/**
* FLAC audio encoder
- * Copyright (c) 2006 Justin Ruggles <jruggle@earthlink.net>
+ * Copyright (c) 2006 Justin Ruggles <justin.ruggles@gmail.com>
*
* This file is part of FFmpeg.
*
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "libavutil/crc.h"
+#include "libavutil/md5.h"
#include "avcodec.h"
-#include "bitstream.h"
-#include "crc.h"
+#include "get_bits.h"
#include "dsputil.h"
#include "golomb.h"
-#include "lls.h"
-
-#define FLAC_MAX_CH 8
-#define FLAC_MIN_BLOCKSIZE 16
-#define FLAC_MAX_BLOCKSIZE 65535
+#include "lpc.h"
+#include "flac.h"
+#include "flacdata.h"
#define FLAC_SUBFRAME_CONSTANT 0
#define FLAC_SUBFRAME_VERBATIM 1
#define FLAC_SUBFRAME_FIXED 8
#define FLAC_SUBFRAME_LPC 32
-#define FLAC_CHMODE_NOT_STEREO 0
-#define FLAC_CHMODE_LEFT_RIGHT 1
-#define FLAC_CHMODE_LEFT_SIDE 8
-#define FLAC_CHMODE_RIGHT_SIDE 9
-#define FLAC_CHMODE_MID_SIDE 10
-
-#define ORDER_METHOD_EST 0
-#define ORDER_METHOD_2LEVEL 1
-#define ORDER_METHOD_4LEVEL 2
-#define ORDER_METHOD_8LEVEL 3
-#define ORDER_METHOD_SEARCH 4
-#define ORDER_METHOD_LOG 5
-
-#define FLAC_STREAMINFO_SIZE 34
-
-#define MIN_LPC_ORDER 1
-#define MAX_LPC_ORDER 32
#define MAX_FIXED_ORDER 4
#define MAX_PARTITION_ORDER 8
#define MAX_PARTITIONS (1 << MAX_PARTITION_ORDER)
} FlacSubframe;
typedef struct FlacFrame {
- FlacSubframe subframes[FLAC_MAX_CH];
+ FlacSubframe subframes[FLAC_MAX_CHANNELS];
int blocksize;
int bs_code[2];
uint8_t crc8;
typedef struct FlacEncodeContext {
PutBitContext pb;
int channels;
- int ch_code;
int samplerate;
int sr_code[2];
- int blocksize;
+ int max_blocksize;
+ int min_framesize;
int max_framesize;
+ int max_encoded_framesize;
uint32_t frame_count;
+ uint64_t sample_count;
+ uint8_t md5sum[16];
FlacFrame frame;
CompressionOptions options;
AVCodecContext *avctx;
DSPContext dsp;
+ struct AVMD5 *md5ctx;
} FlacEncodeContext;
-static const int flac_samplerates[16] = {
- 0, 0, 0, 0,
- 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
- 0, 0, 0, 0
-};
-
-static const int flac_blocksizes[16] = {
- 0,
- 192,
- 576, 1152, 2304, 4608,
- 0, 0,
- 256, 512, 1024, 2048, 4096, 8192, 16384, 32768
-};
-
/**
* Writes streaminfo metadata block to byte array
*/
init_put_bits(&pb, header, FLAC_STREAMINFO_SIZE);
/* streaminfo metadata block */
- put_bits(&pb, 16, s->blocksize);
- put_bits(&pb, 16, s->blocksize);
- put_bits(&pb, 24, 0);
+ put_bits(&pb, 16, s->max_blocksize);
+ put_bits(&pb, 16, s->max_blocksize);
+ put_bits(&pb, 24, s->min_framesize);
put_bits(&pb, 24, s->max_framesize);
put_bits(&pb, 20, s->samplerate);
put_bits(&pb, 3, s->channels-1);
put_bits(&pb, 5, 15); /* bits per sample - 1 */
+ /* write 36-bit sample count in 2 put_bits() calls */
+ put_bits(&pb, 24, (s->sample_count & 0xFFFFFF000LL) >> 12);
+ put_bits(&pb, 12, s->sample_count & 0x000000FFFLL);
flush_put_bits(&pb);
- /* total samples = 0 */
- /* MD5 signature = 0 */
+ memcpy(&header[18], s->md5sum, 16);
}
/**
int blocksize;
assert(samplerate > 0);
- blocksize = flac_blocksizes[1];
+ blocksize = ff_flac_blocksize_table[1];
target = (samplerate * block_time_ms) / 1000;
for(i=0; i<16; i++) {
- if(target >= flac_blocksizes[i] && flac_blocksizes[i] > blocksize) {
- blocksize = flac_blocksizes[i];
+ if(target >= ff_flac_blocksize_table[i] && ff_flac_blocksize_table[i] > blocksize) {
+ blocksize = ff_flac_blocksize_table[i];
}
}
return blocksize;
}
-static int flac_encode_init(AVCodecContext *avctx)
+static av_cold int flac_encode_init(AVCodecContext *avctx)
{
int freq = avctx->sample_rate;
int channels = avctx->channels;
return -1;
}
- if(channels < 1 || channels > FLAC_MAX_CH) {
+ if(channels < 1 || channels > FLAC_MAX_CHANNELS) {
return -1;
}
s->channels = channels;
- s->ch_code = s->channels-1;
/* find samplerate in table */
if(freq < 1)
return -1;
for(i=4; i<12; i++) {
- if(freq == flac_samplerates[i]) {
- s->samplerate = flac_samplerates[i];
+ if(freq == ff_flac_sample_rate_table[i]) {
+ s->samplerate = ff_flac_sample_rate_table[i];
s->sr_code[0] = i;
s->sr_code[1] = 0;
break;
avctx->frame_size);
return -1;
}
- s->blocksize = avctx->frame_size;
} else {
- s->blocksize = select_blocksize(s->samplerate, s->options.block_time_ms);
- avctx->frame_size = s->blocksize;
+ s->avctx->frame_size = select_blocksize(s->samplerate, s->options.block_time_ms);
}
- av_log(avctx, AV_LOG_DEBUG, " block size: %d\n", s->blocksize);
+ s->max_blocksize = s->avctx->frame_size;
+ av_log(avctx, AV_LOG_DEBUG, " block size: %d\n", s->avctx->frame_size);
/* set LPC precision */
if(avctx->lpc_coeff_precision > 0) {
}
s->options.lpc_coeff_precision = avctx->lpc_coeff_precision;
} else {
- /* select LPC precision based on block size */
- if( s->blocksize <= 192) s->options.lpc_coeff_precision = 7;
- else if(s->blocksize <= 384) s->options.lpc_coeff_precision = 8;
- else if(s->blocksize <= 576) s->options.lpc_coeff_precision = 9;
- else if(s->blocksize <= 1152) s->options.lpc_coeff_precision = 10;
- else if(s->blocksize <= 2304) s->options.lpc_coeff_precision = 11;
- else if(s->blocksize <= 4608) s->options.lpc_coeff_precision = 12;
- else if(s->blocksize <= 8192) s->options.lpc_coeff_precision = 13;
- else if(s->blocksize <= 16384) s->options.lpc_coeff_precision = 14;
- else s->options.lpc_coeff_precision = 15;
+ /* default LPC precision */
+ s->options.lpc_coeff_precision = 15;
}
av_log(avctx, AV_LOG_DEBUG, " lpc precision: %d\n",
s->options.lpc_coeff_precision);
/* set maximum encoded frame size in verbatim mode */
- if(s->channels == 2) {
- s->max_framesize = 14 + ((s->blocksize * 33 + 7) >> 3);
- } else {
- s->max_framesize = 14 + (s->blocksize * s->channels * 2);
- }
+ s->max_framesize = ff_flac_get_max_frame_size(s->avctx->frame_size,
+ s->channels, 16);
+
+ /* initialize MD5 context */
+ s->md5ctx = av_malloc(av_md5_size);
+ if(!s->md5ctx)
+ return AVERROR_NOMEM;
+ av_md5_init(s->md5ctx);
streaminfo = av_malloc(FLAC_STREAMINFO_SIZE);
write_streaminfo(s, streaminfo);
avctx->extradata_size = FLAC_STREAMINFO_SIZE;
s->frame_count = 0;
+ s->min_framesize = s->max_framesize;
avctx->coded_frame = avcodec_alloc_frame();
avctx->coded_frame->key_frame = 1;
frame = &s->frame;
for(i=0; i<16; i++) {
- if(s->blocksize == flac_blocksizes[i]) {
- frame->blocksize = flac_blocksizes[i];
+ if(s->avctx->frame_size == ff_flac_blocksize_table[i]) {
+ frame->blocksize = ff_flac_blocksize_table[i];
frame->bs_code[0] = i;
frame->bs_code[1] = 0;
break;
}
}
if(i == 16) {
- frame->blocksize = s->blocksize;
+ frame->blocksize = s->avctx->frame_size;
if(frame->blocksize <= 256) {
frame->bs_code[0] = 6;
frame->bs_code[1] = frame->blocksize-1;
uint32_t all_bits;
part = (1 << porder);
- all_bits = 0;
+ all_bits = 4 * part;
cnt = (n >> porder) - pred_order;
for(i=0; i<part; i++) {
- if(i == 1) cnt = (n >> porder);
k = find_optimal_param(sums[i], cnt);
rc->params[i] = k;
all_bits += rice_encode_count(sums[i], cnt, k);
+ cnt = n >> porder;
}
- all_bits += (4 * part);
rc->porder = porder;
double w;
double c;
+ assert(!(len&1)); //the optimization in r11881 does not support odd len
+ //if someone wants odd len extend the change in r11881
+
n2 = (len >> 1);
c = 2.0 / (len - 1.0);
+
+ w_data+=n2;
+ data+=n2;
for(i=0; i<n2; i++) {
- w = c - i - 1.0;
+ w = c - n2 + i;
w = 1.0 - (w * w);
- w_data[i] = data[i] * w;
- w_data[len-1-i] = data[len-1-i] * w;
+ w_data[-i-1] = data[-i-1] * w;
+ w_data[+i ] = data[+i ] * w;
}
}
for(j=0; j<lag; j+=2){
double sum0 = 1.0, sum1 = 1.0;
- for(i=0; i<len; i++){
+ for(i=j; i<len; i++){
sum0 += data1[i] * data1[i-j];
sum1 += data1[i] * data1[i-j-1];
}
if(j==lag){
double sum = 1.0;
- for(i=0; i<len; i+=2){
+ for(i=j-1; i<len; i+=2){
sum += data1[i ] * data1[i-j ]
+ data1[i+1] * data1[i-j+1];
}
}
}
-/**
- * Levinson-Durbin recursion.
- * Produces LPC coefficients from autocorrelation data.
- */
-static void compute_lpc_coefs(const double *autoc, int max_order,
- double lpc[][MAX_LPC_ORDER], double *ref)
-{
- int i, j, i2;
- double r, err, tmp;
- double lpc_tmp[MAX_LPC_ORDER];
-
- for(i=0; i<max_order; i++) lpc_tmp[i] = 0;
- err = autoc[0];
-
- for(i=0; i<max_order; i++) {
- r = -autoc[i+1];
- for(j=0; j<i; j++) {
- r -= lpc_tmp[j] * autoc[i-j];
- }
- r /= err;
- ref[i] = fabs(r);
-
- err *= 1.0 - (r * r);
-
- i2 = (i >> 1);
- lpc_tmp[i] = r;
- for(j=0; j<i2; j++) {
- tmp = lpc_tmp[j];
- lpc_tmp[j] += r * lpc_tmp[i-1-j];
- lpc_tmp[i-1-j] += r * tmp;
- }
- if(i & 1) {
- lpc_tmp[j] += lpc_tmp[j] * r;
- }
-
- for(j=0; j<=i; j++) {
- lpc[i][j] = -lpc_tmp[j];
- }
- }
-}
-
-/**
- * Quantize LPC coefficients
- */
-static void quantize_lpc_coefs(double *lpc_in, int order, int precision,
- int32_t *lpc_out, int *shift)
-{
- int i;
- double cmax, error;
- int32_t qmax;
- int sh;
-
- /* define maximum levels */
- qmax = (1 << (precision - 1)) - 1;
-
- /* find maximum coefficient value */
- cmax = 0.0;
- for(i=0; i<order; i++) {
- cmax= FFMAX(cmax, fabs(lpc_in[i]));
- }
-
- /* if maximum value quantizes to zero, return all zeros */
- if(cmax * (1 << MAX_LPC_SHIFT) < 1.0) {
- *shift = 0;
- memset(lpc_out, 0, sizeof(int32_t) * order);
- return;
- }
-
- /* calculate level shift which scales max coeff to available bits */
- sh = MAX_LPC_SHIFT;
- while((cmax * (1 << sh) > qmax) && (sh > 0)) {
- sh--;
- }
-
- /* since negative shift values are unsupported in decoder, scale down
- coefficients instead */
- if(sh == 0 && cmax > qmax) {
- double scale = ((double)qmax) / cmax;
- for(i=0; i<order; i++) {
- lpc_in[i] *= scale;
- }
- }
-
- /* output quantized coefficients and level shift */
- error=0;
- for(i=0; i<order; i++) {
- error += lpc_in[i] * (1 << sh);
- lpc_out[i] = av_clip(lrintf(error), -qmax, qmax);
- error -= lpc_out[i];
- }
- *shift = sh;
-}
-
-static int estimate_best_order(double *ref, int max_order)
-{
- int i, est;
-
- est = 1;
- for(i=max_order-1; i>=0; i--) {
- if(ref[i] > 0.10) {
- est = i+1;
- break;
- }
- }
- return est;
-}
-
-/**
- * Calculate LPC coefficients for multiple orders
- */
-static int lpc_calc_coefs(FlacEncodeContext *s,
- const int32_t *samples, int blocksize, int max_order,
- int precision, int32_t coefs[][MAX_LPC_ORDER],
- int *shift, int use_lpc, int omethod)
-{
- double autoc[MAX_LPC_ORDER+1];
- double ref[MAX_LPC_ORDER];
- double lpc[MAX_LPC_ORDER][MAX_LPC_ORDER];
- int i, j, pass;
- int opt_order;
-
- assert(max_order >= MIN_LPC_ORDER && max_order <= MAX_LPC_ORDER);
-
- if(use_lpc == 1){
- s->dsp.flac_compute_autocorr(samples, blocksize, max_order, autoc);
-
- compute_lpc_coefs(autoc, max_order, lpc, ref);
- }else{
- LLSModel m[2];
- double var[MAX_LPC_ORDER+1], weight;
-
- for(pass=0; pass<use_lpc-1; pass++){
- av_init_lls(&m[pass&1], max_order);
-
- weight=0;
- for(i=max_order; i<blocksize; i++){
- for(j=0; j<=max_order; j++)
- var[j]= samples[i-j];
-
- if(pass){
- double eval, inv, rinv;
- eval= av_evaluate_lls(&m[(pass-1)&1], var+1, max_order-1);
- eval= (512>>pass) + fabs(eval - var[0]);
- inv = 1/eval;
- rinv = sqrt(inv);
- for(j=0; j<=max_order; j++)
- var[j] *= rinv;
- weight += inv;
- }else
- weight++;
-
- av_update_lls(&m[pass&1], var, 1.0);
- }
- av_solve_lls(&m[pass&1], 0.001, 0);
- }
-
- for(i=0; i<max_order; i++){
- for(j=0; j<max_order; j++)
- lpc[i][j]= m[(pass-1)&1].coeff[i][j];
- ref[i]= sqrt(m[(pass-1)&1].variance[i] / weight) * (blocksize - max_order) / 4000;
- }
- for(i=max_order-1; i>0; i--)
- ref[i] = ref[i-1] - ref[i];
- }
- opt_order = max_order;
-
- if(omethod == ORDER_METHOD_EST) {
- opt_order = estimate_best_order(ref, max_order);
- i = opt_order-1;
- quantize_lpc_coefs(lpc[i], i+1, precision, coefs[i], &shift[i]);
- } else {
- for(i=0; i<max_order; i++) {
- quantize_lpc_coefs(lpc[i], i+1, precision, coefs[i], &shift[i]);
- }
- }
-
- return opt_order;
-}
-
static void encode_residual_verbatim(int32_t *res, int32_t *smp, int n)
{
for(i=0; i<order; i++) {
res[i] = smp[i];
}
-#ifdef CONFIG_SMALL
+#if CONFIG_SMALL
for(i=order; i<n; i+=2) {
int j;
int s = smp[i];
}
/* LPC */
- opt_order = lpc_calc_coefs(ctx, smp, n, max_order, precision, coefs, shift, ctx->options.use_lpc, omethod);
+ opt_order = ff_lpc_calc_coefs(&ctx->dsp, smp, n, min_order, max_order,
+ precision, coefs, shift, ctx->options.use_lpc,
+ omethod, MAX_LPC_SHIFT, 0);
if(omethod == ORDER_METHOD_2LEVEL ||
omethod == ORDER_METHOD_4LEVEL ||
}
}
if(best == 0) {
- return FLAC_CHMODE_LEFT_RIGHT;
+ return FLAC_CHMODE_INDEPENDENT;
} else if(best == 1) {
return FLAC_CHMODE_LEFT_SIDE;
} else if(best == 2) {
right = frame->subframes[1].samples;
if(ctx->channels != 2) {
- frame->ch_mode = FLAC_CHMODE_NOT_STEREO;
+ frame->ch_mode = FLAC_CHMODE_INDEPENDENT;
return;
}
frame->ch_mode = estimate_stereo_mode(left, right, n);
/* perform decorrelation and adjust bits-per-sample */
- if(frame->ch_mode == FLAC_CHMODE_LEFT_RIGHT) {
+ if(frame->ch_mode == FLAC_CHMODE_INDEPENDENT) {
return;
}
if(frame->ch_mode == FLAC_CHMODE_MID_SIDE) {
}
}
-static void put_sbits(PutBitContext *pb, int bits, int32_t val)
-{
- assert(bits >= 0 && bits <= 31);
-
- put_bits(pb, bits, val & ((1<<bits)-1));
-}
-
static void write_utf8(PutBitContext *pb, uint32_t val)
{
uint8_t tmp;
put_bits(&s->pb, 16, 0xFFF8);
put_bits(&s->pb, 4, frame->bs_code[0]);
put_bits(&s->pb, 4, s->sr_code[0]);
- if(frame->ch_mode == FLAC_CHMODE_NOT_STEREO) {
- put_bits(&s->pb, 4, s->ch_code);
+ if(frame->ch_mode == FLAC_CHMODE_INDEPENDENT) {
+ put_bits(&s->pb, 4, s->channels-1);
} else {
put_bits(&s->pb, 4, frame->ch_mode);
}
put_bits(&s->pb, 16, s->sr_code[1]);
}
flush_put_bits(&s->pb);
- crc = av_crc(av_crc07, 0, s->pb.buf, put_bits_count(&s->pb)>>3);
+ crc = av_crc(av_crc_get_table(AV_CRC_8_ATM), 0,
+ s->pb.buf, put_bits_count(&s->pb)>>3);
put_bits(&s->pb, 8, crc);
}
{
int crc;
flush_put_bits(&s->pb);
- crc = bswap_16(av_crc(av_crc8005, 0, s->pb.buf, put_bits_count(&s->pb)>>3));
+ crc = bswap_16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
+ s->pb.buf, put_bits_count(&s->pb)>>3));
put_bits(&s->pb, 16, crc);
flush_put_bits(&s->pb);
}
+static void update_md5_sum(FlacEncodeContext *s, int16_t *samples)
+{
+#ifdef WORDS_BIGENDIAN
+ int i;
+ for(i = 0; i < s->frame.blocksize*s->channels; i++) {
+ int16_t smp = le2me_16(samples[i]);
+ av_md5_update(s->md5ctx, (uint8_t *)&smp, 2);
+ }
+#else
+ av_md5_update(s->md5ctx, (uint8_t *)samples, s->frame.blocksize*s->channels*2);
+#endif
+}
+
static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
{
FlacEncodeContext *s;
int16_t *samples = data;
int out_bytes;
+ int reencoded=0;
s = avctx->priv_data;
- s->blocksize = avctx->frame_size;
+ if(buf_size < s->max_framesize*2) {
+ av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+ return 0;
+ }
+
+ /* when the last block is reached, update the header in extradata */
+ if (!data) {
+ s->max_framesize = s->max_encoded_framesize;
+ av_md5_final(s->md5ctx, s->md5sum);
+ write_streaminfo(s, avctx->extradata);
+ return 0;
+ }
+
init_frame(s);
copy_samples(s, samples);
for(ch=0; ch<s->channels; ch++) {
encode_residual(s, ch);
}
+
+write_frame:
init_put_bits(&s->pb, frame, buf_size);
output_frame_header(s);
output_subframes(s);
output_frame_footer(s);
out_bytes = put_bits_count(&s->pb) >> 3;
- if(out_bytes > s->max_framesize || out_bytes >= buf_size) {
- /* frame too large. use verbatim mode */
- for(ch=0; ch<s->channels; ch++) {
- encode_residual_v(s, ch);
- }
- init_put_bits(&s->pb, frame, buf_size);
- output_frame_header(s);
- output_subframes(s);
- output_frame_footer(s);
- out_bytes = put_bits_count(&s->pb) >> 3;
-
- if(out_bytes > s->max_framesize || out_bytes >= buf_size) {
+ if(out_bytes > s->max_framesize) {
+ if(reencoded) {
/* still too large. must be an error. */
av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
return -1;
}
+
+ /* frame too large. use verbatim mode */
+ for(ch=0; ch<s->channels; ch++) {
+ encode_residual_v(s, ch);
+ }
+ reencoded = 1;
+ goto write_frame;
}
s->frame_count++;
+ s->sample_count += avctx->frame_size;
+ update_md5_sum(s, samples);
+ if (out_bytes > s->max_encoded_framesize)
+ s->max_encoded_framesize = out_bytes;
+ if (out_bytes < s->min_framesize)
+ s->min_framesize = out_bytes;
+
return out_bytes;
}
-static int flac_encode_close(AVCodecContext *avctx)
+static av_cold int flac_encode_close(AVCodecContext *avctx)
{
+ if (avctx->priv_data) {
+ FlacEncodeContext *s = avctx->priv_data;
+ av_freep(&s->md5ctx);
+ }
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
av_freep(&avctx->coded_frame);
flac_encode_frame,
flac_encode_close,
NULL,
- .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
+ .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+ .long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};
projects / frescor / ffmpeg.git / blobdiff