[FFmpeg-soc] [soc]: r2280 - in aacenc: aac_enc.patch aacenc.c
kostya
subversion at mplayerhq.hu
Sat May 31 10:11:18 CEST 2008
Author: kostya
Date: Sat May 31 10:11:18 2008
New Revision: 2280
Log:
Hier ist der Drachen.
Decoder sceleton which can produce dummy AAC files.
Added:
aacenc/aac_enc.patch
aacenc/aacenc.c
Added: aacenc/aac_enc.patch
==============================================================================
--- (empty file)
+++ aacenc/aac_enc.patch Sat May 31 10:11:18 2008
@@ -0,0 +1,140 @@
+diff --git a/libavcodec/Makefile b/libavcodec/Makefile
+index d4f6d1c..0ed9057 100644
+--- a/libavcodec/Makefile
++++ b/libavcodec/Makefile
+@@ -29,6 +29,7 @@ HEADERS = avcodec.h opt.h
+
+ OBJS-$(CONFIG_ENCODERS) += faandct.o jfdctfst.o jfdctint.o
+
++OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o mdct.o fft.o mpeg4audio.o
+ OBJS-$(CONFIG_AASC_DECODER) += aasc.o
+ OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3tab.o ac3.o mdct.o fft.o
+ OBJS-$(CONFIG_AC3_ENCODER) += ac3enc.o ac3tab.o ac3.o
+diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
+new file mode 100644
+index 0000000..1c44007
+--- /dev/null
++++ b/libavcodec/aacenc.c
+@@ -0,0 +1,110 @@
++/*
++ * AAC encoder
++ * Copyright (C) 2008 Konstantin Shishkov
++ *
++ * This file is part of FFmpeg.
++ *
++ * 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.
++ *
++ * 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 FFmpeg; if not, write to the Free Software
++ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
++ */
++
++/**
++ * @file aacenc.c
++ * AAC encoder
++ */
++
++#include "avcodec.h"
++#include "bitstream.h"
++#include "dsputil.h"
++#include "mpeg4audio.h"
++
++typedef struct {
++ PutBitContext pb;
++ MDCTContext mdct;
++ DECLARE_ALIGNED_16(float, kbd_long_1024[1024]);
++} AACEncContext;
++
++/**
++ * Make AAC audio config object.
++ * @see 1.6.2.1
++ */
++static int put_audio_specific_config(AVCodecContext *avctx)
++{
++ PutBitContext pb;
++ init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
++ put_bits(&pb, 5, 2); //object type - AAC-LC
++ for(i = 0; i < 16; i++)
++ if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
++ break;
++ if(i == 16){
++ av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
++ return -1;
++ }
++ put_bits(&pb, 4, i); //sample rate index
++ put_bits(&pb, 4, avctx->channels); //channel config - stereo
++ //GASpecificConfig
++ put_bits(&pb, 1, 0); //frame length - 1024 samples
++ put_bits(&pb, 1, 0); //does not depend on core coder
++ put_bits(&pb, 1, 0); //is not extension
++ flush_put_bits(&pb);
++}
++
++static int aac_encode_init(AVCodecContext *avctx)
++{
++ AACEncContext *c = avctx->priv_data;
++ int i;
++
++ avctx->frame_size = 1024;
++
++ ff_mdct_init(&c->mdct, 11, 1);
++ // windows init
++ ff_kbd_window_init(c->kbd_long_1024, 4.0, 1024);
++
++ // produce AAC audio config object
++ // see 1.6.2.1
++ avctx->extradata = av_malloc(2);
++ avctx->extradata_size = 2;
++ put_audio_specific_config(avctx);
++ return 0;
++}
++
++static int aac_encode_frame(AVCodecContext *avctx,
++ uint8_t *frame, int buf_size, void *data)
++{
++ int i,k,channel;
++ AACEncContext *s = avctx->priv_data;
++ int16_t *samples = data;
++
++ init_put_bits(&s->pb, frame, buf_size*8);
++ put_bits(&s->pb, 8, 0xAB);
++
++ flush_put_bits(&s->pb);
++ return put_bits_count(&s->pb)>>3;
++}
++
++static void sine_window_init(float *window, int n) {
++ const float alpha = M_PI / n;
++ int i;
++ for(i = 0; i < n/2; i++)
++ window[i] = sin((i + 0.5) * alpha);
++}
++
++AVCodec aac_encoder = {
++ "aac",
++ CODEC_TYPE_AUDIO,
++ CODEC_ID_AAC,
++ sizeof(AACEncContext),
++ aac_encode_init,
++ aac_encode_frame,
++};
+diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
+index 33a4242..6871496 100644
+--- a/libavcodec/allcodecs.c
++++ b/libavcodec/allcodecs.c
+@@ -178,6 +178,7 @@ void avcodec_register_all(void)
+ REGISTER_ENCDEC (ZMBV, zmbv);
+
+ /* audio codecs */
++ REGISTER_ENCODER (AAC, aac);
+ REGISTER_ENCDEC (AC3, ac3);
+ REGISTER_DECODER (ALAC, alac);
+ REGISTER_DECODER (APE, ape);
Added: aacenc/aacenc.c
==============================================================================
--- (empty file)
+++ aacenc/aacenc.c Sat May 31 10:11:18 2008
@@ -0,0 +1,101 @@
+/*
+ * AAC encoder
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file aacenc.c
+ * AAC encoder
+ */
+
+#include "avcodec.h"
+#include "bitstream.h"
+#include "dsputil.h"
+#include "mpeg4audio.h"
+
+typedef struct {
+ PutBitContext pb;
+ MDCTContext mdct;
+ DECLARE_ALIGNED_16(float, kbd_long_1024[1024]);
+} AACEncContext;
+
+/**
+ * Make AAC audio config object.
+ * @see 1.6.2.1
+ */
+static int put_audio_specific_config(AVCodecContext *avctx)
+{
+ PutBitContext pb;
+ init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
+ put_bits(&pb, 5, 2); //object type - AAC-LC
+ for(i = 0; i < 16; i++)
+ if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
+ break;
+ if(i == 16){
+ av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
+ return -1;
+ }
+ put_bits(&pb, 4, i); //sample rate index
+ put_bits(&pb, 4, avctx->channels); //channel config - stereo
+ //GASpecificConfig
+ put_bits(&pb, 1, 0); //frame length - 1024 samples
+ put_bits(&pb, 1, 0); //does not depend on core coder
+ put_bits(&pb, 1, 0); //is not extension
+ flush_put_bits(&pb);
+}
+
+static int aac_encode_init(AVCodecContext *avctx)
+{
+ AACEncContext *c = avctx->priv_data;
+ int i;
+
+ avctx->frame_size = 1024;
+
+ ff_mdct_init(&c->mdct, 11, 1);
+ // window init
+ ff_kbd_window_init(c->kbd_long_1024, 4.0, 1024);
+
+ avctx->extradata = av_malloc(2);
+ avctx->extradata_size = 2;
+ put_audio_specific_config(avctx);
+ return 0;
+}
+
+static int aac_encode_frame(AVCodecContext *avctx,
+ uint8_t *frame, int buf_size, void *data)
+{
+ int i,k,channel;
+ AACEncContext *s = avctx->priv_data;
+ int16_t *samples = data;
+
+ init_put_bits(&s->pb, frame, buf_size*8);
+ put_bits(&s->pb, 8, 0xAB);
+
+ flush_put_bits(&s->pb);
+ return put_bits_count(&s->pb)>>3;
+}
+
+AVCodec aac_encoder = {
+ "aac",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_AAC,
+ sizeof(AACEncContext),
+ aac_encode_init,
+ aac_encode_frame,
+};
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