[FFmpeg-cvslog] aacdec: Allow selecting float output at runtime.

[Reimar Döffinger]

ffmpeg | branch: master | Reimar Döffinger <Reimar.Doeffinger at gmx.de> | Mon Apr 25 12:16:40 2011 +0200| [26d5a4b6b496dce0573bd0f5e4af5150899eb3ec] | committer: Reimar Döffinger

aacdec: Allow selecting float output at runtime.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=26d5a4b6b496dce0573bd0f5e4af5150899eb3ec
---

 libavcodec/aacdec.c |   38 ++++++++++----------------------------
 1 files changed, 10 insertions(+), 28 deletions(-)

diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 7b1e501..96b1323 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -557,12 +557,8 @@ 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
+    avctx->sample_fmt = avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
+                        AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
 
     AAC_INIT_VLC_STATIC( 0, 304);
     AAC_INIT_VLC_STATIC( 1, 270);
@@ -2179,12 +2175,8 @@ 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
+    data_size_tmp = samples * avctx->channels;
+    data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(float) : sizeof(int16_t);
     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",
@@ -2194,12 +2186,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     *data_size = data_size_tmp;
 
     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 (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+            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);
     }
 
     if (ac->output_configured)
@@ -2518,11 +2508,7 @@ 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
+        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
     },
     .channel_layouts = aac_channel_layout,
 };
@@ -2542,11 +2528,7 @@ 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
+        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
     },
     .channel_layouts = aac_channel_layout,
 };