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| author | Justin Ruggles <justin.ruggles@gmail.com> | 2011-04-22 21:30:19 -0400 |
|---|---|---|
| committer | Justin Ruggles <justin.ruggles@gmail.com> | 2011-05-18 17:27:06 -0400 |
| commit | 9aa8193a234ccb6a79cba5cc550531f62ffb0a17 (patch) | |
| tree | 699dce38e0c73e2daf1aa8afb2d31e42da860515 /libavcodec/vorbisdec.c | |
| parent | bc778a0cea3027941afa1ff6bbb424b3159a0b27 (diff) | |
| download | ffmpeg-9aa8193a234ccb6a79cba5cc550531f62ffb0a17.tar.gz | |
Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis
decoders.
Based on patches by clsid2 in ffdshow-tryout.
Diffstat (limited to 'libavcodec/vorbisdec.c')
| -rw-r--r-- | libavcodec/vorbisdec.c | 23 |
1 files changed, 18 insertions, 5 deletions
diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c index 7443e98a64..f6ec74f4a1 100644 --- a/libavcodec/vorbisdec.c +++ b/libavcodec/vorbisdec.c @@ -979,7 +979,13 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext) dsputil_init(&vc->dsp, avccontext); ff_fmt_convert_init(&vc->fmt_conv, avccontext); - vc->scale_bias = 32768.0f; + if (avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT) { + avccontext->sample_fmt = AV_SAMPLE_FMT_FLT; + vc->scale_bias = 1.0f; + } else { + avccontext->sample_fmt = AV_SAMPLE_FMT_S16; + vc->scale_bias = 32768.0f; + } if (!headers_len) { av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n"); @@ -1024,7 +1030,6 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext) avccontext->channels = vc->audio_channels; avccontext->sample_rate = vc->audio_samplerate; avccontext->frame_size = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2; - avccontext->sample_fmt = AV_SAMPLE_FMT_S16; return 0 ; } @@ -1634,9 +1639,14 @@ static int vorbis_decode_frame(AVCodecContext *avccontext, len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i]; } - vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len, - vc->audio_channels); - *data_size = len * 2 * vc->audio_channels; + if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT) + vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels); + else + vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len, + vc->audio_channels); + + *data_size = len * vc->audio_channels * + (av_get_bits_per_sample_fmt(avccontext->sample_fmt) / 8); return buf_size ; } @@ -1663,5 +1673,8 @@ AVCodec ff_vorbis_decoder = { vorbis_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("Vorbis"), .channel_layouts = ff_vorbis_channel_layouts, + .sample_fmts = (const enum AVSampleFormat[]) { + AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE + }, }; |