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author | Michael Niedermayer <michaelni@gmx.at> | 2011-12-03 02:08:55 +0100 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-12-03 03:00:30 +0100 |
commit | e4de71677f3adeac0f74b89ac8df5d417364df2c (patch) | |
tree | 4792dd8d85d24f0f4eaddabb65f6044727907daa /libavcodec/ac3dec.c | |
parent | 12804348f5babf56a315fa01751eea1ffdddf98a (diff) | |
parent | d268b79e3436107c11ee8bcdf9f3645368bb3fcd (diff) | |
download | ffmpeg-streaming-e4de71677f3adeac0f74b89ac8df5d417364df2c.zip ffmpeg-streaming-e4de71677f3adeac0f74b89ac8df5d417364df2c.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/ac3dec.c')
-rw-r--r-- | libavcodec/ac3dec.c | 40 |
1 files changed, 24 insertions, 16 deletions
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index 09b9a31..c650881 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; } @@ -1296,16 +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; - int data_size_orig, data_size_tmp; + 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]; @@ -1322,8 +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_orig = *data_size; - *data_size = 0; err = parse_frame_header(s); if (err) { @@ -1345,6 +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"); @@ -1406,21 +1407,24 @@ 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++) output[ch] = s->output[channel_map[ch]]; - data_size_tmp = s->num_blocks * 256 * avctx->channels; - data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples_s16); - if (data_size_orig < data_size_tmp) - return -1; - *data_size = data_size_tmp; for (blk = 0; blk < s->num_blocks; blk++) { if (!err && decode_audio_block(s, blk)) { av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n"); err = 1; } - if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) { s->fmt_conv.float_interleave(out_samples_flt, output, 256, s->out_channels); @@ -1431,8 +1435,10 @@ 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); } @@ -1477,6 +1483,7 @@ 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 @@ -1499,6 +1506,7 @@ 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 |