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-rw-r--r--audio/out/ao_oss.c714
1 files changed, 229 insertions, 485 deletions
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index fbc535e413..afe5839dd5 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -4,6 +4,7 @@
* Original author: A'rpi
* Support for >2 output channels added 2001-11-25
* - Steve Davies <steve@daviesfam.org>
+ * Rozhuk Ivan <rozhuk.im@gmail.com> 2020-2023
*
* This file is part of mpv.
*
@@ -21,83 +22,53 @@
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
+#include <errno.h>
+#include <fcntl.h>
#include <stdio.h>
-#include <stdlib.h>
+#include <unistd.h>
#include <sys/ioctl.h>
-#include <unistd.h>
-#include <sys/time.h>
-#include <sys/types.h>
+#include <sys/soundcard.h>
#include <sys/stat.h>
-#include <fcntl.h>
-#include <poll.h>
-#include <errno.h>
-#include <string.h>
-#include <strings.h>
+#if defined(__DragonFly__) || defined(__FreeBSD__)
+#include <sys/sysctl.h>
+#endif
+#include <sys/types.h>
-#include "config.h"
-#include "options/options.h"
+#include "audio/format.h"
#include "common/common.h"
#include "common/msg.h"
-#include "osdep/timer.h"
+#include "options/options.h"
#include "osdep/endian.h"
-
-#include <sys/soundcard.h>
-
-#include "audio/format.h"
-
+#include "osdep/io.h"
#include "ao.h"
#include "internal.h"
-#if !HAVE_GPL
-#error GPL only
-#endif
-
-// 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.
-#if defined(SNDCTL_DSP_RESET) && !defined(__NetBSD__)
-#define KEEP_DEVICE 1
-#else
-#define KEEP_DEVICE 0
+#ifndef AFMT_AC3
+#define AFMT_AC3 -1
#endif
#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;
+ int dsp_fd;
+ double bps; /* Bytes per second. */
};
-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
+ {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_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
@@ -117,541 +88,314 @@ static const int format_table[][2] = {
{-1, -1}
};
-#ifndef AFMT_AC3
-#define AFMT_AC3 -1
-#endif
+#define MP_WARN_IOCTL_ERR(__ao) \
+ MP_WARN((__ao), "%s: ioctl() fail, err = %i: %s\n", \
+ __FUNCTION__, errno, mp_strerror(errno))
-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;
-}
+static void uninit(struct ao *ao);
-#ifdef SNDCTL_DSP_GETPLAYVOL
-static int volume_oss4(struct ao *ao, ao_control_vol_t *vol, int cmd)
+static void device_descr_get(size_t dev_idx, char *buf, size_t buf_size)
{
- struct priv *p = ao->priv;
- int v;
+#if defined(__DragonFly__) || defined(__FreeBSD__)
+ char dev_path[32];
+ size_t tmp = (buf_size - 1);
- 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;
+ snprintf(dev_path, sizeof(dev_path), "dev.pcm.%zu.%%desc", dev_idx);
+ if (sysctlbyname(dev_path, buf, &tmp, NULL, 0) != 0) {
+ tmp = 0;
}
+ buf[tmp] = 0x00;
+#elif defined(SOUND_MIXER_INFO)
+ size_t tmp = 0;
+ char dev_path[32];
+ mixer_info mi;
+
+ snprintf(dev_path, sizeof(dev_path), PATH_DEV_MIXER"%zu", dev_idx);
+ int fd = open(dev_path, O_RDONLY);
+ if (ioctl(fd, SOUND_MIXER_INFO, &mi) == 0) {
+ strncpy(buf, mi.name, buf_size - 1);
+ tmp = (buf_size - 1);
}
- 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);
+ close(fd);
+ buf[tmp] = 0x00;
+#else
+ buf[0] = 0x00;
#endif
- close(p->audio_fd);
- p->audio_fd = -1;
}
-// close audio device
-static void uninit(struct ao *ao)
+static int format2oss(int format)
{
- close_device(ao);
+ for (size_t i = 0; format_table[i][0] != -1; i++) {
+ if (format_table[i][1] == format)
+ return format_table[i][0];
+ }
+ return -1;
}
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));
+ 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;
+ return (ioctl(p->dsp_fd, SNDCTL_DSP_SETFMT, &oss_format) != -1);
}
-static int reopen_device(struct ao *ao, bool allow_format_changes)
+static int init(struct ao *ao)
{
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;
+ audio_buf_info info;
+ size_t i;
+ int format, samplerate, nchannels, reqchannels, trig = 0;
+ int best_sample_formats[AF_FORMAT_COUNT + 1];
+ const char *device = ((ao->device) ? ao->device : PATH_DEV_DSP);
+
+ /* Opening device. */
+ MP_VERBOSE(ao, "Using '%s' audio device.\n", device);
+ p->dsp_fd = open(device, (O_WRONLY | O_CLOEXEC));
+ if (p->dsp_fd < 0) {
+ MP_ERR(ao, "Can't open audio device %s: %s.\n",
+ device, mp_strerror(errno));
+ goto err_out;
}
-#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 + 1];
- af_get_best_sample_formats(format, try_formats);
- for (int n = 0; try_formats[n]; n++) {
- format = try_formats[n];
+ /* Selecting sound format. */
+ format = af_fmt_from_planar(ao->format);
+ af_get_best_sample_formats(format, best_sample_formats);
+ for (i = 0; best_sample_formats[i]; i++) {
+ format = best_sample_formats[i];
if (try_format(ao, &format))
break;
}
-
if (!format) {
MP_ERR(ao, "Can't set sample format.\n");
- goto fail;
+ goto err_out;
}
+ MP_VERBOSE(ao, "Sample format: %s\n", af_fmt_to_str(format));
- MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(format));
-
- if (!af_fmt_is_spdif(format)) {
+ /* Channels count. */
+ if (af_fmt_is_spdif(format)) {
+ nchannels = reqchannels = channels.num;
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1) {
+ MP_ERR(ao, "Failed to set audio device to %d channels.\n",
+ reqchannels);
+ goto err_out_ioctl;
+ }
+ } else {
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]);
+ for (i = 0; i < MP_ARRAY_SIZE(oss_layouts); i++) {
+ mp_chmap_sel_add_map(&sel, &oss_layouts[i]);
+ }
if (!ao_chmap_sel_adjust(ao, &sel, &channels))
- goto fail;
- int c, nchannels, reqchannels;
+ goto err_out;
nchannels = reqchannels = channels.num;
- // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
- if (reqchannels > 2) {
- if (ioctl(p->audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1) {
- MP_ERR(ao, "Failed to set audio device to %d channels.\n",
- reqchannels);
- goto fail;
- }
- if (nchannels != reqchannels) {
- // Fallback to stereo
- nchannels = 2;
- goto stereo;
- }
- } else {
-stereo:
- c = nchannels - 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;
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1) {
+ MP_ERR(ao, "Failed to set audio device to %d channels.\n",
+ reqchannels);
+ goto err_out_ioctl;
}
- 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);
+ if (nchannels != reqchannels) {
+ /* Update number of channels to OSS suggested value. */
+ if (!ao_chmap_sel_get_def(ao, &sel, &channels, nchannels))
+ goto err_out;
}
- } 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;
+ MP_VERBOSE(ao, "Using %d channels (requested: %d).\n",
+ channels.num, reqchannels);
}
- 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;
- }
+ /* Sample rate. */
+ samplerate = ao->samplerate;
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_SPEED, &samplerate) == -1)
+ goto err_out_ioctl;
+ MP_VERBOSE(ao, "Using %d Hz samplerate.\n", samplerate);
+
+ /* Get buffer size. */
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_GETOSPACE, &info) == -1)
+ goto err_out_ioctl;
+ /* See ao.c ao->sstride initializations and get_state(). */
+ ao->device_buffer = ((info.fragstotal * info.fragsize) /
+ af_fmt_to_bytes(format));
+ if (!af_fmt_is_planar(format)) {
+ ao->device_buffer /= channels.num;
}
- int sstride = channels.num * af_fmt_to_bytes(format);
- p->outburst -= p->outburst % sstride; // round down
- ao->period_size = p->outburst / sstride;
+ /* Do not start playback after data written. */
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_SETTRIGGER, &trig) == -1)
+ goto err_out_ioctl;
+
+ /* Update sound params. */
+ ao->format = format;
+ ao->samplerate = samplerate;
+ ao->channels = channels;
+ p->bps = (channels.num * samplerate * af_fmt_to_bytes(format));
return 0;
-fail:
- close_device(ao);
+err_out_ioctl:
+ MP_WARN_IOCTL_ERR(ao);
+err_out:
+ uninit(ao);
return -1;
}
-// open & setup audio device
-// return: 0=success -1=fail
-static int init(struct ao *ao)
+static void uninit(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]);
+ if (p->dsp_fd == -1)
+ return;
+ ioctl(p->dsp_fd, SNDCTL_DSP_HALT, NULL);
+ close(p->dsp_fd);
+ p->dsp_fd = -1;
+}
- ao->format = af_fmt_from_planar(ao->format);
+static int control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+ struct priv *p = ao->priv;
+ float *vol = arg;
+ int v;
- if (reopen_device(ao, true) < 0)
- goto fail;
+ if (p->dsp_fd < 0)
+ return CONTROL_ERROR;
- 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;
+ switch (cmd) {
+ case AOCONTROL_GET_VOLUME:
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1) {
+ MP_WARN_IOCTL_ERR(ao);
+ return CONTROL_ERROR;
}
- free(data);
- if (p->buffersize == 0) {
- MP_ERR(ao, "Your OSS audio driver DOES NOT support poll().\n");
- goto fail;
+ *vol = ((v & 0x00ff) + ((v & 0xff00) >> 8)) / 2.0;
+ return CONTROL_OK;
+ case AOCONTROL_SET_VOLUME:
+ v = ((int)*vol << 8) | (int)*vol;
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1) {
+ MP_WARN_IOCTL_ERR(ao);
+ return CONTROL_ERROR;
}
+ return CONTROL_OK;
}
- 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
+ return CONTROL_UNKNOWN;
}
-// 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
+ int trig = 0;
+
+ /* Clear buf and do not start playback after data written. */
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_HALT, NULL) == -1 ||
+ ioctl(p->dsp_fd, SNDCTL_DSP_SETTRIGGER, &trig) == -1)
+ {
+ MP_WARN_IOCTL_ERR(ao);
+ MP_WARN(ao, "Force reinitialize audio device.\n");
+ uninit(ao);
+ init(ao);
+ }
}
-// 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)
+static void start(struct ao *ao)
{
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;
- }
+ int trig = PCM_ENABLE_OUTPUT;
- if (len > p->outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
- len /= p->outburst;
- len *= p->outburst;
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_SETTRIGGER, &trig) == -1) {
+ MP_WARN_IOCTL_ERR(ao);
+ return;
}
- 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)
+static bool audio_write(struct ao *ao, void **data, int samples)
{
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;
-}
-
+ ssize_t rc;
+ const size_t size = (samples * ao->sstride);
-// return: how many samples can be played without blocking
-static int get_space(struct ao *ao)
-{
- struct priv *p = ao->priv;
+ if (size == 0)
+ return true;
- 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;
+ while ((rc = write(p->dsp_fd, data[0], size)) == -1) {
+ if (errno == EINTR)
+ continue;
+ MP_WARN(ao, "audio_write: write() fail, err = %i: %s.\n",
+ errno, mp_strerror(errno));
+ return false;
+ }
+ if ((size_t)rc != size) {
+ MP_WARN(ao, "audio_write: unexpected partial write: required: %zu, written: %zu.\n",
+ size, (size_t)rc);
+ return false;
}
- 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;
+ return true;
}
-// stop playing, keep buffers (for pause)
-static void audio_pause(struct ao *ao)
+static void get_state(struct ao *ao, struct mp_pcm_state *state)
{
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);
+ audio_buf_info info;
+ int odelay;
+
+ if (ioctl(p->dsp_fd, SNDCTL_DSP_GETOSPACE, &info) == -1 ||
+ ioctl(p->dsp_fd, SNDCTL_DSP_GETODELAY, &odelay) == -1)
+ {
+ MP_WARN_IOCTL_ERR(ao);
+ memset(state, 0x00, sizeof(struct mp_pcm_state));
+ state->delay = 0.0;
+ return;
+ }
+ state->free_samples = (info.bytes / ao->sstride);
+ state->queued_samples = (ao->device_buffer - state->free_samples);
+ state->delay = (odelay / p->bps);
+ state->playing = (state->queued_samples != 0);
}
-static int audio_wait(struct ao *ao, pthread_mutex_t *lock)
+static void list_devs(struct ao *ao, struct ao_device_list *list)
{
- struct priv *p = ao->priv;
+ struct stat st;
+ char dev_path[32] = PATH_DEV_DSP, dev_descr[256] = "Default";
+ struct ao_device_desc dev = {.name = dev_path, .desc = dev_descr};
- 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;
-}
+ if (stat(PATH_DEV_DSP, &st) == 0) {
+ ao_device_list_add(list, ao, &dev);
+ }
-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"});
+ /* Auto detect. */
+ for (size_t i = 0, fail_cnt = 0; fail_cnt < 8; i ++, fail_cnt ++) {
+ snprintf(dev_path, sizeof(dev_path), PATH_DEV_DSP"%zu", i);
+ if (stat(dev_path, &st) != 0)
+ continue;
+ device_descr_get(i, dev_descr, sizeof(dev_descr));
+ ao_device_list_add(list, ao, &dev);
+ fail_cnt = 0; /* Reset fail counter. */
+ }
}
-#define OPT_BASE_STRUCT struct priv
-
const struct ao_driver audio_out_oss = {
- .description = "OSS/ioctl audio output",
.name = "oss",
+ .description = "OSS/ioctl audio output",
.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,
+ .start = start,
+ .write = audio_write,
+ .get_state = get_state,
.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, M_OPT_FILE),
- OPT_STRING("mixer-channel", cfg_oss_mixer_channel, 0),
- {0}
+ .dsp_fd = -1,
},
- .options_prefix = "oss",
};
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-23 13:36:34 +0000