_MIXER_NRDEVICES + 1;
break;
}
p->oss_mixer_channel = i;
break;
}
}
if (i == SOUND_MIXER_NRDEVICES) {
MP_ERR(ao, "Audio card mixer does not have "
"channel '%s', using default.\n", mchan);
}
}
} else {
p->oss_mixer_channel = SOUND_MIXER_PCM;
}
MP_VERBOSE(ao, "using '%s' mixer device\n", p->oss_mixer_device);
MP_VERBOSE(ao, "using '%s' mixer channel\n", mixer_channels[p->oss_mixer_channel]);
ao->format = af_fmt_from_planar(ao->format);
if (reopen_device(ao, true) < 0)
goto fail;
if (p->buffersize == -1) {
// Measuring buffer size:
void *data = malloc(p->outburst);
if (!data) {
MP_ERR(ao, "Out of memory, or broken outburst size.\n");
goto fail;
}
p->buffersize = 0;
memset(data, 0, p->outburst);
while (p->buffersize < 0x40000 && device_writable(ao) > 0) {
(void)write(p->audio_fd, data, p->outburst);
p->buffersize += p->outburst;
}
free(data);
if (p->buffersize == 0) {
MP_ERR(ao, "Your OSS audio driver DOES NOT support poll().\n");
goto fail;
}
}
return 0;
fail:
uninit(ao);
return -1;
}
static void drain(struct ao *ao)
{
#ifdef SNDCTL_DSP_SYNC
struct priv *p = ao->priv;
// to get the buffer played
if (p->audio_fd != -1)
ioctl(p->audio_fd, SNDCTL_DSP_SYNC, NULL);
#endif
}
// stop playing and empty buffers (for seeking/pause)
static void reset(struct ao *ao)
{
#if KEEP_DEVICE
struct priv *p = ao->priv;
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
#else
close_device(ao);
#endif
}
// plays 'len' samples of 'data'
// it should round it down to outburst*n
// return: number of samples played
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *p = ao->priv;
int len = samples * ao->sstride;
if (len == 0)
return len;
if (p->audio_fd < 0 && !p->device_failed && reopen_device(ao, false) < 0)
MP_ERR(ao, "Fatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE ***\n");
if (p->audio_fd < 0) {
// Let playback continue normally, even with a closed device.
p->device_failed = true;
double now = mp_time_sec();
if (p->audio_end < now)
p->audio_end = now;
p->audio_end += samples / (double)ao->samplerate;
return samples;
}
if (len > p->outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
len /= p->outburst;
len *= p->outburst;
}
len = write(p->audio_fd, data[0], len);
return len / ao->sstride;
}
// return: delay in seconds between first and last sample in buffer
static double get_delay(struct ao *ao)
{
struct priv *p = ao->priv;
if (p->audio_fd < 0) {
double rest = p->audio_end - mp_time_sec();
if (rest > 0)
return rest;
return 0;
}
/* Calculate how many bytes/second is sent out */
if (p->audio_delay_method == 2) {
#ifdef SNDCTL_DSP_GETODELAY
int r = 0;
if (ioctl(p->audio_fd, SNDCTL_DSP_GETODELAY, &r) != -1)
return r / (double)ao->bps;
#endif
p->audio_delay_method = 1; // fallback if not supported
}
if (p->audio_delay_method == 1) {
audio_buf_info zz = {0};
if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) {
return (p->buffersize - zz.bytes) / (double)ao->bps;
}
p->audio_delay_method = 0; // fallback if not supported
}
return p->buffersize / (double)ao->bps;
}
// return: how many samples can be played without blocking
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
audio_buf_info zz = {0};
if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) {
// calculate exact buffer space:
return zz.fragments * zz.fragsize / ao->sstride;
}
if (p->audio_fd < 0 && p->device_failed && get_delay(ao) > 0.2)
return 0;
if (p->audio_fd < 0 || device_writable(ao) > 0)
return p->outburst / ao->sstride;
return 0;
}
// stop playing, keep buffers (for pause)
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
p->prepause_samples = get_delay(ao) * ao->samplerate;
#if KEEP_DEVICE
ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL);
#else
close_device(ao);
#endif
}
// resume playing, after audio_pause()
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
p->audio_end = 0;
if (p->prepause_samples > 0)
ao_play_silence(ao, p->prepause_samples);
}
static int audio_wait(struct ao *ao, pthread_mutex_t *lock)
{
struct priv *p = ao->priv;
struct pollfd fd = {.fd = p->audio_fd, .events = POLLOUT};
int r = ao_wait_poll(ao, &fd, 1, lock);
if (fd.revents & (POLLERR | POLLNVAL))
return -1;
return r;
}
static void list_devs(struct ao *ao, struct ao_device_list *list)
{
if (stat(PATH_DEV_DSP, &(struct stat){0}) == 0)
ao_device_list_add(list, ao, &(struct ao_device_desc){"", "Default"});
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_oss = {
.description = "OSS/ioctl audio output",
.name = "oss",
.init = init,
.uninit = uninit,
.control = control,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = audio_pause,
.resume = audio_resume,
.reset = reset,
.drain = drain,
.wait = audio_wait,
.wakeup = ao_wakeup_poll,
.list_devs = list_devs,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.audio_fd = -1,
.audio_delay_method = 2,
.buffersize = -1,
.outburst = 512,
.oss_mixer_channel = SOUND_MIXER_PCM,
.oss_mixer_device = PATH_DEV_MIXER,
},
.options = (const struct m_option[]) {
OPT_STRING("mixer-device", oss_mixer_device, 0),
OPT_STRING("mixer-channel", cfg_oss_mixer_channel, 0),
{0}
},
.options_prefix = "oss",
};
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