X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Flibfdk-aacenc.c;h=2cea58f152e11c7f16065ed7de31cc68d3c09481;hb=081961f819c0b16c7a860d7da7d39f1fd91bd2f0;hp=00397d58ed26f137449f65b85cee11a3f1d29c5f;hpb=511cf612ac979f536fd65e14603a87ca5ad435f3;p=ffmpeg

diff --git a/libavcodec/libfdk-aacenc.c b/libavcodec/libfdk-aacenc.c
index 00397d58ed2..2cea58f152e 100644
--- a/libavcodec/libfdk-aacenc.c
+++ b/libavcodec/libfdk-aacenc.c
@@ -4,19 +4,17 @@
  *
  * This file is part of Libav.
  *
- * Libav 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.
+ * Permission to use, copy, modify, and/or distribute this software for any
+ * purpose with or without fee is hereby granted, provided that the above
+ * copyright notice and this permission notice appear in all copies.
  *
- * Libav 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
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ * THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
+ * WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
+ * ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
+ * WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
+ * ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
+ * OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
  */
 
 #include <fdk-aac/aacenc_lib.h>
@@ -97,9 +95,6 @@ static int aac_encode_close(AVCodecContext *avctx)
 
     if (s->handle)
         aacEncClose(&s->handle);
-#if FF_API_OLD_ENCODE_AUDIO
-    av_freep(&avctx->coded_frame);
-#endif
     av_freep(&avctx->extradata);
     ff_af_queue_close(&s->afq);
 
@@ -154,6 +149,20 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     case 4: mode = MODE_1_2_1;   sce = 2; cpe = 1; break;
     case 5: mode = MODE_1_2_2;   sce = 1; cpe = 2; break;
     case 6: mode = MODE_1_2_2_1; sce = 2; cpe = 2; break;
+/* The version macro is introduced the same time as the 7.1 support, so this
+   should suffice. */
+#ifdef AACENCODER_LIB_VL0
+    case 8:
+        sce = 2;
+        cpe = 3;
+        if (avctx->channel_layout == AV_CH_LAYOUT_7POINT1) {
+            mode = MODE_7_1_REAR_SURROUND;
+        } else {
+            // MODE_1_2_2_2_1 and MODE_7_1_FRONT_CENTER use the same channel layout
+            mode = MODE_7_1_FRONT_CENTER;
+        }
+        break;
+#endif
     default:
         av_log(avctx, AV_LOG_ERROR,
                "Unsupported number of channels %d\n", avctx->channels);
@@ -175,7 +184,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
         goto error;
     }
 
-    if (avctx->flags & CODEC_FLAG_QSCALE || s->vbr) {
+    if (avctx->flags & AV_CODEC_FLAG_QSCALE || s->vbr) {
         int mode = s->vbr ? s->vbr : avctx->global_quality;
         if (mode <  1 || mode > 5) {
             av_log(avctx, AV_LOG_WARNING,
@@ -200,6 +209,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
             avctx->bit_rate = (96*sce + 128*cpe) * avctx->sample_rate / 44;
             if (avctx->profile == FF_PROFILE_AAC_HE ||
                 avctx->profile == FF_PROFILE_AAC_HE_V2 ||
+                avctx->profile == FF_PROFILE_MPEG2_AAC_HE ||
                 s->eld_sbr)
                 avctx->bit_rate /= 2;
         }
@@ -214,7 +224,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     /* Choose bitstream format - if global header is requested, use
      * raw access units, otherwise use ADTS. */
     if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX,
-                                   avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : s->latm ? 10 : 2)) != AACENC_OK) {
+                                   avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER ? 0 : s->latm ? 10 : 2)) != AACENC_OK) {
         av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",
                aac_get_error(err));
         goto error;
@@ -233,7 +243,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
      * if using mp4 mode (raw access units, with global header) and
      * implicit signaling if using ADTS. */
     if (s->signaling < 0)
-        s->signaling = avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 2 : 0;
+        s->signaling = avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER ? 2 : 0;
 
     if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE,
                                    s->signaling)) != AACENC_OK) {
@@ -250,7 +260,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     }
 
     if (avctx->cutoff > 0) {
-        if (avctx->cutoff < (avctx->sample_rate + 255) >> 8) {
+        if (avctx->cutoff < (avctx->sample_rate + 255) >> 8 || avctx->cutoff > 20000) {
             av_log(avctx, AV_LOG_ERROR, "cutoff valid range is %d-20000\n",
                    (avctx->sample_rate + 255) >> 8);
             goto error;
@@ -275,21 +285,14 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
         goto error;
     }
 
-#if FF_API_OLD_ENCODE_AUDIO
-    avctx->coded_frame = avcodec_alloc_frame();
-    if (!avctx->coded_frame) {
-        ret = AVERROR(ENOMEM);
-        goto error;
-    }
-#endif
     avctx->frame_size = info.frameLength;
-    avctx->delay      = info.encoderDelay;
+    avctx->initial_padding = info.encoderDelay;
     ff_af_queue_init(avctx, &s->afq);
 
-    if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
+    if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) {
         avctx->extradata_size = info.confSize;
         avctx->extradata      = av_mallocz(avctx->extradata_size +
-                                           FF_INPUT_BUFFER_PADDING_SIZE);
+                                           AV_INPUT_BUFFER_PADDING_SIZE);
         if (!avctx->extradata) {
             ret = AVERROR(ENOMEM);
             goto error;
@@ -334,7 +337,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         in_buf.bufElSizes        = &in_buffer_element_size;
 
         /* add current frame to the queue */
-        if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
+        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
             return ret;
     }
 
@@ -395,6 +398,10 @@ static const uint64_t aac_channel_layout[] = {
     AV_CH_LAYOUT_4POINT0,
     AV_CH_LAYOUT_5POINT0_BACK,
     AV_CH_LAYOUT_5POINT1_BACK,
+#ifdef AACENCODER_LIB_VL0
+    AV_CH_LAYOUT_7POINT1_WIDE_BACK,
+    AV_CH_LAYOUT_7POINT1,
+#endif
     0,
 };
 
@@ -405,16 +412,16 @@ static const int aac_sample_rates[] = {
 
 AVCodec ff_libfdk_aac_encoder = {
     .name                  = "libfdk_aac",
+    .long_name             = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"),
     .type                  = AVMEDIA_TYPE_AUDIO,
     .id                    = AV_CODEC_ID_AAC,
     .priv_data_size        = sizeof(AACContext),
     .init                  = aac_encode_init,
     .encode2               = aac_encode_frame,
     .close                 = aac_encode_close,
-    .capabilities          = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
+    .capabilities          = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
     .sample_fmts           = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                             AV_SAMPLE_FMT_NONE },
-    .long_name             = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"),
     .priv_class            = &aac_enc_class,
     .defaults              = aac_encode_defaults,
     .profiles              = profiles,