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Add ALAC 24 bps decoding support.

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Originally committed as revision 21637 to svn://svn.ffmpeg.org/ffmpeg/trunk
release/0.6
Jai Menon 15 years ago
parent 3102d180bb
commit f430c7b6ac
1 changed files with 103 additions and 16 deletions
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  1. 119
      libavcodec/alac.c

119
libavcodec/alac.c
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@ -77,6 +77,8 @@ typedef struct {
int32_t *outputsamples_buffer[MAX_CHANNELS];
int32_t *wasted_bits_buffer[MAX_CHANNELS];
/* stuff from setinfo */
uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
uint8_t setinfo_sample_size; /* 0x10 */
@ -85,6 +87,7 @@ typedef struct {
uint8_t setinfo_rice_kmodifier; /* 0x0e */
/* end setinfo stuff */
int wasted_bits;
} ALACContext;
static void allocate_buffers(ALACContext *alac)
@ -96,6 +99,8 @@ static void allocate_buffers(ALACContext *alac)
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);
}
}
@ -398,6 +403,56 @@ static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
}
}
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)
{
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];
if (wasted_bits) {
left = (left << wasted_bits) | wasted_bits_buffer[0][i];
right = (right << wasted_bits) | wasted_bits_buffer[1][i];
}
buffer_out[i * numchannels] = left << 8;
buffer_out[i * numchannels + 1] = right << 8;
}
}
}
static int alac_decode_frame(AVCodecContext *avctx,
void *outbuffer, int *outputsize,
AVPacket *avpkt)
@ -410,7 +465,6 @@ static int alac_decode_frame(AVCodecContext *avctx,
unsigned int outputsamples;
int hassize;
unsigned int readsamplesize;
int wasted_bytes;
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
@ -452,7 +506,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
/* the output sample size is stored soon */
hassize = get_bits1(&alac->gb);
wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
/* whether the frame is compressed */
isnotcompressed = get_bits1(&alac->gb);
@ -467,13 +521,25 @@ 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 = SAMPLE_FMT_S16;
alac->bytespersample = channels << 1;
break;
case 24: avctx->sample_fmt = 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;
}
if(outputsamples > *outputsize / alac->bytespersample){
av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
return -1;
}
*outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
return -1;
@ -503,9 +569,13 @@ static int alac_decode_frame(AVCodecContext *avctx,
predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
}
if (wasted_bytes)
av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
if (alac->wasted_bits) {
int i, ch;
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);
}
}
for (chan = 0; chan < channels; chan++) {
bastardized_rice_decompress(alac,
alac->predicterror_buffer[chan],
@ -538,6 +608,7 @@ 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;
@ -546,7 +617,17 @@ static int alac_decode_frame(AVCodecContext *avctx,
alac->outputsamples_buffer[chan][i] = audiobits;
}
/* wasted_bytes = 0; */
} 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);
}
}
}
alac->wasted_bits = 0;
interlacing_shift = 0;
interlacing_leftweight = 0;
}
@ -570,14 +651,21 @@ static int alac_decode_frame(AVCodecContext *avctx,
}
}
break;
case 20:
case 24:
// It is not clear if there exist any encoder that creates 24 bit ALAC
// files. iTunes convert 24 bit raw files to 16 bit before encoding.
case 32:
av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
break;
default:
if (channels == 2) {
decorrelate_stereo_24(alac->outputsamples_buffer,
outbuffer,
alac->wasted_bits_buffer,
alac->wasted_bits,
alac->numchannels,
outputsamples,
interlacing_shift,
interlacing_leftweight);
} else {
int i;
for (i = 0; i < outputsamples; i++)
((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
}
break;
}
@ -594,8 +682,6 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
alac->context_initialized = 0;
alac->numchannels = alac->avctx->channels;
alac->bytespersample = 2 * alac->numchannels;
avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
@ -608,6 +694,7 @@ static av_cold int alac_decode_close(AVCodecContext *avctx)
for (chan = 0; chan < MAX_CHANNELS; chan++) {
av_free(alac->predicterror_buffer[chan]);
av_free(alac->outputsamples_buffer[chan]);
av_freep(&alac->wasted_bits_buffer[chan]);
}
return 0;

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