falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
Browse Source

Merge commit 'cbcd497f384f0f8ef3f76f85b29b644b900d6b9f' 

* commit 'cbcd497f384f0f8ef3f76f85b29b644b900d6b9f':
  adxdec: use planar sample format
  adpcmdec: use planar sample format for adpcm_thp
  adpcmdec: use planar sample format for adpcm_ea_xas
  adpcmdec: use planar sample format for adpcm_ea_r1/r2/r3
  adpcmdec: use planar sample format for adpcm_xa
  adpcmdec: use planar sample format for adpcm_ima_ws for vqa version 3
  adpcmdec: use planar sample format for adpcm_4xm
  adpcmdec: use planar sample format for adpcm_ima_wav
  adpcmdec: use planar sample format for adpcm_ima_qt
  pcmdec: use planar sample format for pcm_lxf
  mace: use planar sample format
  atrac1: use planar sample format
  build: non-x86: Only compile mpegvideo optimizations when necessary
  rtpdec_mpeg4: au_headers is a single array, simple av_free is enough
  avcodec: free extended_data instead address of it
  fate: Add tests of the ff_make_absolute_url function
  url: Handle relative urls starting with two slashes
  url: Handle relative urls being just a new query string
  url: Don't treat slashes in query parameters as directory separators

Conflicts:
	libavcodec/adxdec.c
	libavcodec/mips/Makefile
	libavcodec/pcm.c
	libavcodec/utils.c
	libavformat/Makefile
	libavformat/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
tags/n1.1
Michael Niedermayer 12 years ago
parent
b4e516e30e cbcd497f38
commit
eadba3e94d
18 changed files with 239 additions and 159 deletions
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. +98
    -74
      libavcodec/adpcm.c
  2. +13
    -8
      libavcodec/adxdec.c
  3. +2
    -1
      libavcodec/alpha/Makefile
  4. +5
    -4
      libavcodec/arm/Makefile
  5. +4
    -25
      libavcodec/atrac1.c
  6. +2
    -1
      libavcodec/bfin/Makefile
  7. +15
    -17
      libavcodec/mace.c
  8. +2
    -1
      libavcodec/mips/Makefile
  9. +13
    -14
      libavcodec/pcm.c
  10. +1
    -1
      libavcodec/ppc/Makefile
  11. +1
    -1
      libavcodec/utils.c
  12. +2
    -1
      libavformat/Makefile
  13. +1
    -9
      libavformat/rtpdec_mpeg4.c
  14. +58
    -0
      libavformat/url-test.c
  15. +2
    -2
      libavformat/utils.c
  16. +2
    -0
      tests/Makefile
  17. +5
    -0
      tests/fate/libavformat.mak
  18. +13
    -0
      tests/ref/fate/url

+ 98
- 74
libavcodec/adpcm.c View File

@@ -134,7 +134,26 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
default:
break;
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16;

switch(avctx->codec->id) {
case AV_CODEC_ID_ADPCM_IMA_QT:
case AV_CODEC_ID_ADPCM_IMA_WAV:
case AV_CODEC_ID_ADPCM_4XM:
case AV_CODEC_ID_ADPCM_XA:
case AV_CODEC_ID_ADPCM_EA_R1:
case AV_CODEC_ID_ADPCM_EA_R2:
case AV_CODEC_ID_ADPCM_EA_R3:
case AV_CODEC_ID_ADPCM_EA_XAS:
case AV_CODEC_ID_ADPCM_THP:
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
break;
case AV_CODEC_ID_ADPCM_IMA_WS:
avctx->sample_fmt = c->vqa_version == 3 ? AV_SAMPLE_FMT_S16P :
AV_SAMPLE_FMT_S16;
break;
default:
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
}

avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
@@ -264,17 +283,22 @@ static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned c
return c->predictor;
}

static int xa_decode(AVCodecContext *avctx,
short *out, const unsigned char *in,
ADPCMChannelStatus *left, ADPCMChannelStatus *right, int inc)
static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
const uint8_t *in, ADPCMChannelStatus *left,
ADPCMChannelStatus *right, int channels, int sample_offset)
{
int i, j;
int shift,filter,f0,f1;
int s_1,s_2;
int d,s,t;

for(i=0;i<4;i++) {
out0 += sample_offset;
if (channels == 1)
out1 = out0 + 28;
else
out1 += sample_offset;

for(i=0;i<4;i++) {
shift = 12 - (in[4+i*2] & 15);
filter = in[4+i*2] >> 4;
if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table)) {
@@ -294,16 +318,14 @@ static int xa_decode(AVCodecContext *avctx,
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
*out = s_1;
out += inc;
out0[j] = s_1;
}

if (inc==2) { /* stereo */
if (channels == 2) {
left->sample1 = s_1;
left->sample2 = s_2;
s_1 = right->sample1;
s_2 = right->sample2;
out = out + 1 - 28*2;
}

shift = 12 - (in[5+i*2] & 15);
@@ -323,18 +345,19 @@ static int xa_decode(AVCodecContext *avctx,
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
*out = s_1;
out += inc;
out1[j] = s_1;
}

if (inc==2) { /* stereo */
if (channels == 2) {
right->sample1 = s_1;
right->sample2 = s_2;
out -= 1;
} else {
left->sample1 = s_1;
left->sample2 = s_2;
}

out0 += 28 * (3 - channels);
out1 += 28 * (3 - channels);
}

return 0;
@@ -573,6 +596,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
ADPCMChannelStatus *cs;
int n, m, channel, i;
short *samples;
int16_t **samples_p;
int st; /* stereo */
int count1, count2;
int nb_samples, coded_samples, ret;
@@ -592,6 +616,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
return ret;
}
samples = (short *)c->frame.data[0];
samples_p = (int16_t **)c->frame.extended_data;

/* use coded_samples when applicable */
/* it is always <= nb_samples, so the output buffer will be large enough */
@@ -636,21 +661,19 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
return AVERROR_INVALIDDATA;
}

samples = (short *)c->frame.data[0] + channel;
samples = samples_p[channel];

for (m = 0; m < 32; m++) {
for (m = 0; m < 64; m += 2) {
int byte = bytestream2_get_byteu(&gb);
*samples = adpcm_ima_qt_expand_nibble(cs, byte & 0x0F, 3);
samples += avctx->channels;
*samples = adpcm_ima_qt_expand_nibble(cs, byte >> 4 , 3);
samples += avctx->channels;
samples[m ] = adpcm_ima_qt_expand_nibble(cs, byte & 0x0F, 3);
samples[m + 1] = adpcm_ima_qt_expand_nibble(cs, byte >> 4 , 3);
}
}
break;
case AV_CODEC_ID_ADPCM_IMA_WAV:
for(i=0; i<avctx->channels; i++){
cs = &(c->status[i]);
cs->predictor = *samples++ = sign_extend(bytestream2_get_le16u(&gb), 16);
cs->predictor = samples_p[i][0] = sign_extend(bytestream2_get_le16u(&gb), 16);

cs->step_index = sign_extend(bytestream2_get_le16u(&gb), 16);
if (cs->step_index > 88u){
@@ -660,19 +683,16 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
}
}

for (n = (nb_samples - 1) / 8; n > 0; n--) {
for (n = 0; n < (nb_samples - 1) / 8; n++) {
for (i = 0; i < avctx->channels; i++) {
cs = &c->status[i];
for (m = 0; m < 4; m++) {
samples = &samples_p[i][1 + n * 8];
for (m = 0; m < 8; m += 2) {
int v = bytestream2_get_byteu(&gb);
*samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 3);
samples += avctx->channels;
*samples = adpcm_ima_expand_nibble(cs, v >> 4 , 3);
samples += avctx->channels;
samples[m ] = adpcm_ima_expand_nibble(cs, v & 0x0F, 3);
samples[m + 1] = adpcm_ima_expand_nibble(cs, v >> 4 , 3);
}
samples -= 8 * avctx->channels - 1;
}
samples += 7 * avctx->channels;
}
break;
case AV_CODEC_ID_ADPCM_4XM:
@@ -689,14 +709,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
}

for (i = 0; i < avctx->channels; i++) {
samples = (short *)c->frame.data[0] + i;
samples = (int16_t *)c->frame.data[i];
cs = &c->status[i];
for (n = nb_samples >> 1; n > 0; n--) {
int v = bytestream2_get_byteu(&gb);
*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++ = adpcm_ima_expand_nibble(cs, v & 0x0F, 4);
*samples++ = adpcm_ima_expand_nibble(cs, v >> 4 , 4);
}
}
break;
@@ -858,14 +876,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
case AV_CODEC_ID_ADPCM_IMA_WS:
if (c->vqa_version == 3) {
for (channel = 0; channel < avctx->channels; channel++) {
int16_t *smp = samples + channel;
int16_t *smp = samples_p[channel];

for (n = nb_samples / 2; n > 0; n--) {
int v = bytestream2_get_byteu(&gb);
*smp = adpcm_ima_expand_nibble(&c->status[channel], v >> 4 , 3);
smp += avctx->channels;
*smp = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
smp += avctx->channels;
*smp++ = adpcm_ima_expand_nibble(&c->status[channel], v >> 4 , 3);
*smp++ = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
}
}
} else {
@@ -881,14 +897,21 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
bytestream2_seek(&gb, 0, SEEK_END);
break;
case AV_CODEC_ID_ADPCM_XA:
{
int16_t *out0 = samples_p[0];
int16_t *out1 = samples_p[1];
int samples_per_block = 28 * (3 - avctx->channels) * 4;
int sample_offset = 0;
while (bytestream2_get_bytes_left(&gb) >= 128) {
if ((ret = xa_decode(avctx, samples, buf + bytestream2_tell(&gb), &c->status[0],
&c->status[1], avctx->channels)) < 0)
if ((ret = xa_decode(avctx, out0, out1, buf + bytestream2_tell(&gb),
&c->status[0], &c->status[1],
avctx->channels, sample_offset)) < 0)
return ret;
bytestream2_skipu(&gb, 128);
samples += 28 * 8;
sample_offset += samples_per_block;
}
break;
}
case AV_CODEC_ID_ADPCM_IMA_EA_EACS:
for (i=0; i<=st; i++) {
c->status[i].step_index = bytestream2_get_le32u(&gb);
@@ -1022,7 +1045,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,

for (channel=0; channel<avctx->channels; channel++) {
bytestream2_seek(&gb, offsets[channel], SEEK_SET);
samplesC = samples + channel;
samplesC = samples_p[channel];

if (avctx->codec->id == AV_CODEC_ID_ADPCM_EA_R1) {
current_sample = sign_extend(bytestream2_get_le16(&gb), 16);
@@ -1038,10 +1061,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
current_sample = sign_extend(bytestream2_get_be16(&gb), 16);
previous_sample = sign_extend(bytestream2_get_be16(&gb), 16);

for (count2=0; count2<28; count2++) {
*samplesC = sign_extend(bytestream2_get_be16(&gb), 16);
samplesC += avctx->channels;
}
for (count2=0; count2<28; count2++)
*samplesC++ = sign_extend(bytestream2_get_be16(&gb), 16);
} else {
coeff1 = ea_adpcm_table[ byte >> 4 ];
coeff2 = ea_adpcm_table[(byte >> 4) + 4];
@@ -1061,8 +1082,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,

previous_sample = current_sample;
current_sample = next_sample;
*samplesC = current_sample;
samplesC += avctx->channels;
*samplesC++ = current_sample;
}
}
}
@@ -1086,8 +1106,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
case AV_CODEC_ID_ADPCM_EA_XAS:
for (channel=0; channel<avctx->channels; channel++) {
int coeff[2][4], shift[4];
short *s2, *s = &samples[channel];
for (n=0; n<4; n++, s+=32*avctx->channels) {
int16_t *s = samples_p[channel];
for (n = 0; n < 4; n++, s += 32) {
int val = sign_extend(bytestream2_get_le16u(&gb), 16);
for (i=0; i<2; i++)
coeff[i][n] = ea_adpcm_table[(val&0x0F)+4*i];
@@ -1095,19 +1115,22 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,

val = sign_extend(bytestream2_get_le16u(&gb), 16);
shift[n] = 20 - (val & 0x0F);
s[avctx->channels] = val & ~0x0F;
s[1] = val & ~0x0F;
}

for (m=2; m<32; m+=2) {
s = &samples[m*avctx->channels + channel];
for (n=0; n<4; n++, s+=32*avctx->channels) {
s = &samples_p[channel][m];
for (n = 0; n < 4; n++, s += 32) {
int level, pred;
int byte = bytestream2_get_byteu(&gb);
for (s2=s, i=0; i<8; i+=4, s2+=avctx->channels) {
int level = sign_extend(byte >> (4 - i), 4) << shift[n];
int pred = s2[-1*avctx->channels] * coeff[0][n]
+ s2[-2*avctx->channels] * coeff[1][n];
s2[0] = av_clip_int16((level + pred + 0x80) >> 8);
}

level = sign_extend(byte >> 4, 4) << shift[n];
pred = s[-1] * coeff[0][n] + s[-2] * coeff[1][n];
s[0] = av_clip_int16((level + pred + 0x80) >> 8);

level = sign_extend(byte, 4) << shift[n];
pred = s[0] * coeff[0][n] + s[-1] * coeff[1][n];
s[1] = av_clip_int16((level + pred + 0x80) >> 8);
}
}
}
@@ -1221,7 +1244,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
prev[i][n] = sign_extend(bytestream2_get_be16u(&gb), 16);

for (ch = 0; ch <= st; ch++) {
samples = (short *)c->frame.data[0] + ch;
samples = samples_p[ch];

/* Read in every sample for this channel. */
for (i = 0; i < nb_samples / 14; i++) {
@@ -1247,10 +1270,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
*samples = av_clip_int16(sampledat);
prev[ch][1] = prev[ch][0];
prev[ch][0] = *samples++;

/* In case of stereo, skip one sample, this sample
is for the other channel. */
samples += st;
}
}
}
@@ -1270,6 +1289,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,

static const enum AVSampleFormat sample_fmts_s16[] = { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE };
static const enum AVSampleFormat sample_fmts_s16p[] = { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE };
static const enum AVSampleFormat sample_fmts_both[] = { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE };

#define ADPCM_DECODER(id_, sample_fmts_, name_, long_name_) \
AVCodec ff_ ## name_ ## _decoder = { \
@@ -1285,14 +1309,14 @@ AVCodec ff_ ## name_ ## _decoder = { \
}

/* Note: Do not forget to add new entries to the Makefile as well. */
ADPCM_DECODER(AV_CODEC_ID_ADPCM_4XM, sample_fmts_s16, adpcm_4xm, "ADPCM 4X Movie");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_4XM, sample_fmts_s16p, adpcm_4xm, "ADPCM 4X Movie");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_CT, sample_fmts_s16, adpcm_ct, "ADPCM Creative Technology");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA, sample_fmts_s16, adpcm_ea, "ADPCM Electronic Arts");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_MAXIS_XA, sample_fmts_s16, adpcm_ea_maxis_xa, "ADPCM Electronic Arts Maxis CDROM XA");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R1, sample_fmts_s16, adpcm_ea_r1, "ADPCM Electronic Arts R1");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R2, sample_fmts_s16, adpcm_ea_r2, "ADPCM Electronic Arts R2");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R3, sample_fmts_s16, adpcm_ea_r3, "ADPCM Electronic Arts R3");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_XAS, sample_fmts_s16, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R1, sample_fmts_s16p, adpcm_ea_r1, "ADPCM Electronic Arts R1");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R2, sample_fmts_s16p, adpcm_ea_r2, "ADPCM Electronic Arts R2");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R3, sample_fmts_s16p, adpcm_ea_r3, "ADPCM Electronic Arts R3");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_XAS, sample_fmts_s16p, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_AMV, sample_fmts_s16, adpcm_ima_amv, "ADPCM IMA AMV");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_APC, sample_fmts_s16, adpcm_ima_apc, "ADPCM IMA CRYO APC");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_DK3, sample_fmts_s16, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
@@ -1300,15 +1324,15 @@ ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_DK4, sample_fmts_s16, adpcm_ima_dk4,
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_EACS, sample_fmts_s16, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_SEAD, sample_fmts_s16, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_ISS, sample_fmts_s16, adpcm_ima_iss, "ADPCM IMA Funcom ISS");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_QT, sample_fmts_s16, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_QT, sample_fmts_s16p, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_SMJPEG, sample_fmts_s16, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WAV, sample_fmts_s16, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WS, sample_fmts_s16, adpcm_ima_ws, "ADPCM IMA Westwood");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WAV, sample_fmts_s16p, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WS, sample_fmts_both, adpcm_ima_ws, "ADPCM IMA Westwood");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_MS, sample_fmts_s16, adpcm_ms, "ADPCM Microsoft");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_2, sample_fmts_s16, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_3, sample_fmts_s16, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_4, sample_fmts_s16, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SWF, sample_fmts_s16, adpcm_swf, "ADPCM Shockwave Flash");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_THP, sample_fmts_s16, adpcm_thp, "ADPCM Nintendo Gamecube THP");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_XA, sample_fmts_s16, adpcm_xa, "ADPCM CDROM XA");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_THP, sample_fmts_s16p, adpcm_thp, "ADPCM Nintendo Gamecube THP");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_XA, sample_fmts_s16p, adpcm_xa, "ADPCM CDROM XA");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_YAMAHA, sample_fmts_s16, adpcm_yamaha, "ADPCM Yamaha");

+ 13
- 8
libavcodec/adxdec.c View File

@@ -49,7 +49,7 @@ static av_cold int adx_decode_init(AVCodecContext *avctx)
c->header_parsed = 1;
}

avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;

avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
@@ -64,7 +64,8 @@ static av_cold int adx_decode_init(AVCodecContext *avctx)
* 2nd-order LPC filter applied to it to form the output signal for a single
* channel.
*/
static int adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch)
static int adx_decode(ADXContext *c, int16_t *out, int offset,
const uint8_t *in, int ch)
{
ADXChannelState *prev = &c->prev[ch];
GetBitContext gb;
@@ -77,6 +78,7 @@ static int adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch)
return -1;

init_get_bits(&gb, in + 2, (BLOCK_SIZE - 2) * 8);
out += offset;
s1 = prev->s1;
s2 = prev->s2;
for (i = 0; i < BLOCK_SAMPLES; i++) {
@@ -84,8 +86,7 @@ static int adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch)
s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
s2 = s1;
s1 = av_clip_int16(s0);
*out = s1;
out += c->channels;
*out++ = s1;
}
prev->s1 = s1;
prev->s2 = s2;
@@ -98,7 +99,8 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
{
int buf_size = avpkt->size;
ADXContext *c = avctx->priv_data;
int16_t *samples;
int16_t **samples;
int samples_offset;
const uint8_t *buf = avpkt->data;
const uint8_t *buf_end = buf + avpkt->size;
int num_blocks, ch, ret;
@@ -145,11 +147,12 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)c->frame.data[0];
samples = (int16_t **)c->frame.extended_data;
samples_offset = 0;

while (num_blocks--) {
for (ch = 0; ch < c->channels; ch++) {
if (buf_end - buf < BLOCK_SIZE || adx_decode(c, samples + ch, buf, ch)) {
if (buf_end - buf < BLOCK_SIZE || adx_decode(c, samples[ch], samples_offset, buf, ch)) {
c->eof = 1;
buf = avpkt->data + avpkt->size;
break;
@@ -157,7 +160,7 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
buf_size -= BLOCK_SIZE;
buf += BLOCK_SIZE;
}
samples += BLOCK_SAMPLES * c->channels;
samples_offset += BLOCK_SAMPLES;
}

*got_frame_ptr = 1;
@@ -183,4 +186,6 @@ AVCodec ff_adpcm_adx_decoder = {
.flush = adx_decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};

+ 2
- 1
libavcodec/alpha/Makefile View File

@@ -2,5 +2,6 @@ OBJS += alpha/dsputil_alpha.o \
alpha/dsputil_alpha_asm.o \
alpha/motion_est_alpha.o \
alpha/motion_est_mvi_asm.o \
alpha/mpegvideo_alpha.o \
alpha/simple_idct_alpha.o \

OBJS-$(CONFIG_MPEGVIDEO) += alpha/mpegvideo_alpha.o

+ 5
- 4
libavcodec/arm/Makefile View File

@@ -14,6 +14,7 @@ OBJS-$(CONFIG_FLAC_DECODER) += arm/flacdsp_init_arm.o \
OBJS-$(CONFIG_MPEGAUDIODSP) += arm/mpegaudiodsp_init_arm.o
ARMV6-OBJS-$(CONFIG_MPEGAUDIODSP) += arm/mpegaudiodsp_fixed_armv6.o

OBJS-$(CONFIG_MPEGVIDEO) += arm/mpegvideo_arm.o
OBJS-$(CONFIG_VP3DSP) += arm/vp3dsp_init_arm.o
OBJS-$(CONFIG_VP5_DECODER) += arm/vp56dsp_init_arm.o
OBJS-$(CONFIG_VP6_DECODER) += arm/vp56dsp_init_arm.o
@@ -31,12 +32,12 @@ OBJS += arm/dsputil_init_arm.o \
arm/fft_fixed_init_arm.o \
arm/fmtconvert_init_arm.o \
arm/jrevdct_arm.o \
arm/mpegvideo_arm.o \
arm/simple_idct_arm.o \

ARMV5TE-OBJS += arm/dsputil_init_armv5te.o \
arm/mpegvideo_armv5te.o \
ARMV5TE-OBJS-$(CONFIG_MPEGVIDEO) += arm/mpegvideo_armv5te.o \
arm/mpegvideo_armv5te_s.o \

ARMV5TE-OBJS += arm/dsputil_init_armv5te.o \
arm/simple_idct_armv5te.o \

ARMV6-OBJS += arm/dsputil_init_armv6.o \
@@ -70,6 +71,7 @@ NEON-OBJS-$(CONFIG_AAC_DECODER) += arm/sbrdsp_neon.o \
NEON-OBJS-$(CONFIG_DCA_DECODER) += arm/dcadsp_neon.o \
arm/synth_filter_neon.o \

NEON-OBJS-$(CONFIG_MPEGVIDEO) += arm/mpegvideo_neon.o
NEON-OBJS-$(CONFIG_RV30_DECODER) += arm/rv34dsp_init_neon.o \
arm/rv34dsp_neon.o \

@@ -92,5 +94,4 @@ NEON-OBJS += arm/dsputil_init_neon.o \
arm/dsputil_neon.o \
arm/fmtconvert_neon.o \
arm/int_neon.o \
arm/mpegvideo_neon.o \
arm/simple_idct_neon.o \

+ 4
- 25
libavcodec/atrac1.c View File

@@ -36,7 +36,6 @@
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
#include "sinewin.h"

#include "atrac.h"
@@ -80,11 +79,9 @@ typedef struct {
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
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 */
@@ -281,7 +278,6 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
AT1Ctx *q = avctx->priv_data;
int ch, ret;
GetBitContext gb;
float *samples;


if (buf_size < 212 * q->channels) {
@@ -295,7 +291,6 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float *)q->frame.data[0];

for (ch = 0; ch < q->channels; ch++) {
AT1SUCtx* su = &q->SUs[ch];
@@ -314,13 +309,7 @@ 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->channels == 1 ? samples : q->out_samples[ch]);
}

/* interleave */
if (q->channels == 2) {
q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
AT1_SU_SAMPLES, 2);
at1_subband_synthesis(q, su, (float *)q->frame.extended_data[ch]);
}

*got_frame_ptr = 1;
@@ -334,8 +323,6 @@ 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]);
@@ -349,7 +336,7 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
AT1Ctx *q = avctx->priv_data;
int ret;

avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
@@ -358,15 +345,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
}
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 */
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))) ||
@@ -381,7 +359,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
ff_atrac_generate_tables();

ff_dsputil_init(&q->dsp, avctx);
ff_fmt_convert_init(&q->fmt_conv, avctx);

q->bands[0] = q->low;
q->bands[1] = q->mid;
@@ -410,4 +387,6 @@ AVCodec ff_atrac1_decoder = {
.decode = atrac1_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};

+ 2
- 1
libavcodec/bfin/Makefile View File

@@ -1,7 +1,8 @@
OBJS += bfin/dsputil_bfin.o \
bfin/fdct_bfin.o \
bfin/idct_bfin.o \
bfin/mpegvideo_bfin.o \
bfin/pixels_bfin.o \
bfin/vp3_bfin.o \
bfin/vp3_idct_bfin.o \

OBJS-$(CONFIG_MPEGVIDEOENC) += bfin/mpegvideo_bfin.o

+ 15
- 17
libavcodec/mace.c View File

@@ -187,9 +187,7 @@ static int16_t read_table(ChannelData *chd, uint8_t val, int tab_idx)
return current;
}

static void chomp3(ChannelData *chd, int16_t *output, uint8_t val,
int tab_idx,
uint32_t numChannels)
static void chomp3(ChannelData *chd, int16_t *output, uint8_t val, int tab_idx)
{

int16_t current = read_table(chd, val, tab_idx);
@@ -200,9 +198,7 @@ static void chomp3(ChannelData *chd, int16_t *output, uint8_t val,
*output = QT_8S_2_16S(current);
}

static void chomp6(ChannelData *chd, int16_t *output, uint8_t val,
int tab_idx,
uint32_t numChannels)
static void chomp6(ChannelData *chd, int16_t *output, uint8_t val, int tab_idx)
{
int16_t current = read_table(chd, val, tab_idx);

@@ -222,8 +218,8 @@ static void chomp6(ChannelData *chd, int16_t *output, uint8_t val,

output[0] = QT_8S_2_16S(chd->previous + chd->prev2 -
((chd->prev2-current) >> 2));
output[numChannels] = QT_8S_2_16S(chd->previous + current +
((chd->prev2-current) >> 2));
output[1] = QT_8S_2_16S(chd->previous + current +
((chd->prev2-current) >> 2));
chd->prev2 = chd->previous;
chd->previous = current;
}
@@ -234,7 +230,7 @@ static av_cold int mace_decode_init(AVCodecContext * avctx)

if (avctx->channels > 2 || avctx->channels <= 0)
return -1;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;

avcodec_get_frame_defaults(&ctx->frame);
avctx->coded_frame = &ctx->frame;
@@ -247,7 +243,7 @@ static int mace_decode_frame(AVCodecContext *avctx, void *data,
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *samples;
int16_t **samples;
MACEContext *ctx = avctx->priv_data;
int i, j, k, l, ret;
int is_mace3 = (avctx->codec_id == AV_CODEC_ID_MACE3);
@@ -258,10 +254,10 @@ static int mace_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)ctx->frame.data[0];
samples = (int16_t **)ctx->frame.extended_data;

for(i = 0; i < avctx->channels; i++) {
int16_t *output = samples + i;
int16_t *output = samples[i];

for (j=0; j < buf_size / (avctx->channels << is_mace3); j++)
for (k=0; k < (1 << is_mace3); k++) {
@@ -273,13 +269,11 @@ static int mace_decode_frame(AVCodecContext *avctx, void *data,

for (l=0; l < 3; l++) {
if (is_mace3)
chomp3(&ctx->chd[i], output, val[1][l], l,
avctx->channels);
chomp3(&ctx->chd[i], output, val[1][l], l);
else
chomp6(&ctx->chd[i], output, val[0][l], l,
avctx->channels);
chomp6(&ctx->chd[i], output, val[0][l], l);

output += avctx->channels << (1-is_mace3);
output += 1 << (1-is_mace3);
}
}
}
@@ -299,6 +293,8 @@ AVCodec ff_mace3_decoder = {
.decode = mace_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MACE (Macintosh Audio Compression/Expansion) 3:1"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};

AVCodec ff_mace6_decoder = {
@@ -310,4 +306,6 @@ AVCodec ff_mace6_decoder = {
.decode = mace_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MACE (Macintosh Audio Compression/Expansion) 6:1"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};

+ 2
- 1
libavcodec/mips/Makefile View File

@@ -1,6 +1,7 @@
MMI-OBJS += mips/dsputil_mmi.o \
mips/idct_mmi.o \
mips/mpegvideo_mmi.o

MMI-OBJS-$(CONFIG_MPEGVIDEO) += mips/mpegvideo_mmi.o

MIPSFPU-OBJS-$(CONFIG_AMRNB_DECODER) += mips/acelp_filters_mips.o \
mips/celp_filters_mips.o \


+ 13
- 14
libavcodec/pcm.c View File

@@ -446,22 +446,21 @@ static int pcm_decode_frame(AVCodecContext *avctx, void *data,
{
int i;
n /= avctx->channels;
//unpack
for (c = 0; c < avctx->channels; c++) {
dst_int32_t = (int32_t *)s->frame.data[c];
dst_int32_t = (int32_t *)s->frame.extended_data[c];
for (i = 0; i < n; i++) {
//extract low 20 bits and expand to 32 bits
*dst_int32_t++ = (src[2] << 28) |
(src[1] << 20) |
(src[0] << 12) |
((src[2] & 0xF) << 8) |
src[1];
//extract high 20 bits and expand to 32 bits
*dst_int32_t++ = (src[4] << 24) |
(src[3] << 16) |
((src[2] & 0xF0) << 8) |
(src[4] << 4) |
(src[3] >> 4);
// extract low 20 bits and expand to 32 bits
*dst_int32_t++ = (src[2] << 28) |
(src[1] << 20) |
(src[0] << 12) |
((src[2] & 0x0F) << 8) |
src[1];
// extract high 20 bits and expand to 32 bits
*dst_int32_t++ = (src[4] << 24) |
(src[3] << 16) |
((src[2] & 0xF0) << 8) |
(src[4] << 4) |
(src[3] >> 4);
src += 5;
}
}


+ 1
- 1
libavcodec/ppc/Makefile View File

@@ -7,6 +7,7 @@ ALTIVEC-OBJS-$(CONFIG_FFT) += ppc/fft_altivec.o \
$(FFT-OBJS-yes)
ALTIVEC-OBJS-$(CONFIG_H264DSP) += ppc/h264_altivec.o
ALTIVEC-OBJS-$(CONFIG_MPEGAUDIODSP) += ppc/mpegaudiodec_altivec.o
ALTIVEC-OBJS-$(CONFIG_MPEGVIDEO) += ppc/mpegvideo_altivec.o
ALTIVEC-OBJS-$(CONFIG_VC1_DECODER) += ppc/vc1dsp_altivec.o
ALTIVEC-OBJS-$(CONFIG_VP8_DECODER) += ppc/vp8dsp_altivec.o

@@ -17,4 +18,3 @@ ALTIVEC-OBJS += ppc/dsputil_altivec.o \
ppc/gmc_altivec.o \
ppc/idct_altivec.o \
ppc/int_altivec.o \
ppc/mpegvideo_altivec.o \

+ 1
- 1
libavcodec/utils.c View File

@@ -393,7 +393,7 @@ static int audio_get_buffer(AVCodecContext *avctx, AVFrame *frame)
if (buf->extended_data[0] && buf_size > buf->audio_data_size) {
av_free(buf->extended_data[0]);
if (buf->extended_data != buf->data)
av_freep(&buf->extended_data);
av_free(buf->extended_data);
buf->extended_data = NULL;
buf->data[0] = NULL;
}


+ 2
- 1
libavformat/Makefile View File

@@ -412,7 +412,8 @@ OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o

SKIPHEADERS-$(CONFIG_FFRTMPCRYPT_PROTOCOL) += rtmpdh.h
SKIPHEADERS-$(CONFIG_NETWORK) += network.h rtsp.h
TESTPROGS = seek
TESTPROGS = seek \
url \

TOOLS = aviocat \
ismindex \


+ 1
- 9
libavformat/rtpdec_mpeg4.c View File

@@ -93,15 +93,7 @@ static PayloadContext *new_context(void)

static void free_context(PayloadContext * data)
{
int i;
for (i = 0; i < data->nb_au_headers; i++) {
/* according to rtp_parse_mp4_au, we treat multiple
* au headers as one, so nb_au_headers is always 1.
* loop anyway in case this changes.
* (note: changes done carelessly might lead to a double free)
*/
av_free(&data->au_headers[i]);
}
av_free(data->au_headers);
av_free(data->mode);
av_free(data);
}


+ 58
- 0
libavformat/url-test.c View File

@@ -0,0 +1,58 @@
/*
* Copyright (c) 2012 Martin Storsjo
*
* 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 2.1 of the License, or (at your option) any later version.
*
* 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#include "internal.h"

#undef printf
#undef exit

static void test(const char *base, const char *rel)
{
char buf[200], buf2[200];
ff_make_absolute_url(buf, sizeof(buf), base, rel);
printf("%s\n", buf);
if (base) {
/* Test in-buffer replacement */
snprintf(buf2, sizeof(buf2), "%s", base);
ff_make_absolute_url(buf2, sizeof(buf2), buf2, rel);
if (strcmp(buf, buf2)) {
printf("In-place handling of %s + %s failed\n", base, rel);
exit(1);
}
}
}

int main(void)
{
test(NULL, "baz");
test("/foo/bar", "baz");
test("/foo/bar", "../baz");
test("/foo/bar", "/baz");
test("http://server/foo/", "baz");
test("http://server/foo/bar", "baz");
test("http://server/foo/", "../baz");
test("http://server/foo/bar/123", "../../baz");
test("http://server/foo/bar/123", "/baz");
test("http://server/foo/bar/123", "https://other/url");
test("http://server/foo/bar?param=value/with/slashes", "/baz");
test("http://server/foo/bar?param&otherparam", "?someparam");
test("http://server/foo/bar", "//other/url");
return 0;
}

+ 2
- 2
libavformat/utils.c View File

@@ -3824,9 +3824,9 @@ void ff_make_absolute_url(char *buf, int size, const char *base,
sep = strstr(buf, "://");
if (sep) {
/* Take scheme from base url */
if (rel[1] == '/')
if (rel[1] == '/') {
sep[1] = '\0';
else {
} else {
/* Take scheme and host from base url */
sep += 3;
sep = strchr(sep, '/');


+ 2
- 0
tests/Makefile View File

@@ -69,6 +69,7 @@ include $(SRC_PATH)/tests/fate/h264.mak
include $(SRC_PATH)/tests/fate/image.mak
include $(SRC_PATH)/tests/fate/indeo.mak
include $(SRC_PATH)/tests/fate/libavcodec.mak
include $(SRC_PATH)/tests/fate/libavformat.mak
include $(SRC_PATH)/tests/fate/libavutil.mak
include $(SRC_PATH)/tests/fate/mapchan.mak
include $(SRC_PATH)/tests/fate/lossless-audio.mak
@@ -108,6 +109,7 @@ FATE-$(CONFIG_FFMPEG) += $(FATE_FFMPEG)
FATE-$(CONFIG_FFPROBE) += $(FATE_FFPROBE)

FATE-$(CONFIG_AVCODEC) += $(FATE_LIBAVCODEC)
FATE-$(CONFIG_AVFORMAT) += $(FATE_LIBAVFORMAT)

FATE_EXTERN-$(CONFIG_FFMPEG) += $(FATE_SAMPLES_AVCONV) $(FATE_SAMPLES_FFMPEG)
FATE_EXTERN += $(FATE_EXTERN-yes)


+ 5
- 0
tests/fate/libavformat.mak View File

@@ -0,0 +1,5 @@
FATE_LIBAVFORMAT += fate-url
fate-url: libavformat/url-test$(EXESUF)
fate-url: CMD = run libavformat/url-test

fate-libavformat: $(FATE_LIBAVFORMAT)

+ 13
- 0
tests/ref/fate/url View File

@@ -0,0 +1,13 @@
baz
/foo/baz
/baz
/baz
http://server/foo/baz
http://server/foo/baz
http://server/baz
http://server/baz
http://server/baz
https://other/url
http://server/baz
http://server/foo/bar?someparam
http://other/url

Write Preview
Loading…
Cancel
Save