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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-28 20:53:54 +02:00

alac: move the current samples per frame to the ALACContext

This will simplify the multi-channel implementation.
This commit is contained in:
Justin Ruggles 2012-07-09 13:15:35 -04:00
parent 46043962ea
commit 7a50ec6799

View File

@ -75,6 +75,7 @@ typedef struct {
uint8_t rice_limit;
int extra_bits; /**< number of extra bits beyond 16-bit */
int nb_samples; /**< number of samples in the current frame */
} ALACContext;
static inline int decode_scalar(GetBitContext *gb, int k, int readsamplesize)
@ -295,7 +296,6 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
ALACContext *alac = avctx->priv_data;
int channels;
unsigned int outputsamples;
int hassize;
unsigned int readsamplesize;
int is_compressed;
@ -324,21 +324,18 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
if (hassize) {
/* now read the number of samples as a 32bit integer */
outputsamples = get_bits_long(&alac->gb, 32);
if (outputsamples > alac->max_samples_per_frame) {
av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n",
outputsamples, alac->max_samples_per_frame);
return -1;
uint32_t output_samples = get_bits_long(&alac->gb, 32);
if (!output_samples || output_samples > alac->max_samples_per_frame) {
av_log(avctx, AV_LOG_ERROR, "invalid samples per frame: %d\n",
output_samples);
return AVERROR_INVALIDDATA;
}
alac->nb_samples = output_samples;
} else
outputsamples = alac->max_samples_per_frame;
alac->nb_samples = alac->max_samples_per_frame;
/* get output buffer */
if (outputsamples > INT32_MAX) {
av_log(avctx, AV_LOG_ERROR, "unsupported block size: %u\n", outputsamples);
return AVERROR_INVALIDDATA;
}
alac->frame.nb_samples = outputsamples;
alac->frame.nb_samples = alac->nb_samples;
if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
@ -373,7 +370,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
}
if (alac->extra_bits) {
for (i = 0; i < outputsamples; i++) {
for (i = 0; i < alac->nb_samples; i++) {
for (ch = 0; ch < channels; ch++)
alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
}
@ -381,7 +378,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
for (ch = 0; ch < channels; ch++) {
bastardized_rice_decompress(alac,
alac->predict_error_buffer[ch],
outputsamples,
alac->nb_samples,
readsamplesize,
ricemodifier[ch] * alac->rice_history_mult / 4);
@ -396,7 +393,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
*/
predictor_decompress_fir_adapt(alac->predict_error_buffer[ch],
alac->predict_error_buffer[ch],
outputsamples, readsamplesize,
alac->nb_samples, readsamplesize,
NULL, 31, 0);
} else if (prediction_type[ch] > 0) {
av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
@ -404,14 +401,14 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
}
predictor_decompress_fir_adapt(alac->predict_error_buffer[ch],
alac->output_samples_buffer[ch],
outputsamples, readsamplesize,
alac->nb_samples, readsamplesize,
predictor_coef_table[ch],
predictor_coef_num[ch],
prediction_quantitization[ch]);
}
} else {
/* not compressed, easy case */
for (i = 0; i < outputsamples; i++) {
for (i = 0; i < alac->nb_samples; i++) {
for (ch = 0; ch < channels; ch++) {
alac->output_samples_buffer[ch][i] = get_sbits_long(&alac->gb,
alac->sample_size);
@ -425,23 +422,24 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
if (channels == 2 && interlacing_leftweight) {
decorrelate_stereo(alac->output_samples_buffer, outputsamples,
decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples,
interlacing_shift, interlacing_leftweight);
}
if (alac->extra_bits) {
append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,
alac->extra_bits, alac->channels, outputsamples);
alac->extra_bits, alac->channels, alac->nb_samples);
}
switch(alac->sample_size) {
case 16:
if (channels == 2) {
interleave_stereo_16(alac->output_samples_buffer,
(int16_t *)alac->frame.data[0], outputsamples);
(int16_t *)alac->frame.data[0],
alac->nb_samples);
} else {
int16_t *outbuffer = (int16_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++) {
for (i = 0; i < alac->nb_samples; i++) {
outbuffer[i] = alac->output_samples_buffer[0][i];
}
}
@ -449,10 +447,11 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
case 24:
if (channels == 2) {
interleave_stereo_24(alac->output_samples_buffer,
(int32_t *)alac->frame.data[0], outputsamples);
(int32_t *)alac->frame.data[0],
alac->nb_samples);
} else {
int32_t *outbuffer = (int32_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++)
for (i = 0; i < alac->nb_samples; i++)
outbuffer[i] = alac->output_samples_buffer[0][i] << 8;
}
break;