edje-multisense: Get rid of the old files

Signed-off-by: Daniel Willmann <d.willmann@samsung.com>

SVN revision: 83382
This commit is contained in:
Daniel Willmann 2013-01-28 15:15:37 +00:00 committed by Daniel Willmann
parent ed92993aab
commit d2f926a449
3 changed files with 0 additions and 936 deletions

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@ -1,405 +0,0 @@
/*
* Remix ALSA Player: ALSA audio output
*
* Govindaraju SM <govi.sm@samsung.com>, October 2011
* Prince Kumar Dubey <prince.dubey@samsung.com>, October 2011
*/
#include "config.h"
#include <stdio.h>
#include <remix/remix.h>
#include <alsa/asoundlib.h>
#include <Eina.h>
#ifdef HAVE_LIBSNDFILE
#include <sndfile.h>
#endif
#define ALSA_PLAYER_BUFFERLEN 2048
typedef struct _Alsa_Player_Data Alsa_Player_Data;
typedef short PLAYER_PCM;
struct _Alsa_Player_Data
{
RemixPCM databuffer[ALSA_PLAYER_BUFFERLEN];
snd_pcm_t *alsa_dev;
unsigned int stereo;
unsigned channels;
unsigned int frequency;
};
static int _log_dom = -1;
static int init_count = 0;
#ifdef WRN
# undef WRN
#endif
#define WRN(...) EINA_LOG_DOM_WARN(_log_dom, __VA_ARGS__)
//#define MIXDBG 1
/* Optimisation dependencies: none */
static RemixBase *alsa_player_optimise(RemixEnv *env, RemixBase *base);
static snd_pcm_t *
alsa_open(int channels, unsigned int samplerate, unsigned int *real_samplerate)
{
const char *device = "default";
snd_pcm_t *alsa_dev = NULL;
snd_pcm_hw_params_t *hw_params;
snd_pcm_uframes_t alsa_buffer_frames;
snd_pcm_uframes_t alsa_period_size;
unsigned int samplerate_ret = 0;
int err;
alsa_buffer_frames = ALSA_PLAYER_BUFFERLEN;
alsa_period_size = ALSA_PLAYER_BUFFERLEN / 4;
if ((err = snd_pcm_open(&alsa_dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
WRN("cannot open alsa playback stream (%s)", snd_strerror(err));
goto catch_error;
}
snd_pcm_hw_params_alloca(&hw_params);
if ((err = snd_pcm_hw_params_any(alsa_dev, hw_params)) < 0)
{
WRN("cannot initialize snd hw params (%s)", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_set_access(alsa_dev, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
WRN("cannot set interleaved access (%s)", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_set_format(alsa_dev, hw_params,
SND_PCM_FORMAT_FLOAT)) < 0)
{
WRN("cannot set float sample format (%s)", snd_strerror(err));
goto catch_error;
}
#ifdef MIXDBG // testing/debugging by making output samplerate be 48khz
samplerate_ret = 48000;
if ((err = snd_pcm_hw_params_set_rate_near(alsa_dev, hw_params,
&samplerate_ret, 0)) < 0)
{
WRN("cannot set sample rate (%s)", snd_strerror(err));
goto catch_error;
}
#else
if ((err = snd_pcm_hw_params_set_rate_near(alsa_dev, hw_params,
&samplerate, 0)) < 0)
{
WRN("cannot set sample rate (%s)", snd_strerror(err));
goto catch_error;
}
#endif
if ((err = snd_pcm_hw_params_set_channels(alsa_dev, hw_params, channels)) < 0)
{
WRN("cannot set channel count (%s)", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_set_buffer_size_near(alsa_dev, hw_params,
&alsa_buffer_frames)) < 0)
{
WRN("cannot set buffer size (%s)", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_set_period_size_near(alsa_dev, hw_params,
&alsa_period_size, 0)) < 0)
{
WRN("cannot set period size (%s)", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params(alsa_dev, hw_params)) < 0)
{
WRN("cannot set parameters (%s)", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_get_rate(hw_params, &samplerate_ret, 0)) < 0)
{
WRN("cannot get samplerate (%s)", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_prepare(alsa_dev)) < 0)
{
WRN("cannot prepare audio for use (%s)", snd_strerror(err));
goto catch_error;
}
if (real_samplerate) *real_samplerate = samplerate_ret;
catch_error:
if ((err < 0) && (alsa_dev != NULL))
{
snd_pcm_close(alsa_dev);
return NULL;
}
return alsa_dev;
}
static RemixBase *
alsa_player_reset_device(RemixEnv *env, RemixBase *base)
{
Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
unsigned int real_samplerate = 0;
if (player_data->alsa_dev)
{
snd_pcm_drain(player_data->alsa_dev);
snd_pcm_close(player_data->alsa_dev);
}
player_data->alsa_dev = alsa_open(player_data->channels,
player_data->frequency,
&real_samplerate);
if (!player_data->alsa_dev)
{
remix_set_error(env, REMIX_ERROR_SYSTEM);
return RemixNone;
}
// printf("%i != %i\n", real_samplerate, player_data->frequency);
if (real_samplerate != player_data->frequency)
{
player_data->frequency = real_samplerate;
remix_set_samplerate(env, player_data->frequency);
}
return base;
}
static RemixBase *
alsa_player_init(RemixEnv *env, RemixBase *base, CDSet *parameters __UNUSED__)
{
CDSet *channels;
Alsa_Player_Data *player_data = calloc(1, sizeof(Alsa_Player_Data));
if (!player_data)
{
remix_set_error(env, REMIX_ERROR_SYSTEM);
return RemixNone;
}
init_count++;
if (init_count == 1)
{
eina_init();
_log_dom = eina_log_domain_register("remix-alsa", EINA_COLOR_CYAN);
}
remix_base_set_instance_data(env, base, player_data);
channels = remix_get_channels(env);
player_data->channels = cd_set_size(env, channels);
if (player_data->channels == 1) player_data->stereo = 0;
else if (player_data->channels == 2) player_data->stereo = 1;
player_data->frequency = remix_get_samplerate(env);
alsa_player_reset_device(env, base);
base = alsa_player_optimise(env, base);
return base;
}
static RemixBase *
alsa_player_clone(RemixEnv *env, RemixBase *base __UNUSED__)
{
RemixBase *new_player = remix_base_new(env);
alsa_player_init(env, new_player, NULL);
return new_player;
}
static int
alsa_player_destroy(RemixEnv *env, RemixBase *base)
{
Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
if (player_data->alsa_dev)
{
snd_pcm_drain(player_data->alsa_dev);
snd_pcm_close(player_data->alsa_dev);
}
free(player_data);
init_count--;
if (init_count == 0)
{
eina_log_domain_unregister(_log_dom);
_log_dom = -1;
eina_shutdown();
}
return 0;
}
static int
alsa_player_ready(RemixEnv *env, RemixBase *base)
{
Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
RemixCount nr_channels;
CDSet *channels;
int samplerate;
channels = remix_get_channels(env);
samplerate = (int)remix_get_samplerate(env);
nr_channels = cd_set_size(env, channels);
return ((samplerate == (int)player_data->frequency) &&
(((nr_channels == 1) && (player_data->stereo == 0)) ||
((nr_channels > 1) && (player_data->stereo == 1))));
}
static RemixBase *
alsa_player_prepare(RemixEnv *env, RemixBase *base)
{
alsa_player_reset_device(env, base);
return base;
}
static RemixCount
alsa_player_playbuffer(RemixEnv *env __UNUSED__, Alsa_Player_Data *player, RemixPCM *data, RemixCount count)
{
#ifdef MIXDBG
{
static int total = 0;
static SNDFILE *sfile = NULL;
static SF_INFO sfinfo;
if (total == 0)
{
sfinfo.frames = 0;
sfinfo.samplerate = player->frequency;
sfinfo.channels = 2;
sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16 | SF_ENDIAN_LITTLE;
sfinfo.sections = 0;
sfinfo.seekable = 0;
sfile = sf_open("out.wav", SFM_WRITE, &sfinfo);
}
if (sfile)
{
sf_writef_float(sfile, data, count);
total += count;
}
}
#endif
return snd_pcm_writei(player->alsa_dev, data, count);
}
static RemixCount
alsa_player_chunk(RemixEnv *env, RemixChunk *chunk, RemixCount offset, RemixCount count, int channelname __UNUSED__, void *data)
{
Alsa_Player_Data *player = data;
RemixCount remaining = count, written = 0, n, playcount;
RemixPCM *d;
while (remaining > 0)
{
playcount = MIN(remaining, ALSA_PLAYER_BUFFERLEN);
d = &chunk->data[offset];
n = alsa_player_playbuffer(env, player, d, playcount);
if (n == -1) return -1;
else n /= sizeof(PLAYER_PCM);
offset += n;
written += n;
remaining -= n;
}
return written;
}
static RemixCount
alsa_player_process(RemixEnv *env, RemixBase *base, RemixCount count, RemixStream *input, RemixStream *output __UNUSED__)
{
Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
RemixCount nr_channels = remix_stream_nr_channels(env, input);
RemixCount remaining = count, processed = 0, n, nn;
if ((nr_channels == 1) && (player_data->stereo == 0))
{ /*MONO*/
return remix_stream_chunkfuncify(env, input, count,
alsa_player_chunk, player_data);
}
else if ((nr_channels == 2) && (player_data->stereo == 1))
{ /*STEREO*/
while (remaining > 0)
{
n = MIN(remaining, ALSA_PLAYER_BUFFERLEN / 2);
n = remix_stream_interleave_2(env, input,
REMIX_CHANNEL_LEFT,
REMIX_CHANNEL_RIGHT,
player_data->databuffer, n);
nn = alsa_player_playbuffer(env, player_data,
player_data->databuffer, n);
processed += n;
remaining -= n;
}
return processed;
}
WRN("[alsa_player_process] unsupported stream/output channel "
"combination %ld / %d", nr_channels, player_data->stereo ? 2 : 1);
return -1;
}
static RemixCount
alsa_player_length(RemixEnv *env __UNUSED__, RemixBase *base __UNUSED__)
{
return REMIX_COUNT_INFINITE;
}
static RemixCount
alsa_player_seek(RemixEnv *env __UNUSED__, RemixBase *base __UNUSED__, RemixCount count __UNUSED__)
{
return count;
}
static int
alsa_player_flush(RemixEnv *env, RemixBase *base)
{
alsa_player_reset_device(env, base);
return 0;
}
static struct _RemixMethods _alsa_player_methods =
{
alsa_player_clone,
alsa_player_destroy,
alsa_player_ready,
alsa_player_prepare,
alsa_player_process,
alsa_player_length,
alsa_player_seek,
alsa_player_flush,
};
static RemixBase *
alsa_player_optimise(RemixEnv *env, RemixBase *base)
{
remix_base_set_methods(env, base, &_alsa_player_methods);
return base;
}
static struct _RemixMetaText alsa_player_metatext =
{
"alsa_snd_player",
"ALSA sound player for Remix",
"Output the audio stream into ALSA Driver",
"Copyright (C) 2011, Samsung Electronics Co., Ltd.",
"http://www.samsung.com",
REMIX_ONE_AUTHOR("Govindaraju SM", "prince.dubey@samsung.com"),
};
static struct _RemixPlugin alsa_player_plugin =
{
&alsa_player_metatext,
REMIX_FLAGS_NONE,
CD_EMPTY_SET, /* init scheme */
alsa_player_init,
CD_EMPTY_SET, /* process scheme */
NULL, /* suggests */
NULL, /* plugin data */
NULL /* destroy */
};
EAPI CDList *
remix_load(RemixEnv *env)
{
CDList *plugins = cd_list_new(env);
plugins = cd_list_prepend(env, plugins, CD_POINTER(&alsa_player_plugin));
return plugins;
}

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@ -1,498 +0,0 @@
/*
* RemixSnd_eetfile: a libsnd EET Virtual file handler
*
* Govindaraju SM <govi.sm@samsung.com>, August 2011
* Prince Kumar Dubey <prince.dubey@samsung.com>, August 2011
*/
#include "config.h"
#include <math.h>
#include <sndfile.h>
#include <remix/remix.h>
#include <Eet.h>
#define PATH_KEY 1
#define SOUND_ID_KEY 2
#define SPEED_KEY 3
#define BLOCK_FRAMES 8192
static RemixBase *remix_eet_sndfile_optimise(RemixEnv *env, RemixBase *sndfile);
typedef struct _VIO_DATA VIO_DATA;
typedef struct _SndInstanceData SndInstanceData;
struct _VIO_DATA
{
sf_count_t offset, length;
const char *data;
};
struct _SndInstanceData
{
/* plugin parameters */
char *path;
char *sound_id;
double speed;
/* Edj & Sndfile Reader */
Eet_File *efp;
SNDFILE *pcm_fp;
SF_INFO *snd_info;
VIO_DATA *vio_data;
/* PCM buffers */
RemixPCM *readbuf;
RemixPCM *inbuf;
RemixPCM *outbuf;
/* Resample stuffs */
RemixPCM prevreadbuf[2];
int enable_resample;
double rs_ratio;
RemixCount resample_len;
RemixCount in_avail;
RemixCount out_generated;
RemixCount required_resamples;
};
static sf_count_t
eet_snd_file_get_length(void *user_data)
{
VIO_DATA *vf = user_data;
return vf->length;
}
static sf_count_t
eet_snd_file_seek(sf_count_t offset, int whence, void *user_data)
{
VIO_DATA *vf = user_data;
switch (whence)
{
case SEEK_SET:
vf->offset = offset;
break;
case SEEK_CUR:
vf->offset += offset;
break;
case SEEK_END:
vf->offset = vf->length + offset;
break;
default:
break;
}
return vf->offset;
}
static sf_count_t
eet_snd_file_read(void *ptr, sf_count_t count, void *user_data)
{
VIO_DATA *vf = user_data;
if ((vf->offset + count) > vf->length)
count = vf->length - vf->offset;
memcpy(ptr, vf->data + vf->offset, count);
vf->offset += count;
return count;
}
static sf_count_t
eet_snd_file_tell(void *user_data)
{
VIO_DATA *vf = user_data;
return vf->offset;
}
static int
remix_init_resampler_data(RemixEnv *env, RemixBase *base)
{
SndInstanceData *si = remix_base_get_instance_data(env, base);
si->rs_ratio = remix_get_samplerate(env) / si->snd_info->samplerate;
si->rs_ratio /= si->speed;
si->resample_len = (si->snd_info->frames * si->rs_ratio);
si->outbuf = malloc(sizeof(RemixPCM) * BLOCK_FRAMES * 2);
if (!si->outbuf) return 0;
if ((si->rs_ratio == 1.0)/* && (si->snd_info->channels == 2)*/)
{
si->enable_resample = 0;
return 1;
}
else
si->enable_resample = 1;
si->in_avail = 0;
si->out_generated = 0;
si->inbuf = malloc(sizeof(RemixPCM) * BLOCK_FRAMES *
si->snd_info->channels);
if (!si->inbuf) return 0;
return 1;
}
static RemixBase *
remix_eet_sndfile_create(RemixEnv *env, RemixBase *sndfile, const char *path, const char *sound_id, const double speed)
{
SF_VIRTUAL_IO *eet_vio = NULL;
SndInstanceData *si;
const void *sound_data;
int sound_size;
if ((!path) || (!sound_id)) return NULL;
si = calloc(1, sizeof(SndInstanceData));
if (!si) goto err;
remix_base_set_instance_data(env, sndfile, si);
si->path = strdup(path);
si->sound_id = strdup(sound_id);
si->speed = speed;
si->efp = eet_open(path, EET_FILE_MODE_READ);
if (!si->efp) goto err;
// xxx: eet_read_direct does not work on Threads, using eet_read.
sound_data = eet_read(si->efp, sound_id, &(sound_size));
eet_close(si->efp);
si->efp = NULL;
if (sound_data == NULL) goto err;
eet_vio = calloc(1, sizeof(SF_VIRTUAL_IO));
if (!eet_vio) goto err;
/* Set up func pointers to read snd file directly from EET. */
eet_vio->get_filelen = eet_snd_file_get_length;
eet_vio->seek = eet_snd_file_seek;
eet_vio->read = eet_snd_file_read;
eet_vio->tell = eet_snd_file_tell;
si->vio_data = calloc(1, sizeof(VIO_DATA));
if (!si->vio_data) goto err;
si->vio_data->offset = 0;
si->vio_data->length = sound_size;
si->vio_data->data = sound_data;
si->snd_info = calloc(1, sizeof(SF_INFO));
if (!si->snd_info) goto err;
si->pcm_fp = sf_open_virtual(eet_vio, SFM_READ, si->snd_info, si->vio_data);
if (!si->pcm_fp) goto err;
free(eet_vio);
eet_vio = NULL;
if (!remix_init_resampler_data(env, sndfile)) goto err;
si->out_generated = 0;
return sndfile;
err:
if (eet_vio) free(eet_vio);
remix_set_error(env, REMIX_ERROR_SYSTEM);
remix_destroy(env, (RemixBase *)sndfile);
return RemixNone;
}
static RemixBase *
remix_eet_sndfile_reader_init(RemixEnv *env, RemixBase *base, CDSet *parameters)
{
char *file_path, *sound_id;
double speed;
file_path = (cd_set_find(env, parameters, PATH_KEY)).s_string;
sound_id = (cd_set_find(env, parameters, SOUND_ID_KEY)).s_string;
speed = (cd_set_find(env, parameters, SPEED_KEY)).s_double;
if (!remix_eet_sndfile_create(env, base, file_path, sound_id, speed))
return RemixNone;
remix_eet_sndfile_optimise (env, base);
return base;
}
static RemixBase *
remix_eet_sndfile_clone(RemixEnv *env, RemixBase *base)
{
SndInstanceData *si = remix_base_get_instance_data(env, base);
RemixBase *new_sndfile = remix_base_new(env);
remix_eet_sndfile_create(env, new_sndfile, si->path, si->sound_id, si->speed);
remix_eet_sndfile_optimise(env, new_sndfile);
return new_sndfile;
}
static int
remix_eet_sndfile_destroy(RemixEnv *env, RemixBase *base)
{
SndInstanceData *si = remix_base_get_instance_data(env, base);
if (si)
{
sf_close (si->pcm_fp);
eet_close(si->efp);
if (si->path) free(si->path);
if (si->sound_id) free(si->sound_id);
if (si->snd_info) free(si->snd_info);
if (si->efp) eet_close(si->efp);
if (si->inbuf) free(si->inbuf);
if (si->outbuf) free(si->outbuf);
if (si->vio_data) free(si->vio_data);
free(si);
}
if (base) free (base);
return 0;
}
static int
remix_pcm_resample(SndInstanceData *si)
{
RemixPCM *src, *dst, *srcbase;
int i = 0, in_samples, pos, total, chnum, reqsamp, avail;
int interp = 1;
dst = si->outbuf + (si->out_generated * 2);
in_samples = (double)si->required_resamples / si->rs_ratio;
chnum = si->snd_info->channels;
reqsamp = si->required_resamples;
avail = si->in_avail;
srcbase = si->readbuf;
if ((interp) && (si->rs_ratio >= 1.0))
{
// linear interpolation of resampling for lower quality samples
// so they don't get high requency aliasing effects
for (i = 0; i < reqsamp; i++)
{
float fpos, fpos1;
RemixPCM psam[2];
fpos = (float)(i * in_samples) / (float)reqsamp;
pos = fpos;
if (pos >= avail) break;
fpos -= pos;
fpos1 = 1.0 - fpos;
src = srcbase + (pos * chnum);
if (chnum == 2)
{
if (pos == 0)
{
psam[0] = si->prevreadbuf[0];
psam[1] = si->prevreadbuf[1];
}
else
{
psam[0] = src[0 - 2];
psam[1] = src[1 - 2];
}
*dst++ = (src[0] * fpos) + (psam[0] * fpos1);
*dst++ = (src[1] * fpos) + (psam[1] * fpos1);
}
else
{
if (pos == 0)
psam[0] = si->prevreadbuf[0];
else
psam[0] = src[0 - 1];
*dst++ = (src[0] * fpos) + (psam[0] * fpos1);
}
}
}
else
{
// simple sample-picking/nearest. faster and simpler
for (i = 0; i < reqsamp; i++)
{
pos = (i * in_samples) / reqsamp;
if (pos >= avail) break;
src = srcbase + (pos * chnum);
if (chnum == 2)
{
*dst++ = src[0];
*dst++ = src[1];
}
else
*dst++ = src[0];
}
}
si->out_generated += i;
total = (i * in_samples) / reqsamp;
si->readbuf += total * chnum;
si->in_avail -= total;
return total;
}
/* An RemixChunkFunc for creating sndfile */
static RemixCount
remix_eet_sndfile_read_update(RemixEnv *env, RemixBase *sndfile, RemixCount count)
{
SndInstanceData *si = remix_base_get_instance_data(env, sndfile);
si->out_generated = 0;
if (si->enable_resample)
{
RemixCount gen = 0;
while (gen < count)
{
if (si->in_avail <= 0)
{
si->in_avail = sf_readf_float(si->pcm_fp, si->inbuf, BLOCK_FRAMES);
si->readbuf = si->inbuf;
}
si->required_resamples = (count - gen);
remix_pcm_resample(si);
if (si->snd_info->channels == 2)
{
si->prevreadbuf[0] = si->readbuf[-2];
si->prevreadbuf[1] = si->readbuf[-1];
}
else
{
si->prevreadbuf[0] = si->readbuf[-1];
}
gen += si->out_generated;
}
si->out_generated = gen;
}
else
{
si->out_generated = sf_readf_float(si->pcm_fp, si->outbuf, count);
}
return si->out_generated;
}
static RemixCount
remix_eet_sndfile_read_into_chunk(RemixEnv *env, RemixChunk *chunk, RemixCount offset, RemixCount count, int channelname, void *data)
{
RemixBase *sndfile = data;
SndInstanceData *si = remix_base_get_instance_data(env, sndfile);
RemixPCM *d, *p;
RemixCount remaining = count, written = 0, n, i;
d = &chunk->data[offset];
n = MIN(remaining, BLOCK_FRAMES);
// Need parameter support to advance the data reading
if (channelname == 0)
remix_eet_sndfile_read_update(env, sndfile, n);
n = MIN(si->out_generated, remaining);
p = si->outbuf;
if (si->snd_info->channels > 1) p += channelname;
for (i = 0; i < n; i++)
{
*d++ = *p;
p += si->snd_info->channels;
}
if (n == 0) n = _remix_pcm_set(d, 0.0, remaining);
remaining -= n;
written += n;
return written;
}
static RemixCount
remix_eet_sndfile_reader_process(RemixEnv *env, RemixBase *base, RemixCount count, RemixStream *input __UNUSED__, RemixStream *output)
{
return remix_stream_chunkfuncify(env, output, count,
remix_eet_sndfile_read_into_chunk,
base);
}
static RemixCount
remix_eet_sndfile_length(RemixEnv *env, RemixBase *base)
{
SndInstanceData *si = remix_base_get_instance_data(env, base);
return si->resample_len;
}
static RemixCount
remix_eet_sndfile_seek(RemixEnv *env, RemixBase *base, RemixCount offset)
{
SndInstanceData *si = remix_base_get_instance_data(env, base);
return sf_seek(si->pcm_fp, offset, SEEK_SET);
}
static struct _RemixMethods _remix_eet_sndfile_reader_methods =
{
remix_eet_sndfile_clone,
remix_eet_sndfile_destroy,
NULL, /* ready */
NULL, /* prepare */
remix_eet_sndfile_reader_process,
remix_eet_sndfile_length,
remix_eet_sndfile_seek,
NULL, /* flush */
};
static RemixBase *
remix_eet_sndfile_optimise(RemixEnv *env, RemixBase *sndfile)
{
remix_base_set_methods(env, sndfile, &_remix_eet_sndfile_reader_methods);
return sndfile;
}
static struct _RemixParameterScheme path_scheme =
{
"path",
"Path to sound file",
REMIX_TYPE_STRING,
REMIX_CONSTRAINT_TYPE_NONE,
{NULL},
REMIX_HINT_FILENAME,
};
static struct _RemixParameterScheme sound_id_scheme =
{
"sound_id",
"Sound Id (Key) inside EET",
REMIX_TYPE_STRING,
REMIX_CONSTRAINT_TYPE_NONE,
{NULL},
REMIX_HINT_DEFAULT,
};
static struct _RemixParameterScheme speed_scheme =
{
"speed",
"Sound Play Speed",
REMIX_TYPE_FLOAT,
REMIX_CONSTRAINT_TYPE_NONE,
{NULL},
REMIX_HINT_DEFAULT,
};
static struct _RemixMetaText eet_sndfile_reader_metatext =
{
"eet_sndfile_reader",
"File:: Sound file Reader from EET",
"Reads PCM audio files from EET bundle using libsndfile",
"Copyright (C) 2011, Samsung Electronics Co., Ltd.",
"http://www.samsung.com",
REMIX_ONE_AUTHOR ("govi.sm@samsung.com", "prince.dubey@samsung.com"),
};
static struct _RemixPlugin eet_sndfile_reader_plugin =
{
&eet_sndfile_reader_metatext,
REMIX_FLAGS_NONE,
CD_EMPTY_SET, /* init scheme */
remix_eet_sndfile_reader_init,
CD_EMPTY_SET, /* process scheme */
NULL, /* suggests */
NULL, /* plugin data */
NULL /* destroy */
};
EAPI CDList *
remix_load(RemixEnv *env)
{
CDList *plugins = cd_list_new(env);
eet_sndfile_reader_plugin.init_scheme =
cd_set_insert(env, eet_sndfile_reader_plugin.init_scheme, PATH_KEY,
CD_POINTER(&path_scheme));
eet_sndfile_reader_plugin.init_scheme =
cd_set_insert(env, eet_sndfile_reader_plugin.init_scheme, SOUND_ID_KEY,
CD_POINTER(&sound_id_scheme));
eet_sndfile_reader_plugin.init_scheme =
cd_set_insert(env, eet_sndfile_reader_plugin.init_scheme, SPEED_KEY,
CD_POINTER(&speed_scheme));
plugins = cd_list_prepend(env, plugins,
CD_POINTER(&eet_sndfile_reader_plugin));
return plugins;
}

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@ -1,33 +0,0 @@
#include "config.h"
#include "edje_private.h"
#define DEFAULT_SAMPLERATE 44100
#ifdef HAVE_LIBREMIX
EAPI RemixBase *
multisense_sound_player_get(Edje_Multisense_Env *msenv)
{
RemixEnv *env = msenv->remixenv;
RemixPlugin *player_plugin;
RemixBase *player;
player_plugin = remix_find_plugin(env, "alsa_snd_player");
if (!player_plugin)
{
WRN("ALSA player_plugin init fail\n");
return remix_monitor_new(env);
}
player = remix_new(env, player_plugin, NULL);
return player;
}
#endif
EAPI Eina_Bool
multisense_factory_init(Edje_Multisense_Env *env __UNUSED__)
{
#ifdef HAVE_LIBREMIX
remix_set_samplerate(env->remixenv, DEFAULT_SAMPLERATE);
remix_set_channels(env->remixenv, REMIX_STEREO);
#endif
return EINA_TRUE;
}