Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis

decoders.

Based on patches by clsid2 in ffdshow-tryout.

1 files changed, 24 insertions, 10 deletions

diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 5f9dd834a0..f2d50f4aba 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -186,7 +186,7 @@ static av_cold int che_configure(AACContext *ac,
     if (che_pos[type][id]) {
         if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
             return AVERROR(ENOMEM);
-        ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
+        ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
         if (type != TYPE_CCE) {
             ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
             if (type == TYPE_CPE ||
@@ -546,6 +546,7 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
 static av_cold int aac_decode_init(AVCodecContext *avctx)
 {
     AACContext *ac = avctx->priv_data;
+    float output_scale_factor;
 
     ac->avctx = avctx;
     ac->m4ac.sample_rate = avctx->sample_rate;
@@ -557,7 +558,13 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
             return -1;
     }
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+        output_scale_factor = 1.0 / 32768.0;
+    } else {
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+        output_scale_factor = 1.0;
+    }
 
     AAC_INIT_VLC_STATIC( 0, 304);
     AAC_INIT_VLC_STATIC( 1, 270);
@@ -585,9 +592,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
                     ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
                     352);
 
-    ff_mdct_init(&ac->mdct,       11, 1, 1.0/1024.0);
-    ff_mdct_init(&ac->mdct_small,  8, 1, 1.0/128.0);
-    ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0);
+    ff_mdct_init(&ac->mdct,       11, 1, output_scale_factor/1024.0);
+    ff_mdct_init(&ac->mdct_small,  8, 1, output_scale_factor/128.0);
+    ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0/output_scale_factor);
     // window initialization
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
@@ -2169,7 +2176,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
         avctx->frame_size = samples;
     }
 
-    data_size_tmp = samples * avctx->channels * sizeof(int16_t);
+    data_size_tmp = samples * avctx->channels *
+                    (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
     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",
@@ -2178,8 +2186,14 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     }
     *data_size = data_size_tmp;
 
-    if (samples)
-        ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
+    if (samples) {
+        if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
+            ac->fmt_conv.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)
         ac->output_configured = OC_LOCKED;
@@ -2497,7 +2511,7 @@ AVCodec ff_aac_decoder = {
     aac_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
     .sample_fmts = (const enum AVSampleFormat[]) {
-        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
     },
     .channel_layouts = aac_channel_layout,
 };
@@ -2517,7 +2531,7 @@ 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[]) {
-        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
     },
     .channel_layouts = aac_channel_layout,
 };