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sdk-hwV1.3/external/eyesee-mpp/middleware/sun8iw21/media/librender/audio_render_alsa.c

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/*******************************************************************************
-- --
-- CedarX Multimedia Framework --
-- --
-- the Multimedia Framework for Linux/Android System --
-- --
-- This software is confidential and proprietary and may be used --
-- only as expressly authorized by a licensing agreement from --
-- Softwinner Products. --
-- --
-- (C) COPYRIGHT 2011 SOFTWINNER PRODUCTS --
-- ALL RIGHTS RESERVED --
-- --
-- The entire notice above must be reproduced --
-- on all copies and should not be removed. --
-- --
*******************************************************************************/
#define LOG_NDEBUG 0
#define LOG_TAG "audio_render"
#include <CDX_Debug.h>
#include <CDX_Types.h>
#include "audio_render.h"
#include <errno.h>
#include <sys/time.h>
#include <stdlib.h>
#include <stdarg.h>
#include <ctype.h>
#include <math.h>
#include <string.h>
#include <alloca.h>
#include <CDX_PlayerAPI.h>
//#include "config.h"
//#include "subopt-helper.h"
//#include "mixer.h"
//------- custom define for compile start -----------
#include "help_mp.h"
#define HAVE_ALSA_ASOUNDLIB_H 1
//#define mp_msg(t, l, ...) fprintf(stderr, __VA_ARGS__)
#define mp_msg(t, l, ...) ((void)0)
typedef struct strarg_s {
int len; ///< length of the string determined by the parser
char const * str; ///< pointer to position inside the parse string
} strarg_t;
typedef struct ao_data_s {
int samplerate;
int channels;
int format;
int bps;
int outburst;
int buffersize;
int pts;
} ao_data_t;
#define OUTBURST 512
ao_data_t ao_data = { 0, 0, 0, 0, OUTBURST, -1, 0 };
//------- custom define for compile end -----------
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#ifdef HAVE_SYS_ASOUNDLIB_H
#include <sys/asoundlib.h>
#elif defined(HAVE_ALSA_ASOUNDLIB_H)
#include <alsa/asoundlib.h>
#else
#error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
#endif
static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
static snd_pcm_hw_params_t *alsa_hwparams;
static snd_pcm_sw_params_t *alsa_swparams;
/* 16 sets buffersize to 16 * chunksize is as default 1024
* which seems to be good avarge for most situations
* so buffersize is 16384 frames by default */
static int alsa_fragcount = 16;
static snd_pcm_uframes_t chunk_size = 1024;
static size_t bytes_per_sample;
static int ao_noblock = 0;
static int open_mode;
static int alsa_can_pause = 0;
#define ALSA_DEVICE_SIZE 256
#undef BUFFERTIME
#define SET_CHUNKSIZE
static void alsa_error_handler(const char *file, int line,
const char *function, int err, const char *format, ...) {
char tmp[0xc00];
va_list va;
va_start(va, format);
vsnprintf(tmp, sizeof tmp, format, va);
va_end(va);
tmp[sizeof tmp - 1] = '\0';
if (err)
mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
file, line, function, tmp, snd_strerror(err));
else
mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
file, line, function, tmp);
}
/* to set/get/query special features/parameters */
//static int control(int cmd, void *arg)
//{
// switch(cmd) {
// case AOCONTROL_QUERY_FORMAT:
// return CONTROL_TRUE;
// case AOCONTROL_GET_VOLUME:
// case AOCONTROL_SET_VOLUME:
// {
// ao_control_vol_t *vol = (ao_control_vol_t *)arg;
//
// int err;
// snd_mixer_t *handle;
// snd_mixer_elem_t *elem;
// snd_mixer_selem_id_t *sid;
//
// static char *mix_name = "PCM";
// static char *card = "default";
// static int mix_index = 0;
//
// long pmin, pmax;
// long get_vol, set_vol;
// float f_multi;
//
// if(ao_data.format == AF_FORMAT_AC3)
// return CONTROL_TRUE;
//
// if(mixer_channel) {
// char *test_mix_index;
//
// mix_name = strdup(mixer_channel);
// if ((test_mix_index = strchr(mix_name, ','))){
// *test_mix_index = 0;
// test_mix_index++;
// mix_index = strtol(test_mix_index, &test_mix_index, 0);
//
// if (*test_mix_index){
// mp_msg(MSGT_AO,MSGL_ERR,
// MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
// mix_index = 0 ;
// }
// }
// }
// if(mixer_device) card = mixer_device;
//
// //allocate simple id
// snd_mixer_selem_id_alloca(&sid);
//
// //sets simple-mixer index and name
// snd_mixer_selem_id_set_index(sid, mix_index);
// snd_mixer_selem_id_set_name(sid, mix_name);
//
// if (mixer_channel) {
// free(mix_name);
// mix_name = NULL;
// }
//
// if ((err = snd_mixer_open(&handle, 0)) < 0) {
// mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
// return CONTROL_ERROR;
// }
//
// if ((err = snd_mixer_attach(handle, card)) < 0) {
// mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
// card, snd_strerror(err));
// snd_mixer_close(handle);
// return CONTROL_ERROR;
// }
//
// if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
// mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err));
// snd_mixer_close(handle);
// return CONTROL_ERROR;
// }
// err = snd_mixer_load(handle);
// if (err < 0) {
// mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, snd_strerror(err));
// snd_mixer_close(handle);
// return CONTROL_ERROR;
// }
//
// elem = snd_mixer_find_selem(handle, sid);
// if (!elem) {
// mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl,
// snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
// snd_mixer_close(handle);
// return CONTROL_ERROR;
// }
//
// snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
// f_multi = (100 / (float)(pmax - pmin));
//
// if (cmd == AOCONTROL_SET_VOLUME) {
//
// set_vol = vol->left / f_multi + pmin + 0.5;
//
// //setting channels
// if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
// mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
// snd_strerror(err));
// return CONTROL_ERROR;
// }
// mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
//
// set_vol = vol->right / f_multi + pmin + 0.5;
//
// if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
// mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
// snd_strerror(err));
// return CONTROL_ERROR;
// }
// mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
// set_vol, pmin, pmax, f_multi);
//
// if (snd_mixer_selem_has_playback_switch(elem)) {
// int lmute = (vol->left == 0.0);
// int rmute = (vol->right == 0.0);
// if (snd_mixer_selem_has_playback_switch_joined(elem)) {
// lmute = rmute = lmute && rmute;
// } else {
// snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
// }
// snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
// }
// }
// else {
// snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
// vol->left = (get_vol - pmin) * f_multi;
// snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
// vol->right = (get_vol - pmin) * f_multi;
//
// mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
// }
// snd_mixer_close(handle);
// return CONTROL_OK;
// }
//
// } //end switch
// return CONTROL_UNKNOWN;
//}
static void parse_device(char *dest, const char *src, int len) {
char *tmp;
memmove(dest, src, len);
dest[len] = 0;
while ((tmp = strrchr(dest, '.')))
tmp[0] = ',';
while ((tmp = strrchr(dest, '=')))
tmp[0] = ':';
}
static void print_help(void) {
mp_msg (MSGT_AO, MSGL_FATAL,
MSGTR_AO_ALSA_CommandlineHelp);
}
static int str_maxlen(strarg_t *str) {
if (str->len > ALSA_DEVICE_SIZE)
return 0;
return 1;
}
static int try_open_device(const char *device, int open_mode, int try_ac3) {
int err;
err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
open_mode);
return err;
}
/*
open & setup audio device
return: 1=success 0=fail
*/
static int alsa_init(struct CDX_AudioRenderHAL *handle, int rate_hz, int channels, int format) {
int err;
int block;
strarg_t device;
snd_pcm_uframes_t bufsize;
snd_pcm_uframes_t boundary;
// opt_t subopts[] = {
// {"block", OPT_ARG_BOOL, &block, NULL},
// {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
// {NULL}
// };
char alsa_device[ALSA_DEVICE_SIZE + 1];
// make sure alsa_device is null-terminated even when using strncpy etc.
memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
LOGH;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
channels, format);
alsa_handler = NULL;
#if SND_LIB_VERSION >= 0x010005
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
#else
mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
#endif
snd_lib_error_set_handler(alsa_error_handler);
ao_data.samplerate = rate_hz;
ao_data.format = format;
ao_data.channels = channels;
alsa_format = SND_PCM_FORMAT_S16_BE; //TODO: is it right??
//subdevice parsing
// set defaults
block = 1;
/* switch for spdif
* sets opening sequence for SPDIF
* sets also the playback and other switches 'on the fly'
* while opening the abstract alias for the spdif subdevice
* 'iec958'
*/
// if (format == AF_FORMAT_AC3) {
// device.str = "iec958";
// mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
// }
// else
/* in any case for multichannel playback we should select
* appropriate device
*/
switch (channels) {
case 1:
case 2:
device.str = "default";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
break;
case 4:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
// hack - use the converter plugin
device.str = "plug:surround40";
else
device.str = "surround40";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
break;
case 6:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
device.str = "plug:surround51";
else
device.str = "surround51";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
break;
default:
device.str = "default";
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
}
device.len = strlen(device.str);
// if (subopt_parse(ao_subdevice, subopts) != 0) {
// print_help();
// return 0;
// }
ao_noblock = !block;
parse_device(alsa_device, device.str, device.len);
if(ao_data.format == CDX_AUDIO_OUT_I2S)
strcpy(alsa_device, "hw:1");
else
strcpy(alsa_device, "hw:0");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
//setting modes for block or nonblock-mode
if (ao_noblock) {
open_mode = SND_PCM_NONBLOCK;
} else {
open_mode = 0;
}
//sets buff/chunksize if its set manually
if (ao_data.buffersize) {
switch (ao_data.buffersize) {
case 1:
alsa_fragcount = 16;
chunk_size = 512;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
break;
case 2:
alsa_fragcount = 8;
chunk_size = 1024;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
break;
case 3:
alsa_fragcount = 32;
chunk_size = 512;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
break;
case 4:
alsa_fragcount = 16;
chunk_size = 1024;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
break;
default:
alsa_fragcount = 16;
chunk_size = 1024;
break;
}
}
if (!alsa_handler) {
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
if ((err = try_open_device(alsa_device, open_mode, 0)) < 0) {
if (err != -EBUSY && ao_noblock) {
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
if ((err = try_open_device(alsa_device, 0, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return 0;
}
} else {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return 0;
}
}
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
}
snd_pcm_hw_params_alloca(&alsa_hwparams);
snd_pcm_sw_params_alloca(&alsa_swparams);
// setting hw-parameters
if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
snd_strerror(err));
return 0;
}
err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
snd_strerror(err));
return 0;
}
/* workaround for nonsupported formats
sets default format to S16_LE if the given formats aren't supported */
if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0) {
// mp_msg(MSGT_AO,MSGL_INFO,
// MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
//ao_data.format = AF_FORMAT_S16_LE;
}
if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
snd_strerror(err));
return 0;
}
if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler,
alsa_hwparams, &ao_data.channels)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
snd_strerror(err));
return 0;
}
/* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
prefer our own resampler */
#if SND_LIB_VERSION >= 0x010009
if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
0)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
&ao_data.samplerate, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
snd_strerror(err));
return 0;
}
bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
bytes_per_sample *= ao_data.channels;
ao_data.bps = ao_data.samplerate * bytes_per_sample;
#ifdef BUFFERTIME
{
int alsa_buffer_time = 500000; /* original 60 */
int alsa_period_time;
alsa_period_time = alsa_buffer_time/4;
if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
&alsa_buffer_time, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
snd_strerror(err));
return 0;
} else
alsa_buffer_time = err;
if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
&alsa_period_time, NULL)) < 0)
/* original: alsa_buffer_time/ao_data.bps */
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodTime,
snd_strerror(err));
return 0;
}
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime,
alsa_buffer_time, err);
}
#endif//end SET_BUFFERTIME
#ifdef SET_CHUNKSIZE
{
//set chunksize
if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler,
alsa_hwparams, &chunk_size, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodSize,
chunk_size, snd_strerror(err));
return 0;
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
}
if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler,
alsa_hwparams, &alsa_fragcount, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
snd_strerror(err));
return 0;
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
}
}
#endif//end SET_CHUNKSIZE
/* finally install hardware parameters */
if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
snd_strerror(err));
return 0;
}
// end setting hw-params
// gets buffersize for control
if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize))
< 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
return 0;
} else {
ao_data.buffersize = bufsize * bytes_per_sample;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
}
if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams,
&chunk_size, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
return 0;
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
}
ao_data.outburst = chunk_size * bytes_per_sample;
/* setting software parameters */
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
snd_strerror(err));
return 0;
}
#else
boundary = 0x7fffffff;
#endif
/* start playing when one period has been written */
if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler,
alsa_swparams, chunk_size)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
snd_strerror(err));
return 0;
}
/* disable underrun reporting */
if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler,
alsa_swparams, boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
/* play silence when there is an underrun */
if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
/* end setting sw-params */
mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
snd_pcm_format_description(alsa_format));
} // end switch alsa_handler (spdif)
alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
return 1;
} // end init
/* close audio device */
static void alsa_exit(struct CDX_AudioRenderHAL *handle, int immed) {
if (alsa_handler) {
int err;
if (!immed)
snd_pcm_drain(alsa_handler);
if ((err = snd_pcm_close(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err));
return;
} else {
alsa_handler = NULL;
mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
}
} else {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined);
}
}
static void audio_pause(void) {
int err;
if (alsa_can_pause) {
if ((err = snd_pcm_pause(alsa_handler, 1)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, snd_strerror(err));
return;
}
mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
} else {
if ((err = snd_pcm_drop(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err));
return;
}
}
}
static void audio_resume(void) {
int err;
if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN)
sleep(1);
}
if (alsa_can_pause) {
if ((err = snd_pcm_pause(alsa_handler, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, snd_strerror(err));
return;
}
mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
} else {
if ((err = snd_pcm_prepare(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
return;
}
}
}
/* stop playing and empty buffers (for seeking/pause) */
static void reset(void) {
int err;
if ((err = snd_pcm_drop(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
return;
}
if ((err = snd_pcm_prepare(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
return;
}
return;
}
/*
plays 'len' bytes of 'data'
returns: number of bytes played
modified last at 29.06.02 by jp
thanxs for marius <marius@rospot.com> for giving us the light ;)
*/
static int alsa_play(struct CDX_AudioRenderHAL *handle, void* data, int len) {
int num_frames = len / bytes_per_sample;
snd_pcm_sframes_t res = 0;
//mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
if (!alsa_handler) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError);
return 0;
}
if (num_frames == 0)
return 0;
do {
res = snd_pcm_writei(alsa_handler, data, num_frames);
if (res == -EINTR) {
/* nothing to do */
res = 0;
} else if (res == -ESTRPIPE) { /* suspend */
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
sleep(1);
}
if (res < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res));
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard);
if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res));
return 0;
break;
}
}
} while (res == 0);
return res < 0 ? res : res * bytes_per_sample;
}
/* how many byes are free in the buffer */
static int alsa_get_space(struct CDX_AudioRenderHAL *handle) {
snd_pcm_status_t *status;
int ret;
snd_pcm_status_alloca(&status);
if ((ret = snd_pcm_status(alsa_handler, status)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
return 0;
}
ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
// if (ret > ao_data.buffersize) // Buffer underrun?
// ret = ao_data.buffersize;
return ret;
}
/* delay in seconds between first and last sample in buffer */
static int alsa_get_delay(struct CDX_AudioRenderHAL *handle) {
if (alsa_handler) {
snd_pcm_sframes_t delay;
if (snd_pcm_delay(alsa_handler, &delay) < 0)
return 0;
if (delay < 0) {
/* underrun - move the application pointer forward to catch up */
#if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
snd_pcm_forward(alsa_handler, -delay);
#endif
delay = 0;
}
return ((int)((float) delay * 1000000 / (float) ao_data.samplerate));
} else {
return 0;
}
}
CDX_AudioRenderHAL audio_render_alsa_hal = {
.info = "audio alsa render",
.init = alsa_init,
.exit = alsa_exit,
.render = alsa_play,
.get_space = alsa_get_space,
.get_delay = alsa_get_delay,
.pause = NULL,
};
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