mjpegdec: print nb_components too in pict debug info

[ffmpeg] / libavcodec / aacenc.c

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 9e876ff5dc1d5dab2338ebc0ae75060da3dd5afb..e60a778c6d1dce45c6305274cd79717c9e75e2e6 100644 (file)
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -2,20 +2,20 @@
  * AAC encoder
  * Copyright (C) 2008 Konstantin Shishkov
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -135,6 +135,15 @@ static const uint8_t aac_chan_configs[6][5] = {
  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
 };
 
+static const uint8_t channel_maps[][AAC_MAX_CHANNELS] = {
+    { 0 },
+    { 0, 1 },
+    { 2, 0, 1 },
+    { 2, 0, 1, 3 },
+    { 2, 0, 1, 3, 4 },
+    { 2, 0, 1, 4, 5, 3 },
+};
+
 /**
  * Make AAC audio config object.
  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
@@ -165,6 +174,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     AACEncContext *s = avctx->priv_data;
     int i;
     const uint8_t *sizes[2];
+    uint8_t grouping[AAC_MAX_CHANNELS];
     int lengths[2];
 
     avctx->frame_size = 1024;
@@ -210,7 +220,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     sizes[1]   = swb_size_128[i];
     lengths[0] = ff_aac_num_swb_1024[i];
     lengths[1] = ff_aac_num_swb_128[i];
-    ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], &s->chan_map[1]);
+    for (i = 0; i < s->chan_map[0]; i++)
+        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
+    ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping);
     s->psypp = ff_psy_preprocess_init(avctx);
     s->coder = &ff_aac_coders[2];
 
@@ -499,15 +511,24 @@ static int aac_encode_frame(AVCodecContext *avctx,
         return 0;
     if (data) {
         if (!s->psypp) {
-            memcpy(s->samples + 1024 * avctx->channels, data,
-                   1024 * avctx->channels * sizeof(s->samples[0]));
+            if (avctx->channels <= 2) {
+                memcpy(s->samples + 1024 * avctx->channels, data,
+                       1024 * avctx->channels * sizeof(s->samples[0]));
+            } else {
+                for (i = 0; i < 1024; i++)
+                    for (ch = 0; ch < avctx->channels; ch++)
+                        s->samples[(i + 1024) * avctx->channels + ch] =
+                            ((int16_t*)data)[i * avctx->channels +
+                                             channel_maps[avctx->channels-1][ch]];
+            }
         } else {
             start_ch = 0;
             samples2 = s->samples + 1024 * avctx->channels;
             for (i = 0; i < s->chan_map[0]; i++) {
                 tag = s->chan_map[i+1];
                 chans = tag == TYPE_CPE ? 2 : 1;
-                ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
+                ff_psy_preprocess(s->psypp,
+                                  (uint16_t*)data + channel_maps[avctx->channels-1][start_ch],
                                   samples2 + start_ch, start_ch, chans);
                 start_ch += chans;
             }
@@ -537,6 +558,12 @@ static int aac_encode_frame(AVCodecContext *avctx,
                 wi[ch].window_shape   = 0;
                 wi[ch].num_windows    = 1;
                 wi[ch].grouping[0]    = 1;
+
+                /* Only the lowest 12 coefficients are used in a LFE channel.
+                 * The expression below results in only the bottom 8 coefficients
+                 * being used for 11.025kHz to 16kHz sample rates.
+                 */
+                ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
             } else {
                 wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
                                               ics->window_sequence[0]);
@@ -547,7 +574,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
             ics->use_kb_window[0]   = wi[ch].window_shape;
             ics->num_windows        = wi[ch].num_windows;
             ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
-            ics->num_swb            = tag == TYPE_LFE ? 12 : s->psy.num_bands[ics->num_windows == 8];
+            ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
             for (w = 0; w < ics->num_windows; w++)
                 ics->group_len[w] = wi[ch].grouping[w];