Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavcodec / aacenc.c

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 6021c375bb4709c438b5e5892ae1b5efe21f7d4a..234ebff39aaf467bfb70cab05d6cd5d0d8eb762b 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
  */
 
@@ -145,7 +145,7 @@ static const uint8_t aac_chan_configs[6][5] = {
 };
 
 /**
- * Table to remap channels from Libav's default order to AAC order.
+ * Table to remap channels from libavcodec's default order to AAC order.
  */
 static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
     { 0 },
@@ -475,7 +475,7 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
 
 /*
  * Deinterleave input samples.
- * Channels are reordered from Libav's default order to AAC order.
+ * Channels are reordered from libavcodec's default order to AAC order.
  */
 static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
 {
@@ -571,11 +571,10 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
         start_ch += chans;
     }
-    if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) {
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 768 * s->channels))) {
         av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
         return ret;
     }
-
     do {
         int frame_bits;
 
@@ -776,7 +775,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
         goto fail;
     s->psypp = ff_psy_preprocess_init(avctx);
-    s->coder = &ff_aac_coders[2];
+    s->coder = &ff_aac_coders[s->options.aac_coder];
 
     s->lambda = avctx->global_quality ? avctx->global_quality : 120;
 
@@ -800,6 +799,7 @@ static const AVOption aacenc_options[] = {
         {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.dbl = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
         {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.dbl =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
         {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.dbl =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+    {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.dbl = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS},
     {NULL}
 };
 
@@ -818,6 +818,7 @@ AVCodec ff_aac_encoder = {
     .init           = aac_encode_init,
     .encode2        = aac_encode_frame,
     .close          = aac_encode_end,
+    .supported_samplerates = avpriv_mpeg4audio_sample_rates,
     .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
                       CODEC_CAP_EXPERIMENTAL,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,