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Code Issues Releases Activity

Float output for libavcodec AAC decoder

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git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3770 3b938f2f-1a1a-0410-8054-a526ea5ff92c
This commit is contained in:
clsid2 2011-03-07 00:28:50 +00:00 committed by Michael Niedermayer
parent ba7a28045f
commit 361fa0ed40
1 changed files with 25 additions and 1 deletions
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26
libavcodec/aacdec.c
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@ -549,7 +549,12 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
return -1; return -1;
} }
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
#else
avctx->sample_fmt = AV_SAMPLE_FMT_S16; avctx->sample_fmt = AV_SAMPLE_FMT_S16;
#endif
AAC_INIT_VLC_STATIC( 0, 304); AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270); AAC_INIT_VLC_STATIC( 1, 270);
@ -2166,7 +2171,12 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
avctx->frame_size = samples; 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); data_size_tmp = samples * avctx->channels * sizeof(int16_t);
#endif
if (*data_size < data_size_tmp) { if (*data_size < data_size_tmp) {
av_log(avctx, AV_LOG_ERROR, av_log(avctx, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", "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; *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); ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
#endif
}
if (ac->output_configured) if (ac->output_configured)
ac->output_configured = OC_LOCKED; ac->output_configured = OC_LOCKED;
@ -2494,7 +2510,11 @@ AVCodec ff_aac_decoder = {
aac_decode_frame, aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (const enum AVSampleFormat[]) { .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 AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
#endif
}, },
.channel_layouts = aac_channel_layout, .channel_layouts = aac_channel_layout,
}; };
@ -2514,7 +2534,11 @@ AVCodec ff_aac_latm_decoder = {
.decode = latm_decode_frame, .decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"), .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) { .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 AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
#endif
}, },
.channel_layouts = aac_channel_layout, .channel_layouts = aac_channel_layout,
}; };