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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/apedec.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/apedec.c')
-rw-r--r-- | libavcodec/apedec.c | 37 |
1 files changed, 22 insertions, 15 deletions
diff --git a/libavcodec/apedec.c b/libavcodec/apedec.c index 7702b291c8..2d03c554a6 100644 --- a/libavcodec/apedec.c +++ b/libavcodec/apedec.c @@ -129,6 +129,7 @@ typedef struct APEPredictor { /** Decoder context */ typedef struct APEContext { AVCodecContext *avctx; + AVFrame frame; DSPContext dsp; int channels; int samples; ///< samples left to decode in current frame @@ -215,6 +216,10 @@ static av_cold int ape_decode_init(AVCodecContext *avctx) dsputil_init(&s->dsp, avctx); avctx->sample_fmt = AV_SAMPLE_FMT_S16; avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; filter_alloc_fail: ape_decode_close(avctx); @@ -805,16 +810,15 @@ static void ape_unpack_stereo(APEContext *ctx, int count) } } -static int ape_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int ape_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; APEContext *s = avctx->priv_data; - int16_t *samples = data; - int i; - int blockstodecode, out_size; + int16_t *samples; + int i, ret; + int blockstodecode; int bytes_used = 0; /* this should never be negative, but bad things will happen if it is, so @@ -826,7 +830,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *tmp_data; if (!buf_size) { - *data_size = 0; + *got_frame_ptr = 0; return 0; } if (buf_size < 8) { @@ -874,18 +878,19 @@ static int ape_decode_frame(AVCodecContext *avctx, } if (!s->data) { - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } blockstodecode = FFMIN(BLOCKS_PER_LOOP, s->samples); - out_size = blockstodecode * avctx->channels * - av_get_bytes_per_sample(avctx->sample_fmt); - if (*data_size < out_size) { - av_log(avctx, AV_LOG_ERROR, "Output buffer is too small.\n"); - return AVERROR(EINVAL); + /* get output buffer */ + s->frame.nb_samples = blockstodecode; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)s->frame.data[0]; s->error=0; @@ -909,7 +914,9 @@ static int ape_decode_frame(AVCodecContext *avctx, s->samples -= blockstodecode; - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return bytes_used; } @@ -927,7 +934,7 @@ AVCodec ff_ape_decoder = { .init = ape_decode_init, .close = ape_decode_close, .decode = ape_decode_frame, - .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY, + .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1, .flush = ape_flush, .long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"), }; |