static void process_samples(struct example_sink_userdata *u, uint64_t expected_bytes)
{
pa_assert(u);
while (u->rendered_bytes < expected_bytes) {
/* read chunk from every connected sink input, mix them, allocate
* memblock, fill it with mixed samples, and return it to us.
*/
pa_memchunk chunk;
pa_sink_render(u->sink, 0, &chunk);
/* start reading chunk's memblock */
const char *buf = pa_memblock_acquire(chunk.memblock);
/* write samples from memblock to the file */
ssize_t sz =
write_samples(u->output_fd, buf + chunk.index, chunk.length);
if (sz != chunk.length) {
break;
}
u->rendered_bytes += chunk.length;
/* finish reading memblock */
pa_memblock_release(chunk.memblock);
/* return memblock to the pool */
pa_memblock_unref(chunk.memblock);
}
}
static int genode_run_out(HWVoiceOut *hw)
{
GenodeVoiceOut * const out = (GenodeVoiceOut *)hw;
Pcm_out_buffer & pcm_buf = out->pcm_buf;
int const live = audio_pcm_hw_get_live_out(&out->hw);
if (!live)
return 0;
int const decr = audio_MIN(live, out->hw.samples);
size_t const avail = pcm_buf.read_avail();
if ((avail / (VBOX_SAMPLE_SIZE*VBOX_CHANNELS)) < decr)
Genode::error(__func__, ": avail: ", avail, " < decr ", decr);
char buf[decr*VBOX_SAMPLE_SIZE*VBOX_CHANNELS];
pcm_buf.read(buf, sizeof(buf), true);
int const samples = write_samples(out, (int16_t*)buf, decr);
if (samples == 0) return 0;
pcm_buf.read_advance(samples * (VBOX_SAMPLE_SIZE*VBOX_CHANNELS));
out->hw.rpos = (out->hw.rpos + samples) % out->hw.samples;
return samples;
}
int main (int argc, const char *argv[])
{
CmdArgParser parser(argc, argv);
parser.setHeader("Archive Benchmark\n");
parser.setArgumentsInfo("<index>");
CmdArgFlag old(parser, "old", "Use old directory file");
CmdArgInt samples(parser, "samples", "<count>",
"Number of samples to write");
CmdArgString channel_name(parser, "channel", "<name>",
"Channel Name");
CmdArgFlag do_read(parser, "read",
"Perform read test");
// defaults
samples.set(100000);
channel_name.set("fred");
if (parser.parse() == false)
return -1;
if (parser.getArguments().size() != 1)
{
parser.usage();
return -1;
}
stdString index_name = parser.getArgument(0);
size_t count;
epicsTime start, stop;
try
{
if (do_read)
{
start = epicsTime::getCurrent ();
if (old)
count = old_read_samples(index_name, channel_name.get());
else
count = read_samples(index_name, channel_name.get());
stop = epicsTime::getCurrent ();
}
else
{
count = samples.get();
start = epicsTime::getCurrent ();
if (old)
old_write_samples(index_name, channel_name.get(), count);
else
write_samples(index_name, channel_name.get(), count);
stop = epicsTime::getCurrent ();
}
double secs = stop - start;
printf("%zd values in %g seconds: %g vals/sec\n",
count, secs,
(double)count / secs);
}
catch (GenericException &e)
{
fprintf(stderr, "Error:\n%s\n", e.what());
}
return 0;
}
void end_audio_frame() {
if (frame_offset == 0)
// No audio added; blip_end_frame() dislikes being called with an
// offset of 0
return;
assert(!(is_backwards_frame && frame_offset != get_frame_len()));
// Bring the signal level at the end of the frame to zero as outlined in
// set_audio_signal_level()
set_audio_signal_level(0);
blip_end_frame(blip, frame_offset);
if (playback_started) {
// Fudge playback rate by an amount proportional to the difference
// between the desired and current buffer fill levels to try to steer
// towards it
double const fudge_factor = 1.0 + 2*max_adjust*(0.5 - fill_level());
blip_set_rates(blip, cpu_clock_rate, sample_rate*fudge_factor);
}
else {
if (fill_level() >= 0.5) {
start_audio_playback();
playback_started = true;
}
}
int const n_samples = blip_read_samples(blip, blip_samples, ARRAY_LEN(blip_samples), 0);
// We expect to read all samples from blip_buf. If something goes wrong and
// we don't, clear the buffer to prevent data piling up in blip_buf's
// buffer (which lacks bounds checking).
int const avail = blip_samples_avail(blip);
if (avail != 0) {
printf("Warning: didn't read all samples from blip_buf (%d samples remain) - dropping samples\n",
avail);
blip_clear(blip);
}
#ifdef RECORD_MOVIE
add_movie_audio_frame(blip_samples, n_samples);
#endif
// Save the samples to the audio ring buffer
lock_audio();
write_samples(blip_samples, n_samples);
unlock_audio();
}
TESTABLE_STATIC int logging_sample(struct logging_status *ls,
LoggerMessage *msg)
{
/* If we haven't starting logging yet, then don't log (duh!) */
if (!ls->logging)
return 0;
int attempts = 2;
int rc = WRITE_FAIL;
while (attempts--) {
/*
* Open the log file if not yet done. This is good for
* both lazy opening and easy recovering of failure as
* this method will init the FS.
*/
if (WRITING_ACTIVE != ls->writing_status)
open_log_file(ls);
/* Don't try to write if file isn't open */
if (WRITING_ACTIVE == ls->writing_status) {
rc = write_samples(ls, msg);
if (0 == rc)
break;
}
/* If here, then unmount and try attempts more time */
pr_error("Remounting FS due to write error.\r\n");
close_log_file(ls);
/*
* We yield here because init/f_open/f_close all involve
* locks and can eat up significant resournce. This ensures
* we don't hog resources beyond reason. Or, you know, you
* could keep your damn SD card plugged in. --Stieg
*/
taskYIELD();
}
logging_led_toggle();
return rc;
}
/*
Encode a single frame of audio from 1152 samples
Audio samples are taken from glopts->buffer
Encoded bit stream is placed in to parameter bs
(not intended for use outside the library)
Returns the size of the frame
or -1 if there is an error
*/
static int encode_frame(twolame_options *glopts, bit_stream *bs)
{
int nch = glopts->frame.nch;
int sb, ch, adb, i;
unsigned long frameBits, initial_bits;
short sam[2][1056];
// MEANX
int head1,head2;
// /MEANX
if (!glopts->twolame_init) {
fprintf (stderr, "Please call twolame_init_params() before starting encoding.\n");
return -1;
}
// Scale and mix the input buffer
scale_and_mix_samples( glopts );
// Clear the saved audio buffer
memset ((char *) sam, 0, sizeof (sam));
/* Store the number of bits initially in the bit buffer */
initial_bits = buffer_sstell(bs);
adb = available_bits ( glopts );
/* allow the user to reserve some space at the end of the frame
This will however leave fewer bits for the audio.
Need to do a sanity check here to see that there are *some*
bits left. */
if (glopts->num_ancillary_bits > 0.6*adb) {
/* Trying to reserve more than 60% of the frame.
0.6 is arbitrary. but since most applications
probably only want to reserve a few bytes, this seems fine.
Typical frame size is about 800bytes */
fprintf(stderr,"You're trying to reserve more than 60%% of the mpeg frame for ancillary data\n");
fprintf(stderr,"This is probably an error. But I'll keep going anyway...\n");
}
adb -= glopts->num_ancillary_bits;
/* MFC 26 July 2003
Doing DAB became a bit harder in the reorganisation of the code.
Now there is no guarantee that there is more than one frame in the bitbuffer.
But DAB requires that the CRC for the *current* frame be written at the end
of the *previous* frame.
Workaround: Users (Nicholas?) wanting to implement DAB will have to do some work
in the frontend.
First: Reserve some bits for yourself (options->num_ancillary_bits)
Second: Put the encoder into "single frame mode" i.e. only read 1152 samples
per channel. (frontendoptions->singleFrameMode)
Third: When you receive each mp2 frame back from the library, you'll have to insert
the options->dabCrc[i] values into the end of the frame yourself. (DAB crc calc is
done below)
The frontend will have to keep the previous frame in memory.
As of 26July all that needs to be done is for the frontend to buffer one frame in memory, such that
the CRC for the next frame can be written in at the end of it.
*/
{
int gr, bl, ch;
/* New polyphase filter
Combines windowing and filtering. Ricardo Feb'03 */
for( gr = 0; gr < 3; gr++ )
for ( bl = 0; bl < 12; bl++ )
for ( ch = 0; ch < nch; ch++ )
window_filter_subband( &glopts->smem, &glopts->buffer[ch][gr * 12 * 32 + 32 * bl], ch,
&(*glopts->sb_sample)[ch][gr][bl][0] );
}
scalefactor_calc(*glopts->sb_sample, glopts->scalar, nch, glopts->frame.sblimit);
find_sf_max (glopts, glopts->scalar, glopts->max_sc);
if (glopts->frame.actual_mode == TWOLAME_JOINT_STEREO) {
/* this way we calculate more mono than we need */
/* but it is cheap */
combine_lr (*glopts->sb_sample, *glopts->j_sample, glopts->frame.sblimit);
scalefactor_calc (glopts->j_sample, &glopts->j_scale, 1, glopts->frame.sblimit);
}
if ((glopts->quickmode == TRUE) && (++glopts->psycount % glopts->quickcount != 0)) {
/* We're using quick mode, so we're only calculating the model every
'quickcount' frames. Otherwise, just copy the old ones across */
for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++) {
glopts->smr[ch][sb] = glopts->smrdef[ch][sb];
}
}
} else {
/* calculate the psymodel */
switch (glopts->psymodel) {
case -1:
psycho_n1 (glopts, glopts->smr, nch);
break;
case 0: /* Psy Model A */
psycho_0 (glopts, glopts->smr, glopts->scalar);
break;
case 1:
psycho_1 (glopts, glopts->buffer, glopts->max_sc, glopts->smr);
break;
case 2:
psycho_2 (glopts, glopts->buffer, sam, glopts->smr );
break;
case 3:
/* Modified psy model 1 */
psycho_3 (glopts, glopts->buffer, glopts->max_sc, glopts->smr);
break;
case 4:
psycho_4 (glopts, glopts->buffer, sam, glopts->smr );
break;
default:
fprintf (stderr, "Invalid psy model specification: %i\n", glopts->psymodel);
return -1;
break;
}
if (glopts->quickmode == TRUE) {
/* copy the smr values and reuse them later */
for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++) glopts->smrdef[ch][sb] = glopts->smr[ch][sb];
}
}
}
sf_transmission_pattern (glopts, glopts->scalar, glopts->scfsi);
main_bit_allocation (glopts, glopts->smr, glopts->scfsi, glopts->bit_alloc, &adb);
#if 0 //MEANX
if (glopts->error_protection)
crc_calc (glopts, glopts->bit_alloc, glopts->scfsi, &glopts->crc);
#endif
write_header (glopts, bs);
if (glopts->error_protection)
buffer_putbits (bs, glopts->crc, 16);
head1=write_bit_alloc (glopts, glopts->bit_alloc, bs);
head2=write_scalefactors(glopts, glopts->bit_alloc, glopts->scfsi, glopts->scalar, bs);
//MEANX printf("Bits: %d + %d = %d\n",head1,head2,head1+head2);
subband_quantization (glopts, glopts->scalar, *glopts->sb_sample, glopts->j_scale, *glopts->j_sample, glopts->bit_alloc,
*glopts->subband);
write_samples(glopts, *glopts->subband, glopts->bit_alloc, bs);
/* If not all the bits were used, write out a stack of zeros */
for (i = 0; i < adb; i++) buffer_put1bit (bs, 0);
/* MFC July 03
Write an extra byte for 16/24/32/48 input when padding is on.
Something must be going astray with the frame size calcs.
This fudge works fine for the moment */
if ((glopts->header.samplerate_idx != 0) && (glopts->padding)) // i.e. not a 44.1 or 22kHz input file
buffer_putbits (bs, 0, 8);
if (glopts->do_dab) {
// Do the CRC calc for DAB stuff if required.
// It will be up to the frontend to insert it into the end of the
// previous frame.
for (i=glopts->dab_crc_len-1; i>=0; i--) {
dab_crc_calc (glopts, glopts->bit_alloc, glopts->scfsi, glopts->scalar, &glopts->dab_crc[i], i);
}
}
// Allocate space for the reserved ancillary bits
for (i=0; i<glopts->num_ancillary_bits;i++)
buffer_put1bit(bs, 0);
// Calulate the number of bits in this frame
frameBits = buffer_sstell(bs) - initial_bits;
if (frameBits % 8) { /* a program failure */
fprintf (stderr, "Sent %ld bits = %ld slots plus %ld\n", frameBits, frameBits/8, frameBits%8);
fprintf (stderr, "If you are reading this, the program is broken\n");
fprintf (stderr, "email %s with the command line arguments and other info\n", PACKAGE_BUGREPORT);
return -1;
}
//fprintf(stdout,"Frame size: %li\n\n",frameBits/8);
// *** FIXME currently broken njh Sept 04
if (glopts->do_energy_levels)
do_energy_levels(glopts, bs);
// MEANX: Recompute checksum
if (glopts->error_protection)
{
unsigned char *begin=(initial_bits>>3)+bs->buf;
int protectedBits=head1+head2;
CRC_writeheader(protectedBits, begin);
}
int main( int argc, char *argv[] )
{
int i;
char fn_inspeech[80], fn_outspeech[80];
FILE *fin, *fout;
int numread, nsamp;
float temp[MAXSF];
unsigned long frame_num;
unsigned long total_frames;
unsigned long fer_count;
struct ENCODER_MEM encoder_memory;
struct DECODER_MEM decoder_memory;
struct PACKET packet;
struct CONTROL control;
int input_format;
int output_format;
#if USE_CALLOC
float *in_speech;
float *out_speech;
#else
float in_speech[FSIZE+LPCSIZE-FSIZE+LPCOFFSET];
float out_speech[FSIZE];
#endif
memset(&encoder_memory, 0, sizeof(encoder_memory));
memset(&decoder_memory, 0, sizeof(decoder_memory));
memset(&packet, 0, sizeof(packet));
memset(&control, 0, sizeof(control));
#if USE_CALLOC
alloc_mem_for_speech(&in_speech, &out_speech);
#endif
for(i = 0; i < argc; i++)
fprintf(stdout, "%s ", argv[i]);
fprintf(stdout, "\n");
parse_command_line(argc, argv, fn_inspeech, fn_outspeech, &control);
initialize_encoder_and_decoder(&encoder_memory, &decoder_memory,
&control);
print_welcome_message();
/*** Init TTY/TDD Routines and Varibles ***/
if( tty_debug_flag )
tty_debug();
if( tty_option == TTY_NO_GAIN )
{
fprintf(stdout," TTY OPTION = NO GAIN\n");
init_tty_enc( &tty_enc_char, &tty_enc_header, &tty_enc_baud_rate);
init_tty_dec();
}
else
{
tty_option = 0;
fprintf(stdout," TTY OPTION = OFF\n");
}
if( trans_fname != NULL )
{
fprintf(stdout,"FER SIMULATOR ON: seed = %d\n",fer_sim_seed);
}
for(i = 0; i < LPCORDER; i++)
packet.lsp[i] = packet.lpc[i] = 0;
encoder_memory.frame_num = 0;
frame_num = 0;
fer_count = 0;
if (control.decode_only == YES)
{
/* Input is a CELP file */
switch (control.celp_file_format)
{
case FORMAT_PACKET:
open_binary_input_file(&fin, fn_inspeech);
total_frames = GetNumFrames(fin, sizeof(short)*WORDS_PER_PACKET);
input_format = FORMAT_PACKET;
break;
case FORMAT_QCP:
open_qcp_input_file(&fin, fn_inspeech);
total_frames = get_qcp_packet_count();
input_format = FORMAT_QCP;
break;
default:
fprintf(stderr, "unsupported decode_only input format %d\n", control.celp_file_format);
exit(-2);
}
}
else
{
/* Input is an audio file */
open_binary_input_file(&fin, fn_inspeech);
input_format = FORMAT_RAW_AUDIO;
total_frames = GetNumFrames(fin, sizeof(short)*FSIZE);
}
if ((control.form_res_out == YES)
|| (control.target_after_out == YES)
|| (control.cb_out == YES)
|| (control.pitch_out == YES))
{
/* Output is encoder state for debugging */
open_binary_output_file(&fout, fn_outspeech);
output_format = FORMAT_DEBUG_OUTPUT;
}
else if (control.encode_only == YES)
{
/* Output is a CELP file */
switch (control.celp_file_format)
{
case FORMAT_PACKET:
open_binary_output_file(&fout, fn_outspeech);
output_format = FORMAT_PACKET;
break;
case FORMAT_QCP:
open_qcp_output_file(&fout, fn_outspeech, total_frames);
output_format = FORMAT_QCP;
break;
default:
fprintf(stderr, "unsupported encode_only output format %d\n", control.celp_file_format);
exit(-2);
}
}
else
{
/* Output is an audio file */
open_binary_output_file(&fout, fn_outspeech);
output_format = FORMAT_RAW_AUDIO;
}
if(control.decode_only == NO)
{
#if 0
if (read_samples(fin, in_speech, LPCSIZE-FSIZE+LPCOFFSET)
!=LPCSIZE-FSIZE+LPCOFFSET)
{
printf("Not even enough samples for 1 frame!\n");
usage(&control);
}
#else
for( i=0 ; i < LPCSIZE-FSIZE+LPCOFFSET ; i++ )
{
in_speech[i] = 0;
}
#endif
}
/*-----------------------------------------------
* Main Loop
*------------------------------------------------*/
while( control.num_frames == UNLIMITED || frame_num < control.num_frames )
{
fprintf(stderr,"Processing %lu of %lu FER = %.2f%%\r",
frame_num, total_frames, 100.0*fer_count/(frame_num+1));
if (control.decode_only==NO)
{
nsamp = read_samples(fin, &in_speech[LPCSIZE-FSIZE+LPCOFFSET], FSIZE);
if( nsamp <= 0 )
{
break;
}
else if(nsamp < FSIZE)
{
for (i=nsamp; i<FSIZE; i++)
{
in_speech[LPCSIZE-FSIZE+LPCOFFSET+i]=0;
}
}
encoder(in_speech, &packet, &control,
&encoder_memory, out_speech);
update_snr(ENCODER, in_speech, out_speech, &(control.snr));
}
if (control.decode_only==YES)
{
if (input_format == FORMAT_QCP)
{
numread = read_qcp_packet(fin, packet.data, WORDS_PER_PACKET);
if (numread == 0) break;
}
else
{
/* FORMAT_PACKET */
numread = read_packet(fin, packet.data, WORDS_PER_PACKET);
if( numread != WORDS_PER_PACKET)
{
if(numread != 0)
{
fprintf(stderr,
"%s: Wrong number of words read: %d\n", argv[0], numread);
}
break;
}
}
}
if(control.encode_only==NO)
{
if (control.output_encoder_speech==NO)
{
decoder(out_speech, &packet, &control, &decoder_memory);
if( packet.data[0] == ERASURE )
{
fer_count++;
}
if(control.decode_only==NO)
{
update_snr(DECODER, in_speech, out_speech, &(control.snr));
}
}
i = write_samples(fout,out_speech,FSIZE);
}
else
{
if( trans_fname != NULL )
{
fer_sim( &(packet.data[0]) );
}
if( packet.data[0] == ERASURE )
{
fer_count++;
}
if (output_format == FORMAT_QCP)
{
i = write_qcp_packet(fout, packet.data, WORDS_PER_PACKET);
}
else
{
i = write_packet(fout, packet.data, WORDS_PER_PACKET);
}
}
/***** Update in_speech buffer ***************/
for (i=0; i<LPCSIZE-FSIZE+LPCOFFSET; i++)
{
in_speech[i]=in_speech[i+FSIZE];
}
frame_num++;
encoder_memory.frame_num = frame_num;
} /* end main while() */
if (output_format == FORMAT_QCP)
{
finish_qcp_output_file(fout);
}
fclose(fin);
fclose(fout);
if ((control.encode_only==NO)&&(control.decode_only==NO))
{
compute_snr(&(control.snr), &control);
}
if( control.decode_only == NO )
{
/* calculate the avg. rate for active speech for the entire file */
temp[0] = (encoder_memory.full_cnt + encoder_memory.full_force +
encoder_memory.full_cnt_t + encoder_memory.full_force_t)*14.4+
(encoder_memory.half_cnt + encoder_memory.half_force +
encoder_memory.half_cnt_t + encoder_memory.half_force_t)*7.2+
(encoder_memory.quarter_cnt + encoder_memory.quarter_cnt_t)*3.6;
temp[0] /= (encoder_memory.total_speech_frames+
(STATWINDOW-encoder_memory.block_cnt));
if(control.reduced_rate_flag != 0)
{
printf("\n The target_snr_threshold at the end of the run was %f",
encoder_memory.target_snr_thr);
printf("\n The average rate for the entire file was %f",temp[0]);
i = STATWINDOW-encoder_memory.block_cnt+encoder_memory.total_speech_frames;
temp[1] = i;
printf("\n The # of speech frames in the file is = %d",i);
i = encoder_memory.full_cnt+encoder_memory.full_cnt_t;
printf("\n The percent of frames at 14.4 is %f",100.0*i/temp[1]);
i = encoder_memory.full_force+encoder_memory.full_force_t;
printf("\n The percent of frames forced to 14.4 is %f",100.*i/temp[1]);
i = encoder_memory.half_cnt+encoder_memory.half_cnt_t;
printf("\n The percent of frames at 7.2 is %f",100.*i/temp[1]);
i = encoder_memory.half_force+encoder_memory.half_force_t;
printf("\n The percent of frames forced to 7.2 is %f",100.*i/temp[1]);
i = encoder_memory.quarter_cnt+encoder_memory.quarter_cnt_t;
printf("\n The percent of frames coded at 3.6 is %f\n",100.*i/temp[1]);
}
}
if(control.encode_only == NO)
print_erasure_count();
free_encoder_and_decoder(&encoder_memory, &decoder_memory);
#if USE_CALLOC
free((char*) in_speech);
free((char*) out_speech);
#endif
print_farewell_message();
exit(0);
}/* end of main() */
/*
Encode a single frame of audio from 1152 samples
Audio samples are taken from glopts->buffer
Encoded bit stream is placed in to parameter bs
(not intended for use outside the library)
Returns the size of the frame
or -1 if there is an error
*/
static int encode_frame(twolame_options * glopts, bit_stream * bs)
{
int nch = glopts->num_channels_out;
int sb, ch, adb, i;
unsigned long frameBits, initial_bits;
short sam[2][1056];
if (!glopts->twolame_init) {
fprintf(stderr, "Please call twolame_init_params() before starting encoding.\n");
return -1;
}
// Scale and mix the input buffer
scale_and_mix_samples(glopts);
// Clear the saved audio buffer
memset((char *) sam, 0, sizeof(sam));
// Number of bits to calculate CRC on
glopts->num_crc_bits = 0;
// Store the number of bits initially in the bit buffer
initial_bits = buffer_sstell(bs);
adb = available_bits(glopts);
/* allow the user to reserve some space at the end of the frame This will however leave fewer
bits for the audio. Need to do a sanity check here to see that there are *some* bits left. */
if (glopts->num_ancillary_bits > 0.6 * adb) {
/* Trying to reserve more than 60% of the frame. 0.6 is arbitrary. but since most
applications probably only want to reserve a few bytes, this seems fine. Typical frame
size is about 800bytes */
fprintf(stderr,
"You're trying to reserve more than 60%% of the mpeg frame for ancillary data\n");
fprintf(stderr, "This is probably an error. But I'll keep going anyway...\n");
}
adb -= glopts->num_ancillary_bits;
/* The DAB scf-crc calc is done below. The frontend will have to keep the previous frame in
memory. As of 09May 2014 all that needs to be done is for the frontend to buffer one frame in
memory and call twolame_set_DAB_scf_crc */
{
int gr, bl, ch;
/* New polyphase filter Combines windowing and filtering. Ricardo Feb'03 */
for (gr = 0; gr < 3; gr++)
for (bl = 0; bl < 12; bl++)
for (ch = 0; ch < nch; ch++)
window_filter_subband(&glopts->smem,
&glopts->buffer[ch][gr * 12 * 32 + 32 * bl], ch,
&(*glopts->sb_sample)[ch][gr][bl][0]);
}
scalefactor_calc(*glopts->sb_sample, glopts->scalar, nch, glopts->sblimit);
find_sf_max(glopts, glopts->scalar, glopts->max_sc);
if (glopts->mode == TWOLAME_JOINT_STEREO) {
// this way we calculate more mono than we need but it is cheap
combine_lr(*glopts->sb_sample, *glopts->j_sample, glopts->sblimit);
scalefactor_calc(glopts->j_sample, &glopts->j_scale, 1, glopts->sblimit);
}
if ((glopts->quickmode == TRUE) && (++glopts->psycount % glopts->quickcount != 0)) {
/* We're using quick mode, so we're only calculating the model every 'quickcount' frames.
Otherwise, just copy the old ones across */
for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++) {
glopts->smr[ch][sb] = glopts->smrdef[ch][sb];
}
}
} else {
// calculate the psymodel
switch (glopts->psymodel) {
case -1:
psycho_n1(glopts, glopts->smr, nch);
break;
case 0: // Psy Model A
psycho_0(glopts, glopts->smr, glopts->scalar);
break;
case 1:
psycho_1(glopts, glopts->buffer, glopts->max_sc, glopts->smr);
break;
case 2:
psycho_2(glopts, glopts->buffer, sam, glopts->smr);
break;
case 3:
// Modified psy model 1
psycho_3(glopts, glopts->buffer, glopts->max_sc, glopts->smr);
break;
case 4:
// Modified psy model 2
psycho_4(glopts, glopts->buffer, sam, glopts->smr);
break;
default:
fprintf(stderr, "Invalid psy model specification: %i\n", glopts->psymodel);
return -1;
break;
}
if (glopts->quickmode == TRUE) {
// copy the smr values and reuse them later
for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++)
glopts->smrdef[ch][sb] = glopts->smr[ch][sb];
}
}
}
sf_transmission_pattern(glopts, glopts->scalar, glopts->scfsi);
main_bit_allocation(glopts, glopts->smr, glopts->scfsi, glopts->bit_alloc, &adb);
write_header(glopts, bs);
// Leave space for 2 bytes of CRC to be filled in later
if (glopts->error_protection)
buffer_putbits(bs, 0, 16);
write_bit_alloc(glopts, glopts->bit_alloc, bs);
write_scalefactors(glopts, glopts->bit_alloc, glopts->scfsi, glopts->scalar, bs);
subband_quantization(glopts, glopts->scalar, *glopts->sb_sample, glopts->j_scale,
*glopts->j_sample, glopts->bit_alloc, *glopts->subband);
write_samples(glopts, *glopts->subband, glopts->bit_alloc, bs);
// If not all the bits were used, write out a stack of zeros
for (i = 0; i < adb; i++)
buffer_put1bit(bs, 0);
/* MFC July 03 FIXME Write an extra byte for 16/24/32/48 input when padding is on. Something
must be going astray with the frame size calcs. This fudge works fine for the moment */
if ((glopts->header.samplerate_idx != 0) && (glopts->padding)) // i.e. not a 44.1 or 22kHz
// input file
buffer_putbits(bs, 0, 8);
if (glopts->do_dab) {
// Do the CRC calc for DAB stuff if required.
// It will be up to the frontend to insert it into the end of the
// previous frame.
for (i = glopts->dab_crc_len - 1; i >= 0; i--) {
dab_crc_calc(glopts, glopts->bit_alloc, glopts->scfsi, glopts->scalar,
&glopts->dab_crc[i], i);
}
}
// Allocate space for the reserved ancillary bits
for (i = 0; i < glopts->num_ancillary_bits; i++)
buffer_put1bit(bs, 0);
// Calulate the number of bits in this frame
frameBits = buffer_sstell(bs) - initial_bits;
if (frameBits % 8) { /* a program failure */
fprintf(stderr, "Sent %ld bits = %ld slots plus %ld\n", frameBits, frameBits / 8,
frameBits % 8);
fprintf(stderr, "If you are reading this, the program is broken\n");
fprintf(stderr, "email %s with the command line arguments and other info\n",
PACKAGE_BUGREPORT);
return -1;
}
// Store the energy levels at the end of the frame
if (glopts->do_energy_levels)
do_energy_levels(glopts, bs);
// MEANX: Recompute checksum from bitstream
if (glopts->error_protection) {
unsigned char *frame_ptr = bs->buf + (initial_bits >> 3);
crc_writeheader(frame_ptr, glopts->num_crc_bits);
}
