mirror of
https://github.com/mpv-player/mpv
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c8ae1836e6
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24615 b3059339-0415-0410-9bf9-f77b7e298cf2
301 lines
7.2 KiB
C
301 lines
7.2 KiB
C
/*
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ao_sgi - sgi/irix output plugin for MPlayer
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22oct2001 oliver.schoenbrunner@jku.at
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <errno.h>
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#include <dmedia/audio.h>
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "mp_msg.h"
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#include "help_mp.h"
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#include "libaf/af_format.h"
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static ao_info_t info =
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{
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"sgi audio output",
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"sgi",
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"Oliver Schoenbrunner",
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""
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};
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LIBAO_EXTERN(sgi)
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static ALconfig ao_config;
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static ALport ao_port;
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static int sample_rate;
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static int queue_size;
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static int bytes_per_frame;
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/**
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* \param [in/out] format
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* \param [out] width
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*
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* \return the closest matching SGI AL sample format
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*
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* \note width is set to required per-channel sample width
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* format is updated to match the SGI AL sample format
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*/
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static int fmt2sgial(int *format, int *width) {
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int smpfmt = AL_SAMPFMT_TWOSCOMP;
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/* SGI AL only supports float and signed integers in native
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* endianness. If this is something else, we must rely on the audio
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* filter to convert it to a compatible format. */
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/* 24-bit audio is supported, but only with 32-bit alignment.
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* mplayer's 24-bit format is packed, unfortunately.
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* So we must upgrade 24-bit requests to 32 bits. Then we drop the
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* lowest 8 bits during playback. */
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switch(*format) {
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case AF_FORMAT_U8:
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case AF_FORMAT_S8:
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*width = AL_SAMPLE_8;
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*format = AF_FORMAT_S8;
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break;
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case AF_FORMAT_U16_LE:
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case AF_FORMAT_U16_BE:
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case AF_FORMAT_S16_LE:
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case AF_FORMAT_S16_BE:
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*width = AL_SAMPLE_16;
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*format = AF_FORMAT_S16_NE;
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break;
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case AF_FORMAT_U24_LE:
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case AF_FORMAT_U24_BE:
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case AF_FORMAT_S24_LE:
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case AF_FORMAT_S24_BE:
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case AF_FORMAT_U32_LE:
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case AF_FORMAT_U32_BE:
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case AF_FORMAT_S32_LE:
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case AF_FORMAT_S32_BE:
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*width = AL_SAMPLE_24;
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*format = AF_FORMAT_S32_NE;
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break;
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case AF_FORMAT_FLOAT_LE:
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case AF_FORMAT_FLOAT_BE:
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*width = 4;
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*format = AF_FORMAT_FLOAT_NE;
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smpfmt = AL_SAMPFMT_FLOAT;
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break;
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default:
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*width = AL_SAMPLE_16;
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*format = AF_FORMAT_S16_NE;
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break;
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}
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return smpfmt;
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}
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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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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_INFO);
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switch(cmd) {
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case AOCONTROL_QUERY_FORMAT:
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/* Do not reject any format: return the closest matching
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* format if the request is not supported natively. */
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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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, int channels, int format, int flags) {
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int smpwidth, smpfmt;
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int rv = AL_DEFAULT_OUTPUT;
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smpfmt = fmt2sgial(&format, &smpwidth);
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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
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{ /* from /usr/share/src/dmedia/audio/setrate.c */
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double frate, realrate;
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ALpv x[2];
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if(ao_subdevice) {
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rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE);
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if (!rv) {
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice);
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return 0;
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}
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}
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frate = rate;
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x[0].param = AL_RATE;
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x[0].value.ll = alDoubleToFixed(rate);
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x[1].param = AL_MASTER_CLOCK;
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x[1].value.i = AL_CRYSTAL_MCLK_TYPE;
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if (alSetParams(rv,x, 2)<0) {
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mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror()));
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}
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if (x[0].sizeOut < 0) {
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mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate);
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}
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if (alGetParams(rv,x, 1)<0) {
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mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror()));
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}
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realrate = alFixedToDouble(x[0].value.ll);
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if (frate != realrate) {
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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, realrate, frate);
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}
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sample_rate = (int)realrate;
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}
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bytes_per_frame = channels * smpwidth;
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ao_data.samplerate = sample_rate;
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ao_data.channels = channels;
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ao_data.format = format;
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ao_data.bps = sample_rate * bytes_per_frame;
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ao_data.buffersize=131072;
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ao_data.outburst = ao_data.buffersize/16;
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ao_config = alNewConfig();
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if (!ao_config) {
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
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return 0;
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}
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if(alSetChannels(ao_config, channels) < 0 ||
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alSetWidth(ao_config, smpwidth) < 0 ||
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alSetSampFmt(ao_config, smpfmt) < 0 ||
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alSetQueueSize(ao_config, sample_rate) < 0 ||
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alSetDevice(ao_config, rv) < 0) {
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
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return 0;
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}
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ao_port = alOpenPort("mplayer", "w", ao_config);
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if (!ao_port) {
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, alGetErrorString(oserror()));
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return 0;
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}
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// printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
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queue_size = alGetQueueSize(ao_config);
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return 1;
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}
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// close audio device
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static void uninit(int immed) {
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/* TODO: samplerate should be set back to the value before mplayer was started! */
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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Uninit);
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if (ao_config) {
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alFreeConfig(ao_config);
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ao_config = NULL;
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}
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if (ao_port) {
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if (!immed)
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while(alGetFilled(ao_port) > 0) sginap(1);
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alClosePort(ao_port);
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ao_port = NULL;
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}
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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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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Reset);
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alDiscardFrames(ao_port, queue_size);
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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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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_PauseInfo);
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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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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_ResumeInfo);
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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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// printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_data.outburst);
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// printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
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return alGetFillable(ao_port) * bytes_per_frame;
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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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/* Always process data in quadword-aligned chunks (64-bits). */
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const int plen = len / (sizeof(uint64_t) * bytes_per_frame);
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const int framecount = plen * sizeof(uint64_t);
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// printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config);
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// printf("channels %d\n", ao_data.channels);
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if(ao_data.format == AF_FORMAT_S32_NE) {
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/* The zen of this is explained in fmt2sgial() */
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int32_t *smpls = data;
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const int32_t *smple = smpls + (framecount * ao_data.channels);
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while(smpls < smple)
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*smpls++ >>= 8;
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}
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alWriteFrames(ao_port, data, framecount);
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return framecount * bytes_per_frame;
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}
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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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// printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize);
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// return (float)queue_size/((float)sample_rate);
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const int outstanding = alGetFilled(ao_port);
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return (float)((outstanding < 0) ? queue_size : outstanding) /
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((float)sample_rate);
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}
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