static gboolean tone_play(const gchar *filename, VFSFile *file)
{
GArray *frequencies;
gfloat data[BUF_SAMPLES];
gsize i;
gboolean error = FALSE;
struct
{
gdouble wd;
guint period, t;
} *tone = NULL;
frequencies = tone_filename_parse(filename);
if (frequencies == NULL)
return FALSE;
if (aud_input_open_audio(FMT_FLOAT, OUTPUT_FREQ, 1) == 0)
{
error = TRUE;
goto error_exit;
}
aud_input_set_bitrate(16 * OUTPUT_FREQ);
tone = g_malloc(frequencies->len * sizeof(*tone));
for (i = 0; i < frequencies->len; i++)
{
gdouble f = g_array_index(frequencies, gdouble, i);
tone[i].wd = 2 * PI * f / OUTPUT_FREQ;
tone[i].period = (G_MAXINT * 2U / OUTPUT_FREQ) * (OUTPUT_FREQ / f);
tone[i].t = 0;
}
while (!aud_input_check_stop())
{
for (i = 0; i < BUF_SAMPLES; i++)
{
gsize j;
double sum_sines;
for (sum_sines = 0, j = 0; j < frequencies->len; j++)
{
sum_sines += sin(tone[j].wd * tone[j].t);
if (tone[j].t > tone[j].period)
tone[j].t -= tone[j].period;
tone[j].t++;
}
/* dithering can cause a little bit of clipping */
data[i] = (sum_sines * 0.999 / (gdouble) frequencies->len);
}
aud_input_write_audio(data, BUF_BYTES);
}
error_exit:
g_array_free(frequencies, TRUE);
g_free(tone);
return !error;
}
static bool_t psf2_play(const char * filename, VFSFile * file)
{
void *buffer;
int64_t size;
PSFEngine eng;
bool_t error = FALSE;
const char * slash = strrchr (filename, '/');
if (! slash)
return FALSE;
SNCOPY (dirbuf, filename, slash + 1 - filename);
dirpath = dirbuf;
vfs_file_get_contents (filename, & buffer, & size);
eng = psf_probe(buffer);
if (eng == ENG_NONE || eng == ENG_COUNT)
{
free(buffer);
return FALSE;
}
f = &psf_functor_map[eng];
if (f->start(buffer, size) != AO_SUCCESS)
{
free(buffer);
return FALSE;
}
aud_input_open_audio(FMT_S16_NE, 44100, 2);
aud_input_set_bitrate(44100*2*2*8);
stop_flag = FALSE;
f->execute();
f->stop();
f = NULL;
dirpath = NULL;
free(buffer);
return ! error;
}
static gboolean vorbis_play (const gchar * filename, VFSFile * file)
{
if (file == NULL)
return FALSE;
vorbis_info *vi;
OggVorbis_File vf;
gint last_section = -1;
ReplayGainInfo rg_info;
gfloat pcmout[PCM_BUFSIZE*sizeof(float)], **pcm;
gint bytes, channels, samplerate, br;
gchar * title = NULL;
memset(&vf, 0, sizeof(vf));
gboolean error = FALSE;
if (ov_open_callbacks (file, & vf, NULL, 0, vfs_is_streaming (file) ?
vorbis_callbacks_stream : vorbis_callbacks) < 0)
{
error = TRUE;
goto play_cleanup;
}
vi = ov_info(&vf, -1);
if (vi->channels > 2)
goto play_cleanup;
br = vi->bitrate_nominal;
channels = vi->channels;
samplerate = vi->rate;
aud_input_set_bitrate (br);
if (!aud_input_open_audio(FMT_FLOAT, samplerate, channels)) {
error = TRUE;
goto play_cleanup;
}
vorbis_update_replaygain(&vf, &rg_info);
aud_input_set_gain (& rg_info);
/*
* Note that chaining changes things here; A vorbis file may
* be a mix of different channels, bitrates and sample rates.
* You can fetch the information for any section of the file
* using the ov_ interface.
*/
while (! aud_input_check_stop ())
{
int seek_value = aud_input_check_seek();
if (seek_value >= 0 && ov_time_seek (& vf, (double) seek_value / 1000) < 0)
{
fprintf (stderr, "vorbis: seek failed\n");
error = TRUE;
break;
}
gint current_section = last_section;
bytes = ov_read_float(&vf, &pcm, PCM_FRAMES, ¤t_section);
if (bytes == OV_HOLE)
continue;
if (bytes <= 0)
break;
bytes = vorbis_interleave_buffer (pcm, bytes, channels, pcmout);
{ /* try to detect when metadata has changed */
vorbis_comment * comment = ov_comment (& vf, -1);
const gchar * new_title = (comment == NULL) ? NULL :
vorbis_comment_query (comment, "title", 0);
if (new_title != NULL && (title == NULL || strcmp (title, new_title)))
{
g_free (title);
title = g_strdup (new_title);
aud_input_set_tuple (get_tuple_for_vorbisfile (& vf,
filename));
}
}
if (current_section != last_section)
{
/*
* The info struct is different in each section. vf
* holds them all for the given bitstream. This
* requests the current one
*/
vi = ov_info(&vf, -1);
if (vi->channels > 2)
goto stop_processing;
if (vi->rate != samplerate || vi->channels != channels)
{
samplerate = vi->rate;
channels = vi->channels;
if (!aud_input_open_audio(FMT_FLOAT, vi->rate, vi->channels)) {
error = TRUE;
goto stop_processing;
}
vorbis_update_replaygain(&vf, &rg_info);
aud_input_set_gain (& rg_info); /* audio reopened */
}
}
aud_input_write_audio (pcmout, bytes);
stop_processing:
if (current_section != last_section)
{
aud_input_set_bitrate (br);
last_section = current_section;
}
} /* main loop */
play_cleanup:
ov_clear(&vf);
g_free (title);
return ! error;
}
static bool_t metronom_play (const char * filename, VFSFile * file)
{
metronom_t pmetronom;
int16_t data[BUF_SAMPLES];
int t = 0, tact, num;
int datagoal = 0;
int datamiddle = 0;
int datacurrent = datamiddle;
int datalast = datamiddle;
int data_form[TACT_FORM_MAX];
if (aud_input_open_audio(FMT_S16_NE, AUDIO_FREQ, 1) == 0)
return FALSE;
if (!metronom_get_cp(filename, &pmetronom, NULL))
{
fprintf (stderr, "Invalid metronom tact parameters in URI %s", filename);
return FALSE;
}
aud_input_set_bitrate(sizeof(data[0]) * 8 * AUDIO_FREQ);
tact = 60 * AUDIO_FREQ / pmetronom.bpm;
/* prepare weighted amplitudes */
for (num = 0; num < pmetronom.num; num++)
{
data_form[num] = MAX_AMPL * tact_form[pmetronom.id][num];
}
num = 0;
while (!aud_input_check_stop())
{
int i;
for (i = 0; i < BUF_SAMPLES; i++)
{
if (t == tact)
{
t = 0;
datagoal = data_form[num];
}
else if (t == 10)
{
datagoal = -data_form[num];
}
else if (t == 25)
{
datagoal = data_form[num];
/* circle through weighted amplitudes */
num++;
if (num >= pmetronom.num)
num = 0;
}
/* makes curve a little bit smoother */
data[i] = (datalast + datacurrent + datagoal) / 3;
datalast = datacurrent;
datacurrent = data[i];
if (t > 35)
datagoal = (datamiddle + 7 * datagoal) / 8;
t++;
}
aud_input_write_audio(data, BUF_BYTES);
}
return TRUE;
}
static gboolean vtx_play(const gchar * filename, VFSFile * file)
{
gboolean eof = FALSE;
void *stream; /* pointer to current position in sound buffer */
guchar regs[14];
gint need;
gint left; /* how many sound frames can play with current AY register frame */
gint donow;
gint rate;
left = 0;
rate = chans * (bits / 8);
memset(&ay, 0, sizeof(ay));
if (!ayemu_vtx_open(&vtx, filename))
{
g_print("libvtx: Error read vtx header from %s\n", filename);
return FALSE;
}
else if (!ayemu_vtx_load_data(&vtx))
{
g_print("libvtx: Error read vtx data from %s\n", filename);
return FALSE;
}
ayemu_init(&ay);
ayemu_set_chip_type(&ay, vtx.hdr.chiptype, NULL);
ayemu_set_chip_freq(&ay, vtx.hdr.chipFreq);
ayemu_set_stereo(&ay, vtx.hdr.stereo, NULL);
if (aud_input_open_audio(FMT_S16_NE, freq, chans) == 0)
{
g_print("libvtx: output audio error!\n");
return FALSE;
}
aud_input_set_bitrate(14 * 50 * 8);
while (!aud_input_check_stop() && !eof)
{
/* (time in sec) * 50 = offset in AY register data frames */
int seek_value = aud_input_check_seek();
if (seek_value >= 0)
vtx.pos = seek_value / 20;
/* fill sound buffer */
stream = sndbuf;
for (need = SNDBUFSIZE / rate; need > 0; need -= donow)
{
if (left > 0)
{ /* use current AY register frame */
donow = (need > left) ? left : need;
left -= donow;
stream = ayemu_gen_sound(&ay, (char *)stream, donow * rate);
}
else
{ /* get next AY register frame */
if (ayemu_vtx_get_next_frame(&vtx, (char *)regs) == 0)
{
donow = need;
memset(stream, 0, donow * rate);
eof = TRUE;
}
else
{
left = freq / vtx.hdr.playerFreq;
ayemu_set_regs(&ay, regs);
donow = 0;
}
}
}
aud_input_write_audio(sndbuf, SNDBUFSIZE);
}
ayemu_vtx_free(&vtx);
return TRUE;
}
static gboolean ffaudio_play (const gchar * filename, VFSFile * file)
{
AUDDBG ("Playing %s.\n", filename);
if (! file)
return FALSE;
AVPacket pkt = {.data = NULL};
gint errcount;
gboolean codec_opened = FALSE;
gint out_fmt;
gboolean planar;
gboolean seekable;
gboolean error = FALSE;
void *buf = NULL;
gint bufsize = 0;
AVFormatContext * ic = open_input_file (filename, file);
if (! ic)
return FALSE;
CodecInfo cinfo;
if (! find_codec (ic, & cinfo))
{
fprintf (stderr, "ffaudio: No codec found for %s.\n", filename);
goto error_exit;
}
AUDDBG("got codec %s for stream index %d, opening\n", cinfo.codec->name, cinfo.stream_idx);
if (avcodec_open2 (cinfo.context, cinfo.codec, NULL) < 0)
goto error_exit;
codec_opened = TRUE;
switch (cinfo.context->sample_fmt)
{
case AV_SAMPLE_FMT_U8: out_fmt = FMT_U8; planar = FALSE; break;
case AV_SAMPLE_FMT_S16: out_fmt = FMT_S16_NE; planar = FALSE; break;
case AV_SAMPLE_FMT_S32: out_fmt = FMT_S32_NE; planar = FALSE; break;
case AV_SAMPLE_FMT_FLT: out_fmt = FMT_FLOAT; planar = FALSE; break;
case AV_SAMPLE_FMT_U8P: out_fmt = FMT_U8; planar = TRUE; break;
case AV_SAMPLE_FMT_S16P: out_fmt = FMT_S16_NE; planar = TRUE; break;
case AV_SAMPLE_FMT_S32P: out_fmt = FMT_S32_NE; planar = TRUE; break;
case AV_SAMPLE_FMT_FLTP: out_fmt = FMT_FLOAT; planar = TRUE; break;
default:
fprintf (stderr, "ffaudio: Unsupported audio format %d\n", (int) cinfo.context->sample_fmt);
goto error_exit;
}
/* Open audio output */
AUDDBG("opening audio output\n");
if (aud_input_open_audio(out_fmt, cinfo.context->sample_rate, cinfo.context->channels) <= 0)
{
error = TRUE;
goto error_exit;
}
AUDDBG("setting parameters\n");
aud_input_set_bitrate(ic->bit_rate);
errcount = 0;
seekable = ffaudio_codec_is_seekable(cinfo.codec);
while (! aud_input_check_stop ())
{
int seek_value = aud_input_check_seek ();
if (seek_value >= 0 && seekable)
{
if (av_seek_frame (ic, -1, (gint64) seek_value * AV_TIME_BASE /
1000, AVSEEK_FLAG_ANY) < 0)
{
_ERROR("error while seeking\n");
} else
errcount = 0;
seek_value = -1;
}
AVPacket tmp;
gint ret;
/* Read next frame (or more) of data */
if ((ret = av_read_frame(ic, &pkt)) < 0)
{
if (ret == AVERROR_EOF)
{
AUDDBG("eof reached\n");
break;
}
else
{
if (++errcount > 4)
{
_ERROR("av_read_frame error %d, giving up.\n", ret);
break;
} else
continue;
}
} else
errcount = 0;
/* Ignore any other substreams */
if (pkt.stream_index != cinfo.stream_idx)
{
av_free_packet(&pkt);
continue;
}
/* Decode and play packet/frame */
memcpy(&tmp, &pkt, sizeof(tmp));
while (tmp.size > 0 && ! aud_input_check_stop ())
{
/* Check for seek request and bail out if we have one */
if (seek_value < 0)
seek_value = aud_input_check_seek ();
if (seek_value >= 0)
break;
#if CHECK_LIBAVCODEC_VERSION (55, 28, 1)
AVFrame * frame = av_frame_alloc ();
#else
AVFrame * frame = avcodec_alloc_frame ();
#endif
int decoded = 0;
int len = avcodec_decode_audio4 (cinfo.context, frame, & decoded, & tmp);
if (len < 0)
{
fprintf (stderr, "ffaudio: decode_audio() failed, code %d\n", len);
break;
}
tmp.size -= len;
tmp.data += len;
if (! decoded)
continue;
gint size = FMT_SIZEOF (out_fmt) * cinfo.context->channels * frame->nb_samples;
if (planar)
{
if (bufsize < size)
{
buf = g_realloc (buf, size);
bufsize = size;
}
audio_interlace ((const void * *) frame->data, out_fmt,
cinfo.context->channels, buf, frame->nb_samples);
aud_input_write_audio (buf, size);
}
else
aud_input_write_audio (frame->data[0], size);
#if CHECK_LIBAVCODEC_VERSION (55, 28, 1)
av_frame_free (& frame);
#else
avcodec_free_frame (& frame);
#endif
}
if (pkt.data)
av_free_packet(&pkt);
}
error_exit:
if (pkt.data)
av_free_packet(&pkt);
if (codec_opened)
avcodec_close(cinfo.context);
if (ic != NULL)
close_input_file(ic);
g_free (buf);
return ! error;
}
static bool_t flac_play (const char * filename, VFSFile * file)
{
if (!file)
return FALSE;
void * play_buffer = NULL;
bool_t error = FALSE;
info->fd = file;
if (read_metadata(decoder, info) == FALSE)
{
FLACNG_ERROR("Could not prepare file for playing!\n");
error = TRUE;
goto ERR_NO_CLOSE;
}
play_buffer = g_malloc (BUFFER_SIZE_BYTE);
if (! aud_input_open_audio (SAMPLE_FMT (info->bits_per_sample),
info->sample_rate, info->channels))
{
error = TRUE;
goto ERR_NO_CLOSE;
}
aud_input_set_bitrate(info->bitrate);
while (FLAC__stream_decoder_get_state(decoder) != FLAC__STREAM_DECODER_END_OF_STREAM)
{
if (aud_input_check_stop ())
break;
int seek_value = aud_input_check_seek ();
if (seek_value >= 0)
FLAC__stream_decoder_seek_absolute (decoder, (int64_t)
seek_value * info->sample_rate / 1000);
/* Try to decode a single frame of audio */
if (FLAC__stream_decoder_process_single(decoder) == FALSE)
{
FLACNG_ERROR("Error while decoding!\n");
error = TRUE;
break;
}
squeeze_audio(info->output_buffer, play_buffer, info->buffer_used, info->bits_per_sample);
aud_input_write_audio(play_buffer, info->buffer_used * SAMPLE_SIZE(info->bits_per_sample));
reset_info(info);
}
ERR_NO_CLOSE:
g_free (play_buffer);
reset_info(info);
if (FLAC__stream_decoder_flush(decoder) == FALSE)
FLACNG_ERROR("Could not flush decoder state!\n");
return ! error;
}