* @author 2005 David Hammerton
* @see http://crazney.net/programs/itunes/alac.html
*
- * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
+ * Note: This decoder expects a 36-byte QuickTime atom to be
* passed through the extradata[_size] fields. This atom is tacked onto
* the end of an 'alac' stsd atom and has the following format:
- * bytes 0-3 atom size (0x24), big-endian
- * bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd)
- * bytes 8-35 data bytes needed by decoder
*
- * Extradata:
- * 32bit size
- * 32bit tag (=alac)
- * 32bit zero?
- * 32bit max sample per frame
- * 8bit ?? (zero?)
+ * 32bit atom size
+ * 32bit tag ("alac")
+ * 32bit tag version (0)
+ * 32bit samples per frame (used when not set explicitly in the frames)
+ * 8bit compatible version (0)
* 8bit sample size
- * 8bit history mult
- * 8bit initial history
- * 8bit kmodifier
- * 8bit channels?
- * 16bit ??
- * 32bit max coded frame size
- * 32bit bitrate?
+ * 8bit history mult (40)
+ * 8bit initial history (14)
+ * 8bit kmodifier (10)
+ * 8bit channels
+ * 16bit maxRun (255)
+ * 32bit max coded frame size (0 means unknown)
+ * 32bit average bitrate (0 means unknown)
* 32bit samplerate
*/
typedef struct {
AVCodecContext *avctx;
+ AVFrame frame;
GetBitContext gb;
int numchannels;
- int bytespersample;
/* buffers */
int32_t *predicterror_buffer[MAX_CHANNELS];
}
static void decorrelate_stereo(int32_t *buffer[MAX_CHANNELS],
- int numchannels, int numsamples,
- uint8_t interlacing_shift,
+ int numsamples, uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
{
int i;
}
static void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS],
- int16_t *buffer_out, int numchannels,
- int numsamples)
+ int16_t *buffer_out, int numsamples)
{
int i;
for (i = 0; i < numsamples; i++) {
- int16_t left, right;
-
- left = buffer[0][i];
- right = buffer[1][i];
-
- buffer_out[i*numchannels] = left;
- buffer_out[i*numchannels + 1] = right;
+ *buffer_out++ = buffer[0][i];
+ *buffer_out++ = buffer[1][i];
}
}
static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
- int32_t *buffer_out, int numchannels,
- int numsamples)
+ int32_t *buffer_out, int numsamples)
{
int i;
- for (i = 0; i < numsamples; i++) {
- int32_t left, right;
-
- left = buffer[0][i];
- right = buffer[1][i];
-
- buffer_out[i * numchannels] = left << 8;
- buffer_out[i * numchannels + 1] = right << 8;
- }
+ for (i = 0; i < numsamples; i++) {
+ *buffer_out++ = buffer[0][i] << 8;
+ *buffer_out++ = buffer[1][i] << 8;
+ }
}
-static int alac_decode_frame(AVCodecContext *avctx,
- void *outbuffer, int *outputsize,
- AVPacket *avpkt)
+static int alac_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *inbuffer = avpkt->data;
int input_buffer_size = avpkt->size;
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
-
- /* short-circuit null buffers */
- if (!inbuffer || !input_buffer_size)
- return -1;
+ int i, ch, ret;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
} else
outputsamples = alac->setinfo_max_samples_per_frame;
- alac->bytespersample = channels * av_get_bytes_per_sample(avctx->sample_fmt);
-
- if(outputsamples > *outputsize / alac->bytespersample){
- av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
- return -1;
+ /* 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;
+ if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
}
- *outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - alac->extra_bits + channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
int prediction_type[MAX_CHANNELS];
int prediction_quantitization[MAX_CHANNELS];
int ricemodifier[MAX_CHANNELS];
- int i, chan;
interlacing_shift = get_bits(&alac->gb, 8);
interlacing_leftweight = get_bits(&alac->gb, 8);
- for (chan = 0; chan < channels; chan++) {
- prediction_type[chan] = get_bits(&alac->gb, 4);
- prediction_quantitization[chan] = get_bits(&alac->gb, 4);
+ for (ch = 0; ch < channels; ch++) {
+ prediction_type[ch] = get_bits(&alac->gb, 4);
+ prediction_quantitization[ch] = get_bits(&alac->gb, 4);
- ricemodifier[chan] = get_bits(&alac->gb, 3);
- predictor_coef_num[chan] = get_bits(&alac->gb, 5);
+ ricemodifier[ch] = get_bits(&alac->gb, 3);
+ predictor_coef_num[ch] = get_bits(&alac->gb, 5);
/* read the predictor table */
- for (i = 0; i < predictor_coef_num[chan]; i++)
- predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
+ for (i = 0; i < predictor_coef_num[ch]; i++)
+ predictor_coef_table[ch][i] = (int16_t)get_bits(&alac->gb, 16);
}
if (alac->extra_bits) {
- int i, ch;
for (i = 0; i < outputsamples; i++) {
for (ch = 0; ch < channels; ch++)
alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
}
}
- for (chan = 0; chan < channels; chan++) {
+ for (ch = 0; ch < channels; ch++) {
bastardized_rice_decompress(alac,
- alac->predicterror_buffer[chan],
+ alac->predicterror_buffer[ch],
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
- ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
+ ricemodifier[ch] * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
- if (prediction_type[chan] == 0) {
- /* adaptive fir */
- predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
- alac->outputsamples_buffer[chan],
- outputsamples,
- readsamplesize,
- predictor_coef_table[chan],
- predictor_coef_num[chan],
- prediction_quantitization[chan]);
- } else {
- 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..
+ /* adaptive FIR filter */
+ if (prediction_type[ch] == 15) {
+ /* Prediction type 15 runs the adaptive FIR twice.
+ * The first pass uses the special-case coef_num = 31, while
+ * the second pass uses the coefs from the bitstream.
+ *
+ * However, this prediction type is not currently used by the
+ * reference encoder.
*/
+ predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
+ alac->predicterror_buffer[ch],
+ outputsamples, readsamplesize,
+ NULL, 31, 0);
+ } else if (prediction_type[ch] > 0) {
+ av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
+ prediction_type[ch]);
}
+ predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
+ alac->outputsamples_buffer[ch],
+ outputsamples, readsamplesize,
+ predictor_coef_table[ch],
+ predictor_coef_num[ch],
+ prediction_quantitization[ch]);
}
} 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;
-
- audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
-
- alac->outputsamples_buffer[chan][i] = audiobits;
- }
- } 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);
- }
+ for (i = 0; i < outputsamples; i++) {
+ for (ch = 0; ch < channels; ch++) {
+ alac->outputsamples_buffer[ch][i] = get_sbits_long(&alac->gb,
+ alac->setinfo_sample_size);
}
}
alac->extra_bits = 0;
av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
if (channels == 2 && interlacing_leftweight) {
- decorrelate_stereo(alac->outputsamples_buffer, alac->numchannels,
- outputsamples, interlacing_shift,
- interlacing_leftweight);
+ decorrelate_stereo(alac->outputsamples_buffer, outputsamples,
+ interlacing_shift, interlacing_leftweight);
}
if (alac->extra_bits) {
switch(alac->setinfo_sample_size) {
case 16:
if (channels == 2) {
- interleave_stereo_16(alac->outputsamples_buffer, outbuffer,
- alac->numchannels, outputsamples);
+ interleave_stereo_16(alac->outputsamples_buffer,
+ (int16_t *)alac->frame.data[0], outputsamples);
} else {
- int i;
+ int16_t *outbuffer = (int16_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++) {
- ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
+ outbuffer[i] = alac->outputsamples_buffer[0][i];
}
}
break;
case 24:
if (channels == 2) {
- interleave_stereo_24(alac->outputsamples_buffer, outbuffer,
- alac->numchannels, outputsamples);
+ interleave_stereo_24(alac->outputsamples_buffer,
+ (int32_t *)alac->frame.data[0], outputsamples);
} else {
- int i;
+ int32_t *outbuffer = (int32_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++)
- ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
+ outbuffer[i] = alac->outputsamples_buffer[0][i] << 8;
}
break;
}
if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = alac->frame;
+
return input_buffer_size;
}
{
ALACContext *alac = avctx->priv_data;
- int chan;
- for (chan = 0; chan < alac->numchannels; chan++) {
- av_freep(&alac->predicterror_buffer[chan]);
- av_freep(&alac->outputsamples_buffer[chan]);
- av_freep(&alac->extra_bits_buffer[chan]);
+ int ch;
+ for (ch = 0; ch < alac->numchannels; ch++) {
+ av_freep(&alac->predicterror_buffer[ch]);
+ av_freep(&alac->outputsamples_buffer[ch]);
+ av_freep(&alac->extra_bits_buffer[ch]);
}
return 0;
static int allocate_buffers(ALACContext *alac)
{
- int chan;
- for (chan = 0; chan < alac->numchannels; chan++) {
- alac->predicterror_buffer[chan] =
- av_malloc(alac->setinfo_max_samples_per_frame * 4);
+ int ch;
+ for (ch = 0; ch < alac->numchannels; ch++) {
+ int buf_size = alac->setinfo_max_samples_per_frame * sizeof(int32_t);
- alac->outputsamples_buffer[chan] =
- av_malloc(alac->setinfo_max_samples_per_frame * 4);
+ FF_ALLOC_OR_GOTO(alac->avctx, alac->predicterror_buffer[ch],
+ buf_size, buf_alloc_fail);
- alac->extra_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
+ FF_ALLOC_OR_GOTO(alac->avctx, alac->outputsamples_buffer[ch],
+ buf_size, buf_alloc_fail);
- if (!alac->predicterror_buffer[chan] ||
- !alac->outputsamples_buffer[chan] ||
- !alac->extra_bits_buffer[chan]) {
- alac_decode_close(alac->avctx);
- return AVERROR(ENOMEM);
- }
+ FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch],
+ buf_size, buf_alloc_fail);
}
return 0;
+buf_alloc_fail:
+ alac_decode_close(alac->avctx);
+ return AVERROR(ENOMEM);
}
static int alac_set_info(ALACContext *alac)
ptr += 4; /* size */
ptr += 4; /* alac */
- ptr += 4; /* 0 ? */
+ ptr += 4; /* version */
if(AV_RB32(ptr) >= UINT_MAX/4){
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
/* buffer size / 2 ? */
alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
- ptr++; /* ??? */
+ ptr++; /* compatible version */
alac->setinfo_sample_size = *ptr++;
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
alac->numchannels = *ptr++;
- bytestream_get_be16(&ptr); /* ??? */
+ bytestream_get_be16(&ptr); /* maxRun */
bytestream_get_be32(&ptr); /* max coded frame size */
- bytestream_get_be32(&ptr); /* bitrate ? */
+ bytestream_get_be32(&ptr); /* average bitrate */
bytestream_get_be32(&ptr); /* samplerate */
return 0;
break;
case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
break;
- default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
- alac->setinfo_sample_size);
- return -1;
+ default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
+ alac->setinfo_sample_size);
+ return AVERROR_PATCHWELCOME;
}
if (alac->numchannels < 1) {
return ret;
}
+ avcodec_get_frame_defaults(&alac->frame);
+ avctx->coded_frame = &alac->frame;
+
return 0;
}
.init = alac_decode_init,
.close = alac_decode_close,
.decode = alac_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
projects / ffmpeg / blobdiff