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Merge remote-tracking branch 'qatar/master'

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* qatar/master: (35 commits)
  libopencore-amr: check output buffer size before decoding
  libopencore-amr: remove unneeded buf_size==0 check.
  libopencore-amr: remove unneeded frame_count field.
  aac_latm: remove unneeded check for zero-size packet.
  pcmdec: fix output buffer size check by calculating the actual output size prior to decoding.
  pcmdec: move codec-specific variable declarations to the corresponding codec blocks.
  pcmdec: return buf_size instead of src-buf.
  avcodec: remove the Zork PCM encoder.
  pcm_zork: use AV_SAMPLE_FMT_U8 instead of shifting all samples by 8.
  pcmenc: remove unneeded sample_fmt check.
  pcmdec: move number of channels check to pcm_decode_init()
  pcmdec: remove unnecessary check for sample_fmt change
  pcmdec: move DVD PCM bits_per_coded_sample check near to the code that sets the sample size.
  pcmdec: do not needlessly set *data_size to 0
  alacdec: remove unneeded NULL or zero-size packet checks.
  alacdec: simplify buffer allocation by using FF_ALLOC_OR_GOTO()
  alacdec: ask for a sample for unsupported sample depths.
  alacdec: cosmetics: use 'ch' instead of 'chan' to iterate channels
  alacdec: move some declarations to the top of the function
  alacdec: always use get_sbits_long() for uncompressed samples
  ...

Conflicts:
	libavcodec/pcm.c
	tests/ref/acodec/pcm

Merged-by: Michael Niedermayer <michaelni@gmx.at>
pull/2/head
Michael Niedermayer 13 years ago
parent 1c5670dbb2 4a6a29a7fb
commit 173715d291
12 changed files with 246 additions and 348 deletions
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  1. 2
      doc/general.texi
  2. 1
      libavcodec/Makefile
  3. 3
      libavcodec/aacdec.c
  4. 382
      libavcodec/alac.c
  5. 2
      libavcodec/allcodecs.c
  6. 29
      libavcodec/libopencore-amr.c
  7. 92
      libavcodec/pcm.c
  8. 1
      libavcodec/pthread.c
  9. 24
      libavcodec/x86/dsputil_mmx.c
  10. 1
      tests/codec-regression.sh
  11. 4
      tests/ref/acodec/pcm
  12. 53
      tests/ref/seek/pcm_zork_wav

2
doc/general.texi
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@ -683,7 +683,7 @@ following image formats are supported:
@item PCM unsigned 24-bit little-endian @tab X @tab X
@item PCM unsigned 32-bit big-endian @tab X @tab X
@item PCM unsigned 32-bit little-endian @tab X @tab X
@item PCM Zork @tab X @tab X
@item PCM Zork @tab @tab X
@item QCELP / PureVoice @tab @tab X
@item QDesign Music Codec 2 @tab @tab X
@tab There are still some distortions.

1
libavcodec/Makefile
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@ -508,7 +508,6 @@ OBJS-$(CONFIG_PCM_U32BE_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_U32LE_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_U32LE_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_ZORK_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_ZORK_ENCODER) += pcm.o
OBJS-$(CONFIG_ADPCM_4XM_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_ADX_DECODER) += adxdec.o

3
libavcodec/aacdec.c
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@ -2501,9 +2501,6 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
int muxlength, err;
GetBitContext gb;
if (avpkt->size == 0)
return 0;
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
// check for LOAS sync word

382
libavcodec/alac.c
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@ -72,7 +72,7 @@ typedef struct {
int32_t *outputsamples_buffer[MAX_CHANNELS];
int32_t *wasted_bits_buffer[MAX_CHANNELS];
int32_t *extra_bits_buffer[MAX_CHANNELS];
/* stuff from setinfo */
uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
@ -82,58 +82,9 @@ typedef struct {
uint8_t setinfo_rice_kmodifier; /* 0x0e */
/* end setinfo stuff */
int wasted_bits;
int extra_bits; /**< number of extra bits beyond 16-bit */
} ALACContext;
static void allocate_buffers(ALACContext *alac)
{
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
alac->predicterror_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->outputsamples_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
}
}
static int alac_set_info(ALACContext *alac)
{
const unsigned char *ptr = alac->avctx->extradata;
ptr += 4; /* size */
ptr += 4; /* alac */
ptr += 4; /* 0 ? */
if(AV_RB32(ptr) >= UINT_MAX/4){
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
return -1;
}
/* buffer size / 2 ? */
alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
ptr++; /* ??? */
alac->setinfo_sample_size = *ptr++;
if (alac->setinfo_sample_size > 32) {
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
return -1;
}
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
ptr++; /* channels? */
bytestream_get_be16(&ptr); /* ??? */
bytestream_get_be32(&ptr); /* max coded frame size */
bytestream_get_be32(&ptr); /* bitrate ? */
bytestream_get_be32(&ptr); /* samplerate */
allocate_buffers(alac);
return 0;
}
static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
/* read x - number of 1s before 0 represent the rice */
int x = get_unary_0_9(gb);
@ -347,93 +298,56 @@ static void predictor_decompress_fir_adapt(int32_t *error_buffer,
}
}
static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
int16_t *buffer_out,
int numchannels, int numsamples,
uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
static void decorrelate_stereo(int32_t *buffer[MAX_CHANNELS],
int numsamples, uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
{
int i;
if (numsamples <= 0)
return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t a, b;
a = buffer[0][i];
b = buffer[1][i];
for (i = 0; i < numsamples; i++) {
int32_t a, b;
a -= (b * interlacing_leftweight) >> interlacing_shift;
b += a;
a = buffer[0][i];
b = buffer[1][i];
buffer_out[i*numchannels] = b;
buffer_out[i*numchannels + 1] = a;
}
a -= (b * interlacing_leftweight) >> interlacing_shift;
b += a;
return;
buffer[0][i] = b;
buffer[1][i] = a;
}
}
/* otherwise basic interlacing took place */
for (i = 0; i < numsamples; i++) {
int16_t left, right;
left = buffer[0][i];
right = buffer[1][i];
static void append_extra_bits(int32_t *buffer[MAX_CHANNELS],
int32_t *extra_bits_buffer[MAX_CHANNELS],
int extra_bits, int numchannels, int numsamples)
{
int i, ch;
buffer_out[i*numchannels] = left;
buffer_out[i*numchannels + 1] = right;
}
for (ch = 0; ch < numchannels; ch++)
for (i = 0; i < numsamples; i++)
buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
}
static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
int32_t *buffer_out,
int32_t *wasted_bits_buffer[MAX_CHANNELS],
int wasted_bits,
int numchannels, int numsamples,
uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
static void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS],
int16_t *buffer_out, int numsamples)
{
int i;
if (numsamples <= 0)
return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t a, b;
a = buffer[0][i];
b = buffer[1][i];
a -= (b * interlacing_leftweight) >> interlacing_shift;
b += a;
if (wasted_bits) {
b = (b << wasted_bits) | wasted_bits_buffer[0][i];
a = (a << wasted_bits) | wasted_bits_buffer[1][i];
}
buffer_out[i * numchannels] = b << 8;
buffer_out[i * numchannels + 1] = a << 8;
}
} else {
for (i = 0; i < numsamples; i++) {
int32_t left, right;
left = buffer[0][i];
right = buffer[1][i];
for (i = 0; i < numsamples; i++) {
*buffer_out++ = buffer[0][i];
*buffer_out++ = buffer[1][i];
}
}
if (wasted_bits) {
left = (left << wasted_bits) | wasted_bits_buffer[0][i];
right = (right << wasted_bits) | wasted_bits_buffer[1][i];
}
static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
int32_t *buffer_out, int numsamples)
{
int i;
buffer_out[i * numchannels] = left << 8;
buffer_out[i * numchannels + 1] = right << 8;
}
for (i = 0; i < numsamples; i++) {
*buffer_out++ = buffer[0][i] << 8;
*buffer_out++ = buffer[1][i] << 8;
}
}
@ -452,18 +366,14 @@ static int alac_decode_frame(AVCodecContext *avctx,
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
/* short-circuit null buffers */
if (!inbuffer || !input_buffer_size)
return -1;
int i, ch;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
channels = get_bits(&alac->gb, 3) + 1;
if (channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
MAX_CHANNELS);
return -1;
if (channels != avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "frame header channel count mismatch\n");
return AVERROR_INVALIDDATA;
}
/* 2^result = something to do with output waiting.
@ -476,7 +386,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
/* the output sample size is stored soon */
hassize = get_bits1(&alac->gb);
alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
alac->extra_bits = get_bits(&alac->gb, 2) << 3;
/* whether the frame is compressed */
isnotcompressed = get_bits1(&alac->gb);
@ -491,17 +401,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
} else
outputsamples = alac->setinfo_max_samples_per_frame;
switch (alac->setinfo_sample_size) {
case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
alac->bytespersample = channels << 1;
break;
case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
alac->bytespersample = channels << 2;
break;
default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
alac->setinfo_sample_size);
return -1;
}
alac->bytespersample = channels * av_get_bytes_per_sample(avctx->sample_fmt);
if(outputsamples > *outputsize / alac->bytespersample){
av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
@ -509,7 +409,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
}
*outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1;
readsamplesize = alac->setinfo_sample_size - alac->extra_bits + channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
return -1;
@ -522,51 +422,49 @@ static int alac_decode_frame(AVCodecContext *avctx,
int prediction_type[MAX_CHANNELS];
int prediction_quantitization[MAX_CHANNELS];
int ricemodifier[MAX_CHANNELS];
int i, chan;
interlacing_shift = get_bits(&alac->gb, 8);
interlacing_leftweight = get_bits(&alac->gb, 8);
for (chan = 0; chan < channels; chan++) {
prediction_type[chan] = get_bits(&alac->gb, 4);
prediction_quantitization[chan] = get_bits(&alac->gb, 4);
for (ch = 0; ch < channels; ch++) {
prediction_type[ch] = get_bits(&alac->gb, 4);
prediction_quantitization[ch] = get_bits(&alac->gb, 4);
ricemodifier[chan] = get_bits(&alac->gb, 3);
predictor_coef_num[chan] = get_bits(&alac->gb, 5);
ricemodifier[ch] = get_bits(&alac->gb, 3);
predictor_coef_num[ch] = get_bits(&alac->gb, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num[chan]; i++)
predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
for (i = 0; i < predictor_coef_num[ch]; i++)
predictor_coef_table[ch][i] = (int16_t)get_bits(&alac->gb, 16);
}
if (alac->wasted_bits) {
int i, ch;
if (alac->extra_bits) {
for (i = 0; i < outputsamples; i++) {
for (ch = 0; ch < channels; ch++)
alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits);
alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
}
}
for (chan = 0; chan < channels; chan++) {
for (ch = 0; ch < channels; ch++) {
bastardized_rice_decompress(alac,
alac->predicterror_buffer[chan],
alac->predicterror_buffer[ch],
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
ricemodifier[ch] * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type[chan] == 0) {
if (prediction_type[ch] == 0) {
/* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
alac->outputsamples_buffer[chan],
predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
alac->outputsamples_buffer[ch],
outputsamples,
readsamplesize,
predictor_coef_table[chan],
predictor_coef_num[chan],
prediction_quantitization[chan]);
predictor_coef_table[ch],
predictor_coef_num[ch],
prediction_quantitization[ch]);
} else {
av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[ch]);
/* I think the only other prediction type (or perhaps this is
* just a boolean?) runs adaptive fir twice.. like:
* predictor_decompress_fir_adapt(predictor_error, tempout, ...)
@ -577,44 +475,35 @@ static int alac_decode_frame(AVCodecContext *avctx,
}
} else {
/* not compressed, easy case */
int i, chan;
if (alac->setinfo_sample_size <= 16) {
for (i = 0; i < outputsamples; i++)
for (chan = 0; chan < channels; chan++) {
int32_t audiobits;
audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
alac->outputsamples_buffer[chan][i] = audiobits;
}
} else {
for (i = 0; i < outputsamples; i++) {
for (chan = 0; chan < channels; chan++) {
alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb,
alac->setinfo_sample_size);
alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i],
alac->setinfo_sample_size);
}
for (i = 0; i < outputsamples; i++) {
for (ch = 0; ch < channels; ch++) {
alac->outputsamples_buffer[ch][i] = get_sbits_long(&alac->gb,
alac->setinfo_sample_size);
}
}
alac->wasted_bits = 0;
alac->extra_bits = 0;
interlacing_shift = 0;
interlacing_leftweight = 0;
}
if (get_bits(&alac->gb, 3) != 7)
av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
if (channels == 2 && interlacing_leftweight) {
decorrelate_stereo(alac->outputsamples_buffer, outputsamples,
interlacing_shift, interlacing_leftweight);
}
if (alac->extra_bits) {
append_extra_bits(alac->outputsamples_buffer, alac->extra_bits_buffer,
alac->extra_bits, alac->numchannels, outputsamples);
}
switch(alac->setinfo_sample_size) {
case 16:
if (channels == 2) {
reconstruct_stereo_16(alac->outputsamples_buffer,
(int16_t*)outbuffer,
alac->numchannels,
outputsamples,
interlacing_shift,
interlacing_leftweight);
interleave_stereo_16(alac->outputsamples_buffer, outbuffer,
outputsamples);
} else {
int i;
for (i = 0; i < outputsamples; i++) {
((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
}
@ -622,16 +511,9 @@ static int alac_decode_frame(AVCodecContext *avctx,
break;
case 24:
if (channels == 2) {
decorrelate_stereo_24(alac->outputsamples_buffer,
outbuffer,
alac->wasted_bits_buffer,
alac->wasted_bits,
alac->numchannels,
outputsamples,
interlacing_shift,
interlacing_leftweight);
interleave_stereo_24(alac->outputsamples_buffer, outbuffer,
outputsamples);
} else {
int i;
for (i = 0; i < outputsamples; i++)
((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
}
@ -644,11 +526,75 @@ static int alac_decode_frame(AVCodecContext *avctx,
return input_buffer_size;
}
static av_cold int alac_decode_close(AVCodecContext *avctx)
{
ALACContext *alac = avctx->priv_data;
int ch;
for (ch = 0; ch < alac->numchannels; ch++) {
av_freep(&alac->predicterror_buffer[ch]);
av_freep(&alac->outputsamples_buffer[ch]);
av_freep(&alac->extra_bits_buffer[ch]);
}
return 0;
}
static int allocate_buffers(ALACContext *alac)
{
int ch;
for (ch = 0; ch < alac->numchannels; ch++) {
int buf_size = alac->setinfo_max_samples_per_frame * sizeof(int32_t);
FF_ALLOC_OR_GOTO(alac->avctx, alac->predicterror_buffer[ch],
buf_size, buf_alloc_fail);
FF_ALLOC_OR_GOTO(alac->avctx, alac->outputsamples_buffer[ch],
buf_size, buf_alloc_fail);
FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch],
buf_size, buf_alloc_fail);
}
return 0;
buf_alloc_fail:
alac_decode_close(alac->avctx);
return AVERROR(ENOMEM);
}
static int alac_set_info(ALACContext *alac)
{
const unsigned char *ptr = alac->avctx->extradata;
ptr += 4; /* size */
ptr += 4; /* alac */
ptr += 4; /* 0 ? */
if(AV_RB32(ptr) >= UINT_MAX/4){
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
return -1;
}
/* buffer size / 2 ? */
alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
ptr++; /* ??? */
alac->setinfo_sample_size = *ptr++;
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
alac->numchannels = *ptr++;
bytestream_get_be16(&ptr); /* ??? */
bytestream_get_be32(&ptr); /* max coded frame size */
bytestream_get_be32(&ptr); /* bitrate ? */
bytestream_get_be32(&ptr); /* samplerate */
return 0;
}
static av_cold int alac_decode_init(AVCodecContext * avctx)
{
int ret;
ALACContext *alac = avctx->priv_data;
alac->avctx = avctx;
alac->numchannels = alac->avctx->channels;
/* initialize from the extradata */
if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
@ -661,18 +607,34 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
return -1;
}
return 0;
}
switch (alac->setinfo_sample_size) {
case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
break;
case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
break;
default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
alac->setinfo_sample_size);
return AVERROR_PATCHWELCOME;
}
static av_cold int alac_decode_close(AVCodecContext *avctx)
{
ALACContext *alac = avctx->priv_data;
if (alac->numchannels < 1) {
av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
alac->numchannels = avctx->channels;
} else {
if (alac->numchannels > MAX_CHANNELS)
alac->numchannels = avctx->channels;
else
avctx->channels = alac->numchannels;
}
if (avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
avctx->channels);
return AVERROR_PATCHWELCOME;
}
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
av_freep(&alac->predicterror_buffer[chan]);
av_freep(&alac->outputsamples_buffer[chan]);
av_freep(&alac->wasted_bits_buffer[chan]);
if ((ret = allocate_buffers(alac)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
return ret;
}
return 0;

2
libavcodec/allcodecs.c
Unescape View File

@ -329,7 +329,7 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (PCM_U24LE, pcm_u24le);
REGISTER_ENCDEC (PCM_U32BE, pcm_u32be);
REGISTER_ENCDEC (PCM_U32LE, pcm_u32le);
REGISTER_ENCDEC (PCM_ZORK , pcm_zork);
REGISTER_DECODER (PCM_ZORK , pcm_zork);
/* DPCM codecs */
REGISTER_DECODER (INTERPLAY_DPCM, interplay_dpcm);

29
libavcodec/libopencore-amr.c
Unescape View File

@ -79,7 +79,6 @@ static int get_bitrate_mode(int bitrate, void *log_ctx)
typedef struct AMRContext {
AVClass *av_class;
int frame_count;
void *dec_state;
void *enc_state;
int enc_bitrate;
@ -100,7 +99,6 @@ static av_cold int amr_nb_decode_init(AVCodecContext *avctx)
{
AMRContext *s = avctx->priv_data;
s->frame_count = 0;
s->dec_state = Decoder_Interface_init();
if (!s->dec_state) {
av_log(avctx, AV_LOG_ERROR, "Decoder_Interface_init error\n");
@ -133,10 +131,16 @@ static int amr_nb_decode_frame(AVCodecContext *avctx, void *data,
AMRContext *s = avctx->priv_data;
static const uint8_t block_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };
enum Mode dec_mode;
int packet_size;
int packet_size, out_size;
av_dlog(avctx, "amr_decode_frame buf=%p buf_size=%d frame_count=%d!!\n",
buf, buf_size, s->frame_count);
buf, buf_size, avctx->frame_number);
out_size = 160 * 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);
}
dec_mode = (buf[0] >> 3) & 0x000F;
packet_size = block_size[dec_mode] + 1;
@ -147,12 +151,11 @@ static int amr_nb_decode_frame(AVCodecContext *avctx, void *data,
return AVERROR_INVALIDDATA;
}
s->frame_count++;
av_dlog(avctx, "packet_size=%d buf= 0x%X %X %X %X\n",
packet_size, buf[0], buf[1], buf[2], buf[3]);
/* call decoder */
Decoder_Interface_Decode(s->dec_state, buf, data, 0);
*data_size = 160 * 2;
*data_size = out_size;
return packet_size;
}
@ -172,8 +175,6 @@ static av_cold int amr_nb_encode_init(AVCodecContext *avctx)
{
AMRContext *s = avctx->priv_data;
s->frame_count = 0;
if (avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
return AVERROR(ENOSYS);
@ -276,12 +277,14 @@ static int amr_wb_decode_frame(AVCodecContext *avctx, void *data,
int buf_size = avpkt->size;
AMRWBContext *s = avctx->priv_data;
int mode;
int packet_size;
int packet_size, out_size;
static const uint8_t block_size[16] = {18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1};
if (!buf_size)
/* nothing to do */
return 0;
out_size = 320 * 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);
}
mode = (buf[0] >> 3) & 0x000F;
packet_size = block_size[mode];
@ -293,7 +296,7 @@ static int amr_wb_decode_frame(AVCodecContext *avctx, void *data,
}
D_IF_decode(s->state, buf, data, _good_frame);
*data_size = 320 * 2;
*data_size = out_size;
return packet_size;
}

92
libavcodec/pcm.c
Unescape View File

@ -95,11 +95,6 @@ static int pcm_encode_frame(AVCodecContext *avctx,
samples = data;
dst = frame;
if (avctx->sample_fmt!=avctx->codec->sample_fmts[0]) {
av_log(avctx, AV_LOG_ERROR, "invalid sample_fmt\n");
return -1;
}
switch(avctx->codec->id) {
case CODEC_ID_PCM_U32LE:
ENCODE(uint32_t, le32, samples, dst, n, 0, 0x80000000)
@ -176,14 +171,6 @@ static int pcm_encode_frame(AVCodecContext *avctx,
memcpy(dst, samples, n*sample_size);
dst += n*sample_size;
break;
case CODEC_ID_PCM_ZORK:
for(;n>0;n--) {
v= *samples++ >> 8;
if(v<0) v = -v;
else v+= 128;
*dst++ = v;
}
break;
case CODEC_ID_PCM_ALAW:
for(;n>0;n--) {
v = *samples++;
@ -213,6 +200,11 @@ static av_cold int pcm_decode_init(AVCodecContext * avctx)
PCMDecode *s = avctx->priv_data;
int i;
if (avctx->channels <= 0 || avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n");
return AVERROR(EINVAL);
}
switch(avctx->codec->id) {
case CODEC_ID_PCM_ALAW:
for(i=0;i<256;i++)
@ -255,34 +247,29 @@ static int pcm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
const uint8_t *src = avpkt->data;
int buf_size = avpkt->size;
PCMDecode *s = avctx->priv_data;
int sample_size, c, n, i;
int sample_size, c, n, out_size;
uint8_t *samples;
const uint8_t *src, *src8, *src2[MAX_CHANNELS];
int32_t *dst_int32_t;
samples = data;
src = buf;
if (avctx->sample_fmt!=avctx->codec->sample_fmts[0]) {
av_log(avctx, AV_LOG_ERROR, "invalid sample_fmt\n");
return -1;
}
if(avctx->channels <= 0 || avctx->channels > MAX_CHANNELS){
av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n");
return -1;
}
sample_size = av_get_bits_per_sample(avctx->codec_id)/8;
/* av_get_bits_per_sample returns 0 for CODEC_ID_PCM_DVD */
if (CODEC_ID_PCM_DVD == avctx->codec_id)
if (CODEC_ID_PCM_DVD == avctx->codec_id) {
if (avctx->bits_per_coded_sample != 20 &&
avctx->bits_per_coded_sample != 24) {
av_log(avctx, AV_LOG_ERROR,
"PCM DVD unsupported sample depth %i\n",
avctx->bits_per_coded_sample);
return AVERROR(EINVAL);
}
/* 2 samples are interleaved per block in PCM_DVD */
sample_size = avctx->bits_per_coded_sample * 2 / 8;
else if (avctx->codec_id == CODEC_ID_PCM_LXF)
} else if (avctx->codec_id == CODEC_ID_PCM_LXF)
/* we process 40-bit blocks per channel for LXF */
sample_size = 5;
@ -301,11 +288,17 @@ static int pcm_decode_frame(AVCodecContext *avctx,
buf_size -= buf_size % n;
}
buf_size= FFMIN(buf_size, *data_size/2);
*data_size=0;
n = buf_size/sample_size;
out_size = n * av_get_bytes_per_sample(avctx->sample_fmt);
if (avctx->codec_id == CODEC_ID_PCM_DVD ||
avctx->codec_id == CODEC_ID_PCM_LXF)
out_size *= 2;
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
return AVERROR(EINVAL);
}
switch(avctx->codec->id) {
case CODEC_ID_PCM_U32LE:
DECODE(32, le32, src, samples, n, 0, 0x80000000)
@ -335,6 +328,8 @@ static int pcm_decode_frame(AVCodecContext *avctx,
}
break;
case CODEC_ID_PCM_S16LE_PLANAR:
{
const uint8_t *src2[MAX_CHANNELS];
n /= avctx->channels;
for(c=0;c<avctx->channels;c++)
src2[c] = &src[c*n*2];
@ -343,8 +338,8 @@ static int pcm_decode_frame(AVCodecContext *avctx,
AV_WN16A(samples, bytestream_get_le16(&src2[c]));
samples += 2;
}
src = src2[avctx->channels-1];
break;
}
case CODEC_ID_PCM_U16LE:
DECODE(16, le16, src, samples, n, 0, 0x8000)
break;
@ -389,16 +384,14 @@ static int pcm_decode_frame(AVCodecContext *avctx,
#endif /* HAVE_BIGENDIAN */
case CODEC_ID_PCM_U8:
memcpy(samples, src, n*sample_size);
src += n*sample_size;
samples += n * sample_size;
break;
case CODEC_ID_PCM_ZORK:
for(;n>0;n--) {
int x= *src++;
if(x&128) x-= 128;
else x = -x;
AV_WN16A(samples, x << 8);
samples += 2;
for (; n > 0; n--) {
int v = *src++;
if (v < 128)
v = 128 - v;
*samples++ = v;
}
break;
case CODEC_ID_PCM_ALAW:
@ -409,6 +402,8 @@ static int pcm_decode_frame(AVCodecContext *avctx,
}
break;
case CODEC_ID_PCM_DVD:
{
const uint8_t *src8;
dst_int32_t = data;
n /= avctx->channels;
switch (avctx->bits_per_coded_sample) {
@ -434,15 +429,14 @@ static int pcm_decode_frame(AVCodecContext *avctx,
src = src8;
}
break;
default:
av_log(avctx, AV_LOG_ERROR,
"PCM DVD unsupported sample depth %i\n",
avctx->bits_per_coded_sample);
return -1;
}
samples = (uint8_t *) dst_int32_t;
break;
}
case CODEC_ID_PCM_LXF:
{
int i;
const uint8_t *src8;
dst_int32_t = data;
n /= avctx->channels;
//unpack and de-planerize
@ -459,14 +453,14 @@ static int pcm_decode_frame(AVCodecContext *avctx,
((src8[2] & 0xF0) << 8) | (src8[4] << 4) | (src8[3] >> 4);
}
}
src += n * avctx->channels * 5;
samples = (uint8_t *) dst_int32_t;
break;
}
default:
return -1;
}
*data_size = samples - (uint8_t *)data;
return src - buf;
*data_size = out_size;
return buf_size;
}
#if CONFIG_ENCODERS
@ -529,4 +523,4 @@ PCM_CODEC (CODEC_ID_PCM_U24BE, AV_SAMPLE_FMT_S32, pcm_u24be, "PCM unsigned 24-b
PCM_CODEC (CODEC_ID_PCM_U24LE, AV_SAMPLE_FMT_S32, pcm_u24le, "PCM unsigned 24-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_U32BE, AV_SAMPLE_FMT_S32, pcm_u32be, "PCM unsigned 32-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_U32LE, AV_SAMPLE_FMT_S32, pcm_u32le, "PCM unsigned 32-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_ZORK, AV_SAMPLE_FMT_S16, pcm_zork, "PCM Zork");
PCM_DECODER(CODEC_ID_PCM_ZORK, AV_SAMPLE_FMT_U8, pcm_zork, "PCM Zork");

1
libavcodec/pthread.c
Unescape View File

@ -686,6 +686,7 @@ static void frame_thread_free(AVCodecContext *avctx, int thread_count)
av_freep(&fctx->threads);
pthread_mutex_destroy(&fctx->buffer_mutex);
av_freep(&avctx->thread_opaque);
avctx->has_b_frames -= avctx->thread_count - 1;
}
static int frame_thread_init(AVCodecContext *avctx)

24
libavcodec/x86/dsputil_mmx.c
Unescape View File

@ -2666,18 +2666,18 @@ void dsputil_init_mmx(DSPContext* c, AVCodecContext *avctx)
}
if(mm_flags & AV_CPU_FLAG_SSE2){
if (!high_bit_depth) {
H264_QPEL_FUNCS(0, 1, sse2);
H264_QPEL_FUNCS(0, 2, sse2);
H264_QPEL_FUNCS(0, 3, sse2);
H264_QPEL_FUNCS(1, 1, sse2);
H264_QPEL_FUNCS(1, 2, sse2);
H264_QPEL_FUNCS(1, 3, sse2);
H264_QPEL_FUNCS(2, 1, sse2);
H264_QPEL_FUNCS(2, 2, sse2);
H264_QPEL_FUNCS(2, 3, sse2);
H264_QPEL_FUNCS(3, 1, sse2);
H264_QPEL_FUNCS(3, 2, sse2);
H264_QPEL_FUNCS(3, 3, sse2);
H264_QPEL_FUNCS(0, 1, sse2);
H264_QPEL_FUNCS(0, 2, sse2);
H264_QPEL_FUNCS(0, 3, sse2);
H264_QPEL_FUNCS(1, 1, sse2);
H264_QPEL_FUNCS(1, 2, sse2);
H264_QPEL_FUNCS(1, 3, sse2);
H264_QPEL_FUNCS(2, 1, sse2);
H264_QPEL_FUNCS(2, 2, sse2);
H264_QPEL_FUNCS(2, 3, sse2);
H264_QPEL_FUNCS(3, 1, sse2);
H264_QPEL_FUNCS(3, 2, sse2);
H264_QPEL_FUNCS(3, 3, sse2);
}
#if HAVE_YASM
#define H264_QPEL_FUNCS_10(x, y, CPU)\

1
tests/codec-regression.sh
Unescape View File

@ -393,6 +393,5 @@ do_audio_enc_dec au flt pcm_f32be
do_audio_enc_dec wav flt pcm_f32le
do_audio_enc_dec au dbl pcm_f64be
do_audio_enc_dec wav dbl pcm_f64le
do_audio_enc_dec wav s16 pcm_zork
do_audio_enc_dec 302 s16 pcm_s24daud "-ac 6 -ar 96000"
fi

4
tests/ref/acodec/pcm
Unescape View File

@ -62,10 +62,6 @@ ba17c6d1a270e1333e981f239bf7eb45 *./tests/data/acodec/pcm_f64le.wav
4233680 ./tests/data/acodec/pcm_f64le.wav
64151e4bcc2b717aa5a8454d424d6a1f *./tests/data/pcm.acodec.out.wav
stddev: 0.00 PSNR:999.99 MAXDIFF: 0 bytes: 1058400/ 1058400
ebd38ed390ebdefe0bdf00d21bf12c6b *./tests/data/acodec/pcm_zork.wav
529258 ./tests/data/acodec/pcm_zork.wav
7b02646acdd063650bb3ebc444543ace *./tests/data/pcm.acodec.out.wav
stddev: 633.11 PSNR: 40.30 MAXDIFF:32768 bytes: 1058400/ 1058400
1b75d5198ae789ab3c48f7024e08f4a9 *./tests/data/acodec/pcm_s24daud.302
10368730 ./tests/data/acodec/pcm_s24daud.302
4708f86529c594e29404603c64bb208c *./tests/data/pcm.acodec.out.wav

53
tests/ref/seek/pcm_zork_wav
Unescape View File

@ -1,53 +0,0 @@
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
ret: 0 st:-1 flags:0 ts:-1.000000
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
ret: 0 st:-1 flags:1 ts: 1.894167
ret: 0 st: 0 flags:1 dts: 1.894127 pts: 1.894127 pos: 30364 size: 4096
ret: 0 st: 0 flags:0 ts: 0.788345
ret: 0 st: 0 flags:1 dts: 0.788367 pts: 0.788367 pos: 12672 size: 4096
ret: 0 st: 0 flags:1 ts:-0.317506
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
ret: 0 st:-1 flags:0 ts: 2.576668
ret: 0 st: 0 flags:1 dts: 2.576757 pts: 2.576757 pos: 41286 size: 4096
ret: 0 st:-1 flags:1 ts: 1.470835
ret: 0 st: 0 flags:1 dts: 1.470748 pts: 1.470748 pos: 23590 size: 4096
ret: 0 st: 0 flags:0 ts: 0.365011
ret: 0 st: 0 flags:1 dts: 0.365125 pts: 0.365125 pos: 5900 size: 4096
ret: 0 st: 0 flags:1 ts:-0.740839
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
ret: 0 st:-1 flags:0 ts: 2.153336
ret: 0 st: 0 flags:1 dts: 2.153379 pts: 2.153379 pos: 34512 size: 4096
ret: 0 st:-1 flags:1 ts: 1.047503
ret: 0 st: 0 flags:1 dts: 1.047506 pts: 1.047506 pos: 16818 size: 4096
ret: 0 st: 0 flags:0 ts:-0.058322
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
ret: 0 st: 0 flags:1 ts: 2.835828
ret: 0 st: 0 flags:1 dts: 2.835760 pts: 2.835760 pos: 45430 size: 4096
ret: 0 st:-1 flags:0 ts: 1.730004
ret: 0 st: 0 flags:1 dts: 1.730000 pts: 1.730000 pos: 27738 size: 4096
ret: 0 st:-1 flags:1 ts: 0.624171
ret: 0 st: 0 flags:1 dts: 0.624127 pts: 0.624127 pos: 10044 size: 4096
ret: 0 st: 0 flags:0 ts:-0.481655
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
ret: 0 st: 0 flags:1 ts: 2.412494
ret: 0 st: 0 flags:1 dts: 2.412381 pts: 2.412381 pos: 38656 size: 4096
ret: 0 st:-1 flags:0 ts: 1.306672
ret: 0 st: 0 flags:1 dts: 1.306757 pts: 1.306757 pos: 20966 size: 4096
ret: 0 st:-1 flags:1 ts: 0.200839
ret: 0 st: 0 flags:1 dts: 0.200748 pts: 0.200748 pos: 3270 size: 4096
ret: 0 st: 0 flags:0 ts:-0.904989
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
ret: 0 st: 0 flags:1 ts: 1.989184
ret: 0 st: 0 flags:1 dts: 1.989116 pts: 1.989116 pos: 31884 size: 4096
ret: 0 st:-1 flags:0 ts: 0.883340
ret: 0 st: 0 flags:1 dts: 0.883379 pts: 0.883379 pos: 14192 size: 4096
ret: 0 st:-1 flags:1 ts:-0.222493
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
ret: 0 st: 0 flags:0 ts: 2.671678
ret: 0 st: 0 flags:1 dts: 2.671746 pts: 2.671746 pos: 42806 size: 4096
ret: 0 st: 0 flags:1 ts: 1.565850
ret: 0 st: 0 flags:1 dts: 1.565760 pts: 1.565760 pos: 25110 size: 4096
ret: 0 st:-1 flags:0 ts: 0.460008
ret: 0 st: 0 flags:1 dts: 0.460000 pts: 0.460000 pos: 7418 size: 4096
ret: 0 st:-1 flags:1 ts:-0.645825
ret: 0 st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos: 58 size: 4096
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