Skip to content

F
ffmpeg.wasm-core
Commit d17e7070 authored by Michael Niedermayer's avatar Michael Niedermayer
Browse files
Options

Merge remote-tracking branch 'qatar/master' 

* qatar/master: (51 commits)
  cin audio: use sign_extend() instead of casting to int16_t
  cin audio: restructure decoding loop to avoid a separate counter variable
  cin audio: use local variable for delta value
  cin audio: remove unneeded cast from void*
  cin audio: validate the channel count
  cin audio: remove unneeded AVCodecContext pointer from CinAudioContext
  dsicin: fix several audio-related fields in the CIN demuxer
  flacdec: use av_get_bytes_per_sample() to get sample size
  dca: handle errors from dca_decode_block()
  dca: return error if the frame header is invalid
  dca: return proper error codes instead of -1
  utvideo: handle empty Huffman trees
  binkaudio: change short to int16_t
  binkaudio: only decode one block at a time.
  binkaudio: store interleaved overlap samples in BinkAudioContext.
  binkaudio: pre-calculate quantization factors
  binkaudio: add some buffer overread checks.
  atrac3: support float or int16 output using request_sample_fmt
  atrac3: add CODEC_CAP_SUBFRAMES capability
  atrac3: return appropriate error codes instead of -1
  ...

Conflicts:
	libavcodec/atrac1.c
	libavcodec/dca.c
	libavformat/mov.c
Merged-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents 1af3571e 7d1b17b8
Expand all Hide whitespace changes
Inline Side-by-side
Showing with 493 additions and 277 deletions
libavcodec/atrac1.c
View file @ d17e7070
......@@ -36,6 +36,7 @@
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
#include "sinewin.h"
#include "atrac.h"
......@@ -78,10 +79,11 @@ typedef struct {
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
DECLARE_ALIGNED(32, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
float *out_samples[AT1_MAX_CHANNELS];
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
FmtConvertContext fmt_conv;
} AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */
......@@ -129,7 +131,7 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
nbits = mdct_long_nbits[band_num] - log2_block_count;
if (nbits != 5 && nbits != 7 && nbits != 8)
return -1;
return AVERROR_INVALIDDATA;
} else {
block_size = 32;
nbits = 5;
......@@ -173,14 +175,14 @@ static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
/* low and mid band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp & 1)
return -1;
return AVERROR_INVALIDDATA;
log2_block_cnt[i] = 2 - log2_block_count_tmp;
}
/* high band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
return -1;
return AVERROR_INVALIDDATA;
log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
skip_bits(gb, 2);
......@@ -229,7 +231,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
/* check for bitstream overflow */
if (bits_used > AT1_SU_MAX_BITS)
return -1;
return AVERROR_INVALIDDATA;
/* get the position of the 1st spec according to the block size mode */
pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
......@@ -276,14 +278,21 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
int ch, ret, i;
int ch, ret, out_size;
GetBitContext gb;
float* samples = data;
if (buf_size < 212 * q->channels) {
av_log(avctx, AV_LOG_ERROR,"Not enought data to decode!\n");
return -1;
av_log(avctx,AV_LOG_ERROR,"Not enough data to decode!\n");
return AVERROR_INVALIDDATA;
}
out_size = q->channels * AT1_SU_SAMPLES *
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);
}
for (ch = 0; ch < q->channels; ch++) {
......@@ -303,44 +312,72 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
ret = at1_imdct_block(su, q);
if (ret < 0)
return ret;
at1_subband_synthesis(q, su, q->out_samples[ch]);
at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]);
}
/* interleave; FIXME, should create/use a DSP function */
if (q->channels == 1) {
/* mono */
memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
} else {
/* stereo */
for (i = 0; i < AT1_SU_SAMPLES; i++) {
samples[i * 2] = q->out_samples[0][i];
samples[i * 2 + 1] = q->out_samples[1][i];
}
/* interleave */
if (q->channels == 2) {
q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
AT1_SU_SAMPLES, 2);
}
*data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
*data_size = out_size;
return avctx->block_align;
}
static av_cold int atrac1_decode_end(AVCodecContext * avctx)
{
AT1Ctx *q = avctx->priv_data;
av_freep(&q->out_samples[0]);
ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]);
return 0;
}
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data;
int ret;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
avctx->channels);
return AVERROR(EINVAL);
}
q->channels = avctx->channels;
if (avctx->channels == 2) {
q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0]));
q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES;
if (!q->out_samples[0]) {
av_freep(&q->out_samples[0]);
return AVERROR(ENOMEM);
}
}
/* Init the mdct transforms */
ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
(ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
(ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
atrac1_decode_end(avctx);
return ret;
}
ff_init_ff_sine_windows(5);
atrac_generate_tables();
dsputil_init(&q->dsp, avctx);
ff_fmt_convert_init(&q->fmt_conv, avctx);
q->bands[0] = q->low;
q->bands[1] = q->mid;
......@@ -356,16 +393,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
}
static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
AT1Ctx *q = avctx->priv_data;
ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]);
return 0;
}
AVCodec ff_atrac1_decoder = {
.name = "atrac1",
.type = AVMEDIA_TYPE_AUDIO,
......
libavcodec/atrac3.c
View file @ d17e7070
This diff is collapsed.
libavcodec/binkaudio.c
View file @ d17e7070
......@@ -39,6 +39,8 @@
extern const uint16_t ff_wma_critical_freqs[25];
static float quant_table[95];
#define MAX_CHANNELS 2
#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
......@@ -56,8 +58,11 @@ typedef struct {
unsigned int *bands;
float root;
DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
DECLARE_ALIGNED(16, int16_t, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
DECLARE_ALIGNED(16, int16_t, current)[BINK_BLOCK_MAX_SIZE / 16];
float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
float *prev_ptr[MAX_CHANNELS]; ///< pointers to the overlap points in the coeffs array
uint8_t *packet_buffer;
union {
RDFTContext rdft;
DCTContext dct;
......@@ -107,6 +112,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
s->block_size = (s->frame_len - s->overlap_len) * s->channels;
sample_rate_half = (sample_rate + 1) / 2;
s->root = 2.0 / sqrt(s->frame_len);
for (i = 0; i < 95; i++) {
/* constant is result of 0.066399999/log10(M_E) */
quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
}
/* calculate number of bands */
for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
......@@ -126,8 +135,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
s->first = 1;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
for (i = 0; i < s->channels; i++)
for (i = 0; i < s->channels; i++) {
s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
s->prev_ptr[i] = s->coeffs_ptr[i] + s->frame_len - s->overlap_len;
}
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
......@@ -152,11 +163,18 @@ static const uint8_t rle_length_tab[16] = {
2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
};
#define GET_BITS_SAFE(out, nbits) do { \
if (get_bits_left(gb) < nbits) \
return AVERROR_INVALIDDATA; \
out = get_bits(gb, nbits); \
} while (0)
/**
* Decode Bink Audio block
* @param[out] out Output buffer (must contain s->block_size elements)
* @return 0 on success, negative error code on failure
*/
static void decode_block(BinkAudioContext *s, short *out, int use_dct)
static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
{
int ch, i, j, k;
float q, quant[25];
......@@ -169,17 +187,22 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
for (ch = 0; ch < s->channels; ch++) {
FFTSample *coeffs = s->coeffs_ptr[ch];
if (s->version_b) {
if (get_bits_left(gb) < 64)
return AVERROR_INVALIDDATA;
coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
} else {
if (get_bits_left(gb) < 58)
return AVERROR_INVALIDDATA;
coeffs[0] = get_float(gb) * s->root;
coeffs[1] = get_float(gb) * s->root;
}
if (get_bits_left(gb) < s->num_bands * 8)
return AVERROR_INVALIDDATA;
for (i = 0; i < s->num_bands; i++) {
/* constant is result of 0.066399999/log10(M_E) */
int value = get_bits(gb, 8);
quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
quant[i] = quant_table[FFMIN(value, 95)];
}
k = 0;
......@@ -190,15 +213,20 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
while (i < s->frame_len) {
if (s->version_b) {
j = i + 16;
} else if (get_bits1(gb)) {
j = i + rle_length_tab[get_bits(gb, 4)] * 8;
} else {
j = i + 8;
int v;
GET_BITS_SAFE(v, 1);
if (v) {
GET_BITS_SAFE(v, 4);
j = i + rle_length_tab[v] * 8;
} else {
j = i + 8;
}
}
j = FFMIN(j, s->frame_len);
width = get_bits(gb, 4);
GET_BITS_SAFE(width, 4);
if (width == 0) {
memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
i = j;
......@@ -208,9 +236,11 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
while (i < j) {
if (s->bands[k] == i)
q = quant[k++];
coeff = get_bits(gb, width);
GET_BITS_SAFE(coeff, width);
if (coeff) {
if (get_bits1(gb))
int v;
GET_BITS_SAFE(v, 1);
if (v)
coeffs[i] = -q * coeff;
else
coeffs[i] = q * coeff;
......@@ -231,8 +261,12 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
}
s->fmt_conv.float_to_int16_interleave(s->current,
(const float **)s->prev_ptr,
s->overlap_len, s->channels);
s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
s->frame_len, s->channels);
s->frame_len - s->overlap_len,
s->channels);
if (!s->first) {
int count = s->overlap_len * s->channels;
......@@ -242,16 +276,19 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
}
}
memcpy(s->previous, out + s->block_size,
s->overlap_len * s->channels * sizeof(*out));
memcpy(s->previous, s->current,
s->overlap_len * s->channels * sizeof(*s->previous));
s->first = 0;
return 0;
}
static av_cold int decode_end(AVCodecContext *avctx)
{
BinkAudioContext * s = avctx->priv_data;
av_freep(&s->bands);
av_freep(&s->packet_buffer);
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_end(&s->trans.rdft);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
......@@ -270,25 +307,47 @@ static int decode_frame(AVCodecContext *avctx,
AVPacket *avpkt)
{
BinkAudioContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
short *samples = data;
short *samples_end = (short*)((uint8_t*)data + *data_size);
int reported_size;
int16_t *samples = data;
GetBitContext *gb = &s->gb;
int out_size, consumed = 0;
if (!get_bits_left(gb)) {
uint8_t *buf;
/* handle end-of-stream */
if (!avpkt->size) {
*data_size = 0;
return 0;
}
if (avpkt->size < 4) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!buf)
return AVERROR(ENOMEM);
s->packet_buffer = buf;
memcpy(s->packet_buffer, avpkt->data, avpkt->size);
init_get_bits(gb, s->packet_buffer, avpkt->size * 8);
consumed = avpkt->size;
/* skip reported size */
skip_bits_long(gb, 32);
}
init_get_bits(gb, buf, buf_size * 8);
out_size = s->block_size * 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);
}
reported_size = get_bits_long(gb, 32);
while (get_bits_count(gb) / 8 < buf_size &&
samples + s->block_size <= samples_end) {
decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
samples += s->block_size;
get_bits_align32(gb);
if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) {
av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
return AVERROR_INVALIDDATA;
}
get_bits_align32(gb);
*data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
return buf_size;
*data_size = out_size;
return consumed;
}
AVCodec ff_binkaudio_rdft_decoder = {
......@@ -299,6 +358,7 @@ AVCodec ff_binkaudio_rdft_decoder = {
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.capabilities = CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
};
......@@ -310,5 +370,6 @@ AVCodec ff_binkaudio_dct_decoder = {
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.capabilities = CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
};
libavcodec/cook.c
View file @ d17e7070
This diff is collapsed.
libavcodec/dca.c
View file @ d17e7070
......@@ -528,15 +528,15 @@ static int dca_parse_frame_header(DCAContext * s)
s->sample_blocks = get_bits(&s->gb, 7) + 1;
s->frame_size = get_bits(&s->gb, 14) + 1;
if (s->frame_size < 95)
return -1;
return AVERROR_INVALIDDATA;
s->amode = get_bits(&s->gb, 6);
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
return -1;
return AVERROR_INVALIDDATA;
s->bit_rate_index = get_bits(&s->gb, 5);
s->bit_rate = dca_bit_rates[s->bit_rate_index];
if (!s->bit_rate)
return -1;
return AVERROR_INVALIDDATA;
s->downmix = get_bits(&s->gb, 1);
s->dynrange = get_bits(&s->gb, 1);
......@@ -626,7 +626,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
int j, k;
if (get_bits_left(&s->gb) < 0)
return -1;
return AVERROR_INVALIDDATA;
if (!base_channel) {
s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
......@@ -658,7 +658,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
else if (s->bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
return -1;
return AVERROR_INVALIDDATA;
} else {
s->bitalloc[j][k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
......@@ -667,7 +667,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
if (s->bitalloc[j][k] > 26) {
// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
// j, k, s->bitalloc[j][k]);
return -1;
return AVERROR_INVALIDDATA;
}
}
}
......@@ -685,7 +685,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
}
if (get_bits_left(&s->gb) < 0)
return -1;
return AVERROR_INVALIDDATA;
for (j = base_channel; j < s->prim_channels; j++) {
const uint32_t *scale_table;
......@@ -723,7 +723,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
}
if (get_bits_left(&s->gb) < 0)
return -1;
return AVERROR_INVALIDDATA;
/* Scale factors for joint subband coding */
for (j = base_channel; j < s->prim_channels; j++) {
......@@ -1055,7 +1055,7 @@ static int decode_blockcode(int code, int levels, int *values)
return 0;
else {
av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
return -1;
return AVERROR_INVALIDDATA;
}
}
#endif
......@@ -1096,7 +1096,7 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
for (k = base_channel; k < s->prim_channels; k++) {
if (get_bits_left(&s->gb) < 0)
return -1;
return AVERROR_INVALIDDATA;
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
......@@ -1275,12 +1275,13 @@ static int dca_subframe_footer(DCAContext * s, int base_channel)
static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
{
int ret;
/* Sanity check */
if (s->current_subframe >= s->subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
s->current_subframe, s->subframes);
return -1;
return AVERROR_INVALIDDATA;
}
if (!s->current_subsubframe) {
......@@ -1288,16 +1289,16 @@ static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
/* Read subframe header */
if (dca_subframe_header(s, base_channel, block_index))
return -1;
if ((ret = dca_subframe_header(s, base_channel, block_index)))
return ret;
}
/* Read subsubframe */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
if (dca_subsubframe(s, base_channel, block_index))
return -1;
if ((ret = dca_subsubframe(s, base_channel, block_index)))
return ret;
/* Update state */
s->current_subsubframe++;
......@@ -1310,8 +1311,8 @@ static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
/* Read subframe footer */
if (dca_subframe_footer(s, base_channel))
return -1;
if ((ret = dca_subframe_footer(s, base_channel)))
return ret;
}
return 0;
......@@ -1354,7 +1355,7 @@ static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * ds
flush_put_bits(&pb);
return (put_bits_count(&pb) + 7) >> 3;
default:
return -1;
return AVERROR_INVALIDDATA;
}
}
......@@ -1637,7 +1638,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
int lfe_samples;
int num_core_channels = 0;
int i;
int i, ret;
float *samples_flt = data;
int16_t *samples_s16 = data;
int out_size;
......@@ -1650,16 +1651,15 @@ static int dca_decode_frame(AVCodecContext * avctx,
s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
if (s->dca_buffer_size == -1) {
if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
return -1;
return AVERROR_INVALIDDATA;
}
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
if (dca_parse_frame_header(s) < 0) {
if ((ret = dca_parse_frame_header(s)) < 0) {
//seems like the frame is corrupt, try with the next one
*data_size=0;
return buf_size;
return ret;
}
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
......@@ -1669,7 +1669,10 @@ static int dca_decode_frame(AVCodecContext * avctx,
s->profile = FF_PROFILE_DTS;
for (i = 0; i < (s->sample_blocks / 8); i++) {
dca_decode_block(s, 0, i);
if ((ret = dca_decode_block(s, 0, i))) {
av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
return ret;
}
}
/* record number of core channels incase less than max channels are requested */
......@@ -1725,7 +1728,10 @@ static int dca_decode_frame(AVCodecContext * avctx,
dca_parse_audio_coding_header(s, s->xch_base_channel);
for (i = 0; i < (s->sample_blocks / 8); i++) {
dca_decode_block(s, s->xch_base_channel, i);
if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
continue;
}
}
s->xch_present = 1;
......@@ -1799,7 +1805,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
if (channels > !!s->lfe &&
s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
return -1;
return AVERROR_INVALIDDATA;
if (avctx->request_channels == 2 && s->prim_channels > 2) {
channels = 2;
......@@ -1812,7 +1818,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
}
} else {
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
return -1;
return AVERROR_INVALIDDATA;
}
if (avctx->channels != channels) {
......@@ -1824,7 +1830,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
out_size = 256 / 8 * s->sample_blocks * channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size)
return -1;
return AVERROR(EINVAL);
*data_size = out_size;
/* filter to get final output */
......
libavcodec/dsicinav.c
View file @ d17e7070
......@@ -26,6 +26,7 @@
#include "avcodec.h"
#include "bytestream.h"
#include "mathops.h"
typedef enum CinVideoBitmapIndex {
......@@ -43,7 +44,6 @@ typedef struct CinVideoContext {
} CinVideoContext;
typedef struct CinAudioContext {
AVCodecContext *avctx;
int initial_decode_frame;
int delta;
} CinAudioContext;
......@@ -309,7 +309,11 @@ static av_cold int cinaudio_decode_init(AVCodecContext *avctx)
{
CinAudioContext *cin = avctx->priv_data;
cin->avctx = avctx;
if (avctx->channels != 1) {
av_log_ask_for_sample(avctx, "Number of channels is not supported\n");
return AVERROR_PATCHWELCOME;
}
cin->initial_decode_frame = 1;
cin->delta = 0;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
......@@ -322,29 +326,35 @@ static int cinaudio_decode_frame(AVCodecContext *avctx,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
CinAudioContext *cin = avctx->priv_data;
const uint8_t *src = buf;
int16_t *samples = (int16_t *)data;
buf_size = FFMIN(buf_size, *data_size/2);
const uint8_t *buf_end = buf + avpkt->size;
int16_t *samples = data;
int delta, out_size;
out_size = (avpkt->size - cin->initial_decode_frame) *
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);
}
delta = cin->delta;
if (cin->initial_decode_frame) {
cin->initial_decode_frame = 0;
cin->delta = (int16_t)AV_RL16(src); src += 2;
*samples++ = cin->delta;
buf_size -= 2;
delta = sign_extend(AV_RL16(buf), 16);
buf += 2;
*samples++ = delta;
}
while (buf_size > 0) {
cin->delta += cinaudio_delta16_table[*src++];
cin->delta = av_clip_int16(cin->delta);
*samples++ = cin->delta;
--buf_size;
while (buf < buf_end) {
delta += cinaudio_delta16_table[*buf++];
delta = av_clip_int16(delta);
*samples++ = delta;
}
cin->delta = delta;
*data_size = (uint8_t *)samples - (uint8_t *)data;
*data_size = out_size;
return src - buf;
return avpkt->size;
}
......
libavcodec/flacdec.c
View file @ d17e7070
......@@ -587,7 +587,8 @@ static int flac_decode_frame(AVCodecContext *avctx,
bytes_read = (get_bits_count(&s->gb)+7)/8;
/* check if allocated data size is large enough for output */
output_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
output_size = s->blocksize * s->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (output_size > alloc_data_size) {
av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
"allocated data size\n");
......
libavcodec/h264.c
View file @ d17e7070
......@@ -1815,7 +1815,7 @@ static av_always_inline void hl_decode_mb_predict_luma(H264Context *h, int mb_ty
static const uint8_t dc_mapping[16] = { 0*16, 1*16, 4*16, 5*16, 2*16, 3*16, 6*16, 7*16,
8*16, 9*16,12*16,13*16,10*16,11*16,14*16,15*16};
for(i = 0; i < 16; i++)
dctcoef_set(h->mb+p*256, pixel_shift, dc_mapping[i], dctcoef_get(h->mb_luma_dc[p], pixel_shift, i));
dctcoef_set(h->mb+(p*256 << pixel_shift), pixel_shift, dc_mapping[i], dctcoef_get(h->mb_luma_dc[p], pixel_shift, i));
}
}
}else
......@@ -2033,7 +2033,7 @@ static av_always_inline void hl_decode_mb_internal(H264Context *h, int simple, i
}
if (chroma422) {
for(i=j*16+4; i<j*16+8; i++){
if(h->non_zero_count_cache[ scan8[i] ] || dctcoef_get(h->mb, pixel_shift, i*16))
if(h->non_zero_count_cache[ scan8[i+4] ] || dctcoef_get(h->mb, pixel_shift, i*16))
idct_add (dest[j-1] + block_offset[i+4], h->mb + (i*16 << pixel_shift), uvlinesize);
}
}
......
libavcodec/utvideo.c
View file @ d17e7070
......@@ -66,7 +66,7 @@ static int huff_cmp(const void *a, const void *b)
return (aa->len - bb->len)*256 + aa->sym - bb->sym;
}
static int build_huff(const uint8_t *src, VLC *vlc)
static int build_huff(const uint8_t *src, VLC *vlc, int *fsym)
{
int i;
HuffEntry he[256];
......@@ -76,13 +76,18 @@ static int build_huff(const uint8_t *src, VLC *vlc)
uint8_t syms[256];
uint32_t code;
*fsym = -1;
for (i = 0; i < 256; i++) {
he[i].sym = i;
he[i].len = *src++;
}
qsort(he, 256, sizeof(*he), huff_cmp);
if (!he[0].len || he[0].len > 32)
if (!he[0].len) {
*fsym = he[0].sym;
return 0;
}
if (he[0].len > 32)
return -1;
last = 255;
......@@ -112,12 +117,37 @@ static int decode_plane(UtvideoContext *c, int plane_no,
int sstart, send;
VLC vlc;
GetBitContext gb;
int prev;
int prev, fsym;
const int cmask = ~(!plane_no && c->avctx->pix_fmt == PIX_FMT_YUV420P);
if (build_huff(src, &vlc)) {
if (build_huff(src, &vlc, &fsym)) {
av_log(c->avctx, AV_LOG_ERROR, "Cannot build Huffman codes\n");
return AVERROR_INVALIDDATA;
}
if (fsym >= 0) { // build_huff reported a symbol to fill slices with
send = 0;
for (slice = 0; slice < c->slices; slice++) {
uint8_t *dest;
sstart = send;
send = (height * (slice + 1) / c->slices) & cmask;
dest = dst + sstart * stride;
prev = 0x80;
for (j = sstart; j < send; j++) {
for (i = 0; i < width * step; i += step) {
pix = fsym;
if (use_pred) {
prev += pix;
pix = prev;
}
dest[i] = pix;
}
dest += stride;
}
}
return 0;
}
src += 256;
src_size -= 256;
......@@ -128,7 +158,7 @@ static int decode_plane(UtvideoContext *c, int plane_no,
int slice_data_start, slice_data_end, slice_size;
sstart = send;
send = height * (slice + 1) / c->slices;
send = (height * (slice + 1) / c->slices) & cmask;
dest = dst + sstart * stride;
// slice offset and size validation was done earlier
......@@ -204,16 +234,17 @@ static void restore_rgb_planes(uint8_t *src, int step, int stride, int width, in
}
static void restore_median(uint8_t *src, int step, int stride,
int width, int height, int slices)
int width, int height, int slices, int rmode)
{
int i, j, slice;
int A, B, C;
uint8_t *bsrc;
int slice_start, slice_height;
const int cmask = ~rmode;
for (slice = 0; slice < slices; slice++) {
slice_start = (slice * height) / slices;
slice_height = ((slice + 1) * height) / slices - slice_start;
slice_start = ((slice * height) / slices) & cmask;
slice_height = ((((slice + 1) * height) / slices) & cmask) - slice_start;
bsrc = src + slice_start * stride;
......@@ -337,7 +368,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPac
if (c->frame_pred == PRED_MEDIAN)
restore_median(c->pic.data[0] + rgb_order[i], c->planes,
c->pic.linesize[0], avctx->width, avctx->height,
c->slices);
c->slices, 0);
}
restore_rgb_planes(c->pic.data[0], c->planes, c->pic.linesize[0],
avctx->width, avctx->height);
......@@ -353,7 +384,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPac
if (c->frame_pred == PRED_MEDIAN)
restore_median(c->pic.data[i], 1, c->pic.linesize[i],
avctx->width >> !!i, avctx->height >> !!i,
c->slices);
c->slices, !i);
}
break;
case PIX_FMT_YUV422P:
......@@ -366,7 +397,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPac
return ret;
if (c->frame_pred == PRED_MEDIAN)
restore_median(c->pic.data[i], 1, c->pic.linesize[i],
avctx->width >> !!i, avctx->height, c->slices);
avctx->width >> !!i, avctx->height, c->slices, 0);
}
break;
}
......
libavcodec/vp3.c
View file @ d17e7070
......@@ -45,6 +45,7 @@
#define FRAGMENT_PIXELS 8
static av_cold int vp3_decode_end(AVCodecContext *avctx);
static void vp3_decode_flush(AVCodecContext *avctx);
//FIXME split things out into their own arrays
typedef struct Vp3Fragment {
......@@ -890,7 +891,7 @@ static int unpack_vlcs(Vp3DecodeContext *s, GetBitContext *gb,
/* decode a VLC into a token */
token = get_vlc2(gb, vlc_table, 11, 3);
/* use the token to get a zero run, a coefficient, and an eob run */
if (token <= 6) {
if ((unsigned) token <= 6U) {
eob_run = eob_run_base[token];
if (eob_run_get_bits[token])
eob_run += get_bits(gb, eob_run_get_bits[token]);
......@@ -908,7 +909,7 @@ static int unpack_vlcs(Vp3DecodeContext *s, GetBitContext *gb,
coeff_i += eob_run;
eob_run = 0;
}
} else {
} else if (token >= 0) {
bits_to_get = coeff_get_bits[token];
if (bits_to_get)
bits_to_get = get_bits(gb, bits_to_get);
......@@ -942,6 +943,10 @@ static int unpack_vlcs(Vp3DecodeContext *s, GetBitContext *gb,
for (i = coeff_index+1; i <= coeff_index+zero_run; i++)
s->num_coded_frags[plane][i]--;
coeff_i++;
} else {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid token %d\n", token);
return -1;
}
}
......@@ -991,6 +996,8 @@ static int unpack_dct_coeffs(Vp3DecodeContext *s, GetBitContext *gb)
/* unpack the Y plane DC coefficients */
residual_eob_run = unpack_vlcs(s, gb, &s->dc_vlc[dc_y_table], 0,
0, residual_eob_run);
if (residual_eob_run < 0)
return residual_eob_run;
/* reverse prediction of the Y-plane DC coefficients */
reverse_dc_prediction(s, 0, s->fragment_width[0], s->fragment_height[0]);
......@@ -998,8 +1005,12 @@ static int unpack_dct_coeffs(Vp3DecodeContext *s, GetBitContext *gb)
/* unpack the C plane DC coefficients */
residual_eob_run = unpack_vlcs(s, gb, &s->dc_vlc[dc_c_table], 0,
1, residual_eob_run);
if (residual_eob_run < 0)
return residual_eob_run;
residual_eob_run = unpack_vlcs(s, gb, &s->dc_vlc[dc_c_table], 0,
2, residual_eob_run);
if (residual_eob_run < 0)
return residual_eob_run;
/* reverse prediction of the C-plane DC coefficients */
if (!(s->avctx->flags & CODEC_FLAG_GRAY))
......@@ -1036,11 +1047,17 @@ static int unpack_dct_coeffs(Vp3DecodeContext *s, GetBitContext *gb)
for (i = 1; i <= 63; i++) {
residual_eob_run = unpack_vlcs(s, gb, y_tables[i], i,
0, residual_eob_run);
if (residual_eob_run < 0)
return residual_eob_run;
residual_eob_run = unpack_vlcs(s, gb, c_tables[i], i,
1, residual_eob_run);
if (residual_eob_run < 0)
return residual_eob_run;
residual_eob_run = unpack_vlcs(s, gb, c_tables[i], i,
2, residual_eob_run);
if (residual_eob_run < 0)
return residual_eob_run;
}
return 0;
......@@ -1777,10 +1794,15 @@ static int vp3_update_thread_context(AVCodecContext *dst, const AVCodecContext *
Vp3DecodeContext *s = dst->priv_data, *s1 = src->priv_data;
int qps_changed = 0, i, err;
#define copy_fields(to, from, start_field, end_field) memcpy(&to->start_field, &from->start_field, (char*)&to->end_field - (char*)&to->start_field)
if (!s1->current_frame.data[0]
||s->width != s1->width
||s->height!= s1->height)
||s->height!= s1->height) {
if (s != s1)
copy_fields(s, s1, golden_frame, current_frame);
return -1;
}
if (s != s1) {
// init tables if the first frame hasn't been decoded
......@@ -1796,8 +1818,6 @@ static int vp3_update_thread_context(AVCodecContext *dst, const AVCodecContext *
memcpy(s->motion_val[1], s1->motion_val[1], c_fragment_count * sizeof(*s->motion_val[1]));
}
#define copy_fields(to, from, start_field, end_field) memcpy(&to->start_field, &from->start_field, (char*)&to->end_field - (char*)&to->start_field)
// copy previous frame data
copy_fields(s, s1, golden_frame, dsp);
......@@ -1990,9 +2010,6 @@ static av_cold int vp3_decode_end(AVCodecContext *avctx)
Vp3DecodeContext *s = avctx->priv_data;
int i;
if (avctx->is_copy && !s->current_frame.data[0])
return 0;
av_free(s->superblock_coding);
av_free(s->all_fragments);
av_free(s->coded_fragment_list[0]);
......@@ -2339,6 +2356,23 @@ static void vp3_decode_flush(AVCodecContext *avctx)
ff_thread_release_buffer(avctx, &s->current_frame);
}
static int vp3_init_thread_copy(AVCodecContext *avctx)
{
Vp3DecodeContext *s = avctx->priv_data;
s->superblock_coding = NULL;
s->all_fragments = NULL;
s->coded_fragment_list[0] = NULL;
s->dct_tokens_base = NULL;
s->superblock_fragments = NULL;
s->macroblock_coding = NULL;
s->motion_val[0] = NULL;
s->motion_val[1] = NULL;
s->edge_emu_buffer = NULL;
return 0;
}
AVCodec ff_theora_decoder = {
.name = "theora",
.type = AVMEDIA_TYPE_VIDEO,
......@@ -2350,6 +2384,7 @@ AVCodec ff_theora_decoder = {
.capabilities = CODEC_CAP_DR1 | CODEC_CAP_DRAW_HORIZ_BAND | CODEC_CAP_FRAME_THREADS,
.flush = vp3_decode_flush,
.long_name = NULL_IF_CONFIG_SMALL("Theora"),
.init_thread_copy = ONLY_IF_THREADS_ENABLED(vp3_init_thread_copy),
.update_thread_context = ONLY_IF_THREADS_ENABLED(vp3_update_thread_context)
};
#endif
......@@ -2365,5 +2400,6 @@ AVCodec ff_vp3_decoder = {
.capabilities = CODEC_CAP_DR1 | CODEC_CAP_DRAW_HORIZ_BAND | CODEC_CAP_FRAME_THREADS,
.flush = vp3_decode_flush,
.long_name = NULL_IF_CONFIG_SMALL("On2 VP3"),
.init_thread_copy = ONLY_IF_THREADS_ENABLED(vp3_init_thread_copy),
.update_thread_context = ONLY_IF_THREADS_ENABLED(vp3_update_thread_context)
};
libavcodec/vp8.c
View file @ d17e7070
......@@ -50,7 +50,8 @@ static int vp8_alloc_frame(VP8Context *s, AVFrame *f)
int ret;
if ((ret = ff_thread_get_buffer(s->avctx, f)) < 0)
return ret;
if (!s->maps_are_invalid && s->num_maps_to_be_freed) {
if (s->num_maps_to_be_freed) {
assert(!s->maps_are_invalid);
f->ref_index[0] = s->segmentation_maps[--s->num_maps_to_be_freed];
} else if (!(f->ref_index[0] = av_mallocz(s->mb_width * s->mb_height))) {
ff_thread_release_buffer(s->avctx, f);
......@@ -59,39 +60,50 @@ static int vp8_alloc_frame(VP8Context *s, AVFrame *f)
return 0;
}
static void vp8_release_frame(VP8Context *s, AVFrame *f, int is_close)
static void vp8_release_frame(VP8Context *s, AVFrame *f, int prefer_delayed_free, int can_direct_free)
{
if (!is_close) {
if (f->ref_index[0]) {
assert(s->num_maps_to_be_freed < FF_ARRAY_ELEMS(s->segmentation_maps));
s->segmentation_maps[s->num_maps_to_be_freed++] = f->ref_index[0];
if (f->ref_index[0]) {
if (prefer_delayed_free) {
/* Upon a size change, we want to free the maps but other threads may still
* be using them, so queue them. Upon a seek, all threads are inactive so
* we want to cache one to prevent re-allocation in the next decoding
* iteration, but the rest we can free directly. */
int max_queued_maps = can_direct_free ? 1 : FF_ARRAY_ELEMS(s->segmentation_maps);
if (s->num_maps_to_be_freed < max_queued_maps) {
s->segmentation_maps[s->num_maps_to_be_freed++] = f->ref_index[0];
} else if (can_direct_free) /* vp8_decode_flush(), but our queue is full */ {
av_free(f->ref_index[0]);
} /* else: MEMLEAK (should never happen, but better that than crash) */
f->ref_index[0] = NULL;
} else /* vp8_decode_free() */ {
av_free(f->ref_index[0]);
}
} else {
av_freep(&f->ref_index[0]);
}
ff_thread_release_buffer(s->avctx, f);
}
static void vp8_decode_flush_impl(AVCodecContext *avctx, int force, int is_close)
static void vp8_decode_flush_impl(AVCodecContext *avctx,
int prefer_delayed_free, int can_direct_free, int free_mem)
{
VP8Context *s = avctx->priv_data;
int i;
if (!avctx->is_copy || force) {
if (!avctx->is_copy) {
for (i = 0; i < 5; i++)
if (s->frames[i].data[0])
vp8_release_frame(s, &s->frames[i], is_close);
vp8_release_frame(s, &s->frames[i], prefer_delayed_free, can_direct_free);
}
memset(s->framep, 0, sizeof(s->framep));
free_buffers(s);
s->maps_are_invalid = 1;
if (free_mem) {
free_buffers(s);
s->maps_are_invalid = 1;
}
}
static void vp8_decode_flush(AVCodecContext *avctx)
{
vp8_decode_flush_impl(avctx, 0, 0);
vp8_decode_flush_impl(avctx, 1, 1, 0);
}
static int update_dimensions(VP8Context *s, int width, int height)
......@@ -101,7 +113,7 @@ static int update_dimensions(VP8Context *s, int width, int height)
if (av_image_check_size(width, height, 0, s->avctx))
return AVERROR_INVALIDDATA;
vp8_decode_flush_impl(s->avctx, 1, 0);
vp8_decode_flush_impl(s->avctx, 1, 0, 1);
avcodec_set_dimensions(s->avctx, width, height);
}
......@@ -1581,7 +1593,7 @@ static int vp8_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
&s->frames[i] != s->framep[VP56_FRAME_PREVIOUS] &&
&s->frames[i] != s->framep[VP56_FRAME_GOLDEN] &&
&s->frames[i] != s->framep[VP56_FRAME_GOLDEN2])
vp8_release_frame(s, &s->frames[i], 0);
vp8_release_frame(s, &s->frames[i], 1, 0);
// find a free buffer
for (i = 0; i < 5; i++)
......@@ -1597,7 +1609,7 @@ static int vp8_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
abort();
}
if (curframe->data[0])
ff_thread_release_buffer(avctx, curframe);
vp8_release_frame(s, curframe, 1, 0);
curframe->key_frame = s->keyframe;
curframe->pict_type = s->keyframe ? AV_PICTURE_TYPE_I : AV_PICTURE_TYPE_P;
......@@ -1778,7 +1790,7 @@ static av_cold int vp8_decode_init(AVCodecContext *avctx)
static av_cold int vp8_decode_free(AVCodecContext *avctx)
{
vp8_decode_flush_impl(avctx, 0, 1);
vp8_decode_flush_impl(avctx, 0, 1, 1);
release_queued_segmaps(avctx->priv_data, 1);
return 0;
}
......
libavformat/avformat.h
View file @ d17e7070
......@@ -676,6 +676,7 @@ typedef struct AVStream {
int duration_count;
double duration_error[2][2][MAX_STD_TIMEBASES];
int64_t codec_info_duration;
int nb_decoded_frames;
} *info;
/**
......
libavformat/dsicin.c
View file @ d17e7070
......@@ -131,9 +131,8 @@ static int cin_read_header(AVFormatContext *s, AVFormatParameters *ap)
st->codec->codec_tag = 0; /* no tag */
st->codec->channels = 1;
st->codec->sample_rate = 22050;
st->codec->bits_per_coded_sample = 16;
st->codec->bits_per_coded_sample = 8;
st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample * st->codec->channels;
st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
return 0;
}
......@@ -211,7 +210,8 @@ static int cin_read_packet(AVFormatContext *s, AVPacket *pkt)
pkt->stream_index = cin->audio_stream_index;
pkt->pts = cin->audio_stream_pts;
cin->audio_stream_pts += cin->audio_buffer_size * 2 / cin->file_header.audio_frame_size;
pkt->duration = cin->audio_buffer_size - (pkt->pts == 0);
cin->audio_stream_pts += pkt->duration;
cin->audio_buffer_size = 0;
return 0;
}
......
libavformat/utils.c
View file @ d17e7070
......@@ -2200,7 +2200,7 @@ static int has_codec_parameters(AVCodecContext *avctx)
static int has_decode_delay_been_guessed(AVStream *st)
{
return st->codec->codec_id != CODEC_ID_H264 ||
st->codec_info_nb_frames >= 6 + st->codec->has_b_frames;
st->info->nb_decoded_frames >= 6;
}
static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **options)
......@@ -2226,6 +2226,8 @@ static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **option
avcodec_get_frame_defaults(&picture);
ret = avcodec_decode_video2(st->codec, &picture,
&got_picture, avpkt);
if (got_picture)
st->info->nb_decoded_frames++;
break;
case AVMEDIA_TYPE_AUDIO:
data_size = FFMAX(avpkt->size, AVCODEC_MAX_AUDIO_FRAME_SIZE);
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment