[FFmpeg-cvslog] adpcmdec: calculate actual number of output samples for each decoder.
Justin Ruggles
git at videolan.org
Sun Oct 16 04:20:36 CEST 2011
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sun Oct 2 10:18:17 2011 -0400| [a62c0f94ee5c96e98b1f91d9f1ab9f568037bb00] | committer: Justin Ruggles
adpcmdec: calculate actual number of output samples for each decoder.
This also allows for removing some of the buf_size checks and using the
sample count for some of the decoding loops.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a62c0f94ee5c96e98b1f91d9f1ab9f568037bb00
---
libavcodec/adpcm.c | 351 ++++++++++++++++++++++++++++++++++------------------
1 files changed, 231 insertions(+), 120 deletions(-)
diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c
index 774193e..55f518b 100644
--- a/libavcodec/adpcm.c
+++ b/libavcodec/adpcm.c
@@ -315,6 +315,173 @@ static void xa_decode(short *out, const unsigned char *in,
}
}
+/**
+ * Get the number of samples that will be decoded from the packet.
+ * In one case, this is actually the maximum number of samples possible to
+ * decode with the given buf_size.
+ *
+ * @param[out] coded_samples set to the number of samples as coded in the
+ * packet, or 0 if the codec does not encode the
+ * number of samples in each frame.
+ */
+static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf,
+ int buf_size, int *coded_samples)
+{
+ ADPCMDecodeContext *s = avctx->priv_data;
+ int nb_samples = 0;
+ int ch = avctx->channels;
+ int has_coded_samples = 0;
+ int header_size;
+
+ *coded_samples = 0;
+
+ switch (avctx->codec->id) {
+ /* constant, only check buf_size */
+ case CODEC_ID_ADPCM_EA_XAS:
+ if (buf_size < 76 * ch)
+ return 0;
+ nb_samples = 128;
+ break;
+ case CODEC_ID_ADPCM_IMA_QT:
+ if (buf_size < 34 * ch)
+ return 0;
+ nb_samples = 64;
+ break;
+ /* simple 4-bit adpcm */
+ case CODEC_ID_ADPCM_CT:
+ case CODEC_ID_ADPCM_IMA_EA_SEAD:
+ case CODEC_ID_ADPCM_IMA_WS:
+ case CODEC_ID_ADPCM_YAMAHA:
+ nb_samples = buf_size * 2 / ch;
+ break;
+ }
+ if (nb_samples)
+ return nb_samples;
+
+ /* simple 4-bit adpcm, with header */
+ header_size = 0;
+ switch (avctx->codec->id) {
+ case CODEC_ID_ADPCM_4XM:
+ case CODEC_ID_ADPCM_IMA_ISS: header_size = 4 * ch; break;
+ case CODEC_ID_ADPCM_IMA_AMV: header_size = 8; break;
+ case CODEC_ID_ADPCM_IMA_SMJPEG: header_size = 4; break;
+ }
+ if (header_size > 0)
+ return (buf_size - header_size) * 2 / ch;
+
+ /* more complex formats */
+ switch (avctx->codec->id) {
+ case CODEC_ID_ADPCM_EA:
+ has_coded_samples = 1;
+ if (buf_size < 4)
+ return 0;
+ *coded_samples = AV_RL32(buf);
+ *coded_samples -= *coded_samples % 28;
+ nb_samples = (buf_size - 12) / 30 * 28;
+ break;
+ case CODEC_ID_ADPCM_IMA_EA_EACS:
+ has_coded_samples = 1;
+ if (buf_size < 4)
+ return 0;
+ *coded_samples = AV_RL32(buf);
+ nb_samples = (buf_size - (4 + 8 * ch)) * 2 / ch;
+ break;
+ case CODEC_ID_ADPCM_EA_MAXIS_XA:
+ nb_samples = ((buf_size - ch) / (2 * ch)) * 2 * ch;
+ break;
+ case CODEC_ID_ADPCM_EA_R1:
+ case CODEC_ID_ADPCM_EA_R2:
+ case CODEC_ID_ADPCM_EA_R3:
+ /* maximum number of samples */
+ /* has internal offsets and a per-frame switch to signal raw 16-bit */
+ has_coded_samples = 1;
+ if (buf_size < 4)
+ return 0;
+ switch (avctx->codec->id) {
+ case CODEC_ID_ADPCM_EA_R1:
+ header_size = 4 + 9 * ch;
+ *coded_samples = AV_RL32(buf);
+ break;
+ case CODEC_ID_ADPCM_EA_R2:
+ header_size = 4 + 5 * ch;
+ *coded_samples = AV_RL32(buf);
+ break;
+ case CODEC_ID_ADPCM_EA_R3:
+ header_size = 4 + 5 * ch;
+ *coded_samples = AV_RB32(buf);
+ break;
+ }
+ *coded_samples -= *coded_samples % 28;
+ nb_samples = (buf_size - header_size) * 2 / ch;
+ nb_samples -= nb_samples % 28;
+ break;
+ case CODEC_ID_ADPCM_IMA_DK3:
+ if (avctx->block_align > 0)
+ buf_size = FFMIN(buf_size, avctx->block_align);
+ nb_samples = ((buf_size - 16) * 8 / 3) / ch;
+ break;
+ case CODEC_ID_ADPCM_IMA_DK4:
+ nb_samples = 1 + (buf_size - 4 * ch) * 2 / ch;
+ break;
+ case CODEC_ID_ADPCM_IMA_WAV:
+ if (avctx->block_align > 0)
+ buf_size = FFMIN(buf_size, avctx->block_align);
+ nb_samples = 1 + (buf_size - 4 * ch) / (4 * ch) * 8;
+ break;
+ case CODEC_ID_ADPCM_MS:
+ if (avctx->block_align > 0)
+ buf_size = FFMIN(buf_size, avctx->block_align);
+ nb_samples = 2 + (buf_size - 7 * ch) * 2 / ch;
+ break;
+ case CODEC_ID_ADPCM_SBPRO_2:
+ case CODEC_ID_ADPCM_SBPRO_3:
+ case CODEC_ID_ADPCM_SBPRO_4:
+ {
+ int samples_per_byte;
+ switch (avctx->codec->id) {
+ case CODEC_ID_ADPCM_SBPRO_2: samples_per_byte = 4; break;
+ case CODEC_ID_ADPCM_SBPRO_3: samples_per_byte = 3; break;
+ case CODEC_ID_ADPCM_SBPRO_4: samples_per_byte = 2; break;
+ }
+ if (!s->status[0].step_index) {
+ nb_samples++;
+ buf_size -= ch;
+ }
+ nb_samples += buf_size * samples_per_byte / ch;
+ break;
+ }
+ case CODEC_ID_ADPCM_SWF:
+ {
+ int buf_bits = buf_size * 8 - 2;
+ int nbits = (buf[0] >> 6) + 2;
+ int block_hdr_size = 22 * ch;
+ int block_size = block_hdr_size + nbits * ch * 4095;
+ int nblocks = buf_bits / block_size;
+ int bits_left = buf_bits - nblocks * block_size;
+ nb_samples = nblocks * 4096;
+ if (bits_left >= block_hdr_size)
+ nb_samples += 1 + (bits_left - block_hdr_size) / (nbits * ch);
+ break;
+ }
+ case CODEC_ID_ADPCM_THP:
+ has_coded_samples = 1;
+ if (buf_size < 8)
+ return 0;
+ *coded_samples = AV_RB32(&buf[4]);
+ *coded_samples -= *coded_samples % 14;
+ nb_samples = (buf_size - 80) / (8 * ch) * 14;
+ break;
+ case CODEC_ID_ADPCM_XA:
+ nb_samples = (buf_size / 128) * 224 / ch;
+ break;
+ }
+
+ /* validate coded sample count */
+ if (has_coded_samples && (*coded_samples <= 0 || *coded_samples > nb_samples))
+ return AVERROR_INVALIDDATA;
+
+ return nb_samples;
+}
/* DK3 ADPCM support macro */
#define DK3_GET_NEXT_NIBBLE() \
@@ -344,20 +511,33 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
ADPCMChannelStatus *cs;
int n, m, channel, i;
short *samples;
- short *samples_end;
const uint8_t *src;
int st; /* stereo */
- uint32_t samples_in_chunk;
int count1, count2;
+ int nb_samples, coded_samples, out_bps, out_size;
- //should protect all 4bit ADPCM variants
- //8 is needed for CODEC_ID_ADPCM_IMA_WAV with 2 channels
- //
- if(*data_size/4 < buf_size + 8)
- return -1;
+ nb_samples = get_nb_samples(avctx, buf, buf_size, &coded_samples);
+ if (nb_samples <= 0) {
+ av_log(avctx, AV_LOG_ERROR, "invalid number of samples in packet\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
+ out_size = nb_samples * avctx->channels * out_bps;
+ if (*data_size < out_size) {
+ av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
+ return AVERROR(EINVAL);
+ }
+ /* use coded_samples when applicable */
+ /* it is always <= nb_samples, so the output buffer will be large enough */
+ if (coded_samples) {
+ if (coded_samples != nb_samples)
+ av_log(avctx, AV_LOG_WARNING, "mismatch in coded sample count\n");
+ nb_samples = coded_samples;
+ out_size = nb_samples * avctx->channels * out_bps;
+ }
samples = data;
- samples_end= samples + *data_size/2;
src = buf;
st = avctx->channels == 2 ? 1 : 0;
@@ -366,10 +546,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
case CODEC_ID_ADPCM_IMA_QT:
/* In QuickTime, IMA is encoded by chunks of 34 bytes (=64 samples).
Channel data is interleaved per-chunk. */
- if (buf_size / 34 < avctx->channels) {
- av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
- return AVERROR(EINVAL);
- }
for (channel = 0; channel < avctx->channels; channel++) {
int16_t predictor;
int step_index;
@@ -410,15 +586,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
src ++;
}
}
- if (st)
- samples--;
break;
case CODEC_ID_ADPCM_IMA_WAV:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
-// samples_per_block= (block_align-4*chanels)*8 / (bits_per_sample * chanels) + 1;
-
for(i=0; i<avctx->channels; i++){
cs = &(c->status[i]);
cs->predictor = *samples++ = (int16_t)bytestream_get_le16(&src);
@@ -431,7 +603,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if (*src++) av_log(avctx, AV_LOG_ERROR, "unused byte should be null but is %d!!\n", src[-1]); /* unused */
}
- while (src <= buf + buf_size - (avctx->channels * 4)) {
+ for (n = (nb_samples - 1) / 8; n > 0; n--) {
for (i = 0; i < avctx->channels; i++) {
cs = &c->status[i];
for (m = 0; m < 4; m++) {
@@ -455,20 +627,17 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88);
}
- m= (buf_size - (src - buf))>>st;
-
for (i = 0; i < avctx->channels; i++) {
samples = (short*)data + i;
cs = &c->status[i];
- for (n = 0; n < m; n++) {
- uint8_t v = *src++;
+ for (n = nb_samples >> 1; n > 0; n--, src++) {
+ uint8_t v = *src;
*samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 4);
samples += avctx->channels;
*samples = adpcm_ima_expand_nibble(cs, v >> 4 , 4);
samples += avctx->channels;
}
}
- samples -= (avctx->channels - 1);
break;
case CODEC_ID_ADPCM_MS:
{
@@ -476,9 +645,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
- n = buf_size - 7 * avctx->channels;
- if (n < 0)
- return -1;
block_predictor = av_clip(*src++, 0, 6);
c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor];
@@ -502,10 +668,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if (st) *samples++ = c->status[1].sample2;
*samples++ = c->status[0].sample1;
if (st) *samples++ = c->status[1].sample1;
- for(;n>0;n--) {
+ for(n = (nb_samples - 2) >> (1 - st); n > 0; n--, src++) {
*samples++ = adpcm_ms_expand_nibble(&c->status[0 ], src[0] >> 4 );
*samples++ = adpcm_ms_expand_nibble(&c->status[st], src[0] & 0x0F);
- src ++;
}
break;
}
@@ -513,12 +678,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
- n = buf_size - 4 * avctx->channels;
- if (n < 0) {
- av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
- return AVERROR(EINVAL);
- }
-
for (channel = 0; channel < avctx->channels; channel++) {
cs = &c->status[channel];
cs->predictor = (int16_t)bytestream_get_le16(&src);
@@ -526,8 +685,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
src++;
*samples++ = cs->predictor;
}
- while (n-- > 0) {
- uint8_t v = *src++;
+ for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
+ uint8_t v = *src;
*samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
}
@@ -543,9 +702,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
- if(buf_size + 16 > (samples_end - samples)*3/8)
- return -1;
-
c->status[0].predictor = (int16_t)AV_RL16(src + 10);
c->status[1].predictor = (int16_t)AV_RL16(src + 12);
c->status[0].step_index = src[14];
@@ -586,12 +742,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
break;
}
case CODEC_ID_ADPCM_IMA_ISS:
- n = buf_size - 4 * avctx->channels;
- if (n < 0) {
- av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
- return AVERROR(EINVAL);
- }
-
for (channel = 0; channel < avctx->channels; channel++) {
cs = &c->status[channel];
cs->predictor = (int16_t)bytestream_get_le16(&src);
@@ -599,9 +749,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
src++;
}
- while (n-- > 0) {
+ for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
uint8_t v1, v2;
- uint8_t v = *src++;
+ uint8_t v = *src;
/* nibbles are swapped for mono */
if (st) {
v1 = v >> 4;
@@ -630,28 +780,21 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
buf_size -= 128;
}
break;
- case CODEC_ID_ADPCM_IMA_EA_EACS: {
- unsigned header_size = 4 + (8<<st);
- samples_in_chunk = bytestream_get_le32(&src) >> (1-st);
-
- if (buf_size < header_size || samples_in_chunk > buf_size - header_size) {
- src += buf_size - 4;
- break;
- }
+ case CODEC_ID_ADPCM_IMA_EA_EACS:
+ src += 4; // skip sample count (already read)
for (i=0; i<=st; i++)
c->status[i].step_index = bytestream_get_le32(&src);
for (i=0; i<=st; i++)
c->status[i].predictor = bytestream_get_le32(&src);
- for (; samples_in_chunk; samples_in_chunk--, src++) {
+ for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0], *src>>4, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[st], *src&0x0F, 3);
}
break;
- }
case CODEC_ID_ADPCM_IMA_EA_SEAD:
- for (; src < buf+buf_size; src++) {
+ for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[0] >> 4, 6);
*samples++ = adpcm_ima_expand_nibble(&c->status[st],src[0]&0x0F, 6);
}
@@ -666,22 +809,15 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
/* Each EA ADPCM frame has a 12-byte header followed by 30-byte pieces,
each coding 28 stereo samples. */
- if (buf_size < 12) {
- av_log(avctx, AV_LOG_ERROR, "frame too small\n");
- return AVERROR(EINVAL);
- }
- samples_in_chunk = AV_RL32(src);
- if (samples_in_chunk / 28 > (buf_size - 12) / 30) {
- av_log(avctx, AV_LOG_ERROR, "invalid frame\n");
- return AVERROR(EINVAL);
- }
- src += 4;
+
+ src += 4; // skip sample count (already read)
+
current_left_sample = (int16_t)bytestream_get_le16(&src);
previous_left_sample = (int16_t)bytestream_get_le16(&src);
current_right_sample = (int16_t)bytestream_get_le16(&src);
previous_right_sample = (int16_t)bytestream_get_le16(&src);
- for (count1 = 0; count1 < samples_in_chunk/28;count1++) {
+ for (count1 = 0; count1 < nb_samples / 28; count1++) {
coeff1l = ea_adpcm_table[ *src >> 4 ];
coeff2l = ea_adpcm_table[(*src >> 4 ) + 4];
coeff1r = ea_adpcm_table[*src & 0x0F];
@@ -728,7 +864,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
shift[channel] = (*src & 0x0F) + 8;
src++;
}
- for (count1 = 0; count1 < (buf_size - avctx->channels) / avctx->channels; count1++) {
+ for (count1 = 0; count1 < nb_samples / 2; count1++) {
for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL (mono) */
for(channel = 0; channel < avctx->channels; channel++) {
int32_t sample = (int32_t)(((*(src+channel) >> i) & 0x0F) << 0x1C) >> shift[channel];
@@ -742,6 +878,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
}
src+=avctx->channels;
}
+ /* consume whole packet */
+ src = buf + buf_size;
break;
}
case CODEC_ID_ADPCM_EA_R1:
@@ -759,14 +897,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
uint16_t *samplesC;
const uint8_t *srcC;
const uint8_t *src_end = buf + buf_size;
+ int count = 0;
- samples_in_chunk = (big_endian ? bytestream_get_be32(&src)
- : bytestream_get_le32(&src)) / 28;
- if (samples_in_chunk > UINT32_MAX/(28*avctx->channels) ||
- 28*samples_in_chunk*avctx->channels > samples_end-samples) {
- src += buf_size - 4;
- break;
- }
+ src += 4; // skip sample count (already read)
for (channel=0; channel<avctx->channels; channel++) {
int32_t offset = (big_endian ? bytestream_get_be32(&src)
@@ -785,7 +918,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
previous_sample = c->status[channel].prev_sample;
}
- for (count1=0; count1<samples_in_chunk; count1++) {
+ for (count1 = 0; count1 < nb_samples / 28; count1++) {
if (*srcC == 0xEE) { /* only seen in R2 and R3 */
srcC++;
if (srcC > src_end - 30*2) break;
@@ -819,6 +952,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
}
}
}
+ if (!count) {
+ count = count1;
+ } else if (count != count1) {
+ av_log(avctx, AV_LOG_WARNING, "per-channel sample count mismatch\n");
+ count = FFMAX(count, count1);
+ }
if (avctx->codec->id != CODEC_ID_ADPCM_EA_R1) {
c->status[channel].predictor = current_sample;
@@ -826,16 +965,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
}
}
- src = src + buf_size - (4 + 4*avctx->channels);
- samples += 28 * samples_in_chunk * avctx->channels;
+ out_size = count * 28 * avctx->channels * out_bps;
+ src = src_end;
break;
}
case CODEC_ID_ADPCM_EA_XAS:
- if (samples_end-samples < 32*4*avctx->channels
- || buf_size < (4+15)*4*avctx->channels) {
- src += buf_size;
- break;
- }
for (channel=0; channel<avctx->channels; channel++) {
int coeff[2][4], shift[4];
short *s2, *s = &samples[channel];
@@ -859,7 +993,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
}
}
}
- samples += 32*4*avctx->channels;
break;
case CODEC_ID_ADPCM_IMA_AMV:
case CODEC_ID_ADPCM_IMA_SMJPEG:
@@ -869,7 +1002,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
src+=4;
- while (src < buf + buf_size) {
+ for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
char hi, lo;
lo = *src & 0x0F;
hi = *src >> 4;
@@ -881,12 +1014,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
lo, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
hi, 3);
- src++;
}
break;
case CODEC_ID_ADPCM_CT:
- while (src < buf + buf_size) {
- uint8_t v = *src++;
+ for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
+ uint8_t v = *src;
*samples++ = adpcm_ct_expand_nibble(&c->status[0 ], v >> 4 );
*samples++ = adpcm_ct_expand_nibble(&c->status[st], v & 0x0F);
}
@@ -900,27 +1032,26 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if (st)
*samples++ = 128 * (*src++ - 0x80);
c->status[0].step_index = 1;
+ nb_samples--;
}
if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_4) {
- while (src < buf + buf_size) {
+ for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 4, 4, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
src[0] & 0x0F, 4, 0);
- src++;
}
} else if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_3) {
- while (src < buf + buf_size && samples + 2 < samples_end) {
+ for (n = nb_samples / 3; n > 0; n--, src++) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 5 , 3, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
(src[0] >> 2) & 0x07, 3, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] & 0x03, 2, 0);
- src++;
}
} else {
- while (src < buf + buf_size && samples + 3 < samples_end) {
+ for (n = nb_samples >> (2 - st); n > 0; n--, src++) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 6 , 2, 2);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
@@ -929,7 +1060,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
(src[0] >> 2) & 0x03, 2, 2);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
src[0] & 0x03, 2, 2);
- src++;
}
}
break;
@@ -984,10 +1114,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
c->status[i].predictor = av_clip_int16(c->status[i].predictor);
*samples++ = c->status[i].predictor;
- if (samples >= samples_end) {
- av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
- return -1;
- }
}
}
}
@@ -995,8 +1121,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
break;
}
case CODEC_ID_ADPCM_YAMAHA:
- while (src < buf + buf_size) {
- uint8_t v = *src++;
+ for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
+ uint8_t v = *src;
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0 ], v & 0x0F);
*samples++ = adpcm_yamaha_expand_nibble(&c->status[st], v >> 4 );
}
@@ -1004,17 +1130,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
case CODEC_ID_ADPCM_THP:
{
int table[2][16];
- unsigned int samplecnt;
int prev[2][2];
int ch;
- if (buf_size < 80) {
- av_log(avctx, AV_LOG_ERROR, "frame too small\n");
- return -1;
- }
-
- src+=4;
- samplecnt = bytestream_get_be32(&src);
+ src += 4; // skip channel size
+ src += 4; // skip number of samples (already read)
for (i = 0; i < 32; i++)
table[0][i] = (int16_t)bytestream_get_be16(&src);
@@ -1023,16 +1143,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
for (i = 0; i < 4; i++)
prev[0][i] = (int16_t)bytestream_get_be16(&src);
- if (samplecnt >= (samples_end - samples) / (st + 1)) {
- av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
- return -1;
- }
-
for (ch = 0; ch <= st; ch++) {
samples = (unsigned short *) data + ch;
/* Read in every sample for this channel. */
- for (i = 0; i < samplecnt / 14; i++) {
+ for (i = 0; i < nb_samples / 14; i++) {
int index = (*src >> 4) & 7;
unsigned int exp = 28 - (*src++ & 15);
int factor1 = table[ch][index * 2];
@@ -1056,17 +1171,13 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
}
}
}
-
- /* In the previous loop, in case stereo is used, samples is
- increased exactly one time too often. */
- samples -= st;
break;
}
default:
return -1;
}
- *data_size = (uint8_t *)samples - (uint8_t *)data;
+ *data_size = out_size;
return src - buf;
}
More information about the ffmpeg-cvslog
mailing list