mirror of
https://github.com/mpv-player/mpv.git
synced 2024-09-20 03:52:22 +02:00
ao_oss: switch to new AO API
This commit is contained in:
wm4
parent
f8f4285671
commit
15202ebc76
@ -34,6 +34,7 @@
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#include <string.h>
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#include "config.h"
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#include "core/options.h"
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#include "core/mp_msg.h"
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#include "audio/mixer.h"
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@ -48,17 +49,6 @@
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#include "audio/format.h"
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#include "ao.h"
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#include "audio_out_internal.h"
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static const ao_info_t info =
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{
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"OSS/ioctl audio output",
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"oss",
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"A'rpi",
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""
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};
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LIBAO_EXTERN(oss)
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static int format2oss(int format)
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{
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@ -173,7 +163,7 @@ static int volume_oss4(ao_control_vol_t *vol, int cmd)
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#endif
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// to set/get/query special features/parameters
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static int control(int cmd, void *arg)
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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switch (cmd) {
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case AOCONTROL_GET_VOLUME:
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@ -188,7 +178,7 @@ static int control(int cmd, void *arg)
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return CONTROL_OK;
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#endif
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if (AF_FORMAT_IS_AC3(ao_data.format))
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if (AF_FORMAT_IS_AC3(ao->format))
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return CONTROL_TRUE;
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if ((fd = open(oss_mixer_device, O_RDONLY)) != -1) {
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@ -216,16 +206,16 @@ static int control(int cmd, void *arg)
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}
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(int rate, const struct mp_chmap *channels, int format,
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int flags)
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// return: 0=success -1=fail
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static int init(struct ao *ao, char *params)
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{
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struct MPOpts *opts = ao->opts;
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char *mixer_channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
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int oss_format;
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char *mdev = mixer_device, *mchan = mixer_channel;
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char *mdev = opts->mixer_device, *mchan = opts->mixer_channel;
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mp_msg(MSGT_AO, MSGL_V, "ao2: %d Hz %d chans %s\n", rate,
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ao_data.channels.num, af_fmt2str_short(format));
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mp_msg(MSGT_AO, MSGL_V, "ao2: %d Hz %d chans %s\n", ao->samplerate,
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ao->channels.num, af_fmt2str_short(ao->format));
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if (ao_subdevice) {
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char *m, *c;
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@ -296,7 +286,7 @@ static int init(int rate, const struct mp_chmap *channels, int format,
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mp_tmsg(MSGT_AO, MSGL_ERR,
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"[AO OSS] audio_setup: Can't open audio device %s: %s\n",
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dsp, strerror(errno));
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return 0;
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return -1;
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}
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#ifdef __linux__
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@ -304,7 +294,7 @@ static int init(int rate, const struct mp_chmap *channels, int format,
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if (fcntl(audio_fd, F_SETFL, 0) < 0) {
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mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Can't make file "
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"descriptor blocking: %s\n", strerror(errno));
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return 0;
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return -1;
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}
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#endif
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@ -312,47 +302,45 @@ static int init(int rate, const struct mp_chmap *channels, int format,
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fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
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#endif
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if (AF_FORMAT_IS_AC3(format)) {
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ao_data.samplerate = rate;
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ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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if (AF_FORMAT_IS_AC3(ao->format)) {
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ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
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}
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ac3_retry:
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if (AF_FORMAT_IS_AC3(format))
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format = AF_FORMAT_AC3_NE;
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ao_data.format = format;
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oss_format = format2oss(format);
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if (AF_FORMAT_IS_AC3(ao->format))
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ao->format = AF_FORMAT_AC3_NE;
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oss_format = format2oss(ao->format);
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if (oss_format == -1) {
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#if BYTE_ORDER == BIG_ENDIAN
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oss_format = AFMT_S16_BE;
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#else
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oss_format = AFMT_S16_LE;
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#endif
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format = AF_FORMAT_S16_NE;
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ao->format = AF_FORMAT_S16_NE;
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}
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if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format) < 0 ||
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oss_format != format2oss(format))
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oss_format != format2oss(ao->format))
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{
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mp_tmsg(MSGT_AO, MSGL_WARN, "[AO OSS] Can't set audio device %s to %s "
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"output, trying %s...\n", dsp, af_fmt2str_short(format),
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"output, trying %s...\n", dsp, af_fmt2str_short(ao->format),
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af_fmt2str_short(AF_FORMAT_S16_NE));
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format = AF_FORMAT_S16_NE;
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ao->format = AF_FORMAT_S16_NE;
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goto ac3_retry;
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}
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ao_data.format = oss2format(oss_format);
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if (ao_data.format == -1)
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return 0;
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ao->format = oss2format(oss_format);
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if (ao->format == -1)
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return -1;
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mp_msg(MSGT_AO, MSGL_V, "audio_setup: sample format: %s (requested: %s)\n",
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af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
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mp_msg(MSGT_AO, MSGL_V, "audio_setup: sample format: %s\n",
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af_fmt2str_short(ao->format));
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if (!AF_FORMAT_IS_AC3(format)) {
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if (!AF_FORMAT_IS_AC3(ao->format)) {
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struct mp_chmap_sel sel = {0};
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mp_chmap_sel_add_alsa_def(&sel);
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if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
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return 0;
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int reqchannels = ao_data.channels.num;
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
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return -1;
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int reqchannels = ao->channels.num;
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// We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
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if (reqchannels > 2) {
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int nchannels = reqchannels;
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@ -361,27 +349,25 @@ ac3_retry:
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{
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mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Failed to "
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"set audio device to %d channels.\n", reqchannels);
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return 0;
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return -1;
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}
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} else {
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int c = reqchannels - 1;
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if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
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mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Failed to "
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"set audio device to %d channels.\n", reqchannels);
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return 0;
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return -1;
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}
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if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, c + 1))
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return 0;
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if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, c + 1))
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return -1;
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}
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mp_msg(MSGT_AO, MSGL_V,
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"audio_setup: using %d channels (requested: %d)\n",
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ao_data.channels.num, reqchannels);
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ao->channels.num, reqchannels);
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// set rate
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ao_data.samplerate = rate;
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ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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mp_msg(MSGT_AO, MSGL_V,
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"audio_setup: using %d Hz samplerate (requested: %d)\n",
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ao_data.samplerate, rate);
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ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
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mp_msg(MSGT_AO, MSGL_V, "audio_setup: using %d Hz samplerate\n",
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ao->samplerate);
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}
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if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz) == -1) {
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@ -390,29 +376,29 @@ ac3_retry:
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"support SNDCTL_DSP_GETOSPACE\n");
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if (ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r) == -1)
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mp_msg(MSGT_AO, MSGL_V, "audio_setup: %d bytes/frag (config.h)\n",
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ao_data.outburst);
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ao->outburst);
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else {
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ao_data.outburst = r;
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ao->outburst = r;
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mp_msg(MSGT_AO, MSGL_V, "audio_setup: %d bytes/frag (GETBLKSIZE)\n",
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ao_data.outburst);
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ao->outburst);
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}
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} else {
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mp_msg(MSGT_AO, MSGL_V,
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"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
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zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
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if (ao_data.buffersize == -1)
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ao_data.buffersize = zz.bytes;
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ao_data.outburst = zz.fragsize;
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if (ao->buffersize == -1)
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ao->buffersize = zz.bytes;
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ao->outburst = zz.fragsize;
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}
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if (ao_data.buffersize == -1) {
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if (ao->buffersize == -1) {
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// Measuring buffer size:
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void *data;
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ao_data.buffersize = 0;
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ao->buffersize = 0;
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#ifdef HAVE_AUDIO_SELECT
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data = malloc(ao_data.outburst);
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memset(data, 0, ao_data.outburst);
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while (ao_data.buffersize < 0x40000) {
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data = malloc(ao->outburst);
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memset(data, 0, ao->outburst);
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while (ao->buffersize < 0x40000) {
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fd_set rfds;
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struct timeval tv;
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FD_ZERO(&rfds);
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@ -421,42 +407,28 @@ ac3_retry:
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tv.tv_usec = 0;
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if (!select(audio_fd + 1, NULL, &rfds, NULL, &tv))
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break;
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write(audio_fd, data, ao_data.outburst);
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ao_data.buffersize += ao_data.outburst;
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write(audio_fd, data, ao->outburst);
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ao->buffersize += ao->outburst;
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}
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free(data);
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if (ao_data.buffersize == 0) {
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if (ao->buffersize == 0) {
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mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS]\n *** Your audio driver "
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"DOES NOT support select() ***\n Recompile mpv with "
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"#undef HAVE_AUDIO_SELECT in config.h !\n\n");
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return 0;
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return -1;
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}
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#endif
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}
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ao_data.bps = ao_data.channels.num;
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switch (ao_data.format & AF_FORMAT_BITS_MASK) {
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case AF_FORMAT_8BIT:
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break;
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case AF_FORMAT_16BIT:
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ao_data.bps *= 2;
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break;
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case AF_FORMAT_24BIT:
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ao_data.bps *= 3;
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break;
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case AF_FORMAT_32BIT:
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ao_data.bps *= 4;
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break;
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}
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ao->bps = ao->channels.num * (af_fmt2bits(ao->format) / 8);
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ao->outburst -= ao->outburst % ao->bps; // round down
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ao->bps *= ao->samplerate;
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ao_data.outburst -= ao_data.outburst % ao_data.bps; // round down
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ao_data.bps *= ao_data.samplerate;
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return 1;
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return 0;
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}
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// close audio device
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static void uninit(int immed)
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static void uninit(struct ao *ao, bool immed)
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{
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if (audio_fd == -1)
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return;
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@ -474,10 +446,10 @@ static void uninit(int immed)
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}
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// stop playing and empty buffers (for seeking/pause)
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static void reset(void)
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static void reset(struct ao *ao)
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{
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int oss_format;
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uninit(1);
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uninit(ao, true);
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audio_fd = open(dsp, O_WRONLY);
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if (audio_fd < 0) {
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mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS]\nFatal error: *** CANNOT "
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@ -489,46 +461,25 @@ static void reset(void)
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fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
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#endif
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oss_format = format2oss(ao_data.format);
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if (AF_FORMAT_IS_AC3(ao_data.format))
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ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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oss_format = format2oss(ao->format);
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if (AF_FORMAT_IS_AC3(ao->format))
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ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
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ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
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if (!AF_FORMAT_IS_AC3(ao_data.format)) {
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if (ao_data.channels.num > 2)
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ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels.num);
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if (!AF_FORMAT_IS_AC3(ao->format)) {
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if (ao->channels.num > 2)
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ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao->channels.num);
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else {
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int c = ao_data.channels.num - 1;
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int c = ao->channels.num - 1;
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ioctl(audio_fd, SNDCTL_DSP_STEREO, &c);
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}
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ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate);
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}
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause(void)
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{
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prepause_space = get_space();
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uninit(1);
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}
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// resume playing, after audio_pause()
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static void audio_resume(void)
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{
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int fillcnt;
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reset();
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fillcnt = get_space() - prepause_space;
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if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) {
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void *silence = calloc(fillcnt, 1);
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play(silence, fillcnt, 0);
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free(silence);
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}
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}
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// return: how many bytes can be played without blocking
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static int get_space(void)
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static int get_space(struct ao *ao)
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{
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int playsize = ao_data.outburst;
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int playsize = ao->outburst;
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#ifdef SNDCTL_DSP_GETOSPACE
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if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) {
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@ -552,45 +503,83 @@ static int get_space(void)
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}
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#endif
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return ao_data.outburst;
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return ao->outburst;
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause(struct ao *ao)
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{
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prepause_space = get_space(ao);
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uninit(ao, true);
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}
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// plays 'len' bytes of 'data'
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// it should round it down to outburst*n
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// return: number of bytes played
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static int play(void *data, int len, int flags)
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static int play(struct ao *ao, void *data, int len, int flags)
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{
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if (len == 0)
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return len;
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if (len > ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
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len /= ao_data.outburst;
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len *= ao_data.outburst;
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if (len > ao->outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
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len /= ao->outburst;
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len *= ao->outburst;
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}
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len = write(audio_fd, data, len);
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return len;
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}
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// resume playing, after audio_pause()
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static void audio_resume(struct ao *ao)
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{
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int fillcnt;
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reset(ao);
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fillcnt = get_space(ao) - prepause_space;
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if (fillcnt > 0 && !(ao->format & AF_FORMAT_SPECIAL_MASK)) {
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void *silence = calloc(fillcnt, 1);
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play(ao, silence, fillcnt, 0);
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free(silence);
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}
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}
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static int audio_delay_method = 2;
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// return: delay in seconds between first and last sample in buffer
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static float get_delay(void)
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static float get_delay(struct ao *ao)
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{
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/* Calculate how many bytes/second is sent out */
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if (audio_delay_method == 2) {
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#ifdef SNDCTL_DSP_GETODELAY
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int r = 0;
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if (ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r) != -1)
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return ((float)r) / (float)ao_data.bps;
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return ((float)r) / (float)ao->bps;
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#endif
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audio_delay_method = 1; // fallback if not supported
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}
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if (audio_delay_method == 1) {
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// SNDCTL_DSP_GETOSPACE
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if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) {
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return ((float)(ao_data.buffersize -
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zz.bytes)) / (float)ao_data.bps;
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return ((float)(ao->buffersize -
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zz.bytes)) / (float)ao->bps;
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}
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audio_delay_method = 0; // fallback if not supported
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}
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return ((float)ao_data.buffersize) / (float)ao_data.bps;
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return ((float)ao->buffersize) / (float)ao->bps;
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}
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const struct ao_driver audio_out_oss = {
|
||||
.info = &(const struct ao_info) {
|
||||
"OSS/ioctl audio output",
|
||||
"oss",
|
||||
"A'rpi",
|
||||
""
|
||||
},
|
||||
.init = init,
|
||||
.uninit = uninit,
|
||||
.control = control,
|
||||
.get_space = get_space,
|
||||
.play = play,
|
||||
.get_delay = get_delay,
|
||||
.pause = audio_pause,
|
||||
.resume = audio_resume,
|
||||
.reset = reset,
|
||||
};
|
||||
|
||||
Loading…
Reference in New Issue
Block a user