* available.
*/
-#include <math.h>
-#include <stddef.h>
-#include <stdio.h>
-
#include "libavutil/lfg.h"
-#include "libavutil/random_seed.h"
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
void (* interpolate) (struct cook *q, float* buffer,
int gain_index, int gain_index_next);
- void (* saturate_output) (struct cook *q, int chan, int16_t *out);
+ void (* saturate_output) (struct cook *q, int chan, float *out);
AVCodecContext* avctx;
GetBitContext gb;
}
static av_cold int init_cook_mlt(COOKContext *q) {
- int j;
+ int j, ret;
int mlt_size = q->samples_per_channel;
- if ((q->mlt_window = av_malloc(sizeof(float)*mlt_size)) == 0)
- return -1;
+ if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
+ return AVERROR(ENOMEM);
/* Initialize the MLT window: simple sine window. */
ff_sine_window_init(q->mlt_window, mlt_size);
q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
/* Initialize the MDCT. */
- if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0)) {
- av_free(q->mlt_window);
- return -1;
+ if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0/32768.0))) {
+ av_free(q->mlt_window);
+ return ret;
}
av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
av_log2(mlt_size)+1);
//av_log(q->avctx, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
}
- memset(&exp_index1,0,102*sizeof(int));
- memset(&exp_index2,0,102*sizeof(int));
- memset(&tmp_categorize_array,0,128*2*sizeof(int));
+ memset(&exp_index1, 0, sizeof(exp_index1));
+ memset(&exp_index2, 0, sizeof(exp_index2));
+ memset(&tmp_categorize_array, 0, sizeof(tmp_categorize_array));
bias=-32;
int quant_index_table[102];
int category[128];
- memset(&category, 0, 128*sizeof(int));
- memset(&category_index, 0, 128*sizeof(int));
+ memset(&category, 0, sizeof(category));
+ memset(&category_index, 0, sizeof(category_index));
decode_envelope(q, p, quant_index_table);
q->num_vectors = get_bits(&q->gb,p->log2_numvector_size);
for(i=0 ; i<q->gain_size_factor ; i++){
buffer[i]*=fc1;
}
- return;
} else { //smooth gain
fc2 = q->gain_table[11 + (gain_index_next-gain_index)];
for(i=0 ; i<q->gain_size_factor ; i++){
buffer[i]*=fc1;
fc1*=fc2;
}
- return;
}
}
}
/* Save away the current to be previous block. */
- memcpy(previous_buffer, buffer0, sizeof(float)*q->samples_per_channel);
+ memcpy(previous_buffer, buffer0,
+ q->samples_per_channel * sizeof(*previous_buffer));
}
* @param decouple_tab decoupling array
*
*/
+static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
+{
+ int i;
+ int vlc = get_bits1(&q->gb);
+ int start = cplband[p->js_subband_start];
+ int end = cplband[p->subbands-1];
+ int length = end - start + 1;
-static void decouple_info(COOKContext *q, COOKSubpacket *p, int* decouple_tab){
- int length, i;
-
- if(get_bits1(&q->gb)) {
- if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;
-
- length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
- for (i=0 ; i<length ; i++) {
- decouple_tab[cplband[p->js_subband_start] + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
- }
+ if (start > end)
return;
- }
-
- if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;
- length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
- for (i=0 ; i<length ; i++) {
- decouple_tab[cplband[p->js_subband_start] + i] = get_bits(&q->gb, p->js_vlc_bits);
+ if (vlc) {
+ for (i = 0; i < length; i++)
+ decouple_tab[start + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
+ } else {
+ for (i = 0; i < length; i++)
+ decouple_tab[start + i] = get_bits(&q->gb, p->js_vlc_bits);
}
- return;
}
/*
const float* cplscale;
memset(decouple_tab, 0, sizeof(decouple_tab));
- memset(decode_buffer, 0, sizeof(decode_buffer));
+ memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
/* Make sure the buffers are zeroed out. */
- memset(mlt_buffer1,0, 1024*sizeof(float));
- memset(mlt_buffer2,0, 1024*sizeof(float));
+ memset(mlt_buffer1, 0, 1024 * sizeof(*mlt_buffer1));
+ memset(mlt_buffer2, 0, 1024 * sizeof(*mlt_buffer2));
decouple_info(q, p, decouple_tab);
mono_decode(q, p, decode_buffer);
}
/**
- * Saturate the output signal to signed 16bit integers.
+ * Saturate the output signal and interleave.
*
* @param q pointer to the COOKContext
* @param chan channel to saturate
* @param out pointer to the output vector
*/
-static void
-saturate_output_float (COOKContext *q, int chan, int16_t *out)
+static void saturate_output_float(COOKContext *q, int chan, float *out)
{
int j;
float *output = q->mono_mdct_output + q->samples_per_channel;
- /* Clip and convert floats to 16 bits.
- */
for (j = 0; j < q->samples_per_channel; j++) {
- out[chan + q->nb_channels * j] =
- av_clip_int16(lrintf(output[j]));
+ out[chan + q->nb_channels * j] = av_clipf(output[j], -1.0, 1.0);
}
}
static inline void
mlt_compensate_output(COOKContext *q, float *decode_buffer,
cook_gains *gains_ptr, float *previous_buffer,
- int16_t *out, int chan)
+ float *out, int chan)
{
imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
q->saturate_output (q, chan, out);
* @param inbuffer pointer to the inbuffer
* @param outbuffer pointer to the outbuffer
*/
-static void decode_subpacket(COOKContext *q, COOKSubpacket* p, const uint8_t *inbuffer, int16_t *outbuffer) {
+static void decode_subpacket(COOKContext *q, COOKSubpacket *p,
+ const uint8_t *inbuffer, float *outbuffer)
+{
int sub_packet_size = p->size;
/* packet dump */
// for (i=0 ; i<sub_packet_size ; i++) {
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
COOKContext *q = avctx->priv_data;
- int i;
+ int i, out_size;
int offset = 0;
int chidx = 0;
if (buf_size < avctx->block_align)
return buf_size;
+ out_size = q->nb_channels * q->samples_per_channel *
+ av_get_bytes_per_sample(avctx->sample_fmt);
+ if (*data_size < out_size) {
+ av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+ return AVERROR(EINVAL);
+ }
+
/* estimate subpacket sizes */
q->subpacket[0].size = avctx->block_align;
q->subpacket[0].size -= q->subpacket[i].size + 1;
if (q->subpacket[0].size < 0) {
av_log(avctx,AV_LOG_DEBUG,"frame subpacket size total > avctx->block_align!\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
}
/* decode supbackets */
- *data_size = 0;
for(i=0;i<q->num_subpackets;i++){
q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
q->subpacket[i].ch_idx = chidx;
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
- decode_subpacket(q, &q->subpacket[i], buf + offset, (int16_t*)data);
+ decode_subpacket(q, &q->subpacket[i], buf + offset, data);
offset += q->subpacket[i].size;
chidx += q->subpacket[i].num_channels;
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
}
- *data_size = sizeof(int16_t) * q->nb_channels * q->samples_per_channel;
+ *data_size = out_size;
/* Discard the first two frames: no valid audio. */
if (avctx->frame_number < 2) *data_size = 0;
int extradata_size = avctx->extradata_size;
int s = 0;
unsigned int channel_mask = 0;
+ int ret;
q->avctx = avctx;
/* Take care of the codec specific extradata. */
if (extradata_size <= 0) {
av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
case MONO:
if (q->nb_channels != 1) {
av_log_ask_for_sample(avctx, "Container channels != 1.\n");
- return -1;
+ return AVERROR_PATCHWELCOME;
}
av_log(avctx,AV_LOG_DEBUG,"MONO\n");
break;
case JOINT_STEREO:
if (q->nb_channels != 2) {
av_log_ask_for_sample(avctx, "Container channels != 2.\n");
- return -1;
+ return AVERROR_PATCHWELCOME;
}
av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");
if (avctx->extradata_size >= 16){
break;
default:
av_log_ask_for_sample(avctx, "Unknown Cook version.\n");
- return -1;
+ return AVERROR_PATCHWELCOME;
}
if(s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
av_log(avctx,AV_LOG_ERROR,"different number of samples per channel!\n");
- return -1;
+ return AVERROR_INVALIDDATA;
} else
q->samples_per_channel = q->subpacket[0].samples_per_channel;
/* Try to catch some obviously faulty streams, othervise it might be exploitable */
if (q->subpacket[s].total_subbands > 53) {
av_log_ask_for_sample(avctx, "total_subbands > 53\n");
- return -1;
+ return AVERROR_PATCHWELCOME;
}
- if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 0)) {
- av_log(avctx,AV_LOG_ERROR,"js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->subpacket[s].js_vlc_bits);
- return -1;
+ if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 2*q->subpacket[s].joint_stereo)) {
+ av_log(avctx,AV_LOG_ERROR,"js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
+ q->subpacket[s].js_vlc_bits, 2*q->subpacket[s].joint_stereo);
+ return AVERROR_INVALIDDATA;
}
if (q->subpacket[s].subbands > 50) {
av_log_ask_for_sample(avctx, "subbands > 50\n");
- return -1;
+ return AVERROR_PATCHWELCOME;
}
q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
s++;
if (s > MAX_SUBPACKETS) {
av_log_ask_for_sample(avctx, "Too many subpackets > 5\n");
- return -1;
+ return AVERROR_PATCHWELCOME;
}
}
/* Generate tables */
init_gain_table(q);
init_cplscales_table(q);
- if (init_cook_vlc_tables(q) != 0)
- return -1;
+ if ((ret = init_cook_vlc_tables(q)))
+ return ret;
if(avctx->block_align >= UINT_MAX/2)
- return -1;
+ return AVERROR(EINVAL);
/* Pad the databuffer with:
DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
+ DECODE_BYTES_PAD1(avctx->block_align)
+ FF_INPUT_BUFFER_PADDING_SIZE);
if (q->decoded_bytes_buffer == NULL)
- return -1;
+ return AVERROR(ENOMEM);
/* Initialize transform. */
- if ( init_cook_mlt(q) != 0 )
- return -1;
+ if ((ret = init_cook_mlt(q)))
+ return ret;
/* Initialize COOK signal arithmetic handling */
if (1) {
av_log_ask_for_sample(avctx,
"unknown amount of samples_per_channel = %d\n",
q->samples_per_channel);
- return -1;
+ return AVERROR_PATCHWELCOME;
}
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
if (channel_mask)
avctx->channel_layout = channel_mask;
else
projects / ffmpeg / blobdiff