Remove more code duplication. Based on a patch by Matthieu Castet.

Originally committed as revision 9727 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Vitor Sessak 18 years ago
parent d562ba238e
commit 10fb5763cc
  1. 138
      libavcodec/alac.c

@ -514,122 +514,8 @@ static int alac_decode_frame(AVCodecContext *avctx,
*outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
switch(channels) {
case 1: { /* 1 channel */
int ricemodifier;
if (!isnotcompressed) {
/* so it is compressed */
int16_t predictor_coef_table[32];
int predictor_coef_num;
int prediction_type;
int prediction_quantitization;
int i;
/* FIXME: skip 16 bits, not sure what they are. seem to be used in
* two channel case */
get_bits(&alac->gb, 8);
get_bits(&alac->gb, 8);
prediction_type = get_bits(&alac->gb, 4);
prediction_quantitization = get_bits(&alac->gb, 4);
ricemodifier = get_bits(&alac->gb, 3);
predictor_coef_num = get_bits(&alac->gb, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num; i++) {
predictor_coef_table[i] = (int16_t)get_bits(&alac->gb, 16);
}
if (wasted_bytes) {
/* these bytes seem to have something to do with
* > 2 channel files.
*/
av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
}
bastardized_rice_decompress(alac,
alac->predicterror_buffer[0],
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
ricemodifier * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type == 0) {
/* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer[0],
alac->outputsamples_buffer[0],
outputsamples,
readsamplesize,
predictor_coef_table,
predictor_coef_num,
prediction_quantitization);
} else {
av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type);
/* 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, ...)
* predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
* little strange..
*/
}
} else {
/* not compressed, easy case */
if (readsamplesize <= 16) {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits = get_bits(&alac->gb, readsamplesize);
audiobits = SIGN_EXTENDED32(audiobits, readsamplesize);
alac->outputsamples_buffer[0][i] = audiobits;
}
} else {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits;
audiobits = get_bits(&alac->gb, 16);
/* special case of sign extension..
* as we'll be ORing the low 16bits into this */
audiobits = audiobits << 16;
audiobits = audiobits >> (32 - readsamplesize);
audiobits |= get_bits(&alac->gb, readsamplesize - 16);
alac->outputsamples_buffer[0][i] = audiobits;
}
}
/* wasted_bytes = 0; // unused */
}
switch(alac->setinfo_sample_size) {
case 16: {
int i;
for (i = 0; i < outputsamples; i++) {
int16_t sample = alac->outputsamples_buffer[0][i];
((int16_t*)outbuffer)[i * alac->numchannels] = sample;
}
break;
}
case 20:
case 24:
case 32:
av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
break;
default:
break;
}
break;
}
case 2: { /* 2 channels */
if (!isnotcompressed) {
/* compressed */
int16_t predictor_coef_table[channels][32];
int predictor_coef_num[channels];
int prediction_type[channels];
@ -655,7 +541,6 @@ static int alac_decode_frame(AVCodecContext *avctx,
}
if (wasted_bytes) {
/* see mono case */
av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
}
@ -679,8 +564,13 @@ static int alac_decode_frame(AVCodecContext *avctx,
predictor_coef_num[chan],
prediction_quantitization[chan]);
} else {
/* see mono case */
av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
/* 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, ...)
* predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
* little strange..
*/
}
}
} else {
@ -692,7 +582,8 @@ static int alac_decode_frame(AVCodecContext *avctx,
int32_t audiobits;
audiobits = get_bits(&alac->gb, alac->setinfo_sample_size);
audiobits = SIGN_EXTENDED32(audiobits, alac->setinfo_sample_size);
audiobits = SIGN_EXTENDED32(audiobits, readsamplesize);
alac->outputsamples_buffer[chan][i] = audiobits;
}
}
@ -703,6 +594,8 @@ static int alac_decode_frame(AVCodecContext *avctx,
int32_t audiobits;
audiobits = get_bits(&alac->gb, 16);
/* special case of sign extension..
* as we'll be ORing the low 16bits into this */
audiobits = audiobits << 16;
audiobits = audiobits >> (32 - alac->setinfo_sample_size);
audiobits |= get_bits(&alac->gb, alac->setinfo_sample_size - 16);
@ -718,6 +611,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
switch(alac->setinfo_sample_size) {
case 16: {
if (channels == 2) {
deinterlace_16(alac->outputsamples_buffer[0],
alac->outputsamples_buffer[1],
(int16_t*)outbuffer,
@ -725,6 +619,13 @@ static int alac_decode_frame(AVCodecContext *avctx,
outputsamples,
interlacing_shift,
interlacing_leftweight);
} else {
int i;
for (i = 0; i < outputsamples; i++) {
int16_t sample = alac->outputsamples_buffer[0][i];
((int16_t*)outbuffer)[i * alac->numchannels] = sample;
}
}
break;
}
case 20:
@ -736,9 +637,6 @@ static int alac_decode_frame(AVCodecContext *avctx,
break;
}
break;
}
}
return input_buffer_size;
}

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