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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/qdm2.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/qdm2.c')
-rw-r--r-- | libavcodec/qdm2.c | 32 |
1 files changed, 18 insertions, 14 deletions
diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 5068e675cb..9341c69281 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -130,6 +130,8 @@ typedef struct { * QDM2 decoder context */ typedef struct { + AVFrame frame; + /// Parameters from codec header, do not change during playback int nb_channels; ///< number of channels int channels; ///< number of channels @@ -1875,6 +1877,9 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) avctx->sample_fmt = AV_SAMPLE_FMT_S16; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + // dump_context(s); return 0; } @@ -1952,30 +1957,27 @@ static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) } -static int qdm2_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int qdm2_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; QDM2Context *s = avctx->priv_data; - int16_t *out = data; - int i, out_size; + int16_t *out; + int i, ret; if(!buf) return 0; if(buf_size < s->checksum_size) return -1; - out_size = 16 * s->channels * s->frame_size * - 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 = 16 * s->frame_size; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } - - av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", - buf_size, buf, s->checksum_size, data, *data_size); + out = (int16_t *)s->frame.data[0]; for (i = 0; i < 16; i++) { if (qdm2_decode(s, buf, out) < 0) @@ -1983,7 +1985,8 @@ static int qdm2_decode_frame(AVCodecContext *avctx, out += s->channels * s->frame_size; } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; return s->checksum_size; } @@ -1997,5 +2000,6 @@ AVCodec ff_qdm2_decoder = .init = qdm2_decode_init, .close = qdm2_decode_close, .decode = qdm2_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), }; |