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Add avcodec_decode_audio4().

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Deprecate avcodec_decode_audio3().
Implement audio support in avcodec_default_get_buffer().
Implement the new audio decoder API in all audio decoders.
This commit is contained in:
Justin Ruggles
2011-09-06 12:17:45 -04:00
parent 560f773c7d
commit 0eea212943
64 changed files with 1588 additions and 764 deletions
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43
libavcodec/alsdec.c
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@@ -191,6 +191,7 @@ typedef struct {
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
ALSSpecificConfig sconf;
GetBitContext gb;
DSPContext dsp;
@@ -1415,15 +1416,14 @@ static int read_frame_data(ALSDecContext *ctx, unsigned int ra_frame)
/** Decode an ALS frame.
*/
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
ALSDecContext *ctx = avctx->priv_data;
ALSSpecificConfig *sconf = &ctx->sconf;
const uint8_t *buffer = avpkt->data;
int buffer_size = avpkt->size;
int invalid_frame, size;
int invalid_frame, ret;
unsigned int c, sample, ra_frame, bytes_read, shift;
init_get_bits(&ctx->gb, buffer, buffer_size * 8);
@@ -1448,21 +1448,17 @@ static int decode_frame(AVCodecContext *avctx,
ctx->frame_id++;
// check for size of decoded data
size = ctx->cur_frame_length * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (size > *data_size) {
av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n");
return -1;
/* get output buffer */
ctx->frame.nb_samples = ctx->cur_frame_length;
if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
*data_size = size;
// transform decoded frame into output format
#define INTERLEAVE_OUTPUT(bps) \
{ \
int##bps##_t *dest = (int##bps##_t*) data; \
int##bps##_t *dest = (int##bps##_t*)ctx->frame.data[0]; \
shift = bps - ctx->avctx->bits_per_raw_sample; \
for (sample = 0; sample < ctx->cur_frame_length; sample++) \
for (c = 0; c < avctx->channels; c++) \
@@ -1480,7 +1476,7 @@ static int decode_frame(AVCodecContext *avctx,
int swap = HAVE_BIGENDIAN != sconf->msb_first;
if (ctx->avctx->bits_per_raw_sample == 24) {
int32_t *src = data;
int32_t *src = (int32_t *)ctx->frame.data[0];
for (sample = 0;
sample < ctx->cur_frame_length * avctx->channels;
@@ -1501,22 +1497,25 @@ static int decode_frame(AVCodecContext *avctx,
if (swap) {
if (ctx->avctx->bits_per_raw_sample <= 16) {
int16_t *src = (int16_t*) data;
int16_t *src = (int16_t*) ctx->frame.data[0];
int16_t *dest = (int16_t*) ctx->crc_buffer;
for (sample = 0;
sample < ctx->cur_frame_length * avctx->channels;
sample++)
*dest++ = av_bswap16(src[sample]);
} else {
ctx->dsp.bswap_buf((uint32_t*)ctx->crc_buffer, data,
ctx->dsp.bswap_buf((uint32_t*)ctx->crc_buffer,
(uint32_t *)ctx->frame.data[0],
ctx->cur_frame_length * avctx->channels);
}
crc_source = ctx->crc_buffer;
} else {
crc_source = data;
crc_source = ctx->frame.data[0];
}
ctx->crc = av_crc(ctx->crc_table, ctx->crc, crc_source, size);
ctx->crc = av_crc(ctx->crc_table, ctx->crc, crc_source,
ctx->cur_frame_length * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt));
}
@@ -1527,6 +1526,9 @@ static int decode_frame(AVCodecContext *avctx,
}
}
*got_frame_ptr = 1;
*(AVFrame *)data = ctx->frame;
bytes_read = invalid_frame ? buffer_size :
(get_bits_count(&ctx->gb) + 7) >> 3;
@@ -1724,6 +1726,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
dsputil_init(&ctx->dsp, avctx);
avcodec_get_frame_defaults(&ctx->frame);
avctx->coded_frame = &ctx->frame;
return 0;
}
@@ -1747,7 +1752,7 @@ AVCodec ff_als_decoder = {
.close = decode_end,
.decode = decode_frame,
.flush = flush,
.capabilities = CODEC_CAP_SUBFRAMES,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MPEG-4 Audio Lossless Coding (ALS)"),
};