From 361fa0ed40a042393a2691e3dba9bd7c4bcfe188 Mon Sep 17 00:00:00 2001
From: clsid2 <clsid2@3b938f2f-1a1a-0410-8054-a526ea5ff92c>
Date: Mon, 7 Mar 2011 00:28:50 +0000
Subject: [PATCH] Float output for libavcodec AAC decoder
git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3770 3b938f2f-1a1a-0410-8054-a526ea5ff92c
---
libavcodec/aacdec.c | 26 +++++++++++++++++++++++++-
1 file changed, 25 insertions(+), 1 deletion(-)
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 1399eda126a..e7b312c4154 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,
};
--
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