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authorJustin Ruggles <justin.ruggles@gmail.com>2011-09-06 12:17:45 -0400
committerJustin Ruggles <justin.ruggles@gmail.com>2011-12-02 17:40:40 -0500
commit0eea212943544d40f99b05571aa7159d78667154 (patch)
tree1e6b0271a633bf8a3f92c78bdfbaca275498ee26 /libavcodec/ac3dec.c
parent560f773c7ddd17f66e2621222980c1359a9027be (diff)
downloadffmpeg-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/ac3dec.c')
-rw-r--r--libavcodec/ac3dec.c32
1 files changed, 24 insertions, 8 deletions
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 8e216c039b..7e11cf49ce 100644
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -208,6 +208,9 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
}
s->downmixed = 1;
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
@@ -1296,15 +1299,15 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
/**
* Decode a single AC-3 frame.
*/
-static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
- AVPacket *avpkt)
+static int ac3_decode_frame(AVCodecContext * avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
- float *out_samples_flt = data;
- int16_t *out_samples_s16 = data;
- int blk, ch, err;
+ float *out_samples_flt;
+ int16_t *out_samples_s16;
+ int blk, ch, err, ret;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
@@ -1321,7 +1324,6 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
init_get_bits(&s->gbc, buf, buf_size * 8);
/* parse the syncinfo */
- *data_size = 0;
err = parse_frame_header(s);
if (err) {
@@ -1343,6 +1345,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
/* TODO: add support for substreams and dependent frames */
if(s->frame_type == EAC3_FRAME_TYPE_DEPENDENT || s->substreamid) {
av_log(avctx, AV_LOG_ERROR, "unsupported frame type : skipping frame\n");
+ *got_frame_ptr = 0;
return s->frame_size;
} else {
av_log(avctx, AV_LOG_ERROR, "invalid frame type\n");
@@ -1400,6 +1403,15 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
if (s->bitstream_mode == 0x7 && s->channels > 1)
avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
+ /* get output buffer */
+ s->frame.nb_samples = s->num_blocks * 256;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ out_samples_flt = (float *)s->frame.data[0];
+ out_samples_s16 = (int16_t *)s->frame.data[0];
+
/* decode the audio blocks */
channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
for (ch = 0; ch < s->out_channels; ch++)
@@ -1419,8 +1431,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
out_samples_s16 += 256 * s->out_channels;
}
}
- *data_size = s->num_blocks * 256 * avctx->channels *
- av_get_bytes_per_sample(avctx->sample_fmt);
+
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
return FFMIN(buf_size, s->frame_size);
}
@@ -1458,6 +1472,7 @@ AVCodec ff_ac3_decoder = {
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
@@ -1480,6 +1495,7 @@ AVCodec ff_eac3_decoder = {
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE