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| author | clsid2 <clsid2@3b938f2f-1a1a-0410-8054-a526ea5ff92c> | 2011-03-07 00:28:50 +0000 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2011-04-03 22:52:58 +0200 |
| commit | 361fa0ed40a042393a2691e3dba9bd7c4bcfe188 (patch) | |
| tree | d30bdd0a8edbdcdea0cd81fae3ce2c01fa1e025d /libavcodec/aacdec.c | |
| parent | ba7a28045f0de749f5e84fbefeb626da9952226c (diff) | |
| download | ffmpeg-streaming-361fa0ed40a042393a2691e3dba9bd7c4bcfe188.zip ffmpeg-streaming-361fa0ed40a042393a2691e3dba9bd7c4bcfe188.tar.gz | |
Float output for libavcodec AAC decoder
git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3770 3b938f2f-1a1a-0410-8054-a526ea5ff92c
Diffstat (limited to 'libavcodec/aacdec.c')
| -rw-r--r-- | libavcodec/aacdec.c | 26 |
1 files changed, 25 insertions, 1 deletions
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c index 1399eda..e7b312c 100644 --- a/libavcodec/aacdec.c +++ b/libavcodec/aacdec.c @@ -549,7 +549,12 @@ static av_cold int aac_decode_init(AVCodecContext *avctx) return -1; } + /* ffdshow custom code */ +#if CONFIG_AUDIO_FLOAT + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; +#else avctx->sample_fmt = AV_SAMPLE_FMT_S16; +#endif AAC_INIT_VLC_STATIC( 0, 304); AAC_INIT_VLC_STATIC( 1, 270); @@ -2166,7 +2171,12 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data, avctx->frame_size = samples; } + /* ffdshow custom code */ +#if CONFIG_AUDIO_FLOAT + data_size_tmp = samples * avctx->channels * sizeof(float); +#else data_size_tmp = samples * avctx->channels * sizeof(int16_t); +#endif if (*data_size < data_size_tmp) { av_log(avctx, AV_LOG_ERROR, "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", @@ -2175,8 +2185,14 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data, } *data_size = data_size_tmp; - if (samples) + if (samples) { + /* ffdshow custom code */ +#if CONFIG_AUDIO_FLOAT + float_interleave(data, (const float **)ac->output_data, samples, avctx->channels); +#else ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels); +#endif + } if (ac->output_configured) ac->output_configured = OC_LOCKED; @@ -2494,7 +2510,11 @@ AVCodec ff_aac_decoder = { aac_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), .sample_fmts = (const enum AVSampleFormat[]) { +#if CONFIG_AUDIO_FLOAT + AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE +#else AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE +#endif }, .channel_layouts = aac_channel_layout, }; @@ -2514,7 +2534,11 @@ AVCodec ff_aac_latm_decoder = { .decode = latm_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"), .sample_fmts = (const enum AVSampleFormat[]) { +#if CONFIG_AUDIO_FLOAT + AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE +#else AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE +#endif }, .channel_layouts = aac_channel_layout, }; |
