mirror of https://github.com/mpv-player/mpv
interface to libao2 changed ao_plugin added
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3096 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
anders
parent
becadec644
commit
242aa6ebd4
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@ -4,7 +4,7 @@ include config.mak
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LIBNAME = libao2.a
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# TODO: moveout ao_sdl.c so it's only used when SDL is detected
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SRCS=afmt.c audio_out.c ao_null.c ao_pcm.c ao_mpegpes.c $(OPTIONAL_SRCS)
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SRCS=afmt.c audio_out.c ao_mpegpes.c ao_null.c ao_pcm.c ao_plugin.c $(OPTIONAL_SRCS)
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OBJS=$(SRCS:.c=.o)
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CFLAGS = $(OPTFLAGS) -I. -I.. $(SDL_INC) $(EXTRA_INC)
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@ -38,3 +38,4 @@ depend:
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ifneq ($(wildcard .depend),)
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include .depend
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endif
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@ -32,14 +32,6 @@ static ao_info_t info =
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LIBAO_EXTERN(alsa9)
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/* global variables:
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ao_samplerate
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ao_channels
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ao_format
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ao_bps
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ao_outburst
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ao_buffersize
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*/
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static snd_pcm_t *alsa_handler;
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static snd_pcm_format_t alsa_format;
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@ -67,7 +59,7 @@ static int control(int cmd, int arg)
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strncpy(alsa_device, (char *)arg, ALSA_DEVICE_SIZE);
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uninit();
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ret = init(ao_samplerate, ao_channels, ao_format, 0);
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ret = init(ao_data.samplerate, ao_data.channels, ao_data.format, 0);
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if (ret == 0)
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return(CONTROL_ERROR);
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else
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@ -112,12 +104,12 @@ static int init(int rate_hz, int channels, int format, int flags)
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return(0);
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}
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ao_samplerate = rate_hz;
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ao_bps = channels; /* really this is bytes per frame so bad varname */
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ao_format = format;
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ao_channels = channels;
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ao_outburst = OUTBURST;
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ao_buffersize = 16384;
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ao_data.samplerate = rate_hz;
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ao_data.bps = channels; /* really this is bytes per frame so bad varname */
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ao_data.format = format;
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ao_data.channels = channels;
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ao_data.outburst = OUTBURST;
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ao_data.buffersize = 16384;
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switch (format)
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{
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@ -148,7 +140,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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{
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case SND_PCM_FORMAT_S16_LE:
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case SND_PCM_FORMAT_U16_LE:
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ao_bps *= 2;
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ao_data.bps *= 2;
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break;
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case -1:
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printf("alsa-init: invalid format (%s) requested - output disabled\n",
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@ -164,8 +156,8 @@ static int init(int rate_hz, int channels, int format, int flags)
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return(0);
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}
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if (ao_subdevice != NULL)
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alsa_device = ao_subdevice;
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if (ao_data.subdevice != NULL)
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alsa_device = ao_data.subdevice;
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if (alsa_device == NULL)
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{
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@ -219,14 +211,14 @@ static int init(int rate_hz, int channels, int format, int flags)
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}
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if ((err = snd_pcm_hw_params_set_channels(alsa_handler, alsa_hwparams,
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ao_channels)) < 0)
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ao_data.channels)) < 0)
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{
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printf("alsa-init: unable to set channels: %s\n",
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snd_strerror(err));
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return(0);
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}
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if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_samplerate, 0)) < 0)
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if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_data.samplerate, 0)) < 0)
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/* was originally only snd_pcm_hw_params_set_rate jp*/
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{
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printf("alsa-init: unable to set samplerate-2: %s\n",
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@ -259,9 +251,9 @@ static int init(int rate_hz, int channels, int format, int flags)
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return(0);
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} else
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{
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ao_buffersize = err;
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ao_data.buffersize = err;
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if (verbose)
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printf("alsa-init: got buffersize %i\n", ao_buffersize);
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printf("alsa-init: got buffersize %i\n", ao_data.buffersize);
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}
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#endif
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@ -278,7 +270,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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alsa_buffer_time = err;
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if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, alsa_buffer_time/4, 0)) < 0)
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/* original: alsa_buffer_time/ao_bps */
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/* original: alsa_buffer_time/ao_data.bps */
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{
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printf("alsa-init: unable to set period time: %s\n",
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snd_strerror(err));
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@ -299,7 +291,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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#ifdef sw_params
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{
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chunk_size = snd_pcm_hw_params_get_period_size(alsa_hwparams, 0);
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start_threshold = (double) ao_samplerate * start_delay / 1000000;
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start_threshold = (double) ao_data.samplerate * start_delay / 1000000;
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xfer_align = snd_pcm_sw_params_get_xfer_align(alsa_swparams);
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if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0)
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@ -369,7 +361,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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}
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#endif
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printf("AUDIO: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
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ao_samplerate, ao_channels, ao_bps, ao_buffersize,
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ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
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snd_pcm_format_description(alsa_format));
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return(1);
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}
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@ -488,7 +480,7 @@ static int play(void* data, int len, int flags)
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got_len = snd_pcm_writei(alsa_handler, data, len / 4);
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//if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps)) {
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//if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps)) {
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//SHOULD BE FIXED
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if (got_len == -EPIPE) /* underrun? */
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{
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@ -498,7 +490,7 @@ static int play(void* data, int len, int flags)
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printf("alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
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return(0);
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}
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if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps))
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if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps))
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{
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printf("alsa-play: write error after reset: %s - giving up\n",
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snd_strerror(got_len));
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@ -533,7 +525,7 @@ static int get_space()
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case SND_PCM_STATE_OPEN:
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case SND_PCM_STATE_PREPARED:
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case SND_PCM_STATE_RUNNING:
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ret = snd_pcm_status_get_avail(status) * ao_bps;
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ret = snd_pcm_status_get_avail(status) * ao_data.bps;
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break;
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default:
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ret = 0;
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@ -546,11 +538,11 @@ static int get_space()
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return(ret);
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}
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/* how many unplayed bytes are in the buffer */
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static int get_delay()
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/* delay in seconds between first and last sample in buffer */
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static float get_delay()
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{
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snd_pcm_status_t *status;
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int ret;
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float ret;
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if ((ret = snd_pcm_status_malloc(&status)) < 0)
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{
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@ -569,7 +561,7 @@ static int get_delay()
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case SND_PCM_STATE_OPEN:
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case SND_PCM_STATE_PREPARED:
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case SND_PCM_STATE_RUNNING:
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ret = snd_pcm_status_get_delay(status) * ao_bps;
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ret = (float)snd_pcm_status_get_delay(status)/(float)ao_data.samplerate;
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break;
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default:
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ret = 0;
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@ -27,15 +27,6 @@ static ao_info_t info =
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LIBAO_EXTERN(alsa5)
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/* global variables:
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ao_samplerate
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ao_channels
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ao_format
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ao_bps
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ao_outburst
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ao_buffersize
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*/
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static snd_pcm_t *alsa_handler;
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static snd_pcm_format_t alsa_format;
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static int alsa_rate = SND_PCM_RATE_CONTINUOUS;
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@ -74,12 +65,12 @@ static int init(int rate_hz, int channels, int format, int flags)
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return(0);
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}
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ao_format = format;
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ao_channels = channels - 1;
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ao_samplerate = rate_hz;
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ao_bps = ao_samplerate*(ao_channels+1);
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ao_outburst = OUTBURST;
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ao_buffersize = 16384;
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ao_data.format = format;
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ao_data.channels = channels - 1;
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ao_data.samplerate = rate_hz;
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ao_data.bps = ao_data.samplerate*(ao_data.channels+1);
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ao_data.outburst = OUTBURST;
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ao_data.buffersize = 16384;
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memset(&alsa_format, 0, sizeof(alsa_format));
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switch (format)
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@ -111,7 +102,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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{
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case SND_PCM_SFMT_S16_LE:
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case SND_PCM_SFMT_U16_LE:
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ao_bps *= 2;
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ao_data.bps *= 2;
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break;
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case -1:
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printf("alsa-init: invalid format (%s) requested - output disabled\n",
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@ -161,8 +152,8 @@ static int init(int rate_hz, int channels, int format, int flags)
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break;
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}
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alsa_format.rate = ao_samplerate;
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alsa_format.voices = ao_channels*2;
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alsa_format.rate = ao_data.samplerate;
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alsa_format.voices = ao_data.channels*2;
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alsa_format.interleave = 1;
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if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
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@ -189,13 +180,15 @@ static int init(int rate_hz, int channels, int format, int flags)
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printf("alsa-init: pcm channel info error: %s\n", snd_strerror(err));
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return(0);
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}
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#ifndef __QNX__
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if (chninfo.buffer_size)
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ao_buffersize = chninfo.buffer_size;
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ao_data.buffersize = chninfo.buffer_size;
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#endif
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if (verbose)
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printf("alsa-init: setting preferred buffer size from driver: %d bytes\n",
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ao_buffersize);
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ao_data.buffersize);
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}
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memset(¶ms, 0, sizeof(params));
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@ -204,7 +197,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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params.format = alsa_format;
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params.start_mode = SND_PCM_START_DATA;
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params.stop_mode = SND_PCM_STOP_ROLLOVER;
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params.buf.stream.queue_size = ao_buffersize;
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params.buf.stream.queue_size = ao_data.buffersize;
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params.buf.stream.fill = SND_PCM_FILL_NONE;
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if ((err = snd_pcm_channel_params(alsa_handler, ¶ms)) < 0)
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@ -217,8 +210,8 @@ static int init(int rate_hz, int channels, int format, int flags)
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setup.channel = SND_PCM_CHANNEL_PLAYBACK;
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setup.mode = SND_PCM_MODE_STREAM;
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setup.format = alsa_format;
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setup.buf.stream.queue_size = ao_buffersize;
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setup.msbits_per_sample = ao_bps;
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setup.buf.stream.queue_size = ao_data.buffersize;
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setup.msbits_per_sample = ao_data.bps;
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if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
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{
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@ -233,7 +226,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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}
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printf("AUDIO: %d Hz/%d channels/%d bps/%d bytes buffer/%s\n",
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ao_samplerate, ao_channels+1, ao_bps, ao_buffersize,
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ao_data.samplerate, ao_data.channels+1, ao_data.bps, ao_data.buffersize,
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snd_pcm_get_format_name(alsa_format.format));
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return(1);
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}
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@ -357,15 +350,15 @@ static int get_space()
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return(ch_stat.free);
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}
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/* how many unplayed bytes are in the buffer */
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static int get_delay()
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/* delay in seconds between first and last sample in buffer */
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static float get_delay()
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{
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snd_pcm_channel_status_t ch_stat;
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ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
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if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
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return(ao_buffersize); /* error occured */
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return((float)ao_data.buffersize/(float)ao_data.bps); /* error occured */
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else
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return(ch_stat.count);
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return((float)ch_stat.count/(float)ao_data.bps);
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}
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@ -32,14 +32,6 @@ static ao_info_t info =
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LIBAO_EXTERN(alsa9)
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/* global variables:
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ao_samplerate
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ao_channels
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ao_format
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ao_bps
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ao_outburst
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ao_buffersize
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*/
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static snd_pcm_t *alsa_handler;
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static snd_pcm_format_t alsa_format;
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@ -67,7 +59,7 @@ static int control(int cmd, int arg)
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strncpy(alsa_device, (char *)arg, ALSA_DEVICE_SIZE);
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uninit();
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ret = init(ao_samplerate, ao_channels, ao_format, 0);
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ret = init(ao_data.samplerate, ao_data.channels, ao_data.format, 0);
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if (ret == 0)
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return(CONTROL_ERROR);
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else
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@ -112,12 +104,12 @@ static int init(int rate_hz, int channels, int format, int flags)
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return(0);
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}
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ao_samplerate = rate_hz;
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ao_bps = channels; /* really this is bytes per frame so bad varname */
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ao_format = format;
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ao_channels = channels;
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ao_outburst = OUTBURST;
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ao_buffersize = 16384;
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ao_data.samplerate = rate_hz;
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ao_data.bps = channels; /* really this is bytes per frame so bad varname */
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ao_data.format = format;
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ao_data.channels = channels;
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ao_data.outburst = OUTBURST;
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ao_data.buffersize = 16384;
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switch (format)
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{
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@ -148,7 +140,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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{
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case SND_PCM_FORMAT_S16_LE:
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case SND_PCM_FORMAT_U16_LE:
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ao_bps *= 2;
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ao_data.bps *= 2;
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break;
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case -1:
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printf("alsa-init: invalid format (%s) requested - output disabled\n",
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@ -164,8 +156,8 @@ static int init(int rate_hz, int channels, int format, int flags)
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return(0);
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}
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if (ao_subdevice != NULL)
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alsa_device = ao_subdevice;
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if (ao_data.subdevice != NULL)
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alsa_device = ao_data.subdevice;
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if (alsa_device == NULL)
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{
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@ -219,14 +211,14 @@ static int init(int rate_hz, int channels, int format, int flags)
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}
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if ((err = snd_pcm_hw_params_set_channels(alsa_handler, alsa_hwparams,
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ao_channels)) < 0)
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ao_data.channels)) < 0)
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{
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printf("alsa-init: unable to set channels: %s\n",
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snd_strerror(err));
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return(0);
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}
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if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_samplerate, 0)) < 0)
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if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_data.samplerate, 0)) < 0)
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/* was originally only snd_pcm_hw_params_set_rate jp*/
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{
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printf("alsa-init: unable to set samplerate-2: %s\n",
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@ -259,9 +251,9 @@ static int init(int rate_hz, int channels, int format, int flags)
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return(0);
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} else
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{
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ao_buffersize = err;
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ao_data.buffersize = err;
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if (verbose)
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printf("alsa-init: got buffersize %i\n", ao_buffersize);
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printf("alsa-init: got buffersize %i\n", ao_data.buffersize);
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}
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#endif
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@ -278,7 +270,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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alsa_buffer_time = err;
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if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, alsa_buffer_time/4, 0)) < 0)
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/* original: alsa_buffer_time/ao_bps */
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/* original: alsa_buffer_time/ao_data.bps */
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{
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printf("alsa-init: unable to set period time: %s\n",
|
||||
snd_strerror(err));
|
||||
|
|
@ -299,7 +291,7 @@ static int init(int rate_hz, int channels, int format, int flags)
|
|||
#ifdef sw_params
|
||||
{
|
||||
chunk_size = snd_pcm_hw_params_get_period_size(alsa_hwparams, 0);
|
||||
start_threshold = (double) ao_samplerate * start_delay / 1000000;
|
||||
start_threshold = (double) ao_data.samplerate * start_delay / 1000000;
|
||||
xfer_align = snd_pcm_sw_params_get_xfer_align(alsa_swparams);
|
||||
|
||||
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0)
|
||||
|
|
@ -369,7 +361,7 @@ static int init(int rate_hz, int channels, int format, int flags)
|
|||
}
|
||||
#endif
|
||||
printf("AUDIO: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
|
||||
ao_samplerate, ao_channels, ao_bps, ao_buffersize,
|
||||
ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
|
||||
snd_pcm_format_description(alsa_format));
|
||||
return(1);
|
||||
}
|
||||
|
|
@ -488,7 +480,7 @@ static int play(void* data, int len, int flags)
|
|||
|
||||
got_len = snd_pcm_writei(alsa_handler, data, len / 4);
|
||||
|
||||
//if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps)) {
|
||||
//if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps)) {
|
||||
//SHOULD BE FIXED
|
||||
if (got_len == -EPIPE) /* underrun? */
|
||||
{
|
||||
|
|
@ -498,7 +490,7 @@ static int play(void* data, int len, int flags)
|
|||
printf("alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
|
||||
return(0);
|
||||
}
|
||||
if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps))
|
||||
if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps))
|
||||
{
|
||||
printf("alsa-play: write error after reset: %s - giving up\n",
|
||||
snd_strerror(got_len));
|
||||
|
|
@ -533,7 +525,7 @@ static int get_space()
|
|||
case SND_PCM_STATE_OPEN:
|
||||
case SND_PCM_STATE_PREPARED:
|
||||
case SND_PCM_STATE_RUNNING:
|
||||
ret = snd_pcm_status_get_avail(status) * ao_bps;
|
||||
ret = snd_pcm_status_get_avail(status) * ao_data.bps;
|
||||
break;
|
||||
default:
|
||||
ret = 0;
|
||||
|
|
@ -546,11 +538,11 @@ static int get_space()
|
|||
return(ret);
|
||||
}
|
||||
|
||||
/* how many unplayed bytes are in the buffer */
|
||||
static int get_delay()
|
||||
/* delay in seconds between first and last sample in buffer */
|
||||
static float get_delay()
|
||||
{
|
||||
snd_pcm_status_t *status;
|
||||
int ret;
|
||||
float ret;
|
||||
|
||||
if ((ret = snd_pcm_status_malloc(&status)) < 0)
|
||||
{
|
||||
|
|
@ -569,7 +561,7 @@ static int get_delay()
|
|||
case SND_PCM_STATE_OPEN:
|
||||
case SND_PCM_STATE_PREPARED:
|
||||
case SND_PCM_STATE_RUNNING:
|
||||
ret = snd_pcm_status_get_delay(status) * ao_bps;
|
||||
ret = (float)snd_pcm_status_get_delay(status)/(float)ao_data.samplerate;
|
||||
break;
|
||||
default:
|
||||
ret = 0;
|
||||
|
|
|
|||
|
|
@ -16,13 +16,6 @@ static ao_info_t info =
|
|||
|
||||
LIBAO_EXTERN(mpegpes)
|
||||
|
||||
// there are some globals:
|
||||
// ao_samplerate
|
||||
// ao_channels
|
||||
// ao_format
|
||||
// ao_bps
|
||||
// ao_outburst
|
||||
// ao_buffersize
|
||||
|
||||
// to set/get/query special features/parameters
|
||||
static int control(int cmd,int arg){
|
||||
|
|
@ -33,8 +26,8 @@ static int control(int cmd,int arg){
|
|||
// return: 1=success 0=fail
|
||||
static int init(int rate,int channels,int format,int flags){
|
||||
|
||||
ao_outburst=2000;
|
||||
ao_format=format;
|
||||
ao_data.outburst=2000;
|
||||
ao_data.format=format;
|
||||
|
||||
return 1;
|
||||
}
|
||||
|
|
@ -67,10 +60,10 @@ extern int vo_pts;
|
|||
|
||||
// return: how many bytes can be played without blocking
|
||||
static int get_space(){
|
||||
float x=(float)(vo_pts-ao_pts)/90000.0-0.5;
|
||||
float x=(float)(vo_pts-ao_data.pts)/90000.0-0.5;
|
||||
int y;
|
||||
if(x<=0) return 0;
|
||||
y=48000*4*x;y/=ao_outburst;y*=ao_outburst;
|
||||
y=48000*4*x;y/=ao_data.outburst;y*=ao_data.outburst;
|
||||
// printf("diff: %5.3f -> %d \n",x,y);
|
||||
return y;
|
||||
}
|
||||
|
|
@ -79,20 +72,20 @@ static int get_space(){
|
|||
// it should round it down to outburst*n
|
||||
// return: number of bytes played
|
||||
static int play(void* data,int len,int flags){
|
||||
if(ao_format==AFMT_MPEG)
|
||||
send_pes_packet(data,len,0x1C0,ao_pts);
|
||||
if(ao_data.format==AFMT_MPEG)
|
||||
send_pes_packet(data,len,0x1C0,ao_data.pts);
|
||||
else {
|
||||
int i;
|
||||
unsigned short *s=data;
|
||||
for(i=0;i<len/2;i++) s[i]=(s[i]>>8)|(s[i]<<8); // le<->be
|
||||
send_lpcm_packet(data,len,0xA0,ao_pts);
|
||||
send_lpcm_packet(data,len,0xA0,ao_data.pts);
|
||||
}
|
||||
return len;
|
||||
}
|
||||
|
||||
// return: how many unplayed bytes are in the buffer
|
||||
static int get_delay(){
|
||||
// return: delay in seconds between first and last sample in buffer
|
||||
static float get_delay(){
|
||||
|
||||
return 0;
|
||||
return 0.0;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -14,13 +14,6 @@ static ao_info_t info =
|
|||
|
||||
LIBAO_EXTERN(null)
|
||||
|
||||
// there are some globals:
|
||||
// ao_samplerate
|
||||
// ao_channels
|
||||
// ao_format
|
||||
// ao_bps
|
||||
// ao_outburst
|
||||
// ao_buffersize
|
||||
|
||||
// to set/get/query special features/parameters
|
||||
static int control(int cmd,int arg){
|
||||
|
|
@ -31,7 +24,7 @@ static int control(int cmd,int arg){
|
|||
// return: 1=success 0=fail
|
||||
static int init(int rate,int channels,int format,int flags){
|
||||
|
||||
ao_outburst=4096;
|
||||
ao_data.outburst=4096;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
|
@ -61,7 +54,7 @@ static void audio_resume()
|
|||
// return: how many bytes can be played without blocking
|
||||
static int get_space(){
|
||||
|
||||
return ao_outburst;
|
||||
return ao_data.outburst;
|
||||
}
|
||||
|
||||
// plays 'len' bytes of 'data'
|
||||
|
|
@ -72,10 +65,10 @@ static int play(void* data,int len,int flags){
|
|||
return len;
|
||||
}
|
||||
|
||||
// return: how many unplayed bytes are in the buffer
|
||||
static int get_delay(){
|
||||
// return: delay in seconds between first and last sample in buffer
|
||||
static float get_delay(){
|
||||
|
||||
return 0;
|
||||
return 0.0;
|
||||
}
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -28,14 +28,6 @@ static ao_info_t info =
|
|||
|
||||
LIBAO_EXTERN(oss)
|
||||
|
||||
// there are some globals:
|
||||
// ao_samplerate
|
||||
// ao_channels
|
||||
// ao_format
|
||||
// ao_bps
|
||||
// ao_outburst
|
||||
// ao_buffersize
|
||||
|
||||
static char *dsp="/dev/dsp";
|
||||
static audio_buf_info zz;
|
||||
static int audio_fd=-1;
|
||||
|
|
@ -57,7 +49,7 @@ static int control(int cmd,int arg){
|
|||
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
|
||||
int fd, v, mcmd, devs;
|
||||
|
||||
if(ao_format == AFMT_AC3)
|
||||
if(ao_data.format == AFMT_AC3)
|
||||
return CONTROL_TRUE;
|
||||
|
||||
if ((fd = open("/dev/mixer", O_RDONLY)) > 0)
|
||||
|
|
@ -118,58 +110,62 @@ static int init(int rate,int channels,int format,int flags){
|
|||
return 0;
|
||||
}
|
||||
|
||||
ao_format=format;
|
||||
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_format);
|
||||
if(format == AFMT_AC3 && ao_format != AFMT_AC3) {
|
||||
ao_data.bps=(channels+1)*rate;
|
||||
if(format != AFMT_U8 && format != AFMT_S8)
|
||||
ao_data.bps*=2;
|
||||
|
||||
ao_data.format=format;
|
||||
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
|
||||
if(format == AFMT_AC3 && ao_data.format != AFMT_AC3) {
|
||||
printf("Can't set audio device %s to AC3 output\n", dsp);
|
||||
return 0;
|
||||
}
|
||||
printf("audio_setup: sample format: %s (requested: %s)\n",
|
||||
audio_out_format_name(ao_format), audio_out_format_name(format));
|
||||
audio_out_format_name(ao_data.format), audio_out_format_name(format));
|
||||
|
||||
if(format != AFMT_AC3) {
|
||||
ao_channels=channels-1;
|
||||
ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_channels);
|
||||
ao_data.channels=channels-1;
|
||||
ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_data.channels);
|
||||
|
||||
// set rate
|
||||
ao_samplerate=rate;
|
||||
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_samplerate);
|
||||
printf("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_samplerate,rate);
|
||||
ao_data.samplerate=rate;
|
||||
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
|
||||
printf("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
|
||||
}
|
||||
|
||||
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
|
||||
int r=0;
|
||||
printf("audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
|
||||
if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
|
||||
printf("audio_setup: %d bytes/frag (config.h)\n",ao_outburst);
|
||||
printf("audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
|
||||
} else {
|
||||
ao_outburst=r;
|
||||
printf("audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_outburst);
|
||||
ao_data.outburst=r;
|
||||
printf("audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
|
||||
}
|
||||
} else {
|
||||
printf("audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
|
||||
zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
|
||||
if(ao_buffersize==-1) ao_buffersize=zz.bytes;
|
||||
ao_outburst=zz.fragsize;
|
||||
if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
|
||||
ao_data.outburst=zz.fragsize;
|
||||
}
|
||||
|
||||
if(ao_buffersize==-1){
|
||||
if(ao_data.buffersize==-1){
|
||||
// Measuring buffer size:
|
||||
void* data;
|
||||
ao_buffersize=0;
|
||||
ao_data.buffersize=0;
|
||||
#ifdef HAVE_AUDIO_SELECT
|
||||
data=malloc(ao_outburst); memset(data,0,ao_outburst);
|
||||
while(ao_buffersize<0x40000){
|
||||
data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
|
||||
while(ao_data.buffersize<0x40000){
|
||||
fd_set rfds;
|
||||
struct timeval tv;
|
||||
FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
|
||||
tv.tv_sec=0; tv.tv_usec = 0;
|
||||
if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
|
||||
write(audio_fd,data,ao_outburst);
|
||||
ao_buffersize+=ao_outburst;
|
||||
write(audio_fd,data,ao_data.outburst);
|
||||
ao_data.buffersize+=ao_data.outburst;
|
||||
}
|
||||
free(data);
|
||||
if(ao_buffersize==0){
|
||||
if(ao_data.buffersize==0){
|
||||
printf("\n *** Your audio driver DOES NOT support select() ***\n");
|
||||
printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
|
||||
return 0;
|
||||
|
|
@ -197,10 +193,10 @@ static void reset(){
|
|||
return;
|
||||
}
|
||||
|
||||
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_format);
|
||||
if(ao_format != AFMT_AC3) {
|
||||
ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_channels);
|
||||
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_samplerate);
|
||||
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
|
||||
if(ao_data.format != AFMT_AC3) {
|
||||
ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_data.channels);
|
||||
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -219,7 +215,7 @@ static void audio_resume()
|
|||
|
||||
// return: how many bytes can be played without blocking
|
||||
static int get_space(){
|
||||
int playsize=ao_outburst;
|
||||
int playsize=ao_data.outburst;
|
||||
|
||||
#ifdef SNDCTL_DSP_GETOSPACE
|
||||
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
|
||||
|
|
@ -240,35 +236,38 @@ static int get_space(){
|
|||
}
|
||||
#endif
|
||||
|
||||
return ao_outburst;
|
||||
return ao_data.outburst;
|
||||
}
|
||||
|
||||
// plays 'len' bytes of 'data'
|
||||
// it should round it down to outburst*n
|
||||
// return: number of bytes played
|
||||
static int play(void* data,int len,int flags){
|
||||
len/=ao_outburst;
|
||||
len=write(audio_fd,data,len*ao_outburst);
|
||||
len/=ao_data.outburst;
|
||||
len=write(audio_fd,data,len*ao_data.outburst);
|
||||
return len;
|
||||
}
|
||||
|
||||
static int audio_delay_method=2;
|
||||
|
||||
// return: how many unplayed bytes are in the buffer
|
||||
static int get_delay(){
|
||||
// return: delay in seconds between first and last sample in buffer
|
||||
static float get_delay(){
|
||||
/* Calculate how many bytes/second is sent out */
|
||||
if(audio_delay_method==2){
|
||||
//
|
||||
int r=0;
|
||||
if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
|
||||
return r;
|
||||
return ((float)r)/(float)ao_data.bps;
|
||||
audio_delay_method=1; // fallback if not supported
|
||||
}
|
||||
if(audio_delay_method==1){
|
||||
// SNDCTL_DSP_GETOSPACE
|
||||
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
|
||||
return ao_buffersize-zz.bytes;
|
||||
return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
|
||||
audio_delay_method=0; // fallback if not supported
|
||||
}
|
||||
return ao_buffersize;
|
||||
return ((float)ao_data.buffersize)/(float)ao_data.bps;
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -14,14 +14,6 @@ static ao_info_t info =
|
|||
|
||||
LIBAO_EXTERN(pcm)
|
||||
|
||||
// there are some globals:
|
||||
// ao_samplerate
|
||||
// ao_channels
|
||||
// ao_format
|
||||
// ao_bps
|
||||
// ao_outburst
|
||||
// ao_buffersize
|
||||
|
||||
char *ao_outputfilename = NULL;
|
||||
int ao_pcm_waveheader = 1;
|
||||
|
||||
|
|
@ -90,7 +82,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
printf("PCM: Info - to write WAVE files use -waveheader (default), for RAW PCM -nowaveheader.\n");
|
||||
fp = fopen(ao_outputfilename, "wb");
|
||||
|
||||
ao_outburst = 4096;
|
||||
ao_data.outburst = 4096;
|
||||
|
||||
|
||||
if(fp) {
|
||||
|
|
@ -134,7 +126,7 @@ static void audio_resume()
|
|||
// return: how many bytes can be played without blocking
|
||||
static int get_space(){
|
||||
|
||||
return ao_outburst;
|
||||
return ao_data.outburst;
|
||||
}
|
||||
|
||||
// plays 'len' bytes of 'data'
|
||||
|
|
@ -151,10 +143,10 @@ static int play(void* data,int len,int flags){
|
|||
return len;
|
||||
}
|
||||
|
||||
// return: how many unplayed bytes are in the buffer
|
||||
static int get_delay(){
|
||||
// return: delay in seconds between first and last sample in buffer
|
||||
static float get_delay(){
|
||||
|
||||
return 0;
|
||||
return 0.0;
|
||||
}
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -29,13 +29,6 @@ static ao_info_t info =
|
|||
|
||||
LIBAO_EXTERN(sdl)
|
||||
|
||||
// there are some globals:
|
||||
// ao_samplerate
|
||||
// ao_channels
|
||||
// ao_format
|
||||
// ao_bps
|
||||
// ao_outburst
|
||||
// ao_buffersize
|
||||
|
||||
extern int verbose;
|
||||
|
||||
|
|
@ -150,6 +143,10 @@ static int init(int rate,int channels,int format,int flags){
|
|||
setenv("SDL_AUDIODRIVER", ao_subdevice, 1);
|
||||
printf("SDL: using %s audio driver\n", ao_subdevice);
|
||||
}
|
||||
|
||||
ao_data.bps=(channels+1)*rate;
|
||||
if(format != AFMT_U8 && format != AFMT_S8)
|
||||
ao_data.bps*=2;
|
||||
|
||||
/* The desired audio format (see SDL_AudioSpec) */
|
||||
switch(format) {
|
||||
|
|
@ -205,7 +202,7 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
|
|||
}
|
||||
|
||||
if(verbose) printf("SDL: buf size = %d\n",aspec.size);
|
||||
if(ao_buffersize==-1) ao_buffersize=aspec.size;
|
||||
if(ao_data.buffersize==-1) ao_data.buffersize=aspec.size;
|
||||
|
||||
/* unsilence audio, if callback is ready */
|
||||
SDL_PauseAudio(0);
|
||||
|
|
@ -278,9 +275,9 @@ static int play(void* data,int len,int flags){
|
|||
#endif
|
||||
}
|
||||
|
||||
// return: how many unplayed bytes are in the buffer
|
||||
static int get_delay(){
|
||||
return buffered_bytes + ao_buffersize;
|
||||
// return: delay in seconds between first and last sample in buffer
|
||||
static float get_delay(){
|
||||
return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
|
||||
}
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -22,13 +22,6 @@ static ao_info_t info =
|
|||
|
||||
LIBAO_EXTERN(sgi)
|
||||
|
||||
// there are some globals:
|
||||
// ao_samplerate
|
||||
// ao_channels
|
||||
// ao_format
|
||||
// ao_bps
|
||||
// ao_outburst
|
||||
// ao_buffersize
|
||||
|
||||
static ALconfig ao_config;
|
||||
static ALport ao_port;
|
||||
|
|
@ -86,9 +79,9 @@ static int init(int rate, int channels, int format, int flags) {
|
|||
|
||||
}
|
||||
|
||||
ao_buffersize=131072;
|
||||
ao_outburst = ao_buffersize/16;
|
||||
ao_channels = channels;
|
||||
ao_data.buffersize=131072;
|
||||
ao_data.outburst = ao_data.buffersize/16;
|
||||
ao_data.channels = channels;
|
||||
|
||||
ao_config = alNewConfig();
|
||||
|
||||
|
|
@ -164,7 +157,7 @@ static int get_space() {
|
|||
// printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_outburst);
|
||||
// printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
|
||||
|
||||
return alGetFillable(ao_port)*(2*ao_channels);
|
||||
return alGetFillable(ao_port)*(2*ao_data.channels);
|
||||
|
||||
}
|
||||
|
||||
|
|
@ -177,14 +170,14 @@ static int play(void* data, int len, int flags) {
|
|||
// printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config);
|
||||
// printf("channels %d\n", ao_channels);
|
||||
|
||||
alWriteFrames(ao_port, data, len/(2*ao_channels));
|
||||
alWriteFrames(ao_port, data, len/(2*ao_data.channels));
|
||||
|
||||
return len;
|
||||
|
||||
}
|
||||
|
||||
// return: how many unplayed bytes are in the buffer
|
||||
static int get_delay(){
|
||||
// return: delay in seconds between first and last sample in buffer
|
||||
static float get_delay(){
|
||||
|
||||
// printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize);
|
||||
|
||||
|
|
|
|||
|
|
@ -42,18 +42,11 @@ LIBAO_EXTERN(sun)
|
|||
#endif
|
||||
|
||||
|
||||
// there are some globals:
|
||||
// ao_samplerate
|
||||
// ao_channels
|
||||
// ao_format
|
||||
// ao_bps
|
||||
// ao_outburst
|
||||
// ao_buffersize
|
||||
|
||||
static char *audio_dev = "/dev/audio";
|
||||
static int queued_bursts = 0;
|
||||
static int queued_samples = 0;
|
||||
static int bytes_per_sample = 0;
|
||||
static int byte_per_sec = 0;
|
||||
static int convert_u8_s8;
|
||||
static int audio_fd = -1;
|
||||
static enum {
|
||||
|
|
@ -230,7 +223,6 @@ static int control(int cmd,int arg){
|
|||
static int init(int rate,int channels,int format,int flags){
|
||||
|
||||
audio_info_t info;
|
||||
int byte_per_sec;
|
||||
int ok;
|
||||
|
||||
if (ao_subdevice) audio_dev = ao_subdevice;
|
||||
|
|
@ -252,13 +244,13 @@ static int init(int rate,int channels,int format,int flags){
|
|||
ioctl(audio_fd, AUDIO_DRAIN, 0);
|
||||
|
||||
AUDIO_INITINFO(&info);
|
||||
info.play.encoding = oss2sunfmt(ao_format = format);
|
||||
info.play.encoding = oss2sunfmt(ao_data.format = format);
|
||||
info.play.precision =
|
||||
(format==AFMT_S16_LE || format==AFMT_S16_BE
|
||||
? AUDIO_PRECISION_16
|
||||
: AUDIO_PRECISION_8);
|
||||
info.play.channels = ao_channels = channels;
|
||||
info.play.sample_rate = ao_samplerate = rate;
|
||||
info.play.channels = ao_data.channels = channels;
|
||||
info.play.sample_rate = ao_data.samplerate = rate;
|
||||
convert_u8_s8 = 0;
|
||||
ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
|
||||
if (!ok && info.play.encoding == AUDIO_ENCODING_LINEAR8) {
|
||||
|
|
@ -278,37 +270,37 @@ static int init(int rate,int channels,int format,int flags){
|
|||
|
||||
bytes_per_sample = channels * info.play.precision / 8;
|
||||
byte_per_sec = bytes_per_sample * rate;
|
||||
ao_outburst = byte_per_sec > 100000 ? 16384 : 8192;
|
||||
ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192;
|
||||
|
||||
#ifdef __not_used__
|
||||
/*
|
||||
* hmm, ao_buffersize is currently not used in this driver, do there's
|
||||
* hmm, ao_data.buffersize is currently not used in this driver, do there's
|
||||
* no need to measure it
|
||||
*/
|
||||
if(ao_buffersize==-1){
|
||||
if(ao_data.buffersize==-1){
|
||||
// Measuring buffer size:
|
||||
void* data;
|
||||
ao_buffersize=0;
|
||||
ao_data.buffersize=0;
|
||||
#ifdef HAVE_AUDIO_SELECT
|
||||
data = malloc(ao_outburst);
|
||||
memset(data, format==AFMT_U8 ? 0x80 : 0, ao_outburst);
|
||||
while(ao_buffersize<0x40000){
|
||||
data = malloc(ao_data.outburst);
|
||||
memset(data, format==AFMT_U8 ? 0x80 : 0, ao_data.outburst);
|
||||
while(ao_data.buffersize<0x40000){
|
||||
fd_set rfds;
|
||||
struct timeval tv;
|
||||
FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
|
||||
tv.tv_sec=0; tv.tv_usec = 0;
|
||||
if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
|
||||
write(audio_fd,data,ao_outburst);
|
||||
ao_buffersize+=ao_outburst;
|
||||
write(audio_fd,data,ao_data.outburst);
|
||||
ao_data.buffersize+=ao_data.outburst;
|
||||
}
|
||||
free(data);
|
||||
if(ao_buffersize==0){
|
||||
if(ao_data.buffersize==0){
|
||||
printf("\n *** Your audio driver DOES NOT support select() ***\n");
|
||||
printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
|
||||
return 0;
|
||||
}
|
||||
#ifdef __svr4__
|
||||
// remove the 0 bytes from the above ao_buffersize measurement from the
|
||||
// remove the 0 bytes from the above ao_data.buffersize measurement from the
|
||||
// audio driver's STREAMS queue
|
||||
ioctl(audio_fd, I_FLUSH, FLUSHW);
|
||||
#endif
|
||||
|
|
@ -352,13 +344,13 @@ static void reset(){
|
|||
ioctl(audio_fd, AUDIO_DRAIN, 0);
|
||||
|
||||
AUDIO_INITINFO(&info);
|
||||
info.play.encoding = oss2sunfmt(ao_format);
|
||||
info.play.encoding = oss2sunfmt(ao_data.format);
|
||||
info.play.precision =
|
||||
(ao_format==AFMT_S16_LE || ao_format==AFMT_S16_BE
|
||||
(ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_S16_BE
|
||||
? AUDIO_PRECISION_16
|
||||
: AUDIO_PRECISION_8);
|
||||
info.play.channels = ao_channels;
|
||||
info.play.sample_rate = ao_samplerate;
|
||||
info.play.channels = ao_data.channels;
|
||||
info.play.sample_rate = ao_data.samplerate;
|
||||
info.play.samples = 0;
|
||||
info.play.eof = 0;
|
||||
info.play.error = 0;
|
||||
|
|
@ -388,7 +380,7 @@ static void audio_resume()
|
|||
|
||||
// return: how many bytes can be played without blocking
|
||||
static int get_space(){
|
||||
int playsize = ao_outburst;
|
||||
int playsize = ao_data.outburst;
|
||||
audio_info_t info;
|
||||
|
||||
// check buffer
|
||||
|
|
@ -408,7 +400,7 @@ static int get_space(){
|
|||
if (queued_bursts - info.play.eof > 2)
|
||||
return 0;
|
||||
|
||||
return ao_outburst;
|
||||
return ao_data.outburst;
|
||||
}
|
||||
|
||||
// plays 'len' bytes of 'data'
|
||||
|
|
@ -421,13 +413,13 @@ static int play(void* data,int len,int flags){
|
|||
int native_endian = AFMT_S16_LE;
|
||||
#endif
|
||||
|
||||
if (len < ao_outburst) return 0;
|
||||
len /= ao_outburst;
|
||||
len *= ao_outburst;
|
||||
if (len < ao_data.outburst) return 0;
|
||||
len /= ao_data.outburst;
|
||||
len *= ao_data.outburst;
|
||||
|
||||
/* 16-bit format using the 'wrong' byteorder? swap words */
|
||||
if ((ao_format == AFMT_S16_LE || ao_format == AFMT_S16_BE)
|
||||
&& ao_format != native_endian) {
|
||||
if ((ao_data.format == AFMT_S16_LE || ao_data.format == AFMT_S16_BE)
|
||||
&& ao_data.format != native_endian) {
|
||||
static void *swab_buf;
|
||||
static int swab_len;
|
||||
if (len > swab_len) {
|
||||
|
|
@ -440,7 +432,7 @@ static int play(void* data,int len,int flags){
|
|||
}
|
||||
swab(data, swab_buf, len);
|
||||
data = swab_buf;
|
||||
} else if (ao_format == AFMT_U8 && convert_u8_s8) {
|
||||
} else if (ao_data.format == AFMT_U8 && convert_u8_s8) {
|
||||
int i;
|
||||
unsigned char *p = data;
|
||||
|
||||
|
|
@ -460,13 +452,13 @@ static int play(void* data,int len,int flags){
|
|||
}
|
||||
|
||||
|
||||
// return: how many unplayed bytes are in the buffer
|
||||
static int get_delay(){
|
||||
// return: delay in seconds between first and last sample in buffer
|
||||
static float get_delay(){
|
||||
audio_info_t info;
|
||||
ioctl(audio_fd, AUDIO_GETINFO, &info);
|
||||
if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
|
||||
return (queued_samples - info.play.samples) * bytes_per_sample;
|
||||
return (float)(queued_samples - info.play.samples) / (float)byte_per_sec;
|
||||
else
|
||||
return (queued_bursts - info.play.eof) * ao_outburst;
|
||||
return (flaot)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -6,19 +6,12 @@
|
|||
#include "afmt.h"
|
||||
|
||||
// there are some globals:
|
||||
int ao_samplerate=0;
|
||||
int ao_channels=0;
|
||||
int ao_format=0;
|
||||
int ao_bps=0;
|
||||
int ao_outburst=OUTBURST; // config.h default
|
||||
int ao_buffersize=-1;
|
||||
int ao_pts=0;
|
||||
ao_data_t ao_data={0,0,0,0,OUTBURST,-1,0};
|
||||
char *ao_subdevice = NULL;
|
||||
|
||||
#ifdef USE_OSS_AUDIO
|
||||
extern ao_functions_t audio_out_oss;
|
||||
#endif
|
||||
//extern ao_functions_t audio_out_ossold;
|
||||
extern ao_functions_t audio_out_null;
|
||||
#ifdef HAVE_ALSA5
|
||||
extern ao_functions_t audio_out_alsa5;
|
||||
|
|
@ -42,10 +35,9 @@ extern ao_functions_t audio_out_sgi;
|
|||
extern ao_functions_t audio_out_dxr3;
|
||||
#endif
|
||||
extern ao_functions_t audio_out_pcm;
|
||||
#ifndef USE_LIBVO2
|
||||
extern ao_functions_t audio_out_mpegpes;
|
||||
#endif
|
||||
extern ao_functions_t audio_out_pss;
|
||||
extern ao_functions_t audio_out_plugin;
|
||||
|
||||
ao_functions_t* audio_out_drivers[] =
|
||||
{
|
||||
|
|
@ -75,9 +67,11 @@ ao_functions_t* audio_out_drivers[] =
|
|||
&audio_out_dxr3,
|
||||
#endif
|
||||
&audio_out_pcm,
|
||||
#ifndef USE_LIBVO2
|
||||
&audio_out_mpegpes,
|
||||
#endif
|
||||
&audio_out_plugin,
|
||||
// &audio_out_pss,
|
||||
NULL
|
||||
};
|
||||
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -10,6 +10,7 @@ typedef struct ao_info_s
|
|||
const char *comment;
|
||||
} ao_info_t;
|
||||
|
||||
/* interface towards mplayer and */
|
||||
typedef struct ao_functions_s
|
||||
{
|
||||
ao_info_t *info;
|
||||
|
|
@ -19,25 +20,32 @@ typedef struct ao_functions_s
|
|||
void (*reset)();
|
||||
int (*get_space)();
|
||||
int (*play)(void* data,int len,int flags);
|
||||
int (*get_delay)();
|
||||
float (*get_delay)();
|
||||
void (*pause)();
|
||||
void (*resume)();
|
||||
} ao_functions_t;
|
||||
|
||||
/* global data used by mplayer and plugins */
|
||||
typedef struct ao_data_s
|
||||
{
|
||||
int samplerate;
|
||||
int channels;
|
||||
int format;
|
||||
int bps;
|
||||
int outburst;
|
||||
int buffersize;
|
||||
int pts;
|
||||
} ao_data_t;
|
||||
|
||||
extern char *ao_subdevice;
|
||||
extern ao_data_t ao_data;
|
||||
|
||||
// prototypes
|
||||
extern char *audio_out_format_name(int format);
|
||||
|
||||
// NULL terminated array of all drivers
|
||||
extern ao_functions_t* audio_out_drivers[];
|
||||
|
||||
extern int ao_samplerate;
|
||||
extern int ao_channels;
|
||||
extern int ao_format;
|
||||
extern int ao_bps;
|
||||
extern int ao_outburst;
|
||||
extern int ao_buffersize;
|
||||
extern int ao_pts;
|
||||
extern char *ao_subdevice;
|
||||
|
||||
#define CONTROL_OK 1
|
||||
#define CONTROL_TRUE 1
|
||||
|
|
|
|||
|
|
@ -7,7 +7,7 @@ static void uninit();
|
|||
static void reset();
|
||||
static int get_space();
|
||||
static int play(void* data,int len,int flags);
|
||||
static int get_delay();
|
||||
static float get_delay();
|
||||
static void audio_pause();
|
||||
static void audio_resume();
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue