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author | Justin Ruggles <justin.ruggles@gmail.com> | 2011-09-06 12:17:45 -0400 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2011-12-02 17:40:40 -0500 |
commit | 0eea212943544d40f99b05571aa7159d78667154 (patch) | |
tree | 1e6b0271a633bf8a3f92c78bdfbaca275498ee26 /libavcodec/alsdec.c | |
parent | 560f773c7ddd17f66e2621222980c1359a9027be (diff) | |
download | ffmpeg-0eea212943544d40f99b05571aa7159d78667154.tar.gz |
Add avcodec_decode_audio4().
Deprecate avcodec_decode_audio3().
Implement audio support in avcodec_default_get_buffer().
Implement the new audio decoder API in all audio decoders.
Diffstat (limited to 'libavcodec/alsdec.c')
-rw-r--r-- | libavcodec/alsdec.c | 43 |
1 files changed, 24 insertions, 19 deletions
diff --git a/libavcodec/alsdec.c b/libavcodec/alsdec.c index e7a0de24b1..71495803a3 100644 --- a/libavcodec/alsdec.c +++ b/libavcodec/alsdec.c @@ -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)"), }; |