* Simple free lossless/lossy audio codec
* Copyright (c) 2004 Alex Beregszaszi
*
- * This library is free software; you can redistribute it and/or
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
+#include "bitstream.h"
#include "golomb.h"
/**
typedef struct SonicContext {
int lossless, decorrelation;
-
+
int num_taps, downsampling;
double quantization;
-
+
int channels, samplerate, block_align, frame_size;
int *tap_quant;
int *predictor_state[MAX_CHANNELS];
} SonicContext;
-#define LATTICE_SHIFT 10
-#define SAMPLE_SHIFT 4
-#define LATTICE_FACTOR (1 << LATTICE_SHIFT)
-#define SAMPLE_FACTOR (1 << SAMPLE_SHIFT)
+#define LATTICE_SHIFT 10
+#define SAMPLE_SHIFT 4
+#define LATTICE_FACTOR (1 << LATTICE_SHIFT)
+#define SAMPLE_FACTOR (1 << SAMPLE_SHIFT)
-#define BASE_QUANT 0.6
-#define RATE_VARIATION 3.0
+#define BASE_QUANT 0.6
+#define RATE_VARIATION 3.0
static inline int divide(int a, int b)
{
if (a < 0)
- return -( (-a + b/2)/b );
+ return -( (-a + b/2)/b );
else
- return (a + b/2)/b;
+ return (a + b/2)/b;
}
static inline int shift(int a,int b)
int i;
for (i = 0; i < entries; i++)
- set_se_golomb(pb, buf[i]);
+ set_se_golomb(pb, buf[i]);
return 1;
}
static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
{
int i;
-
+
for (i = 0; i < entries; i++)
- buf[i] = get_se_golomb(gb);
+ buf[i] = get_se_golomb(gb);
return 1;
}
static int bits_to_store(uint64_t x)
{
int res = 0;
-
+
while(x)
{
- res++;
- x >>= 1;
+ res++;
+ x >>= 1;
}
return res;
}
int i, bits;
if (!max)
- return;
+ return;
bits = bits_to_store(max);
for (i = 0; i < bits-1; i++)
- put_bits(pb, 1, value & (1 << i));
+ put_bits(pb, 1, value & (1 << i));
if ( (value | (1 << (bits-1))) <= max)
- put_bits(pb, 1, value & (1 << (bits-1)));
+ put_bits(pb, 1, value & (1 << (bits-1)));
}
static unsigned int read_uint_max(GetBitContext *gb, int max)
{
int i, bits, value = 0;
-
+
if (!max)
- return 0;
+ return 0;
bits = bits_to_store(max);
for (i = 0; i < bits-1; i++)
- if (get_bits1(gb))
- value += 1 << i;
+ if (get_bits1(gb))
+ value += 1 << i;
if ( (value | (1<<(bits-1))) <= max)
- if (get_bits1(gb))
- value += 1 << (bits-1);
+ if (get_bits1(gb))
+ value += 1 << (bits-1);
return value;
}
copy = av_mallocz(4* entries);
if (!copy)
- return -1;
-
+ return -1;
+
if (base_2_part)
{
- int energy = 0;
-
- for (i = 0; i < entries; i++)
- energy += abs(buf[i]);
-
- low_bits = bits_to_store(energy / (entries * 2));
- if (low_bits > 15)
- low_bits = 15;
-
- put_bits(pb, 4, low_bits);
- }
-
+ int energy = 0;
+
+ for (i = 0; i < entries; i++)
+ energy += abs(buf[i]);
+
+ low_bits = bits_to_store(energy / (entries * 2));
+ if (low_bits > 15)
+ low_bits = 15;
+
+ put_bits(pb, 4, low_bits);
+ }
+
for (i = 0; i < entries; i++)
{
- put_bits(pb, low_bits, abs(buf[i]));
- copy[i] = abs(buf[i]) >> low_bits;
- if (copy[i] > max)
- max = abs(copy[i]);
+ put_bits(pb, low_bits, abs(buf[i]));
+ copy[i] = abs(buf[i]) >> low_bits;
+ if (copy[i] > max)
+ max = abs(copy[i]);
}
bits = av_mallocz(4* entries*max);
if (!bits)
{
-// av_free(copy);
- return -1;
+// av_free(copy);
+ return -1;
}
-
+
for (i = 0; i <= max; i++)
{
- for (j = 0; j < entries; j++)
- if (copy[j] >= i)
- bits[x++] = copy[j] > i;
+ for (j = 0; j < entries; j++)
+ if (copy[j] >= i)
+ bits[x++] = copy[j] > i;
}
// store bitstream
while (pos < x)
{
- int steplet = step >> 8;
-
- if (pos + steplet > x)
- steplet = x - pos;
-
- for (i = 0; i < steplet; i++)
- if (bits[i+pos] != dominant)
- any = 1;
-
- put_bits(pb, 1, any);
-
- if (!any)
- {
- pos += steplet;
- step += step / ADAPT_LEVEL;
- }
- else
- {
- int interloper = 0;
-
- while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
- interloper++;
-
- // note change
- write_uint_max(pb, interloper, (step >> 8) - 1);
-
- pos += interloper + 1;
- step -= step / ADAPT_LEVEL;
- }
-
- if (step < 256)
- {
- step = 65536 / step;
- dominant = !dominant;
- }
- }
-
+ int steplet = step >> 8;
+
+ if (pos + steplet > x)
+ steplet = x - pos;
+
+ for (i = 0; i < steplet; i++)
+ if (bits[i+pos] != dominant)
+ any = 1;
+
+ put_bits(pb, 1, any);
+
+ if (!any)
+ {
+ pos += steplet;
+ step += step / ADAPT_LEVEL;
+ }
+ else
+ {
+ int interloper = 0;
+
+ while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
+ interloper++;
+
+ // note change
+ write_uint_max(pb, interloper, (step >> 8) - 1);
+
+ pos += interloper + 1;
+ step -= step / ADAPT_LEVEL;
+ }
+
+ if (step < 256)
+ {
+ step = 65536 / step;
+ dominant = !dominant;
+ }
+ }
+
// store signs
for (i = 0; i < entries; i++)
- if (buf[i])
- put_bits(pb, 1, buf[i] < 0);
+ if (buf[i])
+ put_bits(pb, 1, buf[i] < 0);
// av_free(bits);
// av_free(copy);
int *bits = av_mallocz(4* entries);
if (!bits)
- return -1;
-
+ return -1;
+
if (base_2_part)
{
- low_bits = get_bits(gb, 4);
+ low_bits = get_bits(gb, 4);
- if (low_bits)
- for (i = 0; i < entries; i++)
- buf[i] = get_bits(gb, low_bits);
+ if (low_bits)
+ for (i = 0; i < entries; i++)
+ buf[i] = get_bits(gb, low_bits);
}
// av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits);
while (n_zeros < entries)
{
- int steplet = step >> 8;
-
- if (!get_bits1(gb))
- {
- for (i = 0; i < steplet; i++)
- bits[x++] = dominant;
-
- if (!dominant)
- n_zeros += steplet;
-
- step += step / ADAPT_LEVEL;
- }
- else
- {
- int actual_run = read_uint_max(gb, steplet-1);
-
-// av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run);
-
- for (i = 0; i < actual_run; i++)
- bits[x++] = dominant;
-
- bits[x++] = !dominant;
-
- if (!dominant)
- n_zeros += actual_run;
- else
- n_zeros++;
-
- step -= step / ADAPT_LEVEL;
- }
-
- if (step < 256)
- {
- step = 65536 / step;
- dominant = !dominant;
- }
- }
-
+ int steplet = step >> 8;
+
+ if (!get_bits1(gb))
+ {
+ for (i = 0; i < steplet; i++)
+ bits[x++] = dominant;
+
+ if (!dominant)
+ n_zeros += steplet;
+
+ step += step / ADAPT_LEVEL;
+ }
+ else
+ {
+ int actual_run = read_uint_max(gb, steplet-1);
+
+// av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run);
+
+ for (i = 0; i < actual_run; i++)
+ bits[x++] = dominant;
+
+ bits[x++] = !dominant;
+
+ if (!dominant)
+ n_zeros += actual_run;
+ else
+ n_zeros++;
+
+ step -= step / ADAPT_LEVEL;
+ }
+
+ if (step < 256)
+ {
+ step = 65536 / step;
+ dominant = !dominant;
+ }
+ }
+
// reconstruct unsigned values
n_zeros = 0;
for (i = 0; n_zeros < entries; i++)
{
- while(1)
- {
- if (pos >= entries)
- {
- pos = 0;
- level += 1 << low_bits;
- }
-
- if (buf[pos] >= level)
- break;
-
- pos++;
- }
-
- if (bits[i])
- buf[pos] += 1 << low_bits;
- else
- n_zeros++;
-
- pos++;
+ while(1)
+ {
+ if (pos >= entries)
+ {
+ pos = 0;
+ level += 1 << low_bits;
+ }
+
+ if (buf[pos] >= level)
+ break;
+
+ pos++;
+ }
+
+ if (bits[i])
+ buf[pos] += 1 << low_bits;
+ else
+ n_zeros++;
+
+ pos++;
}
// av_free(bits);
-
+
// read signs
for (i = 0; i < entries; i++)
- if (buf[i] && get_bits1(gb))
- buf[i] = -buf[i];
+ if (buf[i] && get_bits1(gb))
+ buf[i] = -buf[i];
// av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos);
for (i = order-2; i >= 0; i--)
{
- int j, p, x = state[i];
+ int j, p, x = state[i];
- for (j = 0, p = i+1; p < order; j++,p++)
- {
- int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT);
- state[p] += shift_down(k[j]*x, LATTICE_SHIFT);
- x = tmp;
- }
+ for (j = 0, p = i+1; p < order; j++,p++)
+ {
+ int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT);
+ state[p] += shift_down(k[j]*x, LATTICE_SHIFT);
+ x = tmp;
+ }
}
}
#if 1
int *k_ptr = &(k[order-2]),
- *state_ptr = &(state[order-2]);
+ *state_ptr = &(state[order-2]);
for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--)
{
- int k_value = *k_ptr, state_value = *state_ptr;
- x -= shift_down(k_value * state_value, LATTICE_SHIFT);
- state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT);
+ int k_value = *k_ptr, state_value = *state_ptr;
+ x -= shift_down(k_value * state_value, LATTICE_SHIFT);
+ state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT);
}
#else
for (i = order-2; i >= 0; i--)
{
- x -= shift_down(k[i] * state[i], LATTICE_SHIFT);
- state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT);
+ x -= shift_down(k[i] * state[i], LATTICE_SHIFT);
+ state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT);
}
#endif
- // don't drift too far, to avoid overflows
+ // don't drift too far, to avoid overflows
if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16);
if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16);
return x;
}
+#ifdef CONFIG_ENCODERS
// Heavily modified Levinson-Durbin algorithm which
// copes better with quantization, and calculates the
// actual whitened result as it goes.
static void modified_levinson_durbin(int *window, int window_entries,
- int *out, int out_entries, int channels, int *tap_quant)
+ int *out, int out_entries, int channels, int *tap_quant)
{
int i;
int *state = av_mallocz(4* window_entries);
-
+
memcpy(state, window, 4* window_entries);
-
+
for (i = 0; i < out_entries; i++)
{
- int step = (i+1)*channels, k, j;
- double xx = 0.0, xy = 0.0;
+ int step = (i+1)*channels, k, j;
+ double xx = 0.0, xy = 0.0;
#if 1
- int *x_ptr = &(window[step]), *state_ptr = &(state[0]);
- j = window_entries - step;
- for (;j>=0;j--,x_ptr++,state_ptr++)
- {
- double x_value = *x_ptr, state_value = *state_ptr;
- xx += state_value*state_value;
- xy += x_value*state_value;
- }
+ int *x_ptr = &(window[step]), *state_ptr = &(state[0]);
+ j = window_entries - step;
+ for (;j>=0;j--,x_ptr++,state_ptr++)
+ {
+ double x_value = *x_ptr, state_value = *state_ptr;
+ xx += state_value*state_value;
+ xy += x_value*state_value;
+ }
#else
- for (j = 0; j <= (window_entries - step); j++);
- {
- double stepval = window[step+j], stateval = window[j];
-// xx += (double)window[j]*(double)window[j];
-// xy += (double)window[step+j]*(double)window[j];
- xx += stateval*stateval;
- xy += stepval*stateval;
- }
+ for (j = 0; j <= (window_entries - step); j++);
+ {
+ double stepval = window[step+j], stateval = window[j];
+// xx += (double)window[j]*(double)window[j];
+// xy += (double)window[step+j]*(double)window[j];
+ xx += stateval*stateval;
+ xy += stepval*stateval;
+ }
#endif
- if (xx == 0.0)
- k = 0;
- else
- k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5));
-
- if (k > (LATTICE_FACTOR/tap_quant[i]))
- k = LATTICE_FACTOR/tap_quant[i];
- if (-k > (LATTICE_FACTOR/tap_quant[i]))
- k = -(LATTICE_FACTOR/tap_quant[i]);
-
- out[i] = k;
- k *= tap_quant[i];
+ if (xx == 0.0)
+ k = 0;
+ else
+ k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5));
+
+ if (k > (LATTICE_FACTOR/tap_quant[i]))
+ k = LATTICE_FACTOR/tap_quant[i];
+ if (-k > (LATTICE_FACTOR/tap_quant[i]))
+ k = -(LATTICE_FACTOR/tap_quant[i]);
+
+ out[i] = k;
+ k *= tap_quant[i];
#if 1
- x_ptr = &(window[step]);
- state_ptr = &(state[0]);
- j = window_entries - step;
- for (;j>=0;j--,x_ptr++,state_ptr++)
- {
- int x_value = *x_ptr, state_value = *state_ptr;
- *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
- *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
- }
-#else
- for (j=0; j <= (window_entries - step); j++)
- {
- int stepval = window[step+j], stateval=state[j];
- window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
- state[j] += shift_down(k * stepval, LATTICE_SHIFT);
- }
+ x_ptr = &(window[step]);
+ state_ptr = &(state[0]);
+ j = window_entries - step;
+ for (;j>=0;j--,x_ptr++,state_ptr++)
+ {
+ int x_value = *x_ptr, state_value = *state_ptr;
+ *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
+ *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
+ }
+#else
+ for (j=0; j <= (window_entries - step); j++)
+ {
+ int stepval = window[step+j], stateval=state[j];
+ window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
+ state[j] += shift_down(k * stepval, LATTICE_SHIFT);
+ }
#endif
}
-
+
av_free(state);
}
+#endif /* CONFIG_ENCODERS */
static int samplerate_table[] =
{ 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
{
switch (samplerate)
{
- case 44100: return 0;
- case 22050: return 1;
- case 11025: return 2;
- case 96000: return 3;
- case 48000: return 4;
- case 32000: return 5;
- case 24000: return 6;
- case 16000: return 7;
- case 8000: return 8;
+ case 44100: return 0;
+ case 22050: return 1;
+ case 11025: return 2;
+ case 96000: return 3;
+ case 48000: return 4;
+ case 32000: return 5;
+ case 24000: return 6;
+ case 16000: return 7;
+ case 8000: return 8;
}
return -1;
}
if (avctx->channels > MAX_CHANNELS)
{
- av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
+ av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
return -1; /* only stereo or mono for now */
}
if (avctx->channels == 2)
- s->decorrelation = MID_SIDE;
+ s->decorrelation = MID_SIDE;
if (avctx->codec->id == CODEC_ID_SONIC_LS)
{
- s->lossless = 1;
- s->num_taps = 32;
- s->downsampling = 1;
- s->quantization = 0.0;
+ s->lossless = 1;
+ s->num_taps = 32;
+ s->downsampling = 1;
+ s->quantization = 0.0;
}
else
{
- s->num_taps = 128;
- s->downsampling = 2;
- s->quantization = 1.0;
+ s->num_taps = 128;
+ s->downsampling = 2;
+ s->quantization = 1.0;
}
// max tap 2048
if ((s->num_taps < 32) || (s->num_taps > 1024) ||
- ((s->num_taps>>5)<<5 != s->num_taps))
+ ((s->num_taps>>5)<<5 != s->num_taps))
{
- av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
- return -1;
+ av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
+ return -1;
}
// generate taps
s->tap_quant = av_mallocz(4* s->num_taps);
for (i = 0; i < s->num_taps; i++)
- s->tap_quant[i] = (int)(sqrt(i+1));
+ s->tap_quant[i] = (int)(sqrt(i+1));
s->channels = avctx->channels;
s->samplerate = avctx->sample_rate;
s->tail = av_mallocz(4* s->num_taps*s->channels);
if (!s->tail)
- return -1;
+ return -1;
s->tail_size = s->num_taps*s->channels;
s->predictor_k = av_mallocz(4 * s->num_taps);
if (!s->predictor_k)
- return -1;
+ return -1;
for (i = 0; i < s->channels; i++)
{
- s->coded_samples[i] = av_mallocz(4* s->block_align);
- if (!s->coded_samples[i])
- return -1;
+ s->coded_samples[i] = av_mallocz(4* s->block_align);
+ if (!s->coded_samples[i])
+ return -1;
}
-
+
s->int_samples = av_mallocz(4* s->frame_size);
s->window_size = ((2*s->tail_size)+s->frame_size);
s->window = av_mallocz(4* s->window_size);
if (!s->window)
- return -1;
+ return -1;
avctx->extradata = av_mallocz(16);
if (!avctx->extradata)
- return -1;
+ return -1;
init_put_bits(&pb, avctx->extradata, 16*8);
put_bits(&pb, 2, version); // version
if (version == 1)
{
- put_bits(&pb, 2, s->channels);
- put_bits(&pb, 4, code_samplerate(s->samplerate));
+ put_bits(&pb, 2, s->channels);
+ put_bits(&pb, 4, code_samplerate(s->samplerate));
}
put_bits(&pb, 1, s->lossless);
if (!s->lossless)
- put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision
+ put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision
put_bits(&pb, 2, s->decorrelation);
put_bits(&pb, 2, s->downsampling);
put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024
avctx->extradata_size = put_bits_count(&pb)/8;
av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
- version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
+ version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
- return -ENOMEM;
+ return AVERROR(ENOMEM);
avctx->coded_frame->key_frame = 1;
avctx->frame_size = s->block_align*s->downsampling;
av_freep(&avctx->coded_frame);
for (i = 0; i < s->channels; i++)
- av_free(s->coded_samples[i]);
+ av_free(s->coded_samples[i]);
av_free(s->predictor_k);
av_free(s->tail);
}
static int sonic_encode_frame(AVCodecContext *avctx,
- uint8_t *buf, int buf_size, void *data)
+ uint8_t *buf, int buf_size, void *data)
{
SonicContext *s = avctx->priv_data;
PutBitContext pb;
// short -> internal
for (i = 0; i < s->frame_size; i++)
- s->int_samples[i] = samples[i];
+ s->int_samples[i] = samples[i];
if (!s->lossless)
- for (i = 0; i < s->frame_size; i++)
- s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;
+ for (i = 0; i < s->frame_size; i++)
+ s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;
switch(s->decorrelation)
{
- case MID_SIDE:
- for (i = 0; i < s->frame_size; i += s->channels)
- {
- s->int_samples[i] += s->int_samples[i+1];
- s->int_samples[i+1] -= shift(s->int_samples[i], 1);
- }
- break;
- case LEFT_SIDE:
- for (i = 0; i < s->frame_size; i += s->channels)
- s->int_samples[i+1] -= s->int_samples[i];
- break;
- case RIGHT_SIDE:
- for (i = 0; i < s->frame_size; i += s->channels)
- s->int_samples[i] -= s->int_samples[i+1];
- break;
+ case MID_SIDE:
+ for (i = 0; i < s->frame_size; i += s->channels)
+ {
+ s->int_samples[i] += s->int_samples[i+1];
+ s->int_samples[i+1] -= shift(s->int_samples[i], 1);
+ }
+ break;
+ case LEFT_SIDE:
+ for (i = 0; i < s->frame_size; i += s->channels)
+ s->int_samples[i+1] -= s->int_samples[i];
+ break;
+ case RIGHT_SIDE:
+ for (i = 0; i < s->frame_size; i += s->channels)
+ s->int_samples[i] -= s->int_samples[i+1];
+ break;
}
memset(s->window, 0, 4* s->window_size);
-
+
for (i = 0; i < s->tail_size; i++)
- s->window[x++] = s->tail[i];
+ s->window[x++] = s->tail[i];
for (i = 0; i < s->frame_size; i++)
- s->window[x++] = s->int_samples[i];
-
+ s->window[x++] = s->int_samples[i];
+
for (i = 0; i < s->tail_size; i++)
- s->window[x++] = 0;
+ s->window[x++] = 0;
for (i = 0; i < s->tail_size; i++)
- s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i];
+ s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i];
// generate taps
modified_levinson_durbin(s->window, s->window_size,
- s->predictor_k, s->num_taps, s->channels, s->tap_quant);
+ s->predictor_k, s->num_taps, s->channels, s->tap_quant);
if (intlist_write(&pb, s->predictor_k, s->num_taps, 0) < 0)
- return -1;
+ return -1;
for (ch = 0; ch < s->channels; ch++)
{
- x = s->tail_size+ch;
- for (i = 0; i < s->block_align; i++)
- {
- int sum = 0;
- for (j = 0; j < s->downsampling; j++, x += s->channels)
- sum += s->window[x];
- s->coded_samples[ch][i] = sum;
- }
- }
-
- // simple rate control code
+ x = s->tail_size+ch;
+ for (i = 0; i < s->block_align; i++)
+ {
+ int sum = 0;
+ for (j = 0; j < s->downsampling; j++, x += s->channels)
+ sum += s->window[x];
+ s->coded_samples[ch][i] = sum;
+ }
+ }
+
+ // simple rate control code
if (!s->lossless)
{
- double energy1 = 0.0, energy2 = 0.0;
- for (ch = 0; ch < s->channels; ch++)
- {
- for (i = 0; i < s->block_align; i++)
- {
- double sample = s->coded_samples[ch][i];
- energy2 += sample*sample;
- energy1 += fabs(sample);
- }
- }
-
- energy2 = sqrt(energy2/(s->channels*s->block_align));
- energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align);
-
- // increase bitrate when samples are like a gaussian distribution
- // reduce bitrate when samples are like a two-tailed exponential distribution
-
- if (energy2 > energy1)
- energy2 += (energy2-energy1)*RATE_VARIATION;
-
- quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
-// av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2);
-
- if (quant < 1)
- quant = 1;
- if (quant > 65535)
- quant = 65535;
-
- set_ue_golomb(&pb, quant);
-
- quant *= SAMPLE_FACTOR;
+ double energy1 = 0.0, energy2 = 0.0;
+ for (ch = 0; ch < s->channels; ch++)
+ {
+ for (i = 0; i < s->block_align; i++)
+ {
+ double sample = s->coded_samples[ch][i];
+ energy2 += sample*sample;
+ energy1 += fabs(sample);
+ }
+ }
+
+ energy2 = sqrt(energy2/(s->channels*s->block_align));
+ energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align);
+
+ // increase bitrate when samples are like a gaussian distribution
+ // reduce bitrate when samples are like a two-tailed exponential distribution
+
+ if (energy2 > energy1)
+ energy2 += (energy2-energy1)*RATE_VARIATION;
+
+ quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
+// av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2);
+
+ if (quant < 1)
+ quant = 1;
+ if (quant > 65535)
+ quant = 65535;
+
+ set_ue_golomb(&pb, quant);
+
+ quant *= SAMPLE_FACTOR;
}
// write out coded samples
for (ch = 0; ch < s->channels; ch++)
{
- if (!s->lossless)
- for (i = 0; i < s->block_align; i++)
- s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant);
+ if (!s->lossless)
+ for (i = 0; i < s->block_align; i++)
+ s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant);
- if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0)
- return -1;
+ if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0)
+ return -1;
}
// av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);
}
#endif //CONFIG_ENCODERS
+#ifdef CONFIG_DECODERS
static int sonic_decode_init(AVCodecContext *avctx)
{
SonicContext *s = avctx->priv_data;
GetBitContext gb;
int i, version;
-
+
s->channels = avctx->channels;
s->samplerate = avctx->sample_rate;
-
+
if (!avctx->extradata)
{
- av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
- return -1;
+ av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
+ return -1;
}
-
+
init_get_bits(&gb, avctx->extradata, avctx->extradata_size);
-
+
version = get_bits(&gb, 2);
if (version > 1)
{
- av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
- return -1;
+ av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
+ return -1;
}
if (version == 1)
{
- s->channels = get_bits(&gb, 2);
- s->samplerate = samplerate_table[get_bits(&gb, 4)];
- av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n",
- s->channels, s->samplerate);
+ s->channels = get_bits(&gb, 2);
+ s->samplerate = samplerate_table[get_bits(&gb, 4)];
+ av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n",
+ s->channels, s->samplerate);
}
if (s->channels > MAX_CHANNELS)
{
- av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
- return -1;
+ av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
+ return -1;
}
s->lossless = get_bits1(&gb);
if (!s->lossless)
- skip_bits(&gb, 3); // XXX FIXME
+ skip_bits(&gb, 3); // XXX FIXME
s->decorrelation = get_bits(&gb, 2);
s->downsampling = get_bits(&gb, 2);
s->num_taps = (get_bits(&gb, 5)+1)<<5;
if (get_bits1(&gb)) // XXX FIXME
- av_log(avctx, AV_LOG_INFO, "Custom quant table\n");
-
+ av_log(avctx, AV_LOG_INFO, "Custom quant table\n");
+
s->block_align = (int)(2048.0*(s->samplerate/44100))/s->downsampling;
s->frame_size = s->channels*s->block_align*s->downsampling;
// avctx->frame_size = s->block_align;
av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
- version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
+ version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
// generate taps
s->tap_quant = av_mallocz(4* s->num_taps);
for (i = 0; i < s->num_taps; i++)
- s->tap_quant[i] = (int)(sqrt(i+1));
-
+ s->tap_quant[i] = (int)(sqrt(i+1));
+
s->predictor_k = av_mallocz(4* s->num_taps);
-
+
for (i = 0; i < s->channels; i++)
{
- s->predictor_state[i] = av_mallocz(4* s->num_taps);
- if (!s->predictor_state[i])
- return -1;
+ s->predictor_state[i] = av_mallocz(4* s->num_taps);
+ if (!s->predictor_state[i])
+ return -1;
}
for (i = 0; i < s->channels; i++)
{
- s->coded_samples[i] = av_mallocz(4* s->block_align);
- if (!s->coded_samples[i])
- return -1;
+ s->coded_samples[i] = av_mallocz(4* s->block_align);
+ if (!s->coded_samples[i])
+ return -1;
}
s->int_samples = av_mallocz(4* s->frame_size);
{
SonicContext *s = avctx->priv_data;
int i;
-
+
av_free(s->int_samples);
av_free(s->tap_quant);
av_free(s->predictor_k);
-
+
for (i = 0; i < s->channels; i++)
{
- av_free(s->predictor_state[i]);
- av_free(s->coded_samples[i]);
+ av_free(s->predictor_state[i]);
+ av_free(s->coded_samples[i]);
}
-
+
return 0;
}
static int sonic_decode_frame(AVCodecContext *avctx,
- int16_t *data, int *data_size,
- uint8_t *buf, int buf_size)
+ void *data, int *data_size,
+ uint8_t *buf, int buf_size)
{
SonicContext *s = avctx->priv_data;
GetBitContext gb;
if (buf_size == 0) return 0;
// av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);
-
+
init_get_bits(&gb, buf, buf_size*8);
-
+
intlist_read(&gb, s->predictor_k, s->num_taps, 0);
// dequantize
for (i = 0; i < s->num_taps; i++)
- s->predictor_k[i] *= s->tap_quant[i];
+ s->predictor_k[i] *= s->tap_quant[i];
if (s->lossless)
- quant = 1;
+ quant = 1;
else
- quant = get_ue_golomb(&gb) * SAMPLE_FACTOR;
+ quant = get_ue_golomb(&gb) * SAMPLE_FACTOR;
// av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);
for (ch = 0; ch < s->channels; ch++)
{
- int x = ch;
+ int x = ch;
- predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps);
-
- intlist_read(&gb, s->coded_samples[ch], s->block_align, 1);
+ predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps);
- for (i = 0; i < s->block_align; i++)
- {
- for (j = 0; j < s->downsampling - 1; j++)
- {
- s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0);
- x += s->channels;
- }
-
- s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant);
- x += s->channels;
- }
+ intlist_read(&gb, s->coded_samples[ch], s->block_align, 1);
- for (i = 0; i < s->num_taps; i++)
- s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels];
+ for (i = 0; i < s->block_align; i++)
+ {
+ for (j = 0; j < s->downsampling - 1; j++)
+ {
+ s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0);
+ x += s->channels;
+ }
+
+ s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant);
+ x += s->channels;
+ }
+
+ for (i = 0; i < s->num_taps; i++)
+ s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels];
}
-
+
switch(s->decorrelation)
{
- case MID_SIDE:
- for (i = 0; i < s->frame_size; i += s->channels)
- {
- s->int_samples[i+1] += shift(s->int_samples[i], 1);
- s->int_samples[i] -= s->int_samples[i+1];
- }
- break;
- case LEFT_SIDE:
- for (i = 0; i < s->frame_size; i += s->channels)
- s->int_samples[i+1] += s->int_samples[i];
- break;
- case RIGHT_SIDE:
- for (i = 0; i < s->frame_size; i += s->channels)
- s->int_samples[i] += s->int_samples[i+1];
- break;
+ case MID_SIDE:
+ for (i = 0; i < s->frame_size; i += s->channels)
+ {
+ s->int_samples[i+1] += shift(s->int_samples[i], 1);
+ s->int_samples[i] -= s->int_samples[i+1];
+ }
+ break;
+ case LEFT_SIDE:
+ for (i = 0; i < s->frame_size; i += s->channels)
+ s->int_samples[i+1] += s->int_samples[i];
+ break;
+ case RIGHT_SIDE:
+ for (i = 0; i < s->frame_size; i += s->channels)
+ s->int_samples[i] += s->int_samples[i+1];
+ break;
}
if (!s->lossless)
- for (i = 0; i < s->frame_size; i++)
- s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT);
+ for (i = 0; i < s->frame_size; i++)
+ s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT);
// internal -> short
for (i = 0; i < s->frame_size; i++)
{
- if (s->int_samples[i] > 32767)
- samples[i] = 32767;
- else if (s->int_samples[i] < -32768)
- samples[i] = -32768;
- else
- samples[i] = s->int_samples[i];
+ if (s->int_samples[i] > 32767)
+ samples[i] = 32767;
+ else if (s->int_samples[i] < -32768)
+ samples[i] = -32768;
+ else
+ samples[i] = s->int_samples[i];
}
align_get_bits(&gb);
return (get_bits_count(&gb)+7)/8;
}
+#endif
#ifdef CONFIG_ENCODERS
AVCodec sonic_encoder = {
projects / ffmpeg / blobdiff