cosmetics: indentation after last commit

[ffmpeg] / libavcodec / ac3dec.c

diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 4a177aae698dc377acca05025040852b5e06e27b..e2f0ff97ecae081f96a4a3b328c922ff023bef1b 100644 (file)
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -285,6 +285,7 @@ static int parse_frame_header(AC3DecodeContext *s)
     /* get decoding parameters from header info */
     s->bit_alloc_params.sr_code     = hdr.sr_code;
     s->channel_mode                 = hdr.channel_mode;
+    s->channel_layout               = hdr.channel_layout;
     s->lfe_on                       = hdr.lfe_on;
     s->bit_alloc_params.sr_shift    = hdr.sr_shift;
     s->sample_rate                  = hdr.sample_rate;
@@ -1234,6 +1235,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
     int16_t *out_samples = (int16_t *)data;
     int blk, ch, err;
     const uint8_t *channel_map;
+    const float *output[AC3_MAX_CHANNELS];
 
     /* initialize the GetBitContext with the start of valid AC-3 Frame */
     if (s->input_buffer) {
@@ -1307,8 +1309,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
                 avctx->request_channels < s->channels) {
             s->out_channels = avctx->request_channels;
             s->output_mode  = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
+            s->channel_layout = ff_ac3_channel_layout_tab[s->output_mode];
         }
         avctx->channels = s->out_channels;
+        avctx->channel_layout = s->channel_layout;
 
         /* set downmixing coefficients if needed */
         if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
@@ -1323,14 +1327,13 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
 
     /* decode the audio blocks */
     channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
+    for (ch = 0; ch < s->out_channels; ch++)
+        output[ch] = s->output[channel_map[ch]];
     for (blk = 0; blk < s->num_blocks; blk++) {
-        const float *output[s->out_channels];
         if (!err && decode_audio_block(s, blk)) {
             av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
             err = 1;
         }
-        for (ch = 0; ch < s->out_channels; ch++)
-            output[ch] = s->output[channel_map[ch]];
         s->dsp.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
         out_samples += 256 * s->out_channels;
     }