mirror of https://git.ffmpeg.org/ffmpeg.git
Add ALAC 24 bps decoding support.
Originally committed as revision 21637 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Jai Menon
parent
3102d180bb
commit
f430c7b6ac
|
|
@ -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;
|
||||
|
|
|
|||
Loading…
Reference in New Issue