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Merge remote-tracking branch 'qatar/master'
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* 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: Michael Niedermayer <[email protected]>
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@michaelni
michaelni committed Oct 29, 2011
2 parents 1af3571 + 7d1b17b commit d17e707
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Showing 14 changed files with 493 additions and 277 deletions.
93 changes: 60 additions & 33 deletions libavcodec/atrac1.c
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Original file line number Diff line number Diff line change
Expand Up @@ -36,6 +36,7 @@
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
#include "sinewin.h"

#include "atrac.h"
Expand Down Expand Up @@ -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 */
Expand Down Expand Up @@ -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;
Expand Down Expand Up @@ -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);
Expand Down Expand Up @@ -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];
Expand Down Expand Up @@ -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++) {
Expand All @@ -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;
Expand All @@ -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,
Expand Down
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