mpv/libao2/ao_sgi.c

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/*
ao_sgi - sgi/irix output plugin for MPlayer
22oct2001 oliver.schoenbrunner@jku.at
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <dmedia/audio.h>
#include "audio_out.h"
#include "audio_out_internal.h"
#include "mp_msg.h"
#include "help_mp.h"
#include "libaf/af_format.h"
static ao_info_t info =
{
"sgi audio output",
"sgi",
"Oliver Schoenbrunner",
""
};
LIBAO_EXTERN(sgi)
static ALconfig ao_config;
static ALport ao_port;
static int sample_rate;
static int queue_size;
static int bytes_per_frame;
/**
* \param [in/out] format
* \param [out] width
*
* \return the closest matching SGI AL sample format
*
* \note width is set to required per-channel sample width
* format is updated to match the SGI AL sample format
*/
static int fmt2sgial(int *format, int *width) {
int smpfmt = AL_SAMPFMT_TWOSCOMP;
/* SGI AL only supports float and signed integers in native
* endianness. If this is something else, we must rely on the audio
* filter to convert it to a compatible format. */
/* 24-bit audio is supported, but only with 32-bit alignment.
* mplayer's 24-bit format is packed, unfortunately.
* So we must upgrade 24-bit requests to 32 bits. Then we drop the
* lowest 8 bits during playback. */
switch(*format) {
case AF_FORMAT_U8:
case AF_FORMAT_S8:
*width = AL_SAMPLE_8;
*format = AF_FORMAT_S8;
break;
case AF_FORMAT_U16_LE:
case AF_FORMAT_U16_BE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_S16_BE:
*width = AL_SAMPLE_16;
*format = AF_FORMAT_S16_NE;
break;
case AF_FORMAT_U24_LE:
case AF_FORMAT_U24_BE:
case AF_FORMAT_S24_LE:
case AF_FORMAT_S24_BE:
case AF_FORMAT_U32_LE:
case AF_FORMAT_U32_BE:
case AF_FORMAT_S32_LE:
case AF_FORMAT_S32_BE:
*width = AL_SAMPLE_24;
*format = AF_FORMAT_S32_NE;
break;
case AF_FORMAT_FLOAT_LE:
case AF_FORMAT_FLOAT_BE:
*width = 4;
*format = AF_FORMAT_FLOAT_NE;
smpfmt = AL_SAMPFMT_FLOAT;
break;
default:
*width = AL_SAMPLE_16;
*format = AF_FORMAT_S16_NE;
break;
}
return smpfmt;
}
// to set/get/query special features/parameters
static int control(int cmd, void *arg){
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_INFO);
switch(cmd) {
case AOCONTROL_QUERY_FORMAT:
/* Do not reject any format: return the closest matching
* format if the request is not supported natively. */
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {
int smpwidth, smpfmt;
int rv = AL_DEFAULT_OUTPUT;
smpfmt = fmt2sgial(&format, &smpwidth);
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
{ /* from /usr/share/src/dmedia/audio/setrate.c */
double frate, realrate;
ALpv x[2];
if(ao_subdevice) {
rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE);
if (!rv) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice);
return 0;
}
}
frate = rate;
x[0].param = AL_RATE;
x[0].value.ll = alDoubleToFixed(rate);
x[1].param = AL_MASTER_CLOCK;
x[1].value.i = AL_CRYSTAL_MCLK_TYPE;
if (alSetParams(rv,x, 2)<0) {
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror()));
}
if (x[0].sizeOut < 0) {
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate);
}
if (alGetParams(rv,x, 1)<0) {
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror()));
}
realrate = alFixedToDouble(x[0].value.ll);
if (frate != realrate) {
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, realrate, frate);
}
sample_rate = (int)realrate;
}
bytes_per_frame = channels * smpwidth;
ao_data.samplerate = sample_rate;
ao_data.channels = channels;
ao_data.format = format;
ao_data.bps = sample_rate * bytes_per_frame;
ao_data.buffersize=131072;
ao_data.outburst = ao_data.buffersize/16;
ao_config = alNewConfig();
if (!ao_config) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
return 0;
}
if(alSetChannels(ao_config, channels) < 0 ||
alSetWidth(ao_config, smpwidth) < 0 ||
alSetSampFmt(ao_config, smpfmt) < 0 ||
alSetQueueSize(ao_config, sample_rate) < 0 ||
alSetDevice(ao_config, rv) < 0) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
return 0;
}
ao_port = alOpenPort("mplayer", "w", ao_config);
if (!ao_port) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, alGetErrorString(oserror()));
return 0;
}
// printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
queue_size = alGetQueueSize(ao_config);
return 1;
}
// close audio device
static void uninit(int immed) {
/* TODO: samplerate should be set back to the value before mplayer was started! */
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Uninit);
if (ao_config) {
alFreeConfig(ao_config);
ao_config = NULL;
}
if (ao_port) {
if (!immed)
while(alGetFilled(ao_port) > 0) sginap(1);
alClosePort(ao_port);
ao_port = NULL;
}
}
// stop playing and empty buffers (for seeking/pause)
static void reset(void) {
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Reset);
alDiscardFrames(ao_port, queue_size);
}
// stop playing, keep buffers (for pause)
static void audio_pause(void) {
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_PauseInfo);
}
// resume playing, after audio_pause()
static void audio_resume(void) {
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_ResumeInfo);
}
// return: how many bytes can be played without blocking
static int get_space(void) {
// printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_data.outburst);
// printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
return alGetFillable(ao_port) * bytes_per_frame;
}
// 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) {
/* Always process data in quadword-aligned chunks (64-bits). */
const int plen = len / (sizeof(uint64_t) * bytes_per_frame);
const int framecount = plen * sizeof(uint64_t);
// printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config);
// printf("channels %d\n", ao_data.channels);
if(ao_data.format == AF_FORMAT_S32_NE) {
/* The zen of this is explained in fmt2sgial() */
int32_t *smpls = data;
const int32_t *smple = smpls + (framecount * ao_data.channels);
while(smpls < smple)
*smpls++ >>= 8;
}
alWriteFrames(ao_port, data, framecount);
return framecount * bytes_per_frame;
}
// return: delay in seconds between first and last sample in buffer
static float get_delay(void){
// printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize);
// return (float)queue_size/((float)sample_rate);
const int outstanding = alGetFilled(ao_port);
return (float)((outstanding < 0) ? queue_size : outstanding) /
((float)sample_rate);
}