[FFmpeg-cvslog] Merge commit 'b5f19f7478492307e4b4763aeac3180faf50e17f'
James Almer
git at videolan.org
Tue Oct 31 00:13:24 EET 2017
ffmpeg | branch: master | James Almer <jamrial at gmail.com> | Mon Oct 30 18:56:45 2017 -0300| [b9d3def9b2cb77eb83542086aa3ac883b4d7efa4] | committer: James Almer
Merge commit 'b5f19f7478492307e4b4763aeac3180faf50e17f'
* commit 'b5f19f7478492307e4b4763aeac3180faf50e17f':
aac: Split function to parse ADTS header data into public and private part
Merged-by: James Almer <jamrial at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b9d3def9b2cb77eb83542086aa3ac883b4d7efa4
---
configure | 8 ++++--
doc/developer.texi | 2 +-
libavcodec/Makefile | 13 +++++----
libavcodec/aac_adtstoasc_bsf.c | 13 +++++----
libavcodec/aac_parser.c | 10 ++++---
libavcodec/aacdec.c | 2 +-
libavcodec/aacdec_fixed.c | 2 +-
libavcodec/aacdec_template.c | 2 +-
libavcodec/{aacadtsdec.c => adts_header.c} | 7 +++--
libavcodec/{aacadtsdec.h => adts_header.h} | 12 +++-----
libavcodec/adts_parser.c | 44 ++++++++++++++++++++++++++++++
libavcodec/adts_parser.h | 37 +++++++++++++++++++++++++
libavformat/spdifdec.c | 24 +++++++++-------
libavformat/spdifenc.c | 17 ++++++------
14 files changed, 141 insertions(+), 52 deletions(-)
diff --git a/configure b/configure
index 3380c87432..0414347e3b 100755
--- a/configure
+++ b/configure
@@ -2135,6 +2135,7 @@ HAVE_LIST="
CONFIG_EXTRA="
aandcttables
ac3dsp
+ adts_header
audio_frame_queue
audiodsp
blockdsp
@@ -2421,7 +2422,7 @@ vc1dsp_select="h264chroma qpeldsp startcode"
rdft_select="fft"
# decoders / encoders
-aac_decoder_select="mdct15 mdct sinewin"
+aac_decoder_select="adts_header mdct15 mdct sinewin"
aac_fixed_decoder_select="mdct sinewin"
aac_encoder_select="audio_frame_queue iirfilter lpc mdct sinewin"
aac_latm_decoder_select="aac_decoder aac_latm_parser"
@@ -2873,6 +2874,7 @@ vp9_v4l2m2m_decoder_deps="v4l2_m2m vp9_v4l2_m2m"
wmv3_crystalhd_decoder_select="crystalhd"
# parsers
+aac_parser_select="adts_header"
h264_parser_select="golomb h264dsp h264parse"
hevc_parser_select="hevcparse"
mpegaudio_parser_select="mpegaudioheader"
@@ -2881,6 +2883,7 @@ mpeg4video_parser_select="h263dsp mpegvideo qpeldsp"
vc1_parser_select="vc1dsp"
# bitstream_filters
+aac_adtstoasc_bsf_select="adts_header"
h264_metadata_bsf_select="cbs_h264"
h264_redundant_pps_bsf_select="cbs_h264"
hevc_metadata_bsf_select="cbs_h265"
@@ -3047,7 +3050,8 @@ sap_demuxer_select="sdp_demuxer"
sap_muxer_select="rtp_muxer rtp_protocol rtpenc_chain"
sdp_demuxer_select="rtpdec"
smoothstreaming_muxer_select="ismv_muxer"
-spdif_muxer_select="aac_parser"
+spdif_demuxer_select="adts_header"
+spdif_muxer_select="adts_header"
spx_muxer_select="ogg_muxer"
swf_demuxer_suggest="zlib"
tak_demuxer_select="tak_parser"
diff --git a/doc/developer.texi b/doc/developer.texi
index 98540c8f99..a7b4f1d737 100644
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -184,7 +184,7 @@ e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
- at samp{avpriv_aac_parse_header}.
+ at samp{avpriv_report_missing_feature}.
@item
Each library has its own prefix for public symbols, in addition to the
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index e4e7a4adbf..3a33361f33 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -2,6 +2,7 @@ NAME = avcodec
DESC = FFmpeg codec library
HEADERS = ac3_parser.h \
+ adts_parser.h \
avcodec.h \
avdct.h \
avfft.h \
@@ -20,6 +21,7 @@ HEADERS = ac3_parser.h \
xvmc.h \
OBJS = ac3_parser.o \
+ adts_parser.o \
allcodecs.o \
avdct.o \
avpacket.o \
@@ -52,6 +54,7 @@ OBJS = ac3_parser.o \
# subsystems
OBJS-$(CONFIG_AANDCTTABLES) += aandcttab.o
OBJS-$(CONFIG_AC3DSP) += ac3dsp.o ac3.o ac3tab.o
+OBJS-$(CONFIG_ADTS_HEADER) += adts_header.o mpeg4audio.o
OBJS-$(CONFIG_AUDIO_FRAME_QUEUE) += audio_frame_queue.o
OBJS-$(CONFIG_AUDIODSP) += audiodsp.o
OBJS-$(CONFIG_BLOCKDSP) += blockdsp.o
@@ -148,10 +151,10 @@ OBJS-$(CONFIG_ZERO12V_DECODER) += 012v.o
OBJS-$(CONFIG_A64MULTI_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps_float.o \
- aacadtsdec.o mpeg4audio.o kbdwin.o \
+ mpeg4audio.o kbdwin.o \
sbrdsp.o aacpsdsp_float.o cbrt_data.o
OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o aacps_fixed.o \
- aacadtsdec.o mpeg4audio.o kbdwin.o \
+ mpeg4audio.o kbdwin.o \
sbrdsp_fixed.o aacpsdsp_fixed.o cbrt_data_fixed.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o aacenctab.o \
aacpsy.o aactab.o \
@@ -880,7 +883,6 @@ OBJS-$(CONFIG_MXF_MUXER) += dnxhddata.o
OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
OBJS-$(CONFIG_NUT_DEMUXER) += mpegaudiodata.o mpeg4audio.o
OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o
-OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_SPDIF_MUXER) += dca.o
OBJS-$(CONFIG_TAK_DEMUXER) += tak.o
OBJS-$(CONFIG_WEBM_MUXER) += mpeg4audio.o
@@ -957,7 +959,7 @@ OBJS-$(CONFIG_LIBZVBI_TELETEXT_DECODER) += libzvbi-teletextdec.o ass.o
# parsers
OBJS-$(CONFIG_AAC_LATM_PARSER) += latm_parser.o
OBJS-$(CONFIG_AAC_PARSER) += aac_parser.o aac_ac3_parser.o \
- aacadtsdec.o mpeg4audio.o
+ mpeg4audio.o
OBJS-$(CONFIG_AC3_PARSER) += ac3tab.o aac_ac3_parser.o
OBJS-$(CONFIG_ADX_PARSER) += adx_parser.o adx.o
OBJS-$(CONFIG_BMP_PARSER) += bmp_parser.o
@@ -1003,8 +1005,7 @@ OBJS-$(CONFIG_VP9_PARSER) += vp9_parser.o
OBJS-$(CONFIG_XMA_PARSER) += xma_parser.o
# bitstream filters
-OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o aacadtsdec.o \
- mpeg4audio.o
+OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o mpeg4audio.o
OBJS-$(CONFIG_CHOMP_BSF) += chomp_bsf.o
OBJS-$(CONFIG_DUMP_EXTRADATA_BSF) += dump_extradata_bsf.o
OBJS-$(CONFIG_DCA_CORE_BSF) += dca_core_bsf.o
diff --git a/libavcodec/aac_adtstoasc_bsf.c b/libavcodec/aac_adtstoasc_bsf.c
index d92779ed23..49f1f095e6 100644
--- a/libavcodec/aac_adtstoasc_bsf.c
+++ b/libavcodec/aac_adtstoasc_bsf.c
@@ -19,8 +19,9 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "adts_header.h"
+#include "adts_parser.h"
#include "avcodec.h"
-#include "aacadtsdec.h"
#include "bsf.h"
#include "put_bits.h"
#include "get_bits.h"
@@ -52,12 +53,12 @@ static int aac_adtstoasc_filter(AVBSFContext *bsfc, AVPacket *out)
if (bsfc->par_in->extradata && in->size >= 2 && (AV_RB16(in->data) >> 4) != 0xfff)
goto finish;
- if (in->size < AAC_ADTS_HEADER_SIZE)
+ if (in->size < AV_AAC_ADTS_HEADER_SIZE)
goto packet_too_small;
- init_get_bits(&gb, in->data, AAC_ADTS_HEADER_SIZE * 8);
+ init_get_bits(&gb, in->data, AV_AAC_ADTS_HEADER_SIZE * 8);
- if (avpriv_aac_parse_header(&gb, &hdr) < 0) {
+ if (ff_adts_header_parse(&gb, &hdr) < 0) {
av_log(bsfc, AV_LOG_ERROR, "Error parsing ADTS frame header!\n");
ret = AVERROR_INVALIDDATA;
goto fail;
@@ -70,10 +71,10 @@ static int aac_adtstoasc_filter(AVBSFContext *bsfc, AVPacket *out)
goto fail;
}
- in->size -= AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
+ in->size -= AV_AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
if (in->size <= 0)
goto packet_too_small;
- in->data += AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
+ in->data += AV_AAC_ADTS_HEADER_SIZE + 2 * !hdr.crc_absent;
if (!ctx->first_frame_done) {
int pce_size = 0;
diff --git a/libavcodec/aac_parser.c b/libavcodec/aac_parser.c
index 0b868edcb2..b8692625f3 100644
--- a/libavcodec/aac_parser.c
+++ b/libavcodec/aac_parser.c
@@ -22,7 +22,8 @@
#include "parser.h"
#include "aac_ac3_parser.h"
-#include "aacadtsdec.h"
+#include "adts_header.h"
+#include "adts_parser.h"
#include "get_bits.h"
#include "mpeg4audio.h"
@@ -38,9 +39,10 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
} tmp;
tmp.u64 = av_be2ne64(state);
- init_get_bits(&bits, tmp.u8+8-AAC_ADTS_HEADER_SIZE, AAC_ADTS_HEADER_SIZE * 8);
+ init_get_bits(&bits, tmp.u8 + 8 - AV_AAC_ADTS_HEADER_SIZE,
+ AV_AAC_ADTS_HEADER_SIZE * 8);
- if ((size = avpriv_aac_parse_header(&bits, &hdr)) < 0)
+ if ((size = ff_adts_header_parse(&bits, &hdr)) < 0)
return 0;
*need_next_header = 0;
*new_frame_start = 1;
@@ -54,7 +56,7 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
static av_cold int aac_parse_init(AVCodecParserContext *s1)
{
AACAC3ParseContext *s = s1->priv_data;
- s->header_size = AAC_ADTS_HEADER_SIZE;
+ s->header_size = AV_AAC_ADTS_HEADER_SIZE;
s->sync = aac_sync;
return 0;
}
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index fe50871476..44352764a7 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -50,11 +50,11 @@
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
+#include "adts_header.h"
#include "cbrt_data.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
-#include "aacadtsdec.h"
#include "profiles.h"
#include "libavutil/intfloat.h"
diff --git a/libavcodec/aacdec_fixed.c b/libavcodec/aacdec_fixed.c
index d802f3834f..ffd577c789 100644
--- a/libavcodec/aacdec_fixed.c
+++ b/libavcodec/aacdec_fixed.c
@@ -75,11 +75,11 @@
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
+#include "adts_header.h"
#include "cbrt_data.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
-#include "aacadtsdec.h"
#include "profiles.h"
#include "libavutil/intfloat.h"
diff --git a/libavcodec/aacdec_template.c b/libavcodec/aacdec_template.c
index 082cc908d2..6c6cdd84af 100644
--- a/libavcodec/aacdec_template.c
+++ b/libavcodec/aacdec_template.c
@@ -2955,7 +2955,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags, ret;
- size = avpriv_aac_parse_header(gb, &hdr_info);
+ size = ff_adts_header_parse(gb, &hdr_info);
if (size > 0) {
if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
// This is 2 for "VLB " audio in NSV files.
diff --git a/libavcodec/aacadtsdec.c b/libavcodec/adts_header.c
similarity index 93%
rename from libavcodec/aacadtsdec.c
rename to libavcodec/adts_header.c
index d0814ac27e..0889820f8a 100644
--- a/libavcodec/aacadtsdec.c
+++ b/libavcodec/adts_header.c
@@ -22,11 +22,12 @@
*/
#include "aac_ac3_parser.h"
-#include "aacadtsdec.h"
+#include "adts_header.h"
+#include "adts_parser.h"
#include "get_bits.h"
#include "mpeg4audio.h"
-int avpriv_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
+int ff_adts_header_parse(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
{
int size, rdb, ch, sr;
int aot, crc_abs;
@@ -51,7 +52,7 @@ int avpriv_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
skip_bits1(gbc); /* copyright_identification_bit */
skip_bits1(gbc); /* copyright_identification_start */
size = get_bits(gbc, 13); /* aac_frame_length */
- if (size < AAC_ADTS_HEADER_SIZE)
+ if (size < AV_AAC_ADTS_HEADER_SIZE)
return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
skip_bits(gbc, 11); /* adts_buffer_fullness */
diff --git a/libavcodec/aacadtsdec.h b/libavcodec/adts_header.h
similarity index 86%
rename from libavcodec/aacadtsdec.h
rename to libavcodec/adts_header.h
index d0584ef36a..f615f6a9f9 100644
--- a/libavcodec/aacadtsdec.h
+++ b/libavcodec/adts_header.h
@@ -20,14 +20,11 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#ifndef AVCODEC_AACADTSDEC_H
-#define AVCODEC_AACADTSDEC_H
+#ifndef AVCODEC_ADTS_HEADER_H
+#define AVCODEC_ADTS_HEADER_H
-#include <stdint.h>
#include "get_bits.h"
-#define AAC_ADTS_HEADER_SIZE 7
-
typedef struct AACADTSHeaderInfo {
uint32_t sample_rate;
uint32_t samples;
@@ -40,7 +37,6 @@ typedef struct AACADTSHeaderInfo {
} AACADTSHeaderInfo;
/**
- * Parse AAC frame header.
* Parse the ADTS frame header to the end of the variable header, which is
* the first 54 bits.
* @param[in] gbc BitContext containing the first 54 bits of the frame.
@@ -49,6 +45,6 @@ typedef struct AACADTSHeaderInfo {
* -2 if the version element is invalid, -3 if the sample rate
* element is invalid, or -4 if the bit rate element is invalid.
*/
-int avpriv_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr);
+int ff_adts_header_parse(GetBitContext *gbc, AACADTSHeaderInfo *hdr);
-#endif /* AVCODEC_AACADTSDEC_H */
+#endif /* AVCODEC_ADTS_HEADER_H */
diff --git a/libavcodec/adts_parser.c b/libavcodec/adts_parser.c
new file mode 100644
index 0000000000..5c9f8ff6f2
--- /dev/null
+++ b/libavcodec/adts_parser.c
@@ -0,0 +1,44 @@
+/*
+ * 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
+ */
+
+#include "config.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "adts_header.h"
+#include "adts_parser.h"
+
+int av_adts_header_parse(const uint8_t *buf, uint32_t *samples, uint8_t *frames)
+{
+#if CONFIG_ADTS_HEADER
+ GetBitContext gb;
+ AACADTSHeaderInfo hdr;
+ int err = init_get_bits8(&gb, buf, AV_AAC_ADTS_HEADER_SIZE);
+ if (err < 0)
+ return err;
+ err = ff_adts_header_parse(&gb, &hdr);
+ if (err < 0)
+ return err;
+ *samples = hdr.samples;
+ *frames = hdr.num_aac_frames;
+ return 0;
+#else
+ return AVERROR(ENOSYS);
+#endif
+}
diff --git a/libavcodec/adts_parser.h b/libavcodec/adts_parser.h
new file mode 100644
index 0000000000..f85becd131
--- /dev/null
+++ b/libavcodec/adts_parser.h
@@ -0,0 +1,37 @@
+/*
+ * 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
+ */
+
+#ifndef AVCODEC_ADTS_PARSER_H
+#define AVCODEC_ADTS_PARSER_H
+
+#include <stddef.h>
+#include <stdint.h>
+
+#define AV_AAC_ADTS_HEADER_SIZE 7
+
+/**
+ * Extract the number of samples and frames from AAC data.
+ * @param[in] buf pointer to AAC data buffer
+ * @param[out] samples Pointer to where number of samples is written
+ * @param[out] frames Pointer to where number of frames is written
+ * @return Returns 0 on success, error code on failure.
+ */
+int av_adts_header_parse(const uint8_t *buf, uint32_t *samples,
+ uint8_t *frames);
+
+#endif /* AVCODEC_ADTS_PARSER_H */
diff --git a/libavformat/spdifdec.c b/libavformat/spdifdec.c
index f7288376f6..21bfce4226 100644
--- a/libavformat/spdifdec.c
+++ b/libavformat/spdifdec.c
@@ -25,18 +25,22 @@
* @author Anssi Hannula
*/
+#include "libavutil/bswap.h"
+
+#include "libavcodec/ac3.h"
+#include "libavcodec/adts_parser.h"
+
#include "avformat.h"
#include "spdif.h"
-#include "libavcodec/ac3.h"
-#include "libavcodec/aacadtsdec.h"
static int spdif_get_offset_and_codec(AVFormatContext *s,
enum IEC61937DataType data_type,
const char *buf, int *offset,
enum AVCodecID *codec)
{
- AACADTSHeaderInfo aac_hdr;
- GetBitContext gbc;
+ uint32_t samples;
+ uint8_t frames;
+ int ret;
switch (data_type & 0xff) {
case IEC61937_AC3:
@@ -56,13 +60,13 @@ static int spdif_get_offset_and_codec(AVFormatContext *s,
*codec = AV_CODEC_ID_MP3;
break;
case IEC61937_MPEG2_AAC:
- init_get_bits(&gbc, buf, AAC_ADTS_HEADER_SIZE * 8);
- if (avpriv_aac_parse_header(&gbc, &aac_hdr) < 0) {
+ ret = av_adts_header_parse(buf, &samples, &frames);
+ if (ret < 0) {
if (s) /* be silent during a probe */
av_log(s, AV_LOG_ERROR, "Invalid AAC packet in IEC 61937\n");
- return AVERROR_INVALIDDATA;
+ return ret;
}
- *offset = aac_hdr.samples << 2;
+ *offset = samples << 2;
*codec = AV_CODEC_ID_AAC;
break;
case IEC61937_MPEG2_LAYER1_LSF:
@@ -100,7 +104,7 @@ static int spdif_get_offset_and_codec(AVFormatContext *s,
}
/* Largest offset between bursts we currently handle, i.e. AAC with
- aac_hdr.samples = 4096 */
+ samples = 4096 */
#define SPDIF_MAX_OFFSET 16384
static int spdif_probe(AVProbeData *p)
@@ -132,7 +136,7 @@ int ff_spdif_probe(const uint8_t *p_buf, int buf_size, enum AVCodecID *codec)
} else
consecutive_codes = 0;
- if (buf + 4 + AAC_ADTS_HEADER_SIZE > p_buf + buf_size)
+ if (buf + 4 + AV_AAC_ADTS_HEADER_SIZE > p_buf + buf_size)
break;
/* continue probing to find more sync codes */
diff --git a/libavformat/spdifenc.c b/libavformat/spdifenc.c
index b47ec123e8..3a50aebbef 100644
--- a/libavformat/spdifenc.c
+++ b/libavformat/spdifenc.c
@@ -50,9 +50,9 @@
#include "avio_internal.h"
#include "spdif.h"
#include "libavcodec/ac3.h"
+#include "libavcodec/adts_parser.h"
#include "libavcodec/dca.h"
#include "libavcodec/dca_syncwords.h"
-#include "libavcodec/aacadtsdec.h"
#include "libavutil/opt.h"
typedef struct IEC61937Context {
@@ -349,19 +349,18 @@ static int spdif_header_mpeg(AVFormatContext *s, AVPacket *pkt)
static int spdif_header_aac(AVFormatContext *s, AVPacket *pkt)
{
IEC61937Context *ctx = s->priv_data;
- AACADTSHeaderInfo hdr;
- GetBitContext gbc;
+ uint32_t samples;
+ uint8_t frames;
int ret;
- init_get_bits(&gbc, pkt->data, AAC_ADTS_HEADER_SIZE * 8);
- ret = avpriv_aac_parse_header(&gbc, &hdr);
+ ret = av_adts_header_parse(pkt->data, &samples, &frames);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "Wrong AAC file format\n");
- return AVERROR_INVALIDDATA;
+ return ret;
}
- ctx->pkt_offset = hdr.samples << 2;
- switch (hdr.num_aac_frames) {
+ ctx->pkt_offset = samples << 2;
+ switch (frames) {
case 1:
ctx->data_type = IEC61937_MPEG2_AAC;
break;
@@ -373,7 +372,7 @@ static int spdif_header_aac(AVFormatContext *s, AVPacket *pkt)
break;
default:
av_log(s, AV_LOG_ERROR,
- "%"PRIu32" samples in AAC frame not supported\n", hdr.samples);
+ "%"PRIu32" samples in AAC frame not supported\n", samples);
return AVERROR(EINVAL);
}
//TODO Data type dependent info (LC profile/SBR)
======================================================================
diff --cc configure
index 3380c87432,6f696c9ab5..0414347e3b
--- a/configure
+++ b/configure
@@@ -2415,18 -1962,15 +2416,18 @@@ me_cmp_select="fdctdsp idctdsp pixblock
mpeg_er_select="error_resilience"
mpegaudio_select="mpegaudiodsp mpegaudioheader"
mpegaudiodsp_select="dct"
-mpegvideo_select="blockdsp hpeldsp idctdsp me_cmp mpeg_er videodsp"
+mpegvideo_select="blockdsp h264chroma hpeldsp idctdsp me_cmp mpeg_er videodsp"
mpegvideoenc_select="me_cmp mpegvideo pixblockdsp qpeldsp"
vc1dsp_select="h264chroma qpeldsp startcode"
+rdft_select="fft"
# decoders / encoders
- aac_decoder_select="mdct15 mdct sinewin"
-aac_decoder_select="adts_header imdct15 mdct sinewin"
-aac_encoder_select="audio_frame_queue iirfilter mdct sinewin"
++aac_decoder_select="adts_header mdct15 mdct sinewin"
+aac_fixed_decoder_select="mdct sinewin"
+aac_encoder_select="audio_frame_queue iirfilter lpc mdct sinewin"
aac_latm_decoder_select="aac_decoder aac_latm_parser"
ac3_decoder_select="ac3_parser ac3dsp bswapdsp fmtconvert mdct"
+ac3_fixed_decoder_select="ac3_parser ac3dsp bswapdsp mdct"
ac3_encoder_select="ac3dsp audiodsp mdct me_cmp"
ac3_fixed_encoder_select="ac3dsp audiodsp mdct me_cmp"
adpcm_g722_decoder_select="g722dsp"
@@@ -2839,93 -2265,42 +2840,95 @@@ mpeg2_qsv_decoder_select="qsvdec mpeg2_
mpeg2_qsv_encoder_deps="libmfx"
mpeg2_qsv_encoder_select="qsvenc"
mpeg2_vaapi_encoder_deps="VAEncPictureParameterBufferMPEG2"
-mpeg2_vaapi_encoder_select="vaapi_encode"
+mpeg2_vaapi_encoder_select="cbs_mpeg2 vaapi_encode"
+mpeg2_v4l2m2m_decoder_deps="v4l2_m2m mpeg2_v4l2_m2m"
+mpeg4_crystalhd_decoder_select="crystalhd"
+mpeg4_cuvid_decoder_deps="cuda cuvid"
+mpeg4_mediacodec_decoder_deps="mediacodec"
+mpeg4_mmal_decoder_deps="mmal"
mpeg4_omx_encoder_deps="omx"
+mpeg4_v4l2m2m_decoder_deps="v4l2_m2m mpeg4_v4l2_m2m"
+mpeg4_v4l2m2m_encoder_deps="v4l2_m2m mpeg4_v4l2_m2m"
+msmpeg4_crystalhd_decoder_select="crystalhd"
+nvenc_h264_encoder_select="h264_nvenc_encoder"
+nvenc_hevc_encoder_select="hevc_nvenc_encoder"
+vc1_crystalhd_decoder_select="crystalhd"
+vc1_cuvid_decoder_deps="cuda cuvid"
vc1_mmal_decoder_deps="mmal"
-vc1_qsv_decoder_deps="libmfx"
-vc1_qsv_decoder_select="qsvdec vc1_qsv_hwaccel vc1_parser"
+vc1_v4l2m2m_decoder_deps="v4l2_m2m vc1_v4l2_m2m"
+vp8_cuvid_decoder_deps="cuda cuvid"
+vp8_mediacodec_decoder_deps="mediacodec"
vp8_qsv_decoder_deps="libmfx"
vp8_qsv_decoder_select="qsvdec vp8_qsv_hwaccel vp8_parser"
+vp8_rkmpp_decoder_deps="rkmpp"
vp8_vaapi_encoder_deps="VAEncPictureParameterBufferVP8"
vp8_vaapi_encoder_select="vaapi_encode"
+vp8_v4l2m2m_decoder_deps="v4l2_m2m vp8_v4l2_m2m"
+vp8_v4l2m2m_encoder_deps="v4l2_m2m vp8_v4l2_m2m"
+vp9_cuvid_decoder_deps="cuda cuvid"
+vp9_mediacodec_decoder_deps="mediacodec"
+vp9_rkmpp_decoder_deps="rkmpp"
vp9_vaapi_encoder_deps="VAEncPictureParameterBufferVP9"
vp9_vaapi_encoder_select="vaapi_encode"
-
-nvenc_h264_encoder_select="h264_nvenc_encoder"
-nvenc_hevc_encoder_select="hevc_nvenc_encoder"
+vp9_v4l2m2m_decoder_deps="v4l2_m2m vp9_v4l2_m2m"
+wmv3_crystalhd_decoder_select="crystalhd"
# parsers
+ aac_parser_select="adts_header"
h264_parser_select="golomb h264dsp h264parse"
-hevc_parser_select="hevc_ps"
+hevc_parser_select="hevcparse"
mpegaudio_parser_select="mpegaudioheader"
mpegvideo_parser_select="mpegvideo"
-mpeg4video_parser_select="error_resilience h263dsp mpegvideo qpeldsp"
+mpeg4video_parser_select="h263dsp mpegvideo qpeldsp"
vc1_parser_select="vc1dsp"
# bitstream_filters
+ aac_adtstoasc_bsf_select="adts_header"
+h264_metadata_bsf_select="cbs_h264"
+h264_redundant_pps_bsf_select="cbs_h264"
+hevc_metadata_bsf_select="cbs_h265"
mjpeg2jpeg_bsf_select="jpegtables"
+mpeg2_metadata_bsf_select="cbs_mpeg2"
+trace_headers_bsf_select="cbs_h264 cbs_h265 cbs_mpeg2"
# external libraries
-avisynth_deps="LoadLibrary"
-avxsynth_deps="libdl"
-avisynth_demuxer_deps_any="avisynth avxsynth"
+aac_at_decoder_deps="audiotoolbox"
+ac3_at_decoder_deps="audiotoolbox"
+ac3_at_decoder_select="ac3_parser"
+adpcm_ima_qt_at_decoder_deps="audiotoolbox"
+alac_at_decoder_deps="audiotoolbox"
+amr_nb_at_decoder_deps="audiotoolbox"
+avisynth_deps_any="libdl LoadLibrary"
+avisynth_demuxer_deps="avisynth"
avisynth_demuxer_select="riffdec"
-libdcadec_decoder_deps="libdcadec"
-libfaac_encoder_deps="libfaac"
-libfaac_encoder_select="audio_frame_queue"
+eac3_at_decoder_deps="audiotoolbox"
+eac3_at_decoder_select="ac3_parser"
+gsm_ms_at_decoder_deps="audiotoolbox"
+ilbc_at_decoder_deps="audiotoolbox"
+mp1_at_decoder_deps="audiotoolbox"
+mp2_at_decoder_deps="audiotoolbox"
+mp3_at_decoder_deps="audiotoolbox"
+mp1_at_decoder_select="mpegaudioheader"
+mp2_at_decoder_select="mpegaudioheader"
+mp3_at_decoder_select="mpegaudioheader"
+pcm_alaw_at_decoder_deps="audiotoolbox"
+pcm_mulaw_at_decoder_deps="audiotoolbox"
+qdmc_at_decoder_deps="audiotoolbox"
+qdm2_at_decoder_deps="audiotoolbox"
+aac_at_encoder_deps="audiotoolbox"
+aac_at_encoder_select="audio_frame_queue"
+alac_at_encoder_deps="audiotoolbox"
+alac_at_encoder_select="audio_frame_queue"
+ilbc_at_encoder_deps="audiotoolbox"
+ilbc_at_encoder_select="audio_frame_queue"
+pcm_alaw_at_encoder_deps="audiotoolbox"
+pcm_alaw_at_encoder_select="audio_frame_queue"
+pcm_mulaw_at_encoder_deps="audiotoolbox"
+pcm_mulaw_at_encoder_select="audio_frame_queue"
+chromaprint_muxer_deps="chromaprint"
+h264_videotoolbox_encoder_deps="pthreads"
+h264_videotoolbox_encoder_select="videotoolbox_encoder"
+libcelt_decoder_deps="libcelt"
libfdk_aac_decoder_deps="libfdk_aac"
libfdk_aac_encoder_deps="libfdk_aac"
libfdk_aac_encoder_select="audio_frame_queue"
diff --cc doc/developer.texi
index 98540c8f99,824fba4591..a7b4f1d737
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@@ -184,16 -218,11 +184,16 @@@ e.g. @samp{ff_w64_demuxer}
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
- @samp{avpriv_aac_parse_header}.
+ @samp{avpriv_report_missing_feature}.
@item
-For externally visible symbols, each library has its own prefix. Check
-the existing code and choose names accordingly.
+Each library has its own prefix for public symbols, in addition to the
+commonly used @code{av_} (@code{avformat_} for libavformat,
+ at code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
+Check the existing code and choose names accordingly.
+Note that some symbols without these prefixes are also exported for
+retro-compatibility reasons. These exceptions are declared in the
+ at code{lib<name>/lib<name>.v} files.
@end itemize
Furthermore, name space reserved for the system should not be invaded.
diff --cc libavcodec/Makefile
index e4e7a4adbf,a1bb4b5950..3a33361f33
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@@ -1,27 -1,24 +1,29 @@@
NAME = avcodec
-DESC = Libav codec library
+DESC = FFmpeg codec library
HEADERS = ac3_parser.h \
+ adts_parser.h \
avcodec.h \
+ avdct.h \
avfft.h \
d3d11va.h \
dirac.h \
dv_profile.h \
dxva2.h \
+ jni.h \
+ mediacodec.h \
qsv.h \
vaapi.h \
- vda.h \
vdpau.h \
version.h \
+ videotoolbox.h \
vorbis_parser.h \
+ xvmc.h \
OBJS = ac3_parser.o \
+ adts_parser.o \
allcodecs.o \
+ avdct.o \
avpacket.o \
avpicture.o \
bitstream.o \
@@@ -51,7 -46,8 +53,8 @@@
# subsystems
OBJS-$(CONFIG_AANDCTTABLES) += aandcttab.o
-OBJS-$(CONFIG_AC3DSP) += ac3dsp.o
+OBJS-$(CONFIG_AC3DSP) += ac3dsp.o ac3.o ac3tab.o
+ OBJS-$(CONFIG_ADTS_HEADER) += adts_header.o mpeg4audio.o
OBJS-$(CONFIG_AUDIO_FRAME_QUEUE) += audio_frame_queue.o
OBJS-$(CONFIG_AUDIODSP) += audiodsp.o
OBJS-$(CONFIG_BLOCKDSP) += blockdsp.o
@@@ -144,25 -126,16 +147,25 @@@ OBJS-$(CONFIG_WMA_FREQS)
OBJS-$(CONFIG_WMV2DSP) += wmv2dsp.o
# decoders/encoders
+OBJS-$(CONFIG_ZERO12V_DECODER) += 012v.o
OBJS-$(CONFIG_A64MULTI_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o
-OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
+OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps_float.o \
- aacadtsdec.o mpeg4audio.o kbdwin.o \
+ mpeg4audio.o kbdwin.o \
- sbrdsp.o aacpsdsp.o
-OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
+ sbrdsp.o aacpsdsp_float.o cbrt_data.o
+OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o aacps_fixed.o \
- aacadtsdec.o mpeg4audio.o kbdwin.o \
++ mpeg4audio.o kbdwin.o \
+ sbrdsp_fixed.o aacpsdsp_fixed.o cbrt_data_fixed.o
+OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o aacenctab.o \
aacpsy.o aactab.o \
- psymodel.o mpeg4audio.o kbdwin.o
+ aacenc_is.o \
+ aacenc_tns.o \
+ aacenc_ltp.o \
+ aacenc_pred.o \
+ psymodel.o mpeg4audio.o kbdwin.o cbrt_data.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
-OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
+OBJS-$(CONFIG_AC3_DECODER) += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o ac3tab.o
+OBJS-$(CONFIG_AC3_FIXED_DECODER) += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o ac3tab.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \
ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
@@@ -876,11 -668,7 +879,10 @@@ OBJS-$(CONFIG_LATM_MUXER)
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_MATROSKA_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_MOV_DEMUXER) += ac3tab.o
+OBJS-$(CONFIG_MXF_MUXER) += dnxhddata.o
OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
+OBJS-$(CONFIG_NUT_DEMUXER) += mpegaudiodata.o mpeg4audio.o
+OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o
- OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_SPDIF_MUXER) += dca.o
OBJS-$(CONFIG_TAK_DEMUXER) += tak.o
OBJS-$(CONFIG_WEBM_MUXER) += mpeg4audio.o
@@@ -999,21 -760,14 +1001,20 @@@ OBJS-$(CONFIG_VC1_PARSER)
simple_idct.o wmv2data.o
OBJS-$(CONFIG_VP3_PARSER) += vp3_parser.o
OBJS-$(CONFIG_VP8_PARSER) += vp8_parser.o
+OBJS-$(CONFIG_VP9_PARSER) += vp9_parser.o
+OBJS-$(CONFIG_XMA_PARSER) += xma_parser.o
# bitstream filters
- OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o aacadtsdec.o \
- mpeg4audio.o
+ OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o mpeg4audio.o
OBJS-$(CONFIG_CHOMP_BSF) += chomp_bsf.o
OBJS-$(CONFIG_DUMP_EXTRADATA_BSF) += dump_extradata_bsf.o
+OBJS-$(CONFIG_DCA_CORE_BSF) += dca_core_bsf.o
OBJS-$(CONFIG_EXTRACT_EXTRADATA_BSF) += extract_extradata_bsf.o \
h2645_parse.o
+OBJS-$(CONFIG_H264_METADATA_BSF) += h264_metadata_bsf.o
OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF) += h264_mp4toannexb_bsf.o
+OBJS-$(CONFIG_H264_REDUNDANT_PPS_BSF) += h264_redundant_pps_bsf.o
+OBJS-$(CONFIG_HEVC_METADATA_BSF) += h265_metadata_bsf.o
OBJS-$(CONFIG_HEVC_MP4TOANNEXB_BSF) += hevc_mp4toannexb_bsf.o
OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF) += imx_dump_header_bsf.o
OBJS-$(CONFIG_MJPEG2JPEG_BSF) += mjpeg2jpeg_bsf.o
diff --cc libavcodec/aac_adtstoasc_bsf.c
index d92779ed23,778387cabb..49f1f095e6
--- a/libavcodec/aac_adtstoasc_bsf.c
+++ b/libavcodec/aac_adtstoasc_bsf.c
@@@ -49,15 -50,15 +50,15 @@@ static int aac_adtstoasc_filter(AVBSFCo
if (ret < 0)
return ret;
+ if (bsfc->par_in->extradata && in->size >= 2 && (AV_RB16(in->data) >> 4) != 0xfff)
+ goto finish;
+
- if (in->size < AAC_ADTS_HEADER_SIZE)
+ if (in->size < AV_AAC_ADTS_HEADER_SIZE)
goto packet_too_small;
- init_get_bits(&gb, in->data, AAC_ADTS_HEADER_SIZE * 8);
+ init_get_bits(&gb, in->data, AV_AAC_ADTS_HEADER_SIZE * 8);
- if (avpriv_aac_parse_header(&gb, &hdr) < 0) {
- if (bsfc->par_in->extradata && show_bits(&gb, 12) != 0xfff)
- goto finish;
-
+ if (ff_adts_header_parse(&gb, &hdr) < 0) {
av_log(bsfc, AV_LOG_ERROR, "Error parsing ADTS frame header!\n");
ret = AVERROR_INVALIDDATA;
goto fail;
diff --cc libavcodec/aacdec.c
index fe50871476,e436b4f2f7..44352764a7
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@@ -50,33 -93,1380 +50,33 @@@
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
+ #include "adts_header.h"
-#include "cbrt_tablegen.h"
+#include "cbrt_data.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
- #include "aacadtsdec.h"
-#include "libavutil/intfloat.h"
-
-#include <assert.h>
-#include <errno.h>
-#include <math.h>
-#include <stdint.h>
-#include <string.h>
-
-#if ARCH_ARM
-# include "arm/aac.h"
-#endif
-
-#include "libavutil/thread.h"
-
-static VLC vlc_scalefactors;
-static VLC vlc_spectral[11];
-
-static const char overread_err[] = "Input buffer exhausted before END element found\n";
-
-static int count_channels(uint8_t (*layout)[3], int tags)
-{
- int i, sum = 0;
- for (i = 0; i < tags; i++) {
- int syn_ele = layout[i][0];
- int pos = layout[i][2];
- sum += (1 + (syn_ele == TYPE_CPE)) *
- (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
- }
- return sum;
-}
-
-/**
- * Check for the channel element in the current channel position configuration.
- * If it exists, make sure the appropriate element is allocated and map the
- * channel order to match the internal Libav channel layout.
- *
- * @param che_pos current channel position configuration
- * @param type channel element type
- * @param id channel element id
- * @param channels count of the number of channels in the configuration
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static av_cold int che_configure(AACContext *ac,
- enum ChannelPosition che_pos,
- int type, int id, int *channels)
-{
- if (che_pos) {
- if (!ac->che[type][id]) {
- if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
- return AVERROR(ENOMEM);
- ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
- }
- if (type != TYPE_CCE) {
- if (*channels >= MAX_CHANNELS - 2)
- return AVERROR_INVALIDDATA;
- ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
- if (type == TYPE_CPE ||
- (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
- ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
- }
- }
- } else {
- if (ac->che[type][id])
- ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
- av_freep(&ac->che[type][id]);
- }
- return 0;
-}
-
-static int frame_configure_elements(AVCodecContext *avctx)
-{
- AACContext *ac = avctx->priv_data;
- int type, id, ch, ret;
-
- /* set channel pointers to internal buffers by default */
- for (type = 0; type < 4; type++) {
- for (id = 0; id < MAX_ELEM_ID; id++) {
- ChannelElement *che = ac->che[type][id];
- if (che) {
- che->ch[0].ret = che->ch[0].ret_buf;
- che->ch[1].ret = che->ch[1].ret_buf;
- }
- }
- }
-
- /* get output buffer */
- av_frame_unref(ac->frame);
- ac->frame->nb_samples = 2048;
- if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
- return ret;
- }
-
- /* map output channel pointers to AVFrame data */
- for (ch = 0; ch < avctx->channels; ch++) {
- if (ac->output_element[ch])
- ac->output_element[ch]->ret = (float *)ac->frame->extended_data[ch];
- }
-
- return 0;
-}
-
-struct elem_to_channel {
- uint64_t av_position;
- uint8_t syn_ele;
- uint8_t elem_id;
- uint8_t aac_position;
-};
-
-static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
- uint8_t (*layout_map)[3], int offset, uint64_t left,
- uint64_t right, int pos)
-{
- if (layout_map[offset][0] == TYPE_CPE) {
- e2c_vec[offset] = (struct elem_to_channel) {
- .av_position = left | right,
- .syn_ele = TYPE_CPE,
- .elem_id = layout_map[offset][1],
- .aac_position = pos
- };
- return 1;
- } else {
- e2c_vec[offset] = (struct elem_to_channel) {
- .av_position = left,
- .syn_ele = TYPE_SCE,
- .elem_id = layout_map[offset][1],
- .aac_position = pos
- };
- e2c_vec[offset + 1] = (struct elem_to_channel) {
- .av_position = right,
- .syn_ele = TYPE_SCE,
- .elem_id = layout_map[offset + 1][1],
- .aac_position = pos
- };
- return 2;
- }
-}
-
-static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
- int *current)
-{
- int num_pos_channels = 0;
- int first_cpe = 0;
- int sce_parity = 0;
- int i;
- for (i = *current; i < tags; i++) {
- if (layout_map[i][2] != pos)
- break;
- if (layout_map[i][0] == TYPE_CPE) {
- if (sce_parity) {
- if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
- sce_parity = 0;
- } else {
- return -1;
- }
- }
- num_pos_channels += 2;
- first_cpe = 1;
- } else {
- num_pos_channels++;
- sce_parity ^= 1;
- }
- }
- if (sce_parity &&
- ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
- return -1;
- *current = i;
- return num_pos_channels;
-}
-
-static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
-{
- int i, n, total_non_cc_elements;
- struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
- int num_front_channels, num_side_channels, num_back_channels;
- uint64_t layout;
-
- if (FF_ARRAY_ELEMS(e2c_vec) < tags)
- return 0;
-
- i = 0;
- num_front_channels =
- count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
- if (num_front_channels < 0)
- return 0;
- num_side_channels =
- count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
- if (num_side_channels < 0)
- return 0;
- num_back_channels =
- count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
- if (num_back_channels < 0)
- return 0;
-
- if (num_side_channels == 0 && num_back_channels >= 4) {
- num_side_channels = 2;
- num_back_channels -= 2;
- }
-
- i = 0;
- if (num_front_channels & 1) {
- e2c_vec[i] = (struct elem_to_channel) {
- .av_position = AV_CH_FRONT_CENTER,
- .syn_ele = TYPE_SCE,
- .elem_id = layout_map[i][1],
- .aac_position = AAC_CHANNEL_FRONT
- };
- i++;
- num_front_channels--;
- }
- if (num_front_channels >= 4) {
- i += assign_pair(e2c_vec, layout_map, i,
- AV_CH_FRONT_LEFT_OF_CENTER,
- AV_CH_FRONT_RIGHT_OF_CENTER,
- AAC_CHANNEL_FRONT);
- num_front_channels -= 2;
- }
- if (num_front_channels >= 2) {
- i += assign_pair(e2c_vec, layout_map, i,
- AV_CH_FRONT_LEFT,
- AV_CH_FRONT_RIGHT,
- AAC_CHANNEL_FRONT);
- num_front_channels -= 2;
- }
- while (num_front_channels >= 2) {
- i += assign_pair(e2c_vec, layout_map, i,
- UINT64_MAX,
- UINT64_MAX,
- AAC_CHANNEL_FRONT);
- num_front_channels -= 2;
- }
-
- if (num_side_channels >= 2) {
- i += assign_pair(e2c_vec, layout_map, i,
- AV_CH_SIDE_LEFT,
- AV_CH_SIDE_RIGHT,
- AAC_CHANNEL_FRONT);
- num_side_channels -= 2;
- }
- while (num_side_channels >= 2) {
- i += assign_pair(e2c_vec, layout_map, i,
- UINT64_MAX,
- UINT64_MAX,
- AAC_CHANNEL_SIDE);
- num_side_channels -= 2;
- }
-
- while (num_back_channels >= 4) {
- i += assign_pair(e2c_vec, layout_map, i,
- UINT64_MAX,
- UINT64_MAX,
- AAC_CHANNEL_BACK);
- num_back_channels -= 2;
- }
- if (num_back_channels >= 2) {
- i += assign_pair(e2c_vec, layout_map, i,
- AV_CH_BACK_LEFT,
- AV_CH_BACK_RIGHT,
- AAC_CHANNEL_BACK);
- num_back_channels -= 2;
- }
- if (num_back_channels) {
- e2c_vec[i] = (struct elem_to_channel) {
- .av_position = AV_CH_BACK_CENTER,
- .syn_ele = TYPE_SCE,
- .elem_id = layout_map[i][1],
- .aac_position = AAC_CHANNEL_BACK
- };
- i++;
- num_back_channels--;
- }
-
- if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
- e2c_vec[i] = (struct elem_to_channel) {
- .av_position = AV_CH_LOW_FREQUENCY,
- .syn_ele = TYPE_LFE,
- .elem_id = layout_map[i][1],
- .aac_position = AAC_CHANNEL_LFE
- };
- i++;
- }
- while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
- e2c_vec[i] = (struct elem_to_channel) {
- .av_position = UINT64_MAX,
- .syn_ele = TYPE_LFE,
- .elem_id = layout_map[i][1],
- .aac_position = AAC_CHANNEL_LFE
- };
- i++;
- }
-
- // Must choose a stable sort
- total_non_cc_elements = n = i;
- do {
- int next_n = 0;
- for (i = 1; i < n; i++)
- if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
- FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
- next_n = i;
- }
- n = next_n;
- } while (n > 0);
-
- layout = 0;
- for (i = 0; i < total_non_cc_elements; i++) {
- layout_map[i][0] = e2c_vec[i].syn_ele;
- layout_map[i][1] = e2c_vec[i].elem_id;
- layout_map[i][2] = e2c_vec[i].aac_position;
- if (e2c_vec[i].av_position != UINT64_MAX) {
- layout |= e2c_vec[i].av_position;
- }
- }
-
- return layout;
-}
-
-/**
- * Save current output configuration if and only if it has been locked.
- */
-static void push_output_configuration(AACContext *ac) {
- if (ac->oc[1].status == OC_LOCKED) {
- ac->oc[0] = ac->oc[1];
- }
- ac->oc[1].status = OC_NONE;
-}
-
-/**
- * Restore the previous output configuration if and only if the current
- * configuration is unlocked.
- */
-static void pop_output_configuration(AACContext *ac) {
- if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
- ac->oc[1] = ac->oc[0];
- ac->avctx->channels = ac->oc[1].channels;
- ac->avctx->channel_layout = ac->oc[1].channel_layout;
- }
-}
-
-/**
- * Configure output channel order based on the current program
- * configuration element.
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static int output_configure(AACContext *ac,
- uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
- enum OCStatus oc_type, int get_new_frame)
-{
- AVCodecContext *avctx = ac->avctx;
- int i, channels = 0, ret;
- uint64_t layout = 0;
- uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
- uint8_t type_counts[TYPE_END] = { 0 };
-
- if (ac->oc[1].layout_map != layout_map) {
- memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
- ac->oc[1].layout_map_tags = tags;
- }
- for (i = 0; i < tags; i++) {
- int type = layout_map[i][0];
- int id = layout_map[i][1];
- id_map[type][id] = type_counts[type]++;
- }
- // Try to sniff a reasonable channel order, otherwise output the
- // channels in the order the PCE declared them.
- if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
- layout = sniff_channel_order(layout_map, tags);
- for (i = 0; i < tags; i++) {
- int type = layout_map[i][0];
- int id = layout_map[i][1];
- int iid = id_map[type][id];
- int position = layout_map[i][2];
- // Allocate or free elements depending on if they are in the
- // current program configuration.
- ret = che_configure(ac, position, type, iid, &channels);
- if (ret < 0)
- return ret;
- ac->tag_che_map[type][id] = ac->che[type][iid];
- }
- if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
- if (layout == AV_CH_FRONT_CENTER) {
- layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
- } else {
- layout = 0;
- }
- }
-
- avctx->channel_layout = ac->oc[1].channel_layout = layout;
- avctx->channels = ac->oc[1].channels = channels;
- ac->oc[1].status = oc_type;
-
- if (get_new_frame) {
- if ((ret = frame_configure_elements(ac->avctx)) < 0)
- return ret;
- }
-
- return 0;
-}
-
-/**
- * Set up channel positions based on a default channel configuration
- * as specified in table 1.17.
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static int set_default_channel_config(AVCodecContext *avctx,
- uint8_t (*layout_map)[3],
- int *tags,
- int channel_config)
-{
- if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
- channel_config > 12) {
- av_log(avctx, AV_LOG_ERROR,
- "invalid default channel configuration (%d)\n",
- channel_config);
- return AVERROR_INVALIDDATA;
- }
- *tags = tags_per_config[channel_config];
- memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
- *tags * sizeof(*layout_map));
- return 0;
-}
-
-static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
-{
- /* For PCE based channel configurations map the channels solely based
- * on tags. */
- if (!ac->oc[1].m4ac.chan_config) {
- return ac->tag_che_map[type][elem_id];
- }
- // Allow single CPE stereo files to be signalled with mono configuration.
- if (!ac->tags_mapped && type == TYPE_CPE &&
- ac->oc[1].m4ac.chan_config == 1) {
- uint8_t layout_map[MAX_ELEM_ID*4][3];
- int layout_map_tags;
- push_output_configuration(ac);
-
- if (set_default_channel_config(ac->avctx, layout_map,
- &layout_map_tags, 2) < 0)
- return NULL;
- if (output_configure(ac, layout_map, layout_map_tags,
- OC_TRIAL_FRAME, 1) < 0)
- return NULL;
-
- ac->oc[1].m4ac.chan_config = 2;
- ac->oc[1].m4ac.ps = 0;
- }
- // And vice-versa
- if (!ac->tags_mapped && type == TYPE_SCE &&
- ac->oc[1].m4ac.chan_config == 2) {
- uint8_t layout_map[MAX_ELEM_ID * 4][3];
- int layout_map_tags;
- push_output_configuration(ac);
-
- if (set_default_channel_config(ac->avctx, layout_map,
- &layout_map_tags, 1) < 0)
- return NULL;
- if (output_configure(ac, layout_map, layout_map_tags,
- OC_TRIAL_FRAME, 1) < 0)
- return NULL;
-
- ac->oc[1].m4ac.chan_config = 1;
- if (ac->oc[1].m4ac.sbr)
- ac->oc[1].m4ac.ps = -1;
- }
- /* For indexed channel configurations map the channels solely based
- * on position. */
- switch (ac->oc[1].m4ac.chan_config) {
- case 12:
- case 7:
- if (ac->tags_mapped == 3 && type == TYPE_CPE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
- }
- case 11:
- if (ac->tags_mapped == 2 &&
- ac->oc[1].m4ac.chan_config == 11 &&
- type == TYPE_SCE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
- }
- case 6:
- /* Some streams incorrectly code 5.1 audio as
- * SCE[0] CPE[0] CPE[1] SCE[1]
- * instead of
- * SCE[0] CPE[0] CPE[1] LFE[0].
- * If we seem to have encountered such a stream, transfer
- * the LFE[0] element to the SCE[1]'s mapping */
- if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
- ac->tags_mapped++;
- return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
- }
- case 5:
- if (ac->tags_mapped == 2 && type == TYPE_CPE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
- }
- case 4:
- if (ac->tags_mapped == 2 &&
- ac->oc[1].m4ac.chan_config == 4 &&
- type == TYPE_SCE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
- }
- case 3:
- case 2:
- if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
- type == TYPE_CPE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
- } else if (ac->oc[1].m4ac.chan_config == 2) {
- return NULL;
- }
- case 1:
- if (!ac->tags_mapped && type == TYPE_SCE) {
- ac->tags_mapped++;
- return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
- }
- default:
- return NULL;
- }
-}
-
-/**
- * Decode an array of 4 bit element IDs, optionally interleaved with a
- * stereo/mono switching bit.
- *
- * @param type speaker type/position for these channels
- */
-static void decode_channel_map(uint8_t layout_map[][3],
- enum ChannelPosition type,
- GetBitContext *gb, int n)
-{
- while (n--) {
- enum RawDataBlockType syn_ele;
- switch (type) {
- case AAC_CHANNEL_FRONT:
- case AAC_CHANNEL_BACK:
- case AAC_CHANNEL_SIDE:
- syn_ele = get_bits1(gb);
- break;
- case AAC_CHANNEL_CC:
- skip_bits1(gb);
- syn_ele = TYPE_CCE;
- break;
- case AAC_CHANNEL_LFE:
- syn_ele = TYPE_LFE;
- break;
- default:
- // AAC_CHANNEL_OFF has no channel map
- return;
- }
- layout_map[0][0] = syn_ele;
- layout_map[0][1] = get_bits(gb, 4);
- layout_map[0][2] = type;
- layout_map++;
- }
-}
-
-/**
- * Decode program configuration element; reference: table 4.2.
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
- uint8_t (*layout_map)[3],
- GetBitContext *gb)
-{
- int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
- int sampling_index;
- int comment_len;
- int tags;
-
- skip_bits(gb, 2); // object_type
-
- sampling_index = get_bits(gb, 4);
- if (m4ac->sampling_index != sampling_index)
- av_log(avctx, AV_LOG_WARNING,
- "Sample rate index in program config element does not "
- "match the sample rate index configured by the container.\n");
-
- num_front = get_bits(gb, 4);
- num_side = get_bits(gb, 4);
- num_back = get_bits(gb, 4);
- num_lfe = get_bits(gb, 2);
- num_assoc_data = get_bits(gb, 3);
- num_cc = get_bits(gb, 4);
-
- if (get_bits1(gb))
- skip_bits(gb, 4); // mono_mixdown_tag
- if (get_bits1(gb))
- skip_bits(gb, 4); // stereo_mixdown_tag
-
- if (get_bits1(gb))
- skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
-
- decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
- tags = num_front;
- decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
- tags += num_side;
- decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
- tags += num_back;
- decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
- tags += num_lfe;
-
- skip_bits_long(gb, 4 * num_assoc_data);
-
- decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
- tags += num_cc;
-
- align_get_bits(gb);
-
- /* comment field, first byte is length */
- comment_len = get_bits(gb, 8) * 8;
- if (get_bits_left(gb) < comment_len) {
- av_log(avctx, AV_LOG_ERROR, overread_err);
- return AVERROR_INVALIDDATA;
- }
- skip_bits_long(gb, comment_len);
- return tags;
-}
-
-/**
- * Decode GA "General Audio" specific configuration; reference: table 4.1.
- *
- * @param ac pointer to AACContext, may be null
- * @param avctx pointer to AVCCodecContext, used for logging
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
- GetBitContext *gb,
- MPEG4AudioConfig *m4ac,
- int channel_config)
-{
- int extension_flag, ret, ep_config, res_flags;
- uint8_t layout_map[MAX_ELEM_ID*4][3];
- int tags = 0;
-
- if (get_bits1(gb)) { // frameLengthFlag
- avpriv_request_sample(avctx, "960/120 MDCT window");
- return AVERROR_PATCHWELCOME;
- }
- m4ac->frame_length_short = 0;
-
- if (get_bits1(gb)) // dependsOnCoreCoder
- skip_bits(gb, 14); // coreCoderDelay
- extension_flag = get_bits1(gb);
-
- if (m4ac->object_type == AOT_AAC_SCALABLE ||
- m4ac->object_type == AOT_ER_AAC_SCALABLE)
- skip_bits(gb, 3); // layerNr
-
- if (channel_config == 0) {
- skip_bits(gb, 4); // element_instance_tag
- tags = decode_pce(avctx, m4ac, layout_map, gb);
- if (tags < 0)
- return tags;
- } else {
- if ((ret = set_default_channel_config(avctx, layout_map,
- &tags, channel_config)))
- return ret;
- }
-
- if (count_channels(layout_map, tags) > 1) {
- m4ac->ps = 0;
- } else if (m4ac->sbr == 1 && m4ac->ps == -1)
- m4ac->ps = 1;
-
- if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
- return ret;
-
- if (extension_flag) {
- switch (m4ac->object_type) {
- case AOT_ER_BSAC:
- skip_bits(gb, 5); // numOfSubFrame
- skip_bits(gb, 11); // layer_length
- break;
- case AOT_ER_AAC_LC:
- case AOT_ER_AAC_LTP:
- case AOT_ER_AAC_SCALABLE:
- case AOT_ER_AAC_LD:
- res_flags = get_bits(gb, 3);
- if (res_flags) {
- avpriv_report_missing_feature(avctx,
- "AAC data resilience (flags %x)",
- res_flags);
- return AVERROR_PATCHWELCOME;
- }
- break;
- }
- skip_bits1(gb); // extensionFlag3 (TBD in version 3)
- }
- switch (m4ac->object_type) {
- case AOT_ER_AAC_LC:
- case AOT_ER_AAC_LTP:
- case AOT_ER_AAC_SCALABLE:
- case AOT_ER_AAC_LD:
- ep_config = get_bits(gb, 2);
- if (ep_config) {
- avpriv_report_missing_feature(avctx,
- "epConfig %d", ep_config);
- return AVERROR_PATCHWELCOME;
- }
- }
- return 0;
-}
-
-static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
- GetBitContext *gb,
- MPEG4AudioConfig *m4ac,
- int channel_config)
-{
- int ret, ep_config, res_flags;
- uint8_t layout_map[MAX_ELEM_ID*4][3];
- int tags = 0;
- const int ELDEXT_TERM = 0;
-
- m4ac->ps = 0;
- m4ac->sbr = 0;
-
- m4ac->frame_length_short = get_bits1(gb);
- res_flags = get_bits(gb, 3);
- if (res_flags) {
- avpriv_report_missing_feature(avctx,
- "AAC data resilience (flags %x)",
- res_flags);
- return AVERROR_PATCHWELCOME;
- }
-
- if (get_bits1(gb)) { // ldSbrPresentFlag
- avpriv_report_missing_feature(avctx,
- "Low Delay SBR");
- return AVERROR_PATCHWELCOME;
- }
-
- while (get_bits(gb, 4) != ELDEXT_TERM) {
- int len = get_bits(gb, 4);
- if (len == 15)
- len += get_bits(gb, 8);
- if (len == 15 + 255)
- len += get_bits(gb, 16);
- if (get_bits_left(gb) < len * 8 + 4) {
- av_log(avctx, AV_LOG_ERROR, overread_err);
- return AVERROR_INVALIDDATA;
- }
- skip_bits_long(gb, 8 * len);
- }
-
- if ((ret = set_default_channel_config(avctx, layout_map,
- &tags, channel_config)))
- return ret;
-
- if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
- return ret;
-
- ep_config = get_bits(gb, 2);
- if (ep_config) {
- avpriv_report_missing_feature(avctx,
- "epConfig %d", ep_config);
- return AVERROR_PATCHWELCOME;
- }
- return 0;
-}
-
-/**
- * Decode audio specific configuration; reference: table 1.13.
- *
- * @param ac pointer to AACContext, may be null
- * @param avctx pointer to AVCCodecContext, used for logging
- * @param m4ac pointer to MPEG4AudioConfig, used for parsing
- * @param data pointer to buffer holding an audio specific config
- * @param bit_size size of audio specific config or data in bits
- * @param sync_extension look for an appended sync extension
- *
- * @return Returns error status or number of consumed bits. <0 - error
- */
-static int decode_audio_specific_config(AACContext *ac,
- AVCodecContext *avctx,
- MPEG4AudioConfig *m4ac,
- const uint8_t *data, int bit_size,
- int sync_extension)
-{
- GetBitContext gb;
- int i, ret;
-
- ff_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
- for (i = 0; i < avctx->extradata_size; i++)
- ff_dlog(avctx, "%02x ", avctx->extradata[i]);
- ff_dlog(avctx, "\n");
-
- if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
- return ret;
-
- if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
- sync_extension)) < 0)
- return AVERROR_INVALIDDATA;
- if (m4ac->sampling_index > 12) {
- av_log(avctx, AV_LOG_ERROR,
- "invalid sampling rate index %d\n",
- m4ac->sampling_index);
- return AVERROR_INVALIDDATA;
- }
- if (m4ac->object_type == AOT_ER_AAC_LD &&
- (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
- av_log(avctx, AV_LOG_ERROR,
- "invalid low delay sampling rate index %d\n",
- m4ac->sampling_index);
- return AVERROR_INVALIDDATA;
- }
-
- skip_bits_long(&gb, i);
-
- switch (m4ac->object_type) {
- case AOT_AAC_MAIN:
- case AOT_AAC_LC:
- case AOT_AAC_LTP:
- case AOT_ER_AAC_LC:
- case AOT_ER_AAC_LD:
- if ((ret = decode_ga_specific_config(ac, avctx, &gb,
- m4ac, m4ac->chan_config)) < 0)
- return ret;
- break;
- case AOT_ER_AAC_ELD:
- if ((ret = decode_eld_specific_config(ac, avctx, &gb,
- m4ac, m4ac->chan_config)) < 0)
- return ret;
- break;
- default:
- avpriv_report_missing_feature(avctx,
- "Audio object type %s%d",
- m4ac->sbr == 1 ? "SBR+" : "",
- m4ac->object_type);
- return AVERROR(ENOSYS);
- }
-
- ff_dlog(avctx,
- "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
- m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
- m4ac->sample_rate, m4ac->sbr,
- m4ac->ps);
-
- return get_bits_count(&gb);
-}
-
-/**
- * linear congruential pseudorandom number generator
- *
- * @param previous_val pointer to the current state of the generator
- *
- * @return Returns a 32-bit pseudorandom integer
- */
-static av_always_inline int lcg_random(int previous_val)
-{
- union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
- return v.s;
-}
-
-static av_always_inline void reset_predict_state(PredictorState *ps)
-{
- ps->r0 = 0.0f;
- ps->r1 = 0.0f;
- ps->cor0 = 0.0f;
- ps->cor1 = 0.0f;
- ps->var0 = 1.0f;
- ps->var1 = 1.0f;
-}
-
-static void reset_all_predictors(PredictorState *ps)
-{
- int i;
- for (i = 0; i < MAX_PREDICTORS; i++)
- reset_predict_state(&ps[i]);
-}
-
-static int sample_rate_idx (int rate)
-{
- if (92017 <= rate) return 0;
- else if (75132 <= rate) return 1;
- else if (55426 <= rate) return 2;
- else if (46009 <= rate) return 3;
- else if (37566 <= rate) return 4;
- else if (27713 <= rate) return 5;
- else if (23004 <= rate) return 6;
- else if (18783 <= rate) return 7;
- else if (13856 <= rate) return 8;
- else if (11502 <= rate) return 9;
- else if (9391 <= rate) return 10;
- else return 11;
-}
-
-static void reset_predictor_group(PredictorState *ps, int group_num)
-{
- int i;
- for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
- reset_predict_state(&ps[i]);
-}
-
-#define AAC_INIT_VLC_STATIC(num, size) \
- INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
- ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
- sizeof(ff_aac_spectral_bits[num][0]), \
- ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
- sizeof(ff_aac_spectral_codes[num][0]), \
- size);
-
-static av_cold void aac_static_table_init(void)
-{
- AAC_INIT_VLC_STATIC( 0, 304);
- AAC_INIT_VLC_STATIC( 1, 270);
- AAC_INIT_VLC_STATIC( 2, 550);
- AAC_INIT_VLC_STATIC( 3, 300);
- AAC_INIT_VLC_STATIC( 4, 328);
- AAC_INIT_VLC_STATIC( 5, 294);
- AAC_INIT_VLC_STATIC( 6, 306);
- AAC_INIT_VLC_STATIC( 7, 268);
- AAC_INIT_VLC_STATIC( 8, 510);
- AAC_INIT_VLC_STATIC( 9, 366);
- AAC_INIT_VLC_STATIC(10, 462);
-
- ff_aac_sbr_init();
-
- ff_aac_tableinit();
-
- INIT_VLC_STATIC(&vlc_scalefactors, 7,
- FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
- ff_aac_scalefactor_bits,
- sizeof(ff_aac_scalefactor_bits[0]),
- sizeof(ff_aac_scalefactor_bits[0]),
- ff_aac_scalefactor_code,
- sizeof(ff_aac_scalefactor_code[0]),
- sizeof(ff_aac_scalefactor_code[0]),
- 352);
-
-
- // window initialization
- ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
- ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
- ff_init_ff_sine_windows(10);
- ff_init_ff_sine_windows( 9);
- ff_init_ff_sine_windows( 7);
-
- cbrt_tableinit();
-}
-
-static AVOnce aac_init = AV_ONCE_INIT;
-
-static av_cold int aac_decode_init(AVCodecContext *avctx)
-{
- AACContext *ac = avctx->priv_data;
- int ret;
-
- ret = ff_thread_once(&aac_init, &aac_static_table_init);
- if (ret != 0)
- return AVERROR_UNKNOWN;
-
- ac->avctx = avctx;
- ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
-
- avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
-
- if (avctx->extradata_size > 0) {
- if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
- avctx->extradata,
- avctx->extradata_size * 8,
- 1)) < 0)
- return ret;
- } else {
- int sr, i;
- uint8_t layout_map[MAX_ELEM_ID*4][3];
- int layout_map_tags;
-
- sr = sample_rate_idx(avctx->sample_rate);
- ac->oc[1].m4ac.sampling_index = sr;
- ac->oc[1].m4ac.channels = avctx->channels;
- ac->oc[1].m4ac.sbr = -1;
- ac->oc[1].m4ac.ps = -1;
-
- for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
- if (ff_mpeg4audio_channels[i] == avctx->channels)
- break;
- if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
- i = 0;
- }
- ac->oc[1].m4ac.chan_config = i;
-
- if (ac->oc[1].m4ac.chan_config) {
- int ret = set_default_channel_config(avctx, layout_map,
- &layout_map_tags, ac->oc[1].m4ac.chan_config);
- if (!ret)
- output_configure(ac, layout_map, layout_map_tags,
- OC_GLOBAL_HDR, 0);
- else if (avctx->err_recognition & AV_EF_EXPLODE)
- return AVERROR_INVALIDDATA;
- }
- }
-
- avpriv_float_dsp_init(&ac->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
-
- ac->random_state = 0x1f2e3d4c;
-
- ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
- ff_mdct_init(&ac->mdct_ld, 10, 1, 1.0 / (32768.0 * 512.0));
- ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
- ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
- ret = ff_imdct15_init(&ac->mdct480, 5);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-/**
- * Skip data_stream_element; reference: table 4.10.
- */
-static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
-{
- int byte_align = get_bits1(gb);
- int count = get_bits(gb, 8);
- if (count == 255)
- count += get_bits(gb, 8);
- if (byte_align)
- align_get_bits(gb);
-
- if (get_bits_left(gb) < 8 * count) {
- av_log(ac->avctx, AV_LOG_ERROR, overread_err);
- return AVERROR_INVALIDDATA;
- }
- skip_bits_long(gb, 8 * count);
- return 0;
-}
-
-static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
- GetBitContext *gb)
-{
- int sfb;
- if (get_bits1(gb)) {
- ics->predictor_reset_group = get_bits(gb, 5);
- if (ics->predictor_reset_group == 0 ||
- ics->predictor_reset_group > 30) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "Invalid Predictor Reset Group.\n");
- return AVERROR_INVALIDDATA;
- }
- }
- for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
- ics->prediction_used[sfb] = get_bits1(gb);
- }
- return 0;
-}
-
-/**
- * Decode Long Term Prediction data; reference: table 4.xx.
- */
-static void decode_ltp(LongTermPrediction *ltp,
- GetBitContext *gb, uint8_t max_sfb)
-{
- int sfb;
-
- ltp->lag = get_bits(gb, 11);
- ltp->coef = ltp_coef[get_bits(gb, 3)];
- for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
- ltp->used[sfb] = get_bits1(gb);
-}
-
-/**
- * Decode Individual Channel Stream info; reference: table 4.6.
- */
-static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
- GetBitContext *gb)
-{
- const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
- const int aot = m4ac->object_type;
- const int sampling_index = m4ac->sampling_index;
- if (aot != AOT_ER_AAC_ELD) {
- if (get_bits1(gb)) {
- av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
- if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
- return AVERROR_INVALIDDATA;
- }
- ics->window_sequence[1] = ics->window_sequence[0];
- ics->window_sequence[0] = get_bits(gb, 2);
- if (aot == AOT_ER_AAC_LD &&
- ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
- "window sequence %d found.\n", ics->window_sequence[0]);
- ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
- return AVERROR_INVALIDDATA;
- }
- ics->use_kb_window[1] = ics->use_kb_window[0];
- ics->use_kb_window[0] = get_bits1(gb);
- }
- ics->num_window_groups = 1;
- ics->group_len[0] = 1;
- if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
- int i;
- ics->max_sfb = get_bits(gb, 4);
- for (i = 0; i < 7; i++) {
- if (get_bits1(gb)) {
- ics->group_len[ics->num_window_groups - 1]++;
- } else {
- ics->num_window_groups++;
- ics->group_len[ics->num_window_groups - 1] = 1;
- }
- }
- ics->num_windows = 8;
- ics->swb_offset = ff_swb_offset_128[sampling_index];
- ics->num_swb = ff_aac_num_swb_128[sampling_index];
- ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
- ics->predictor_present = 0;
- } else {
- ics->max_sfb = get_bits(gb, 6);
- ics->num_windows = 1;
- if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
- if (m4ac->frame_length_short) {
- ics->swb_offset = ff_swb_offset_480[sampling_index];
- ics->num_swb = ff_aac_num_swb_480[sampling_index];
- ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
- } else {
- ics->swb_offset = ff_swb_offset_512[sampling_index];
- ics->num_swb = ff_aac_num_swb_512[sampling_index];
- ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
- }
- if (!ics->num_swb || !ics->swb_offset)
- return AVERROR_BUG;
- } else {
- ics->swb_offset = ff_swb_offset_1024[sampling_index];
- ics->num_swb = ff_aac_num_swb_1024[sampling_index];
- ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
- }
- if (aot != AOT_ER_AAC_ELD) {
- ics->predictor_present = get_bits1(gb);
- ics->predictor_reset_group = 0;
- }
- if (ics->predictor_present) {
- if (aot == AOT_AAC_MAIN) {
- if (decode_prediction(ac, ics, gb)) {
- return AVERROR_INVALIDDATA;
- }
- } else if (aot == AOT_AAC_LC ||
- aot == AOT_ER_AAC_LC) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "Prediction is not allowed in AAC-LC.\n");
- return AVERROR_INVALIDDATA;
- } else {
- if (aot == AOT_ER_AAC_LD) {
- avpriv_report_missing_feature(ac->avctx, "LTP in ER AAC LD");
- return AVERROR_PATCHWELCOME;
- }
- if ((ics->ltp.present = get_bits(gb, 1)))
- decode_ltp(&ics->ltp, gb, ics->max_sfb);
- }
- }
- }
-
- if (ics->max_sfb > ics->num_swb) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "Number of scalefactor bands in group (%d) "
- "exceeds limit (%d).\n",
- ics->max_sfb, ics->num_swb);
- return AVERROR_INVALIDDATA;
- }
-
- return 0;
-}
-
-/**
- * Decode band types (section_data payload); reference: table 4.46.
- *
- * @param band_type array of the used band type
- * @param band_type_run_end array of the last scalefactor band of a band type run
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static int decode_band_types(AACContext *ac, enum BandType band_type[120],
- int band_type_run_end[120], GetBitContext *gb,
- IndividualChannelStream *ics)
-{
- int g, idx = 0;
- const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
- for (g = 0; g < ics->num_window_groups; g++) {
- int k = 0;
- while (k < ics->max_sfb) {
- uint8_t sect_end = k;
- int sect_len_incr;
- int sect_band_type = get_bits(gb, 4);
- if (sect_band_type == 12) {
- av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
- return AVERROR_INVALIDDATA;
- }
- do {
- sect_len_incr = get_bits(gb, bits);
- sect_end += sect_len_incr;
- if (get_bits_left(gb) < 0) {
- av_log(ac->avctx, AV_LOG_ERROR, overread_err);
- return AVERROR_INVALIDDATA;
- }
- if (sect_end > ics->max_sfb) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "Number of bands (%d) exceeds limit (%d).\n",
- sect_end, ics->max_sfb);
- return AVERROR_INVALIDDATA;
- }
- } while (sect_len_incr == (1 << bits) - 1);
- for (; k < sect_end; k++) {
- band_type [idx] = sect_band_type;
- band_type_run_end[idx++] = sect_end;
- }
- }
- }
- return 0;
-}
-
-/**
- * Decode scalefactors; reference: table 4.47.
- *
- * @param global_gain first scalefactor value as scalefactors are differentially coded
- * @param band_type array of the used band type
- * @param band_type_run_end array of the last scalefactor band of a band type run
- * @param sf array of scalefactors or intensity stereo positions
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
- unsigned int global_gain,
- IndividualChannelStream *ics,
- enum BandType band_type[120],
- int band_type_run_end[120])
-{
- int g, i, idx = 0;
- int offset[3] = { global_gain, global_gain - 90, 0 };
- int clipped_offset;
- int noise_flag = 1;
- for (g = 0; g < ics->num_window_groups; g++) {
- for (i = 0; i < ics->max_sfb;) {
- int run_end = band_type_run_end[idx];
- if (band_type[idx] == ZERO_BT) {
- for (; i < run_end; i++, idx++)
- sf[idx] = 0.0;
- } else if ((band_type[idx] == INTENSITY_BT) ||
- (band_type[idx] == INTENSITY_BT2)) {
- for (; i < run_end; i++, idx++) {
- offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- clipped_offset = av_clip(offset[2], -155, 100);
- if (offset[2] != clipped_offset) {
- avpriv_request_sample(ac->avctx,
- "If you heard an audible artifact, there may be a bug in the decoder. "
- "Clipped intensity stereo position (%d -> %d)",
- offset[2], clipped_offset);
- }
- sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
- }
- } else if (band_type[idx] == NOISE_BT) {
- for (; i < run_end; i++, idx++) {
- if (noise_flag-- > 0)
- offset[1] += get_bits(gb, 9) - 256;
- else
- offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- clipped_offset = av_clip(offset[1], -100, 155);
- if (offset[1] != clipped_offset) {
- avpriv_request_sample(ac->avctx,
- "If you heard an audible artifact, there may be a bug in the decoder. "
- "Clipped noise gain (%d -> %d)",
- offset[1], clipped_offset);
- }
- sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
- }
- } else {
- for (; i < run_end; i++, idx++) {
- offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- if (offset[0] > 255U) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "Scalefactor (%d) out of range.\n", offset[0]);
- return AVERROR_INVALIDDATA;
- }
- sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
- }
- }
- }
- }
- return 0;
-}
-
-/**
- * Decode pulse data; reference: table 4.7.
- */
-static int decode_pulses(Pulse *pulse, GetBitContext *gb,
- const uint16_t *swb_offset, int num_swb)
-{
- int i, pulse_swb;
- pulse->num_pulse = get_bits(gb, 2) + 1;
- pulse_swb = get_bits(gb, 6);
- if (pulse_swb >= num_swb)
- return -1;
- pulse->pos[0] = swb_offset[pulse_swb];
- pulse->pos[0] += get_bits(gb, 5);
- if (pulse->pos[0] > 1023)
- return -1;
- pulse->amp[0] = get_bits(gb, 4);
- for (i = 1; i < pulse->num_pulse; i++) {
- pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
- if (pulse->pos[i] > 1023)
- return -1;
- pulse->amp[i] = get_bits(gb, 4);
- }
- return 0;
-}
-
-/**
- * Decode Temporal Noise Shaping data; reference: table 4.48.
- *
- * @return Returns error status. 0 - OK, !0 - error
- */
-static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
- GetBitContext *gb, const IndividualChannelStream *ics)
-{
- int w, filt, i, coef_len, coef_res, coef_compress;
- const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
- const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
- for (w = 0; w < ics->num_windows; w++) {
- if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
- coef_res = get_bits1(gb);
-
- for (filt = 0; filt < tns->n_filt[w]; filt++) {
- int tmp2_idx;
- tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
-
- if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "TNS filter order %d is greater than maximum %d.\n",
- tns->order[w][filt], tns_max_order);
- tns->order[w][filt] = 0;
- return AVERROR_INVALIDDATA;
- }
- if (tns->order[w][filt]) {
- tns->direction[w][filt] = get_bits1(gb);
- coef_compress = get_bits1(gb);
- coef_len = coef_res + 3 - coef_compress;
- tmp2_idx = 2 * coef_compress + coef_res;
-
- for (i = 0; i < tns->order[w][filt]; i++)
- tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
- }
- }
- }
- }
- return 0;
-}
+#include "profiles.h"
+#include "libavutil/intfloat.h"
-/**
- * Decode Mid/Side data; reference: table 4.54.
- *
- * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
- * [1] mask is decoded from bitstream; [2] mask is all 1s;
- * [3] reserved for scalable AAC
- */
-static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
- int ms_present)
+#include <errno.h>
+#include <math.h>
+#include <stdint.h>
+#include <string.h>
+
+#if ARCH_ARM
+# include "arm/aac.h"
+#elif ARCH_MIPS
+# include "mips/aacdec_mips.h"
+#endif
+
+static av_always_inline void reset_predict_state(PredictorState *ps)
{
- int idx;
- int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
- if (ms_present == 1) {
- for (idx = 0; idx < max_idx; idx++)
- cpe->ms_mask[idx] = get_bits1(gb);
- } else if (ms_present == 2) {
- memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
- }
+ ps->r0 = 0.0f;
+ ps->r1 = 0.0f;
+ ps->cor0 = 0.0f;
+ ps->cor1 = 0.0f;
+ ps->var0 = 1.0f;
+ ps->var1 = 1.0f;
}
#ifndef VMUL2
diff --cc libavcodec/aacdec_fixed.c
index d802f3834f,0000000000..ffd577c789
mode 100644,000000..100644
--- a/libavcodec/aacdec_fixed.c
+++ b/libavcodec/aacdec_fixed.c
@@@ -1,463 -1,0 +1,463 @@@
+/*
+ * Copyright (c) 2013
+ * MIPS Technologies, Inc., California.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * AAC decoder fixed-point implementation
+ *
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * 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
+ * AAC decoder
+ * @author Oded Shimon ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * Fixed point implementation
+ * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
+ */
+
+#define FFT_FLOAT 0
+#define FFT_FIXED_32 1
+#define USE_FIXED 1
+
+#include "libavutil/fixed_dsp.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "fft.h"
+#include "lpc.h"
+#include "kbdwin.h"
+#include "sinewin.h"
+
+#include "aac.h"
+#include "aactab.h"
+#include "aacdectab.h"
++#include "adts_header.h"
+#include "cbrt_data.h"
+#include "sbr.h"
+#include "aacsbr.h"
+#include "mpeg4audio.h"
- #include "aacadtsdec.h"
+#include "profiles.h"
+#include "libavutil/intfloat.h"
+
+#include <math.h>
+#include <string.h>
+
+static av_always_inline void reset_predict_state(PredictorState *ps)
+{
+ ps->r0.mant = 0;
+ ps->r0.exp = 0;
+ ps->r1.mant = 0;
+ ps->r1.exp = 0;
+ ps->cor0.mant = 0;
+ ps->cor0.exp = 0;
+ ps->cor1.mant = 0;
+ ps->cor1.exp = 0;
+ ps->var0.mant = 0x20000000;
+ ps->var0.exp = 1;
+ ps->var1.mant = 0x20000000;
+ ps->var1.exp = 1;
+}
+
+static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) }; // 2^0, 2^0.25, 2^0.5, 2^0.75
+
+static inline int *DEC_SPAIR(int *dst, unsigned idx)
+{
+ dst[0] = (idx & 15) - 4;
+ dst[1] = (idx >> 4 & 15) - 4;
+
+ return dst + 2;
+}
+
+static inline int *DEC_SQUAD(int *dst, unsigned idx)
+{
+ dst[0] = (idx & 3) - 1;
+ dst[1] = (idx >> 2 & 3) - 1;
+ dst[2] = (idx >> 4 & 3) - 1;
+ dst[3] = (idx >> 6 & 3) - 1;
+
+ return dst + 4;
+}
+
+static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
+{
+ dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
+ dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
+
+ return dst + 2;
+}
+
+static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
+{
+ unsigned nz = idx >> 12;
+
+ dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
+ sign <<= nz & 1;
+ nz >>= 1;
+ dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
+ sign <<= nz & 1;
+ nz >>= 1;
+ dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
+ sign <<= nz & 1;
+ nz >>= 1;
+ dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
+
+ return dst + 4;
+}
+
+static void vector_pow43(int *coefs, int len)
+{
+ int i, coef;
+
+ for (i=0; i<len; i++) {
+ coef = coefs[i];
+ if (coef < 0)
+ coef = -(int)ff_cbrt_tab_fixed[-coef];
+ else
+ coef = (int)ff_cbrt_tab_fixed[coef];
+ coefs[i] = coef;
+ }
+}
+
+static void subband_scale(int *dst, int *src, int scale, int offset, int len)
+{
+ int ssign = scale < 0 ? -1 : 1;
+ int s = FFABS(scale);
+ unsigned int round;
+ int i, out, c = exp2tab[s & 3];
+
+ s = offset - (s >> 2);
+
+ if (s > 31) {
+ for (i=0; i<len; i++) {
+ dst[i] = 0;
+ }
+ } else if (s > 0) {
+ round = 1 << (s-1);
+ for (i=0; i<len; i++) {
+ out = (int)(((int64_t)src[i] * c) >> 32);
+ dst[i] = ((int)(out+round) >> s) * ssign;
+ }
+ } else if (s > -32) {
+ s = s + 32;
+ round = 1U << (s-1);
+ for (i=0; i<len; i++) {
+ out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
+ dst[i] = out * (unsigned)ssign;
+ }
+ } else {
+ av_log(NULL, AV_LOG_ERROR, "Overflow in subband_scale()\n");
+ }
+}
+
+static void noise_scale(int *coefs, int scale, int band_energy, int len)
+{
+ int ssign = scale < 0 ? -1 : 1;
+ int s = FFABS(scale);
+ unsigned int round;
+ int i, out, c = exp2tab[s & 3];
+ int nlz = 0;
+
+ while (band_energy > 0x7fff) {
+ band_energy >>= 1;
+ nlz++;
+ }
+ c /= band_energy;
+ s = 21 + nlz - (s >> 2);
+
+ if (s > 31) {
+ for (i=0; i<len; i++) {
+ coefs[i] = 0;
+ }
+ } else if (s >= 0) {
+ round = s ? 1 << (s-1) : 0;
+ for (i=0; i<len; i++) {
+ out = (int)(((int64_t)coefs[i] * c) >> 32);
+ coefs[i] = ((int)(out+round) >> s) * ssign;
+ }
+ }
+ else {
+ s = s + 32;
+ round = 1 << (s-1);
+ for (i=0; i<len; i++) {
+ out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
+ coefs[i] = out * ssign;
+ }
+ }
+}
+
+static av_always_inline SoftFloat flt16_round(SoftFloat pf)
+{
+ SoftFloat tmp;
+ int s;
+
+ tmp.exp = pf.exp;
+ s = pf.mant >> 31;
+ tmp.mant = (pf.mant ^ s) - s;
+ tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
+ tmp.mant = (tmp.mant ^ s) - s;
+
+ return tmp;
+}
+
+static av_always_inline SoftFloat flt16_even(SoftFloat pf)
+{
+ SoftFloat tmp;
+ int s;
+
+ tmp.exp = pf.exp;
+ s = pf.mant >> 31;
+ tmp.mant = (pf.mant ^ s) - s;
+ tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
+ tmp.mant = (tmp.mant ^ s) - s;
+
+ return tmp;
+}
+
+static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
+{
+ SoftFloat pun;
+ int s;
+
+ pun.exp = pf.exp;
+ s = pf.mant >> 31;
+ pun.mant = (pf.mant ^ s) - s;
+ pun.mant = pun.mant & 0xFFC00000U;
+ pun.mant = (pun.mant ^ s) - s;
+
+ return pun;
+}
+
+static av_always_inline void predict(PredictorState *ps, int *coef,
+ int output_enable)
+{
+ const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
+ const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
+ SoftFloat e0, e1;
+ SoftFloat pv;
+ SoftFloat k1, k2;
+ SoftFloat r0 = ps->r0, r1 = ps->r1;
+ SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
+ SoftFloat var0 = ps->var0, var1 = ps->var1;
+ SoftFloat tmp;
+
+ if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
+ k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
+ }
+ else {
+ k1.mant = 0;
+ k1.exp = 0;
+ }
+
+ if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
+ k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
+ }
+ else {
+ k2.mant = 0;
+ k2.exp = 0;
+ }
+
+ tmp = av_mul_sf(k1, r0);
+ pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
+ if (output_enable) {
+ int shift = 28 - pv.exp;
+
+ if (shift < 31) {
+ if (shift > 0) {
+ *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
+ } else
+ *coef += (unsigned)(pv.mant << -shift);
+ }
+ }
+
+ e0 = av_int2sf(*coef, 2);
+ e1 = av_sub_sf(e0, tmp);
+
+ ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
+ tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
+ tmp.exp--;
+ ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
+ ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
+ tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
+ tmp.exp--;
+ ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
+
+ ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
+ ps->r0 = flt16_trunc(av_mul_sf(a, e0));
+}
+
+
+static const int cce_scale_fixed[8] = {
+ Q30(1.0), //2^(0/8)
+ Q30(1.0905077327), //2^(1/8)
+ Q30(1.1892071150), //2^(2/8)
+ Q30(1.2968395547), //2^(3/8)
+ Q30(1.4142135624), //2^(4/8)
+ Q30(1.5422108254), //2^(5/8)
+ Q30(1.6817928305), //2^(6/8)
+ Q30(1.8340080864), //2^(7/8)
+};
+
+/**
+ * Apply dependent channel coupling (applied before IMDCT).
+ *
+ * @param index index into coupling gain array
+ */
+static void apply_dependent_coupling_fixed(AACContext *ac,
+ SingleChannelElement *target,
+ ChannelElement *cce, int index)
+{
+ IndividualChannelStream *ics = &cce->ch[0].ics;
+ const uint16_t *offsets = ics->swb_offset;
+ int *dest = target->coeffs;
+ const int *src = cce->ch[0].coeffs;
+ int g, i, group, k, idx = 0;
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Dependent coupling is not supported together with LTP\n");
+ return;
+ }
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ if (cce->ch[0].band_type[idx] != ZERO_BT) {
+ const int gain = cce->coup.gain[index][idx];
+ int shift, round, c, tmp;
+
+ if (gain < 0) {
+ c = -cce_scale_fixed[-gain & 7];
+ shift = (-gain-1024) >> 3;
+ }
+ else {
+ c = cce_scale_fixed[gain & 7];
+ shift = (gain-1024) >> 3;
+ }
+
+ if (shift < -31) {
+ // Nothing to do
+ } else if (shift < 0) {
+ shift = -shift;
+ round = 1 << (shift - 1);
+
+ for (group = 0; group < ics->group_len[g]; group++) {
+ for (k = offsets[i]; k < offsets[i + 1]; k++) {
+ tmp = (int)(((int64_t)src[group * 128 + k] * c + \
+ (int64_t)0x1000000000) >> 37);
+ dest[group * 128 + k] += (tmp + round) >> shift;
+ }
+ }
+ }
+ else {
+ for (group = 0; group < ics->group_len[g]; group++) {
+ for (k = offsets[i]; k < offsets[i + 1]; k++) {
+ tmp = (int)(((int64_t)src[group * 128 + k] * c + \
+ (int64_t)0x1000000000) >> 37);
+ dest[group * 128 + k] += tmp * (1U << shift);
+ }
+ }
+ }
+ }
+ }
+ dest += ics->group_len[g] * 128;
+ src += ics->group_len[g] * 128;
+ }
+}
+
+/**
+ * Apply independent channel coupling (applied after IMDCT).
+ *
+ * @param index index into coupling gain array
+ */
+static void apply_independent_coupling_fixed(AACContext *ac,
+ SingleChannelElement *target,
+ ChannelElement *cce, int index)
+{
+ int i, c, shift, round, tmp;
+ const int gain = cce->coup.gain[index][0];
+ const int *src = cce->ch[0].ret;
+ int *dest = target->ret;
+ const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
+
+ c = cce_scale_fixed[gain & 7];
+ shift = (gain-1024) >> 3;
+ if (shift < -31) {
+ return;
+ } else if (shift < 0) {
+ shift = -shift;
+ round = 1 << (shift - 1);
+
+ for (i = 0; i < len; i++) {
+ tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
+ dest[i] += (tmp + round) >> shift;
+ }
+ }
+ else {
+ for (i = 0; i < len; i++) {
+ tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
+ dest[i] += tmp * (1 << shift);
+ }
+ }
+}
+
+#include "aacdec_template.c"
+
+AVCodec ff_aac_fixed_decoder = {
+ .name = "aac_fixed",
+ .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_AAC,
+ .priv_data_size = sizeof(AACContext),
+ .init = aac_decode_init,
+ .close = aac_decode_close,
+ .decode = aac_decode_frame,
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
+ },
+ .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+ .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
+ .channel_layouts = aac_channel_layout,
+ .profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
+ .flush = flush,
+};
diff --cc libavcodec/aacdec_template.c
index 082cc908d2,0000000000..6c6cdd84af
mode 100644,000000..100644
--- a/libavcodec/aacdec_template.c
+++ b/libavcodec/aacdec_template.c
@@@ -1,3412 -1,0 +1,3412 @@@
+/*
+ * AAC decoder
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ * Copyright (c) 2008-2013 Alex Converse <alex.converse at gmail.com>
+ *
+ * AAC LATM decoder
+ * Copyright (c) 2008-2010 Paul Kendall <paul at kcbbs.gen.nz>
+ * Copyright (c) 2010 Janne Grunau <janne-libav at jannau.net>
+ *
+ * AAC decoder fixed-point implementation
+ * Copyright (c) 2013
+ * MIPS Technologies, Inc., California.
+ *
+ * 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
+ * AAC decoder
+ * @author Oded Shimon ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * AAC decoder fixed-point implementation
+ * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
+ * @author Nedeljko Babic ( nedeljko.babic imgtec com )
+ */
+
+/*
+ * supported tools
+ *
+ * Support? Name
+ * N (code in SoC repo) gain control
+ * Y block switching
+ * Y window shapes - standard
+ * N window shapes - Low Delay
+ * Y filterbank - standard
+ * N (code in SoC repo) filterbank - Scalable Sample Rate
+ * Y Temporal Noise Shaping
+ * Y Long Term Prediction
+ * Y intensity stereo
+ * Y channel coupling
+ * Y frequency domain prediction
+ * Y Perceptual Noise Substitution
+ * Y Mid/Side stereo
+ * N Scalable Inverse AAC Quantization
+ * N Frequency Selective Switch
+ * N upsampling filter
+ * Y quantization & coding - AAC
+ * N quantization & coding - TwinVQ
+ * N quantization & coding - BSAC
+ * N AAC Error Resilience tools
+ * N Error Resilience payload syntax
+ * N Error Protection tool
+ * N CELP
+ * N Silence Compression
+ * N HVXC
+ * N HVXC 4kbits/s VR
+ * N Structured Audio tools
+ * N Structured Audio Sample Bank Format
+ * N MIDI
+ * N Harmonic and Individual Lines plus Noise
+ * N Text-To-Speech Interface
+ * Y Spectral Band Replication
+ * Y (not in this code) Layer-1
+ * Y (not in this code) Layer-2
+ * Y (not in this code) Layer-3
+ * N SinuSoidal Coding (Transient, Sinusoid, Noise)
+ * Y Parametric Stereo
+ * N Direct Stream Transfer
+ * Y (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
+ *
+ * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
+ * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
+ Parametric Stereo.
+ */
+
+#include "libavutil/thread.h"
+
+static VLC vlc_scalefactors;
+static VLC vlc_spectral[11];
+
+static int output_configure(AACContext *ac,
+ uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
+ enum OCStatus oc_type, int get_new_frame);
+
+#define overread_err "Input buffer exhausted before END element found\n"
+
+static int count_channels(uint8_t (*layout)[3], int tags)
+{
+ int i, sum = 0;
+ for (i = 0; i < tags; i++) {
+ int syn_ele = layout[i][0];
+ int pos = layout[i][2];
+ sum += (1 + (syn_ele == TYPE_CPE)) *
+ (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
+ }
+ return sum;
+}
+
+/**
+ * Check for the channel element in the current channel position configuration.
+ * If it exists, make sure the appropriate element is allocated and map the
+ * channel order to match the internal FFmpeg channel layout.
+ *
+ * @param che_pos current channel position configuration
+ * @param type channel element type
+ * @param id channel element id
+ * @param channels count of the number of channels in the configuration
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static av_cold int che_configure(AACContext *ac,
+ enum ChannelPosition che_pos,
+ int type, int id, int *channels)
+{
+ if (*channels >= MAX_CHANNELS)
+ return AVERROR_INVALIDDATA;
+ if (che_pos) {
+ if (!ac->che[type][id]) {
+ if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
+ return AVERROR(ENOMEM);
+ AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr, type);
+ }
+ if (type != TYPE_CCE) {
+ if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
+ return AVERROR_INVALIDDATA;
+ }
+ ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
+ if (type == TYPE_CPE ||
+ (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
+ ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
+ }
+ }
+ } else {
+ if (ac->che[type][id])
+ AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr);
+ av_freep(&ac->che[type][id]);
+ }
+ return 0;
+}
+
+static int frame_configure_elements(AVCodecContext *avctx)
+{
+ AACContext *ac = avctx->priv_data;
+ int type, id, ch, ret;
+
+ /* set channel pointers to internal buffers by default */
+ for (type = 0; type < 4; type++) {
+ for (id = 0; id < MAX_ELEM_ID; id++) {
+ ChannelElement *che = ac->che[type][id];
+ if (che) {
+ che->ch[0].ret = che->ch[0].ret_buf;
+ che->ch[1].ret = che->ch[1].ret_buf;
+ }
+ }
+ }
+
+ /* get output buffer */
+ av_frame_unref(ac->frame);
+ if (!avctx->channels)
+ return 1;
+
+ ac->frame->nb_samples = 2048;
+ if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
+ return ret;
+
+ /* map output channel pointers to AVFrame data */
+ for (ch = 0; ch < avctx->channels; ch++) {
+ if (ac->output_element[ch])
+ ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
+ }
+
+ return 0;
+}
+
+struct elem_to_channel {
+ uint64_t av_position;
+ uint8_t syn_ele;
+ uint8_t elem_id;
+ uint8_t aac_position;
+};
+
+static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
+ uint8_t (*layout_map)[3], int offset, uint64_t left,
+ uint64_t right, int pos)
+{
+ if (layout_map[offset][0] == TYPE_CPE) {
+ e2c_vec[offset] = (struct elem_to_channel) {
+ .av_position = left | right,
+ .syn_ele = TYPE_CPE,
+ .elem_id = layout_map[offset][1],
+ .aac_position = pos
+ };
+ return 1;
+ } else {
+ e2c_vec[offset] = (struct elem_to_channel) {
+ .av_position = left,
+ .syn_ele = TYPE_SCE,
+ .elem_id = layout_map[offset][1],
+ .aac_position = pos
+ };
+ e2c_vec[offset + 1] = (struct elem_to_channel) {
+ .av_position = right,
+ .syn_ele = TYPE_SCE,
+ .elem_id = layout_map[offset + 1][1],
+ .aac_position = pos
+ };
+ return 2;
+ }
+}
+
+static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
+ int *current)
+{
+ int num_pos_channels = 0;
+ int first_cpe = 0;
+ int sce_parity = 0;
+ int i;
+ for (i = *current; i < tags; i++) {
+ if (layout_map[i][2] != pos)
+ break;
+ if (layout_map[i][0] == TYPE_CPE) {
+ if (sce_parity) {
+ if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
+ sce_parity = 0;
+ } else {
+ return -1;
+ }
+ }
+ num_pos_channels += 2;
+ first_cpe = 1;
+ } else {
+ num_pos_channels++;
+ sce_parity ^= 1;
+ }
+ }
+ if (sce_parity &&
+ ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
+ return -1;
+ *current = i;
+ return num_pos_channels;
+}
+
+static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
+{
+ int i, n, total_non_cc_elements;
+ struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
+ int num_front_channels, num_side_channels, num_back_channels;
+ uint64_t layout;
+
+ if (FF_ARRAY_ELEMS(e2c_vec) < tags)
+ return 0;
+
+ i = 0;
+ num_front_channels =
+ count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
+ if (num_front_channels < 0)
+ return 0;
+ num_side_channels =
+ count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
+ if (num_side_channels < 0)
+ return 0;
+ num_back_channels =
+ count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
+ if (num_back_channels < 0)
+ return 0;
+
+ if (num_side_channels == 0 && num_back_channels >= 4) {
+ num_side_channels = 2;
+ num_back_channels -= 2;
+ }
+
+ i = 0;
+ if (num_front_channels & 1) {
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = AV_CH_FRONT_CENTER,
+ .syn_ele = TYPE_SCE,
+ .elem_id = layout_map[i][1],
+ .aac_position = AAC_CHANNEL_FRONT
+ };
+ i++;
+ num_front_channels--;
+ }
+ if (num_front_channels >= 4) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ AV_CH_FRONT_LEFT_OF_CENTER,
+ AV_CH_FRONT_RIGHT_OF_CENTER,
+ AAC_CHANNEL_FRONT);
+ num_front_channels -= 2;
+ }
+ if (num_front_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ AV_CH_FRONT_LEFT,
+ AV_CH_FRONT_RIGHT,
+ AAC_CHANNEL_FRONT);
+ num_front_channels -= 2;
+ }
+ while (num_front_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ UINT64_MAX,
+ UINT64_MAX,
+ AAC_CHANNEL_FRONT);
+ num_front_channels -= 2;
+ }
+
+ if (num_side_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ AV_CH_SIDE_LEFT,
+ AV_CH_SIDE_RIGHT,
+ AAC_CHANNEL_FRONT);
+ num_side_channels -= 2;
+ }
+ while (num_side_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ UINT64_MAX,
+ UINT64_MAX,
+ AAC_CHANNEL_SIDE);
+ num_side_channels -= 2;
+ }
+
+ while (num_back_channels >= 4) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ UINT64_MAX,
+ UINT64_MAX,
+ AAC_CHANNEL_BACK);
+ num_back_channels -= 2;
+ }
+ if (num_back_channels >= 2) {
+ i += assign_pair(e2c_vec, layout_map, i,
+ AV_CH_BACK_LEFT,
+ AV_CH_BACK_RIGHT,
+ AAC_CHANNEL_BACK);
+ num_back_channels -= 2;
+ }
+ if (num_back_channels) {
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = AV_CH_BACK_CENTER,
+ .syn_ele = TYPE_SCE,
+ .elem_id = layout_map[i][1],
+ .aac_position = AAC_CHANNEL_BACK
+ };
+ i++;
+ num_back_channels--;
+ }
+
+ if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = AV_CH_LOW_FREQUENCY,
+ .syn_ele = TYPE_LFE,
+ .elem_id = layout_map[i][1],
+ .aac_position = AAC_CHANNEL_LFE
+ };
+ i++;
+ }
+ while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
+ e2c_vec[i] = (struct elem_to_channel) {
+ .av_position = UINT64_MAX,
+ .syn_ele = TYPE_LFE,
+ .elem_id = layout_map[i][1],
+ .aac_position = AAC_CHANNEL_LFE
+ };
+ i++;
+ }
+
+ // Must choose a stable sort
+ total_non_cc_elements = n = i;
+ do {
+ int next_n = 0;
+ for (i = 1; i < n; i++)
+ if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
+ FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
+ next_n = i;
+ }
+ n = next_n;
+ } while (n > 0);
+
+ layout = 0;
+ for (i = 0; i < total_non_cc_elements; i++) {
+ layout_map[i][0] = e2c_vec[i].syn_ele;
+ layout_map[i][1] = e2c_vec[i].elem_id;
+ layout_map[i][2] = e2c_vec[i].aac_position;
+ if (e2c_vec[i].av_position != UINT64_MAX) {
+ layout |= e2c_vec[i].av_position;
+ }
+ }
+
+ return layout;
+}
+
+/**
+ * Save current output configuration if and only if it has been locked.
+ */
+static int push_output_configuration(AACContext *ac) {
+ int pushed = 0;
+
+ if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
+ ac->oc[0] = ac->oc[1];
+ pushed = 1;
+ }
+ ac->oc[1].status = OC_NONE;
+ return pushed;
+}
+
+/**
+ * Restore the previous output configuration if and only if the current
+ * configuration is unlocked.
+ */
+static void pop_output_configuration(AACContext *ac) {
+ if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
+ ac->oc[1] = ac->oc[0];
+ ac->avctx->channels = ac->oc[1].channels;
+ ac->avctx->channel_layout = ac->oc[1].channel_layout;
+ output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
+ ac->oc[1].status, 0);
+ }
+}
+
+/**
+ * Configure output channel order based on the current program
+ * configuration element.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int output_configure(AACContext *ac,
+ uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
+ enum OCStatus oc_type, int get_new_frame)
+{
+ AVCodecContext *avctx = ac->avctx;
+ int i, channels = 0, ret;
+ uint64_t layout = 0;
+ uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
+ uint8_t type_counts[TYPE_END] = { 0 };
+
+ if (ac->oc[1].layout_map != layout_map) {
+ memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
+ ac->oc[1].layout_map_tags = tags;
+ }
+ for (i = 0; i < tags; i++) {
+ int type = layout_map[i][0];
+ int id = layout_map[i][1];
+ id_map[type][id] = type_counts[type]++;
+ if (id_map[type][id] >= MAX_ELEM_ID) {
+ avpriv_request_sample(ac->avctx, "Too large remapped id");
+ return AVERROR_PATCHWELCOME;
+ }
+ }
+ // Try to sniff a reasonable channel order, otherwise output the
+ // channels in the order the PCE declared them.
+ if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
+ layout = sniff_channel_order(layout_map, tags);
+ for (i = 0; i < tags; i++) {
+ int type = layout_map[i][0];
+ int id = layout_map[i][1];
+ int iid = id_map[type][id];
+ int position = layout_map[i][2];
+ // Allocate or free elements depending on if they are in the
+ // current program configuration.
+ ret = che_configure(ac, position, type, iid, &channels);
+ if (ret < 0)
+ return ret;
+ ac->tag_che_map[type][id] = ac->che[type][iid];
+ }
+ if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
+ if (layout == AV_CH_FRONT_CENTER) {
+ layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
+ } else {
+ layout = 0;
+ }
+ }
+
+ if (layout) avctx->channel_layout = layout;
+ ac->oc[1].channel_layout = layout;
+ avctx->channels = ac->oc[1].channels = channels;
+ ac->oc[1].status = oc_type;
+
+ if (get_new_frame) {
+ if ((ret = frame_configure_elements(ac->avctx)) < 0)
+ return ret;
+ }
+
+ return 0;
+}
+
+static void flush(AVCodecContext *avctx)
+{
+ AACContext *ac= avctx->priv_data;
+ int type, i, j;
+
+ for (type = 3; type >= 0; type--) {
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *che = ac->che[type][i];
+ if (che) {
+ for (j = 0; j <= 1; j++) {
+ memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
+ }
+ }
+ }
+ }
+}
+
+/**
+ * Set up channel positions based on a default channel configuration
+ * as specified in table 1.17.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int set_default_channel_config(AVCodecContext *avctx,
+ uint8_t (*layout_map)[3],
+ int *tags,
+ int channel_config)
+{
+ if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
+ channel_config > 12) {
+ av_log(avctx, AV_LOG_ERROR,
+ "invalid default channel configuration (%d)\n",
+ channel_config);
+ return AVERROR_INVALIDDATA;
+ }
+ *tags = tags_per_config[channel_config];
+ memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
+ *tags * sizeof(*layout_map));
+
+ /*
+ * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
+ * However, at least Nero AAC encoder encodes 7.1 streams using the default
+ * channel config 7, mapping the side channels of the original audio stream
+ * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
+ * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
+ * the incorrect streams as if they were correct (and as the encoder intended).
+ *
+ * As actual intended 7.1(wide) streams are very rare, default to assuming a
+ * 7.1 layout was intended.
+ */
+ if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
+ av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
+ " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
+ " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
+ layout_map[2][2] = AAC_CHANNEL_SIDE;
+ }
+
+ return 0;
+}
+
+static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
+{
+ /* For PCE based channel configurations map the channels solely based
+ * on tags. */
+ if (!ac->oc[1].m4ac.chan_config) {
+ return ac->tag_che_map[type][elem_id];
+ }
+ // Allow single CPE stereo files to be signalled with mono configuration.
+ if (!ac->tags_mapped && type == TYPE_CPE &&
+ ac->oc[1].m4ac.chan_config == 1) {
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags;
+ push_output_configuration(ac);
+
+ av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
+
+ if (set_default_channel_config(ac->avctx, layout_map,
+ &layout_map_tags, 2) < 0)
+ return NULL;
+ if (output_configure(ac, layout_map, layout_map_tags,
+ OC_TRIAL_FRAME, 1) < 0)
+ return NULL;
+
+ ac->oc[1].m4ac.chan_config = 2;
+ ac->oc[1].m4ac.ps = 0;
+ }
+ // And vice-versa
+ if (!ac->tags_mapped && type == TYPE_SCE &&
+ ac->oc[1].m4ac.chan_config == 2) {
+ uint8_t layout_map[MAX_ELEM_ID * 4][3];
+ int layout_map_tags;
+ push_output_configuration(ac);
+
+ av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
+
+ if (set_default_channel_config(ac->avctx, layout_map,
+ &layout_map_tags, 1) < 0)
+ return NULL;
+ if (output_configure(ac, layout_map, layout_map_tags,
+ OC_TRIAL_FRAME, 1) < 0)
+ return NULL;
+
+ ac->oc[1].m4ac.chan_config = 1;
+ if (ac->oc[1].m4ac.sbr)
+ ac->oc[1].m4ac.ps = -1;
+ }
+ /* For indexed channel configurations map the channels solely based
+ * on position. */
+ switch (ac->oc[1].m4ac.chan_config) {
+ case 12:
+ case 7:
+ if (ac->tags_mapped == 3 && type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
+ }
+ case 11:
+ if (ac->tags_mapped == 2 &&
+ ac->oc[1].m4ac.chan_config == 11 &&
+ type == TYPE_SCE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+ }
+ case 6:
+ /* Some streams incorrectly code 5.1 audio as
+ * SCE[0] CPE[0] CPE[1] SCE[1]
+ * instead of
+ * SCE[0] CPE[0] CPE[1] LFE[0].
+ * If we seem to have encountered such a stream, transfer
+ * the LFE[0] element to the SCE[1]'s mapping */
+ if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+ if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
+ av_log(ac->avctx, AV_LOG_WARNING,
+ "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
+ type == TYPE_SCE ? "SCE" : "LFE", elem_id);
+ ac->warned_remapping_once++;
+ }
+ ac->tags_mapped++;
+ return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
+ }
+ case 5:
+ if (ac->tags_mapped == 2 && type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
+ }
+ case 4:
+ /* Some streams incorrectly code 4.0 audio as
+ * SCE[0] CPE[0] LFE[0]
+ * instead of
+ * SCE[0] CPE[0] SCE[1].
+ * If we seem to have encountered such a stream, transfer
+ * the SCE[1] element to the LFE[0]'s mapping */
+ if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+ if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
+ av_log(ac->avctx, AV_LOG_WARNING,
+ "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
+ type == TYPE_SCE ? "SCE" : "LFE", elem_id);
+ ac->warned_remapping_once++;
+ }
+ ac->tags_mapped++;
+ return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
+ }
+ if (ac->tags_mapped == 2 &&
+ ac->oc[1].m4ac.chan_config == 4 &&
+ type == TYPE_SCE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+ }
+ case 3:
+ case 2:
+ if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
+ type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
+ } else if (ac->oc[1].m4ac.chan_config == 2) {
+ return NULL;
+ }
+ case 1:
+ if (!ac->tags_mapped && type == TYPE_SCE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
+ }
+ default:
+ return NULL;
+ }
+}
+
+/**
+ * Decode an array of 4 bit element IDs, optionally interleaved with a
+ * stereo/mono switching bit.
+ *
+ * @param type speaker type/position for these channels
+ */
+static void decode_channel_map(uint8_t layout_map[][3],
+ enum ChannelPosition type,
+ GetBitContext *gb, int n)
+{
+ while (n--) {
+ enum RawDataBlockType syn_ele;
+ switch (type) {
+ case AAC_CHANNEL_FRONT:
+ case AAC_CHANNEL_BACK:
+ case AAC_CHANNEL_SIDE:
+ syn_ele = get_bits1(gb);
+ break;
+ case AAC_CHANNEL_CC:
+ skip_bits1(gb);
+ syn_ele = TYPE_CCE;
+ break;
+ case AAC_CHANNEL_LFE:
+ syn_ele = TYPE_LFE;
+ break;
+ default:
+ // AAC_CHANNEL_OFF has no channel map
+ av_assert0(0);
+ }
+ layout_map[0][0] = syn_ele;
+ layout_map[0][1] = get_bits(gb, 4);
+ layout_map[0][2] = type;
+ layout_map++;
+ }
+}
+
+static inline void relative_align_get_bits(GetBitContext *gb,
+ int reference_position) {
+ int n = (reference_position - get_bits_count(gb) & 7);
+ if (n)
+ skip_bits(gb, n);
+}
+
+/**
+ * Decode program configuration element; reference: table 4.2.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
+ uint8_t (*layout_map)[3],
+ GetBitContext *gb, int byte_align_ref)
+{
+ int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
+ int sampling_index;
+ int comment_len;
+ int tags;
+
+ skip_bits(gb, 2); // object_type
+
+ sampling_index = get_bits(gb, 4);
+ if (m4ac->sampling_index != sampling_index)
+ av_log(avctx, AV_LOG_WARNING,
+ "Sample rate index in program config element does not "
+ "match the sample rate index configured by the container.\n");
+
+ num_front = get_bits(gb, 4);
+ num_side = get_bits(gb, 4);
+ num_back = get_bits(gb, 4);
+ num_lfe = get_bits(gb, 2);
+ num_assoc_data = get_bits(gb, 3);
+ num_cc = get_bits(gb, 4);
+
+ if (get_bits1(gb))
+ skip_bits(gb, 4); // mono_mixdown_tag
+ if (get_bits1(gb))
+ skip_bits(gb, 4); // stereo_mixdown_tag
+
+ if (get_bits1(gb))
+ skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
+
+ if (get_bits_left(gb) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(num_lfe + num_assoc_data + num_cc)) {
+ av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
+ return -1;
+ }
+ decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
+ tags = num_front;
+ decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
+ tags += num_side;
+ decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
+ tags += num_back;
+ decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
+ tags += num_lfe;
+
+ skip_bits_long(gb, 4 * num_assoc_data);
+
+ decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
+ tags += num_cc;
+
+ relative_align_get_bits(gb, byte_align_ref);
+
+ /* comment field, first byte is length */
+ comment_len = get_bits(gb, 8) * 8;
+ if (get_bits_left(gb) < comment_len) {
+ av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
+ return AVERROR_INVALIDDATA;
+ }
+ skip_bits_long(gb, comment_len);
+ return tags;
+}
+
+/**
+ * Decode GA "General Audio" specific configuration; reference: table 4.1.
+ *
+ * @param ac pointer to AACContext, may be null
+ * @param avctx pointer to AVCCodecContext, used for logging
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
+ GetBitContext *gb,
+ int get_bit_alignment,
+ MPEG4AudioConfig *m4ac,
+ int channel_config)
+{
+ int extension_flag, ret, ep_config, res_flags;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int tags = 0;
+
+#if USE_FIXED
+ if (get_bits1(gb)) { // frameLengthFlag
+ avpriv_report_missing_feature(avctx, "Fixed point 960/120 MDCT window");
+ return AVERROR_PATCHWELCOME;
+ }
+ m4ac->frame_length_short = 0;
+#else
+ m4ac->frame_length_short = get_bits1(gb);
+ if (m4ac->frame_length_short && m4ac->sbr == 1) {
+ avpriv_report_missing_feature(avctx, "SBR with 960 frame length");
+ if (ac) ac->warned_960_sbr = 1;
+ m4ac->sbr = 0;
+ m4ac->ps = 0;
+ }
+#endif
+
+ if (get_bits1(gb)) // dependsOnCoreCoder
+ skip_bits(gb, 14); // coreCoderDelay
+ extension_flag = get_bits1(gb);
+
+ if (m4ac->object_type == AOT_AAC_SCALABLE ||
+ m4ac->object_type == AOT_ER_AAC_SCALABLE)
+ skip_bits(gb, 3); // layerNr
+
+ if (channel_config == 0) {
+ skip_bits(gb, 4); // element_instance_tag
+ tags = decode_pce(avctx, m4ac, layout_map, gb, get_bit_alignment);
+ if (tags < 0)
+ return tags;
+ } else {
+ if ((ret = set_default_channel_config(avctx, layout_map,
+ &tags, channel_config)))
+ return ret;
+ }
+
+ if (count_channels(layout_map, tags) > 1) {
+ m4ac->ps = 0;
+ } else if (m4ac->sbr == 1 && m4ac->ps == -1)
+ m4ac->ps = 1;
+
+ if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
+ return ret;
+
+ if (extension_flag) {
+ switch (m4ac->object_type) {
+ case AOT_ER_BSAC:
+ skip_bits(gb, 5); // numOfSubFrame
+ skip_bits(gb, 11); // layer_length
+ break;
+ case AOT_ER_AAC_LC:
+ case AOT_ER_AAC_LTP:
+ case AOT_ER_AAC_SCALABLE:
+ case AOT_ER_AAC_LD:
+ res_flags = get_bits(gb, 3);
+ if (res_flags) {
+ avpriv_report_missing_feature(avctx,
+ "AAC data resilience (flags %x)",
+ res_flags);
+ return AVERROR_PATCHWELCOME;
+ }
+ break;
+ }
+ skip_bits1(gb); // extensionFlag3 (TBD in version 3)
+ }
+ switch (m4ac->object_type) {
+ case AOT_ER_AAC_LC:
+ case AOT_ER_AAC_LTP:
+ case AOT_ER_AAC_SCALABLE:
+ case AOT_ER_AAC_LD:
+ ep_config = get_bits(gb, 2);
+ if (ep_config) {
+ avpriv_report_missing_feature(avctx,
+ "epConfig %d", ep_config);
+ return AVERROR_PATCHWELCOME;
+ }
+ }
+ return 0;
+}
+
+static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
+ GetBitContext *gb,
+ MPEG4AudioConfig *m4ac,
+ int channel_config)
+{
+ int ret, ep_config, res_flags;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int tags = 0;
+ const int ELDEXT_TERM = 0;
+
+ m4ac->ps = 0;
+ m4ac->sbr = 0;
+#if USE_FIXED
+ if (get_bits1(gb)) { // frameLengthFlag
+ avpriv_request_sample(avctx, "960/120 MDCT window");
+ return AVERROR_PATCHWELCOME;
+ }
+#else
+ m4ac->frame_length_short = get_bits1(gb);
+#endif
+ res_flags = get_bits(gb, 3);
+ if (res_flags) {
+ avpriv_report_missing_feature(avctx,
+ "AAC data resilience (flags %x)",
+ res_flags);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ if (get_bits1(gb)) { // ldSbrPresentFlag
+ avpriv_report_missing_feature(avctx,
+ "Low Delay SBR");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ while (get_bits(gb, 4) != ELDEXT_TERM) {
+ int len = get_bits(gb, 4);
+ if (len == 15)
+ len += get_bits(gb, 8);
+ if (len == 15 + 255)
+ len += get_bits(gb, 16);
+ if (get_bits_left(gb) < len * 8 + 4) {
+ av_log(avctx, AV_LOG_ERROR, overread_err);
+ return AVERROR_INVALIDDATA;
+ }
+ skip_bits_long(gb, 8 * len);
+ }
+
+ if ((ret = set_default_channel_config(avctx, layout_map,
+ &tags, channel_config)))
+ return ret;
+
+ if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
+ return ret;
+
+ ep_config = get_bits(gb, 2);
+ if (ep_config) {
+ avpriv_report_missing_feature(avctx,
+ "epConfig %d", ep_config);
+ return AVERROR_PATCHWELCOME;
+ }
+ return 0;
+}
+
+/**
+ * Decode audio specific configuration; reference: table 1.13.
+ *
+ * @param ac pointer to AACContext, may be null
+ * @param avctx pointer to AVCCodecContext, used for logging
+ * @param m4ac pointer to MPEG4AudioConfig, used for parsing
+ * @param gb buffer holding an audio specific config
+ * @param get_bit_alignment relative alignment for byte align operations
+ * @param sync_extension look for an appended sync extension
+ *
+ * @return Returns error status or number of consumed bits. <0 - error
+ */
+static int decode_audio_specific_config_gb(AACContext *ac,
+ AVCodecContext *avctx,
+ MPEG4AudioConfig *m4ac,
+ GetBitContext *gb,
+ int get_bit_alignment,
+ int sync_extension)
+{
+ int i, ret;
+ GetBitContext gbc = *gb;
+
+ if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension)) < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (m4ac->sampling_index > 12) {
+ av_log(avctx, AV_LOG_ERROR,
+ "invalid sampling rate index %d\n",
+ m4ac->sampling_index);
+ return AVERROR_INVALIDDATA;
+ }
+ if (m4ac->object_type == AOT_ER_AAC_LD &&
+ (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
+ av_log(avctx, AV_LOG_ERROR,
+ "invalid low delay sampling rate index %d\n",
+ m4ac->sampling_index);
+ return AVERROR_INVALIDDATA;
+ }
+
+ skip_bits_long(gb, i);
+
+ switch (m4ac->object_type) {
+ case AOT_AAC_MAIN:
+ case AOT_AAC_LC:
+ case AOT_AAC_LTP:
+ case AOT_ER_AAC_LC:
+ case AOT_ER_AAC_LD:
+ if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,
+ m4ac, m4ac->chan_config)) < 0)
+ return ret;
+ break;
+ case AOT_ER_AAC_ELD:
+ if ((ret = decode_eld_specific_config(ac, avctx, gb,
+ m4ac, m4ac->chan_config)) < 0)
+ return ret;
+ break;
+ default:
+ avpriv_report_missing_feature(avctx,
+ "Audio object type %s%d",
+ m4ac->sbr == 1 ? "SBR+" : "",
+ m4ac->object_type);
+ return AVERROR(ENOSYS);
+ }
+
+ ff_dlog(avctx,
+ "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
+ m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
+ m4ac->sample_rate, m4ac->sbr,
+ m4ac->ps);
+
+ return get_bits_count(gb);
+}
+
+static int decode_audio_specific_config(AACContext *ac,
+ AVCodecContext *avctx,
+ MPEG4AudioConfig *m4ac,
+ const uint8_t *data, int64_t bit_size,
+ int sync_extension)
+{
+ int i, ret;
+ GetBitContext gb;
+
+ if (bit_size < 0 || bit_size > INT_MAX) {
+ av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
+ for (i = 0; i < bit_size >> 3; i++)
+ ff_dlog(avctx, "%02x ", data[i]);
+ ff_dlog(avctx, "\n");
+
+ if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
+ return ret;
+
+ return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
+ sync_extension);
+}
+
+/**
+ * linear congruential pseudorandom number generator
+ *
+ * @param previous_val pointer to the current state of the generator
+ *
+ * @return Returns a 32-bit pseudorandom integer
+ */
+static av_always_inline int lcg_random(unsigned previous_val)
+{
+ union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
+ return v.s;
+}
+
+static void reset_all_predictors(PredictorState *ps)
+{
+ int i;
+ for (i = 0; i < MAX_PREDICTORS; i++)
+ reset_predict_state(&ps[i]);
+}
+
+static int sample_rate_idx (int rate)
+{
+ if (92017 <= rate) return 0;
+ else if (75132 <= rate) return 1;
+ else if (55426 <= rate) return 2;
+ else if (46009 <= rate) return 3;
+ else if (37566 <= rate) return 4;
+ else if (27713 <= rate) return 5;
+ else if (23004 <= rate) return 6;
+ else if (18783 <= rate) return 7;
+ else if (13856 <= rate) return 8;
+ else if (11502 <= rate) return 9;
+ else if (9391 <= rate) return 10;
+ else return 11;
+}
+
+static void reset_predictor_group(PredictorState *ps, int group_num)
+{
+ int i;
+ for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
+ reset_predict_state(&ps[i]);
+}
+
+#define AAC_INIT_VLC_STATIC(num, size) \
+ INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
+ ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
+ sizeof(ff_aac_spectral_bits[num][0]), \
+ ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
+ sizeof(ff_aac_spectral_codes[num][0]), \
+ size);
+
+static void aacdec_init(AACContext *ac);
+
+static av_cold void aac_static_table_init(void)
+{
+ AAC_INIT_VLC_STATIC( 0, 304);
+ AAC_INIT_VLC_STATIC( 1, 270);
+ AAC_INIT_VLC_STATIC( 2, 550);
+ AAC_INIT_VLC_STATIC( 3, 300);
+ AAC_INIT_VLC_STATIC( 4, 328);
+ AAC_INIT_VLC_STATIC( 5, 294);
+ AAC_INIT_VLC_STATIC( 6, 306);
+ AAC_INIT_VLC_STATIC( 7, 268);
+ AAC_INIT_VLC_STATIC( 8, 510);
+ AAC_INIT_VLC_STATIC( 9, 366);
+ AAC_INIT_VLC_STATIC(10, 462);
+
+ AAC_RENAME(ff_aac_sbr_init)();
+
+ ff_aac_tableinit();
+
+ INIT_VLC_STATIC(&vlc_scalefactors, 7,
+ FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
+ ff_aac_scalefactor_bits,
+ sizeof(ff_aac_scalefactor_bits[0]),
+ sizeof(ff_aac_scalefactor_bits[0]),
+ ff_aac_scalefactor_code,
+ sizeof(ff_aac_scalefactor_code[0]),
+ sizeof(ff_aac_scalefactor_code[0]),
+ 352);
+
+ // window initialization
+ AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
+ AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
+#if !USE_FIXED
+ AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_960), 4.0, 960);
+ AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_120), 6.0, 120);
+ AAC_RENAME(ff_sine_window_init)(AAC_RENAME(ff_sine_960), 960);
+ AAC_RENAME(ff_sine_window_init)(AAC_RENAME(ff_sine_120), 120);
+#endif
+ AAC_RENAME(ff_init_ff_sine_windows)(10);
+ AAC_RENAME(ff_init_ff_sine_windows)( 9);
+ AAC_RENAME(ff_init_ff_sine_windows)( 7);
+
+ AAC_RENAME(ff_cbrt_tableinit)();
+}
+
+static AVOnce aac_table_init = AV_ONCE_INIT;
+
+static av_cold int aac_decode_init(AVCodecContext *avctx)
+{
+ AACContext *ac = avctx->priv_data;
+ int ret;
+
+ ret = ff_thread_once(&aac_table_init, &aac_static_table_init);
+ if (ret != 0)
+ return AVERROR_UNKNOWN;
+
+ ac->avctx = avctx;
+ ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
+
+ aacdec_init(ac);
+#if USE_FIXED
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+#else
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+#endif /* USE_FIXED */
+
+ if (avctx->extradata_size > 0) {
+ if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+ avctx->extradata,
+ avctx->extradata_size * 8LL,
+ 1)) < 0)
+ return ret;
+ } else {
+ int sr, i;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags;
+
+ sr = sample_rate_idx(avctx->sample_rate);
+ ac->oc[1].m4ac.sampling_index = sr;
+ ac->oc[1].m4ac.channels = avctx->channels;
+ ac->oc[1].m4ac.sbr = -1;
+ ac->oc[1].m4ac.ps = -1;
+
+ for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
+ if (ff_mpeg4audio_channels[i] == avctx->channels)
+ break;
+ if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
+ i = 0;
+ }
+ ac->oc[1].m4ac.chan_config = i;
+
+ if (ac->oc[1].m4ac.chan_config) {
+ int ret = set_default_channel_config(avctx, layout_map,
+ &layout_map_tags, ac->oc[1].m4ac.chan_config);
+ if (!ret)
+ output_configure(ac, layout_map, layout_map_tags,
+ OC_GLOBAL_HDR, 0);
+ else if (avctx->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ if (avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+#if USE_FIXED
+ ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+#else
+ ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+#endif /* USE_FIXED */
+ if (!ac->fdsp) {
+ return AVERROR(ENOMEM);
+ }
+
+ ac->random_state = 0x1f2e3d4c;
+
+ AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
+ AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
+ AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
+ AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
+#if !USE_FIXED
+ ret = ff_mdct15_init(&ac->mdct120, 1, 3, 1.0f/(16*1024*120*2));
+ if (ret < 0)
+ return ret;
+ ret = ff_mdct15_init(&ac->mdct480, 1, 5, 1.0f/(16*1024*960));
+ if (ret < 0)
+ return ret;
+ ret = ff_mdct15_init(&ac->mdct960, 1, 6, 1.0f/(16*1024*960*2));
+ if (ret < 0)
+ return ret;
+#endif
+
+ return 0;
+}
+
+/**
+ * Skip data_stream_element; reference: table 4.10.
+ */
+static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
+{
+ int byte_align = get_bits1(gb);
+ int count = get_bits(gb, 8);
+ if (count == 255)
+ count += get_bits(gb, 8);
+ if (byte_align)
+ align_get_bits(gb);
+
+ if (get_bits_left(gb) < 8 * count) {
+ av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
+ return AVERROR_INVALIDDATA;
+ }
+ skip_bits_long(gb, 8 * count);
+ return 0;
+}
+
+static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
+ GetBitContext *gb)
+{
+ int sfb;
+ if (get_bits1(gb)) {
+ ics->predictor_reset_group = get_bits(gb, 5);
+ if (ics->predictor_reset_group == 0 ||
+ ics->predictor_reset_group > 30) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Invalid Predictor Reset Group.\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+ for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
+ ics->prediction_used[sfb] = get_bits1(gb);
+ }
+ return 0;
+}
+
+/**
+ * Decode Long Term Prediction data; reference: table 4.xx.
+ */
+static void decode_ltp(LongTermPrediction *ltp,
+ GetBitContext *gb, uint8_t max_sfb)
+{
+ int sfb;
+
+ ltp->lag = get_bits(gb, 11);
+ ltp->coef = ltp_coef[get_bits(gb, 3)];
+ for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
+ ltp->used[sfb] = get_bits1(gb);
+}
+
+/**
+ * Decode Individual Channel Stream info; reference: table 4.6.
+ */
+static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
+ GetBitContext *gb)
+{
+ const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
+ const int aot = m4ac->object_type;
+ const int sampling_index = m4ac->sampling_index;
+ int ret_fail = AVERROR_INVALIDDATA;
+
+ if (aot != AOT_ER_AAC_ELD) {
+ if (get_bits1(gb)) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
+ if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
+ return AVERROR_INVALIDDATA;
+ }
+ ics->window_sequence[1] = ics->window_sequence[0];
+ ics->window_sequence[0] = get_bits(gb, 2);
+ if (aot == AOT_ER_AAC_LD &&
+ ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
+ "window sequence %d found.\n", ics->window_sequence[0]);
+ ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
+ return AVERROR_INVALIDDATA;
+ }
+ ics->use_kb_window[1] = ics->use_kb_window[0];
+ ics->use_kb_window[0] = get_bits1(gb);
+ }
+ ics->num_window_groups = 1;
+ ics->group_len[0] = 1;
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ int i;
+ ics->max_sfb = get_bits(gb, 4);
+ for (i = 0; i < 7; i++) {
+ if (get_bits1(gb)) {
+ ics->group_len[ics->num_window_groups - 1]++;
+ } else {
+ ics->num_window_groups++;
+ ics->group_len[ics->num_window_groups - 1] = 1;
+ }
+ }
+ ics->num_windows = 8;
+ if (m4ac->frame_length_short) {
+ ics->swb_offset = ff_swb_offset_120[sampling_index];
+ ics->num_swb = ff_aac_num_swb_120[sampling_index];
+ } else {
+ ics->swb_offset = ff_swb_offset_128[sampling_index];
+ ics->num_swb = ff_aac_num_swb_128[sampling_index];
+ }
+ ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
+ ics->predictor_present = 0;
+ } else {
+ ics->max_sfb = get_bits(gb, 6);
+ ics->num_windows = 1;
+ if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
+ if (m4ac->frame_length_short) {
+ ics->swb_offset = ff_swb_offset_480[sampling_index];
+ ics->num_swb = ff_aac_num_swb_480[sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
+ } else {
+ ics->swb_offset = ff_swb_offset_512[sampling_index];
+ ics->num_swb = ff_aac_num_swb_512[sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
+ }
+ if (!ics->num_swb || !ics->swb_offset) {
+ ret_fail = AVERROR_BUG;
+ goto fail;
+ }
+ } else {
+ if (m4ac->frame_length_short) {
+ ics->num_swb = ff_aac_num_swb_960[sampling_index];
+ ics->swb_offset = ff_swb_offset_960[sampling_index];
+ } else {
+ ics->num_swb = ff_aac_num_swb_1024[sampling_index];
+ ics->swb_offset = ff_swb_offset_1024[sampling_index];
+ }
+ ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
+ }
+ if (aot != AOT_ER_AAC_ELD) {
+ ics->predictor_present = get_bits1(gb);
+ ics->predictor_reset_group = 0;
+ }
+ if (ics->predictor_present) {
+ if (aot == AOT_AAC_MAIN) {
+ if (decode_prediction(ac, ics, gb)) {
+ goto fail;
+ }
+ } else if (aot == AOT_AAC_LC ||
+ aot == AOT_ER_AAC_LC) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Prediction is not allowed in AAC-LC.\n");
+ goto fail;
+ } else {
+ if (aot == AOT_ER_AAC_LD) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "LTP in ER AAC LD not yet implemented.\n");
+ ret_fail = AVERROR_PATCHWELCOME;
+ goto fail;
+ }
+ if ((ics->ltp.present = get_bits(gb, 1)))
+ decode_ltp(&ics->ltp, gb, ics->max_sfb);
+ }
+ }
+ }
+
+ if (ics->max_sfb > ics->num_swb) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Number of scalefactor bands in group (%d) "
+ "exceeds limit (%d).\n",
+ ics->max_sfb, ics->num_swb);
+ goto fail;
+ }
+
+ return 0;
+fail:
+ ics->max_sfb = 0;
+ return ret_fail;
+}
+
+/**
+ * Decode band types (section_data payload); reference: table 4.46.
+ *
+ * @param band_type array of the used band type
+ * @param band_type_run_end array of the last scalefactor band of a band type run
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_band_types(AACContext *ac, enum BandType band_type[120],
+ int band_type_run_end[120], GetBitContext *gb,
+ IndividualChannelStream *ics)
+{
+ int g, idx = 0;
+ const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ int k = 0;
+ while (k < ics->max_sfb) {
+ uint8_t sect_end = k;
+ int sect_len_incr;
+ int sect_band_type = get_bits(gb, 4);
+ if (sect_band_type == 12) {
+ av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
+ return AVERROR_INVALIDDATA;
+ }
+ do {
+ sect_len_incr = get_bits(gb, bits);
+ sect_end += sect_len_incr;
+ if (get_bits_left(gb) < 0) {
+ av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
+ return AVERROR_INVALIDDATA;
+ }
+ if (sect_end > ics->max_sfb) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Number of bands (%d) exceeds limit (%d).\n",
+ sect_end, ics->max_sfb);
+ return AVERROR_INVALIDDATA;
+ }
+ } while (sect_len_incr == (1 << bits) - 1);
+ for (; k < sect_end; k++) {
+ band_type [idx] = sect_band_type;
+ band_type_run_end[idx++] = sect_end;
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * Decode scalefactors; reference: table 4.47.
+ *
+ * @param global_gain first scalefactor value as scalefactors are differentially coded
+ * @param band_type array of the used band type
+ * @param band_type_run_end array of the last scalefactor band of a band type run
+ * @param sf array of scalefactors or intensity stereo positions
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
+ unsigned int global_gain,
+ IndividualChannelStream *ics,
+ enum BandType band_type[120],
+ int band_type_run_end[120])
+{
+ int g, i, idx = 0;
+ int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
+ int clipped_offset;
+ int noise_flag = 1;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb;) {
+ int run_end = band_type_run_end[idx];
+ if (band_type[idx] == ZERO_BT) {
+ for (; i < run_end; i++, idx++)
+ sf[idx] = FIXR(0.);
+ } else if ((band_type[idx] == INTENSITY_BT) ||
+ (band_type[idx] == INTENSITY_BT2)) {
+ for (; i < run_end; i++, idx++) {
+ offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
+ clipped_offset = av_clip(offset[2], -155, 100);
+ if (offset[2] != clipped_offset) {
+ avpriv_request_sample(ac->avctx,
+ "If you heard an audible artifact, there may be a bug in the decoder. "
+ "Clipped intensity stereo position (%d -> %d)",
+ offset[2], clipped_offset);
+ }
+#if USE_FIXED
+ sf[idx] = 100 - clipped_offset;
+#else
+ sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
+#endif /* USE_FIXED */
+ }
+ } else if (band_type[idx] == NOISE_BT) {
+ for (; i < run_end; i++, idx++) {
+ if (noise_flag-- > 0)
+ offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
+ else
+ offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
+ clipped_offset = av_clip(offset[1], -100, 155);
+ if (offset[1] != clipped_offset) {
+ avpriv_request_sample(ac->avctx,
+ "If you heard an audible artifact, there may be a bug in the decoder. "
+ "Clipped noise gain (%d -> %d)",
+ offset[1], clipped_offset);
+ }
+#if USE_FIXED
+ sf[idx] = -(100 + clipped_offset);
+#else
+ sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
+#endif /* USE_FIXED */
+ }
+ } else {
+ for (; i < run_end; i++, idx++) {
+ offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
+ if (offset[0] > 255U) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Scalefactor (%d) out of range.\n", offset[0]);
+ return AVERROR_INVALIDDATA;
+ }
+#if USE_FIXED
+ sf[idx] = -offset[0];
+#else
+ sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
+#endif /* USE_FIXED */
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * Decode pulse data; reference: table 4.7.
+ */
+static int decode_pulses(Pulse *pulse, GetBitContext *gb,
+ const uint16_t *swb_offset, int num_swb)
+{
+ int i, pulse_swb;
+ pulse->num_pulse = get_bits(gb, 2) + 1;
+ pulse_swb = get_bits(gb, 6);
+ if (pulse_swb >= num_swb)
+ return -1;
+ pulse->pos[0] = swb_offset[pulse_swb];
+ pulse->pos[0] += get_bits(gb, 5);
+ if (pulse->pos[0] >= swb_offset[num_swb])
+ return -1;
+ pulse->amp[0] = get_bits(gb, 4);
+ for (i = 1; i < pulse->num_pulse; i++) {
+ pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
+ if (pulse->pos[i] >= swb_offset[num_swb])
+ return -1;
+ pulse->amp[i] = get_bits(gb, 4);
+ }
+ return 0;
+}
+
+/**
+ * Decode Temporal Noise Shaping data; reference: table 4.48.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
+ GetBitContext *gb, const IndividualChannelStream *ics)
+{
+ int w, filt, i, coef_len, coef_res, coef_compress;
+ const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+ const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
+ for (w = 0; w < ics->num_windows; w++) {
+ if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
+ coef_res = get_bits1(gb);
+
+ for (filt = 0; filt < tns->n_filt[w]; filt++) {
+ int tmp2_idx;
+ tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
+
+ if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "TNS filter order %d is greater than maximum %d.\n",
+ tns->order[w][filt], tns_max_order);
+ tns->order[w][filt] = 0;
+ return AVERROR_INVALIDDATA;
+ }
+ if (tns->order[w][filt]) {
+ tns->direction[w][filt] = get_bits1(gb);
+ coef_compress = get_bits1(gb);
+ coef_len = coef_res + 3 - coef_compress;
+ tmp2_idx = 2 * coef_compress + coef_res;
+
+ for (i = 0; i < tns->order[w][filt]; i++)
+ tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * Decode Mid/Side data; reference: table 4.54.
+ *
+ * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
+ * [1] mask is decoded from bitstream; [2] mask is all 1s;
+ * [3] reserved for scalable AAC
+ */
+static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
+ int ms_present)
+{
+ int idx;
+ int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
+ if (ms_present == 1) {
+ for (idx = 0; idx < max_idx; idx++)
+ cpe->ms_mask[idx] = get_bits1(gb);
+ } else if (ms_present == 2) {
+ memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
+ }
+}
+
+/**
+ * Decode spectral data; reference: table 4.50.
+ * Dequantize and scale spectral data; reference: 4.6.3.3.
+ *
+ * @param coef array of dequantized, scaled spectral data
+ * @param sf array of scalefactors or intensity stereo positions
+ * @param pulse_present set if pulses are present
+ * @param pulse pointer to pulse data struct
+ * @param band_type array of the used band type
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
+ GetBitContext *gb, const INTFLOAT sf[120],
+ int pulse_present, const Pulse *pulse,
+ const IndividualChannelStream *ics,
+ enum BandType band_type[120])
+{
+ int i, k, g, idx = 0;
+ const int c = 1024 / ics->num_windows;
+ const uint16_t *offsets = ics->swb_offset;
+ INTFLOAT *coef_base = coef;
+
+ for (g = 0; g < ics->num_windows; g++)
+ memset(coef + g * 128 + offsets[ics->max_sfb], 0,
+ sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
+
+ for (g = 0; g < ics->num_window_groups; g++) {
+ unsigned g_len = ics->group_len[g];
+
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ const unsigned cbt_m1 = band_type[idx] - 1;
+ INTFLOAT *cfo = coef + offsets[i];
+ int off_len = offsets[i + 1] - offsets[i];
+ int group;
+
+ if (cbt_m1 >= INTENSITY_BT2 - 1) {
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ memset(cfo, 0, off_len * sizeof(*cfo));
+ }
+ } else if (cbt_m1 == NOISE_BT - 1) {
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+#if !USE_FIXED
+ float scale;
+#endif /* !USE_FIXED */
+ INTFLOAT band_energy;
+
+ for (k = 0; k < off_len; k++) {
+ ac->random_state = lcg_random(ac->random_state);
+#if USE_FIXED
+ cfo[k] = ac->random_state >> 3;
+#else
+ cfo[k] = ac->random_state;
+#endif /* USE_FIXED */
+ }
+
+#if USE_FIXED
+ band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
+ band_energy = fixed_sqrt(band_energy, 31);
+ noise_scale(cfo, sf[idx], band_energy, off_len);
+#else
+ band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
+ scale = sf[idx] / sqrtf(band_energy);
+ ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
+#endif /* USE_FIXED */
+ }
+ } else {
+#if !USE_FIXED
+ const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
+#endif /* !USE_FIXED */
+ const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
+ VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
+ OPEN_READER(re, gb);
+
+ switch (cbt_m1 >> 1) {
+ case 0:
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ INTFLOAT *cf = cfo;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned cb_idx;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+ cb_idx = cb_vector_idx[code];
+#if USE_FIXED
+ cf = DEC_SQUAD(cf, cb_idx);
+#else
+ cf = VMUL4(cf, vq, cb_idx, sf + idx);
+#endif /* USE_FIXED */
+ } while (len -= 4);
+ }
+ break;
+
+ case 1:
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ INTFLOAT *cf = cfo;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned nnz;
+ unsigned cb_idx;
+ uint32_t bits;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+ cb_idx = cb_vector_idx[code];
+ nnz = cb_idx >> 8 & 15;
+ bits = nnz ? GET_CACHE(re, gb) : 0;
+ LAST_SKIP_BITS(re, gb, nnz);
+#if USE_FIXED
+ cf = DEC_UQUAD(cf, cb_idx, bits);
+#else
+ cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
+#endif /* USE_FIXED */
+ } while (len -= 4);
+ }
+ break;
+
+ case 2:
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ INTFLOAT *cf = cfo;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned cb_idx;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+ cb_idx = cb_vector_idx[code];
+#if USE_FIXED
+ cf = DEC_SPAIR(cf, cb_idx);
+#else
+ cf = VMUL2(cf, vq, cb_idx, sf + idx);
+#endif /* USE_FIXED */
+ } while (len -= 2);
+ }
+ break;
+
+ case 3:
+ case 4:
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+ INTFLOAT *cf = cfo;
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned nnz;
+ unsigned cb_idx;
+ unsigned sign;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+ cb_idx = cb_vector_idx[code];
+ nnz = cb_idx >> 8 & 15;
+ sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
+ LAST_SKIP_BITS(re, gb, nnz);
+#if USE_FIXED
+ cf = DEC_UPAIR(cf, cb_idx, sign);
+#else
+ cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
+#endif /* USE_FIXED */
+ } while (len -= 2);
+ }
+ break;
+
+ default:
+ for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
+#if USE_FIXED
+ int *icf = cfo;
+ int v;
+#else
+ float *cf = cfo;
+ uint32_t *icf = (uint32_t *) cf;
+#endif /* USE_FIXED */
+ int len = off_len;
+
+ do {
+ int code;
+ unsigned nzt, nnz;
+ unsigned cb_idx;
+ uint32_t bits;
+ int j;
+
+ UPDATE_CACHE(re, gb);
+ GET_VLC(code, re, gb, vlc_tab, 8, 2);
+
+ if (!code) {
+ *icf++ = 0;
+ *icf++ = 0;
+ continue;
+ }
+
+ cb_idx = cb_vector_idx[code];
+ nnz = cb_idx >> 12;
+ nzt = cb_idx >> 8;
+ bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
+ LAST_SKIP_BITS(re, gb, nnz);
+
+ for (j = 0; j < 2; j++) {
+ if (nzt & 1<<j) {
+ uint32_t b;
+ int n;
+ /* The total length of escape_sequence must be < 22 bits according
+ to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
+ UPDATE_CACHE(re, gb);
+ b = GET_CACHE(re, gb);
+ b = 31 - av_log2(~b);
+
+ if (b > 8) {
+ av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ SKIP_BITS(re, gb, b + 1);
+ b += 4;
+ n = (1 << b) + SHOW_UBITS(re, gb, b);
+ LAST_SKIP_BITS(re, gb, b);
+#if USE_FIXED
+ v = n;
+ if (bits & 1U<<31)
+ v = -v;
+ *icf++ = v;
+#else
+ *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
+#endif /* USE_FIXED */
+ bits <<= 1;
+ } else {
+#if USE_FIXED
+ v = cb_idx & 15;
+ if (bits & 1U<<31)
+ v = -v;
+ *icf++ = v;
+#else
+ unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
+ *icf++ = (bits & 1U<<31) | v;
+#endif /* USE_FIXED */
+ bits <<= !!v;
+ }
+ cb_idx >>= 4;
+ }
+ } while (len -= 2);
+#if !USE_FIXED
+ ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
+#endif /* !USE_FIXED */
+ }
+ }
+
+ CLOSE_READER(re, gb);
+ }
+ }
+ coef += g_len << 7;
+ }
+
+ if (pulse_present) {
+ idx = 0;
+ for (i = 0; i < pulse->num_pulse; i++) {
+ INTFLOAT co = coef_base[ pulse->pos[i] ];
+ while (offsets[idx + 1] <= pulse->pos[i])
+ idx++;
+ if (band_type[idx] != NOISE_BT && sf[idx]) {
+ INTFLOAT ico = -pulse->amp[i];
+#if USE_FIXED
+ if (co) {
+ ico = co + (co > 0 ? -ico : ico);
+ }
+ coef_base[ pulse->pos[i] ] = ico;
+#else
+ if (co) {
+ co /= sf[idx];
+ ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
+ }
+ coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
+#endif /* USE_FIXED */
+ }
+ }
+ }
+#if USE_FIXED
+ coef = coef_base;
+ idx = 0;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ unsigned g_len = ics->group_len[g];
+
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ const unsigned cbt_m1 = band_type[idx] - 1;
+ int *cfo = coef + offsets[i];
+ int off_len = offsets[i + 1] - offsets[i];
+ int group;
+
+ if (cbt_m1 < NOISE_BT - 1) {
+ for (group = 0; group < (int)g_len; group++, cfo+=128) {
+ ac->vector_pow43(cfo, off_len);
+ ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
+ }
+ }
+ }
+ coef += g_len << 7;
+ }
+#endif /* USE_FIXED */
+ return 0;
+}
+
+/**
+ * Apply AAC-Main style frequency domain prediction.
+ */
+static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
+{
+ int sfb, k;
+
+ if (!sce->ics.predictor_initialized) {
+ reset_all_predictors(sce->predictor_state);
+ sce->ics.predictor_initialized = 1;
+ }
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+ for (sfb = 0;
+ sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
+ sfb++) {
+ for (k = sce->ics.swb_offset[sfb];
+ k < sce->ics.swb_offset[sfb + 1];
+ k++) {
+ predict(&sce->predictor_state[k], &sce->coeffs[k],
+ sce->ics.predictor_present &&
+ sce->ics.prediction_used[sfb]);
+ }
+ }
+ if (sce->ics.predictor_reset_group)
+ reset_predictor_group(sce->predictor_state,
+ sce->ics.predictor_reset_group);
+ } else
+ reset_all_predictors(sce->predictor_state);
+}
+
+/**
+ * Decode an individual_channel_stream payload; reference: table 4.44.
+ *
+ * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
+ * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_ics(AACContext *ac, SingleChannelElement *sce,
+ GetBitContext *gb, int common_window, int scale_flag)
+{
+ Pulse pulse;
+ TemporalNoiseShaping *tns = &sce->tns;
+ IndividualChannelStream *ics = &sce->ics;
+ INTFLOAT *out = sce->coeffs;
+ int global_gain, eld_syntax, er_syntax, pulse_present = 0;
+ int ret;
+
+ eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+ er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
+ ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
+ ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
+ ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+
+ /* This assignment is to silence a GCC warning about the variable being used
+ * uninitialized when in fact it always is.
+ */
+ pulse.num_pulse = 0;
+
+ global_gain = get_bits(gb, 8);
+
+ if (!common_window && !scale_flag) {
+ ret = decode_ics_info(ac, ics, gb);
+ if (ret < 0)
+ goto fail;
+ }
+
+ if ((ret = decode_band_types(ac, sce->band_type,
+ sce->band_type_run_end, gb, ics)) < 0)
+ goto fail;
+ if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
+ sce->band_type, sce->band_type_run_end)) < 0)
+ goto fail;
+
+ pulse_present = 0;
+ if (!scale_flag) {
+ if (!eld_syntax && (pulse_present = get_bits1(gb))) {
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Pulse tool not allowed in eight short sequence.\n");
+ ret = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "Pulse data corrupt or invalid.\n");
+ ret = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ }
+ tns->present = get_bits1(gb);
+ if (tns->present && !er_syntax) {
+ ret = decode_tns(ac, tns, gb, ics);
+ if (ret < 0)
+ goto fail;
+ }
+ if (!eld_syntax && get_bits1(gb)) {
+ avpriv_request_sample(ac->avctx, "SSR");
+ ret = AVERROR_PATCHWELCOME;
+ goto fail;
+ }
+ // I see no textual basis in the spec for this occurring after SSR gain
+ // control, but this is what both reference and real implmentations do
+ if (tns->present && er_syntax) {
+ ret = decode_tns(ac, tns, gb, ics);
+ if (ret < 0)
+ goto fail;
+ }
+ }
+
+ ret = decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
+ &pulse, ics, sce->band_type);
+ if (ret < 0)
+ goto fail;
+
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
+ apply_prediction(ac, sce);
+
+ return 0;
+fail:
+ tns->present = 0;
+ return ret;
+}
+
+/**
+ * Mid/Side stereo decoding; reference: 4.6.8.1.3.
+ */
+static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
+{
+ const IndividualChannelStream *ics = &cpe->ch[0].ics;
+ INTFLOAT *ch0 = cpe->ch[0].coeffs;
+ INTFLOAT *ch1 = cpe->ch[1].coeffs;
+ int g, i, group, idx = 0;
+ const uint16_t *offsets = ics->swb_offset;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ if (cpe->ms_mask[idx] &&
+ cpe->ch[0].band_type[idx] < NOISE_BT &&
+ cpe->ch[1].band_type[idx] < NOISE_BT) {
+#if USE_FIXED
+ for (group = 0; group < ics->group_len[g]; group++) {
+ ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
+ ch1 + group * 128 + offsets[i],
+ offsets[i+1] - offsets[i]);
+#else
+ for (group = 0; group < ics->group_len[g]; group++) {
+ ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
+ ch1 + group * 128 + offsets[i],
+ offsets[i+1] - offsets[i]);
+#endif /* USE_FIXED */
+ }
+ }
+ }
+ ch0 += ics->group_len[g] * 128;
+ ch1 += ics->group_len[g] * 128;
+ }
+}
+
+/**
+ * intensity stereo decoding; reference: 4.6.8.2.3
+ *
+ * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
+ * [1] mask is decoded from bitstream; [2] mask is all 1s;
+ * [3] reserved for scalable AAC
+ */
+static void apply_intensity_stereo(AACContext *ac,
+ ChannelElement *cpe, int ms_present)
+{
+ const IndividualChannelStream *ics = &cpe->ch[1].ics;
+ SingleChannelElement *sce1 = &cpe->ch[1];
+ INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
+ const uint16_t *offsets = ics->swb_offset;
+ int g, group, i, idx = 0;
+ int c;
+ INTFLOAT scale;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb;) {
+ if (sce1->band_type[idx] == INTENSITY_BT ||
+ sce1->band_type[idx] == INTENSITY_BT2) {
+ const int bt_run_end = sce1->band_type_run_end[idx];
+ for (; i < bt_run_end; i++, idx++) {
+ c = -1 + 2 * (sce1->band_type[idx] - 14);
+ if (ms_present)
+ c *= 1 - 2 * cpe->ms_mask[idx];
+ scale = c * sce1->sf[idx];
+ for (group = 0; group < ics->group_len[g]; group++)
+#if USE_FIXED
+ ac->subband_scale(coef1 + group * 128 + offsets[i],
+ coef0 + group * 128 + offsets[i],
+ scale,
+ 23,
+ offsets[i + 1] - offsets[i]);
+#else
+ ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
+ coef0 + group * 128 + offsets[i],
+ scale,
+ offsets[i + 1] - offsets[i]);
+#endif /* USE_FIXED */
+ }
+ } else {
+ int bt_run_end = sce1->band_type_run_end[idx];
+ idx += bt_run_end - i;
+ i = bt_run_end;
+ }
+ }
+ coef0 += ics->group_len[g] * 128;
+ coef1 += ics->group_len[g] * 128;
+ }
+}
+
+/**
+ * Decode a channel_pair_element; reference: table 4.4.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
+{
+ int i, ret, common_window, ms_present = 0;
+ int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+
+ common_window = eld_syntax || get_bits1(gb);
+ if (common_window) {
+ if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
+ return AVERROR_INVALIDDATA;
+ i = cpe->ch[1].ics.use_kb_window[0];
+ cpe->ch[1].ics = cpe->ch[0].ics;
+ cpe->ch[1].ics.use_kb_window[1] = i;
+ if (cpe->ch[1].ics.predictor_present &&
+ (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
+ if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
+ decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
+ ms_present = get_bits(gb, 2);
+ if (ms_present == 3) {
+ av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
+ return AVERROR_INVALIDDATA;
+ } else if (ms_present)
+ decode_mid_side_stereo(cpe, gb, ms_present);
+ }
+ if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
+ return ret;
+ if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
+ return ret;
+
+ if (common_window) {
+ if (ms_present)
+ apply_mid_side_stereo(ac, cpe);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
+ apply_prediction(ac, &cpe->ch[0]);
+ apply_prediction(ac, &cpe->ch[1]);
+ }
+ }
+
+ apply_intensity_stereo(ac, cpe, ms_present);
+ return 0;
+}
+
+static const float cce_scale[] = {
+ 1.09050773266525765921, //2^(1/8)
+ 1.18920711500272106672, //2^(1/4)
+ M_SQRT2,
+ 2,
+};
+
+/**
+ * Decode coupling_channel_element; reference: table 4.8.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
+{
+ int num_gain = 0;
+ int c, g, sfb, ret;
+ int sign;
+ INTFLOAT scale;
+ SingleChannelElement *sce = &che->ch[0];
+ ChannelCoupling *coup = &che->coup;
+
+ coup->coupling_point = 2 * get_bits1(gb);
+ coup->num_coupled = get_bits(gb, 3);
+ for (c = 0; c <= coup->num_coupled; c++) {
+ num_gain++;
+ coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
+ coup->id_select[c] = get_bits(gb, 4);
+ if (coup->type[c] == TYPE_CPE) {
+ coup->ch_select[c] = get_bits(gb, 2);
+ if (coup->ch_select[c] == 3)
+ num_gain++;
+ } else
+ coup->ch_select[c] = 2;
+ }
+ coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
+
+ sign = get_bits(gb, 1);
+#if USE_FIXED
+ scale = get_bits(gb, 2);
+#else
+ scale = cce_scale[get_bits(gb, 2)];
+#endif
+
+ if ((ret = decode_ics(ac, sce, gb, 0, 0)))
+ return ret;
+
+ for (c = 0; c < num_gain; c++) {
+ int idx = 0;
+ int cge = 1;
+ int gain = 0;
+ INTFLOAT gain_cache = FIXR10(1.);
+ if (c) {
+ cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
+ gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
+ gain_cache = GET_GAIN(scale, gain);
+#if USE_FIXED
+ if ((abs(gain_cache)-1024) >> 3 > 30)
+ return AVERROR(ERANGE);
+#endif
+ }
+ if (coup->coupling_point == AFTER_IMDCT) {
+ coup->gain[c][0] = gain_cache;
+ } else {
+ for (g = 0; g < sce->ics.num_window_groups; g++) {
+ for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
+ if (sce->band_type[idx] != ZERO_BT) {
+ if (!cge) {
+ int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+ if (t) {
+ int s = 1;
+ t = gain += t;
+ if (sign) {
+ s -= 2 * (t & 0x1);
+ t >>= 1;
+ }
+ gain_cache = GET_GAIN(scale, t) * s;
+#if USE_FIXED
+ if ((abs(gain_cache)-1024) >> 3 > 30)
+ return AVERROR(ERANGE);
+#endif
+ }
+ }
+ coup->gain[c][idx] = gain_cache;
+ }
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
+ *
+ * @return Returns number of bytes consumed.
+ */
+static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
+ GetBitContext *gb)
+{
+ int i;
+ int num_excl_chan = 0;
+
+ do {
+ for (i = 0; i < 7; i++)
+ che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
+ } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
+
+ return num_excl_chan / 7;
+}
+
+/**
+ * Decode dynamic range information; reference: table 4.52.
+ *
+ * @return Returns number of bytes consumed.
+ */
+static int decode_dynamic_range(DynamicRangeControl *che_drc,
+ GetBitContext *gb)
+{
+ int n = 1;
+ int drc_num_bands = 1;
+ int i;
+
+ /* pce_tag_present? */
+ if (get_bits1(gb)) {
+ che_drc->pce_instance_tag = get_bits(gb, 4);
+ skip_bits(gb, 4); // tag_reserved_bits
+ n++;
+ }
+
+ /* excluded_chns_present? */
+ if (get_bits1(gb)) {
+ n += decode_drc_channel_exclusions(che_drc, gb);
+ }
+
+ /* drc_bands_present? */
+ if (get_bits1(gb)) {
+ che_drc->band_incr = get_bits(gb, 4);
+ che_drc->interpolation_scheme = get_bits(gb, 4);
+ n++;
+ drc_num_bands += che_drc->band_incr;
+ for (i = 0; i < drc_num_bands; i++) {
+ che_drc->band_top[i] = get_bits(gb, 8);
+ n++;
+ }
+ }
+
+ /* prog_ref_level_present? */
+ if (get_bits1(gb)) {
+ che_drc->prog_ref_level = get_bits(gb, 7);
+ skip_bits1(gb); // prog_ref_level_reserved_bits
+ n++;
+ }
+
+ for (i = 0; i < drc_num_bands; i++) {
+ che_drc->dyn_rng_sgn[i] = get_bits1(gb);
+ che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
+ n++;
+ }
+
+ return n;
+}
+
+static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
+ uint8_t buf[256];
+ int i, major, minor;
+
+ if (len < 13+7*8)
+ goto unknown;
+
+ get_bits(gb, 13); len -= 13;
+
+ for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
+ buf[i] = get_bits(gb, 8);
+
+ buf[i] = 0;
+ if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
+ av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
+
+ if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
+ ac->avctx->internal->skip_samples = 1024;
+ }
+
+unknown:
+ skip_bits_long(gb, len);
+
+ return 0;
+}
+
+/**
+ * Decode extension data (incomplete); reference: table 4.51.
+ *
+ * @param cnt length of TYPE_FIL syntactic element in bytes
+ *
+ * @return Returns number of bytes consumed
+ */
+static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
+ ChannelElement *che, enum RawDataBlockType elem_type)
+{
+ int crc_flag = 0;
+ int res = cnt;
+ int type = get_bits(gb, 4);
+
+ if (ac->avctx->debug & FF_DEBUG_STARTCODE)
+ av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
+
+ switch (type) { // extension type
+ case EXT_SBR_DATA_CRC:
+ crc_flag++;
+ case EXT_SBR_DATA:
+ if (!che) {
+ av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
+ return res;
+ } else if (ac->oc[1].m4ac.frame_length_short) {
+ if (!ac->warned_960_sbr)
+ avpriv_report_missing_feature(ac->avctx,
+ "SBR with 960 frame length");
+ ac->warned_960_sbr = 1;
+ skip_bits_long(gb, 8 * cnt - 4);
+ return res;
+ } else if (!ac->oc[1].m4ac.sbr) {
+ av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
+ skip_bits_long(gb, 8 * cnt - 4);
+ return res;
+ } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
+ av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
+ skip_bits_long(gb, 8 * cnt - 4);
+ return res;
+ } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
+ ac->oc[1].m4ac.sbr = 1;
+ ac->oc[1].m4ac.ps = 1;
+ ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
+ output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
+ ac->oc[1].status, 1);
+ } else {
+ ac->oc[1].m4ac.sbr = 1;
+ ac->avctx->profile = FF_PROFILE_AAC_HE;
+ }
+ res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
+ break;
+ case EXT_DYNAMIC_RANGE:
+ res = decode_dynamic_range(&ac->che_drc, gb);
+ break;
+ case EXT_FILL:
+ decode_fill(ac, gb, 8 * cnt - 4);
+ break;
+ case EXT_FILL_DATA:
+ case EXT_DATA_ELEMENT:
+ default:
+ skip_bits_long(gb, 8 * cnt - 4);
+ break;
+ };
+ return res;
+}
+
+/**
+ * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
+ *
+ * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
+ * @param coef spectral coefficients
+ */
+static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns,
+ IndividualChannelStream *ics, int decode)
+{
+ const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
+ int w, filt, m, i;
+ int bottom, top, order, start, end, size, inc;
+ INTFLOAT lpc[TNS_MAX_ORDER];
+ INTFLOAT tmp[TNS_MAX_ORDER+1];
+ UINTFLOAT *coef = coef_param;
+
+ for (w = 0; w < ics->num_windows; w++) {
+ bottom = ics->num_swb;
+ for (filt = 0; filt < tns->n_filt[w]; filt++) {
+ top = bottom;
+ bottom = FFMAX(0, top - tns->length[w][filt]);
+ order = tns->order[w][filt];
+ if (order == 0)
+ continue;
+
+ // tns_decode_coef
+ AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
+
+ start = ics->swb_offset[FFMIN(bottom, mmm)];
+ end = ics->swb_offset[FFMIN( top, mmm)];
+ if ((size = end - start) <= 0)
+ continue;
+ if (tns->direction[w][filt]) {
+ inc = -1;
+ start = end - 1;
+ } else {
+ inc = 1;
+ }
+ start += w * 128;
+
+ if (decode) {
+ // ar filter
+ for (m = 0; m < size; m++, start += inc)
+ for (i = 1; i <= FFMIN(m, order); i++)
+ coef[start] -= AAC_MUL26((INTFLOAT)coef[start - i * inc], lpc[i - 1]);
+ } else {
+ // ma filter
+ for (m = 0; m < size; m++, start += inc) {
+ tmp[0] = coef[start];
+ for (i = 1; i <= FFMIN(m, order); i++)
+ coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
+ for (i = order; i > 0; i--)
+ tmp[i] = tmp[i - 1];
+ }
+ }
+ }
+ }
+}
+
+/**
+ * Apply windowing and MDCT to obtain the spectral
+ * coefficient from the predicted sample by LTP.
+ */
+static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
+ INTFLOAT *in, IndividualChannelStream *ics)
+{
+ const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
+ const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
+ const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
+ const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
+
+ if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
+ ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
+ } else {
+ memset(in, 0, 448 * sizeof(*in));
+ ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
+ }
+ if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
+ ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
+ } else {
+ ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
+ memset(in + 1024 + 576, 0, 448 * sizeof(*in));
+ }
+ ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
+}
+
+/**
+ * Apply the long term prediction
+ */
+static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
+{
+ const LongTermPrediction *ltp = &sce->ics.ltp;
+ const uint16_t *offsets = sce->ics.swb_offset;
+ int i, sfb;
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+ INTFLOAT *predTime = sce->ret;
+ INTFLOAT *predFreq = ac->buf_mdct;
+ int16_t num_samples = 2048;
+
+ if (ltp->lag < 1024)
+ num_samples = ltp->lag + 1024;
+ for (i = 0; i < num_samples; i++)
+ predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
+ memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
+
+ ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
+
+ if (sce->tns.present)
+ ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
+
+ for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
+ if (ltp->used[sfb])
+ for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
+ sce->coeffs[i] += predFreq[i];
+ }
+}
+
+/**
+ * Update the LTP buffer for next frame
+ */
+static void update_ltp(AACContext *ac, SingleChannelElement *sce)
+{
+ IndividualChannelStream *ics = &sce->ics;
+ INTFLOAT *saved = sce->saved;
+ INTFLOAT *saved_ltp = sce->coeffs;
+ const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
+ const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
+ int i;
+
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
+ memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
+ ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+
+ for (i = 0; i < 64; i++)
+ saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
+ } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
+ memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
+ memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
+ ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+
+ for (i = 0; i < 64; i++)
+ saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
+ } else { // LONG_STOP or ONLY_LONG
+ ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
+
+ for (i = 0; i < 512; i++)
+ saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
+ }
+
+ memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
+}
+
+/**
+ * Conduct IMDCT and windowing.
+ */
+static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
+{
+ IndividualChannelStream *ics = &sce->ics;
+ INTFLOAT *in = sce->coeffs;
+ INTFLOAT *out = sce->ret;
+ INTFLOAT *saved = sce->saved;
+ const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
+ const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
+ const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
+ INTFLOAT *buf = ac->buf_mdct;
+ INTFLOAT *temp = ac->temp;
+ int i;
+
+ // imdct
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ for (i = 0; i < 1024; i += 128)
+ ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
+ } else {
+ ac->mdct.imdct_half(&ac->mdct, buf, in);
+#if USE_FIXED
+ for (i=0; i<1024; i++)
+ buf[i] = (buf[i] + 4) >> 3;
+#endif /* USE_FIXED */
+ }
+
+ /* window overlapping
+ * NOTE: To simplify the overlapping code, all 'meaningless' short to long
+ * and long to short transitions are considered to be short to short
+ * transitions. This leaves just two cases (long to long and short to short)
+ * with a little special sauce for EIGHT_SHORT_SEQUENCE.
+ */
+ if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
+ (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
+ ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
+ } else {
+ memcpy( out, saved, 448 * sizeof(*out));
+
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
+ ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
+ memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
+ } else {
+ ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
+ memcpy( out + 576, buf + 64, 448 * sizeof(*out));
+ }
+ }
+
+ // buffer update
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ memcpy( saved, temp + 64, 64 * sizeof(*saved));
+ ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
+ ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
+ memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
+ } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
+ memcpy( saved, buf + 512, 448 * sizeof(*saved));
+ memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
+ } else { // LONG_STOP or ONLY_LONG
+ memcpy( saved, buf + 512, 512 * sizeof(*saved));
+ }
+}
+
+/**
+ * Conduct IMDCT and windowing.
+ */
+static void imdct_and_windowing_960(AACContext *ac, SingleChannelElement *sce)
+{
+#if !USE_FIXED
+ IndividualChannelStream *ics = &sce->ics;
+ INTFLOAT *in = sce->coeffs;
+ INTFLOAT *out = sce->ret;
+ INTFLOAT *saved = sce->saved;
+ const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
+ const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_960) : AAC_RENAME(ff_sine_960);
+ const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
+ INTFLOAT *buf = ac->buf_mdct;
+ INTFLOAT *temp = ac->temp;
+ int i;
+
+ // imdct
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ for (i = 0; i < 8; i++)
+ ac->mdct120->imdct_half(ac->mdct120, buf + i * 120, in + i * 128, 1);
+ } else {
+ ac->mdct960->imdct_half(ac->mdct960, buf, in, 1);
+ }
+
+ /* window overlapping
+ * NOTE: To simplify the overlapping code, all 'meaningless' short to long
+ * and long to short transitions are considered to be short to short
+ * transitions. This leaves just two cases (long to long and short to short)
+ * with a little special sauce for EIGHT_SHORT_SEQUENCE.
+ */
+
+ if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
+ (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
+ ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 480);
+ } else {
+ memcpy( out, saved, 420 * sizeof(*out));
+
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ ac->fdsp->vector_fmul_window(out + 420 + 0*120, saved + 420, buf + 0*120, swindow_prev, 60);
+ ac->fdsp->vector_fmul_window(out + 420 + 1*120, buf + 0*120 + 60, buf + 1*120, swindow, 60);
+ ac->fdsp->vector_fmul_window(out + 420 + 2*120, buf + 1*120 + 60, buf + 2*120, swindow, 60);
+ ac->fdsp->vector_fmul_window(out + 420 + 3*120, buf + 2*120 + 60, buf + 3*120, swindow, 60);
+ ac->fdsp->vector_fmul_window(temp, buf + 3*120 + 60, buf + 4*120, swindow, 60);
+ memcpy( out + 420 + 4*120, temp, 60 * sizeof(*out));
+ } else {
+ ac->fdsp->vector_fmul_window(out + 420, saved + 420, buf, swindow_prev, 60);
+ memcpy( out + 540, buf + 60, 420 * sizeof(*out));
+ }
+ }
+
+ // buffer update
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ memcpy( saved, temp + 60, 60 * sizeof(*saved));
+ ac->fdsp->vector_fmul_window(saved + 60, buf + 4*120 + 60, buf + 5*120, swindow, 60);
+ ac->fdsp->vector_fmul_window(saved + 180, buf + 5*120 + 60, buf + 6*120, swindow, 60);
+ ac->fdsp->vector_fmul_window(saved + 300, buf + 6*120 + 60, buf + 7*120, swindow, 60);
+ memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
+ } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
+ memcpy( saved, buf + 480, 420 * sizeof(*saved));
+ memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
+ } else { // LONG_STOP or ONLY_LONG
+ memcpy( saved, buf + 480, 480 * sizeof(*saved));
+ }
+#endif
+}
+static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
+{
+ IndividualChannelStream *ics = &sce->ics;
+ INTFLOAT *in = sce->coeffs;
+ INTFLOAT *out = sce->ret;
+ INTFLOAT *saved = sce->saved;
+ INTFLOAT *buf = ac->buf_mdct;
+#if USE_FIXED
+ int i;
+#endif /* USE_FIXED */
+
+ // imdct
+ ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+
+#if USE_FIXED
+ for (i = 0; i < 1024; i++)
+ buf[i] = (buf[i] + 2) >> 2;
+#endif /* USE_FIXED */
+
+ // window overlapping
+ if (ics->use_kb_window[1]) {
+ // AAC LD uses a low overlap sine window instead of a KBD window
+ memcpy(out, saved, 192 * sizeof(*out));
+ ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
+ memcpy( out + 320, buf + 64, 192 * sizeof(*out));
+ } else {
+ ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
+ }
+
+ // buffer update
+ memcpy(saved, buf + 256, 256 * sizeof(*saved));
+}
+
+static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
+{
+ INTFLOAT *in = sce->coeffs;
+ INTFLOAT *out = sce->ret;
+ INTFLOAT *saved = sce->saved;
+ INTFLOAT *buf = ac->buf_mdct;
+ int i;
+ const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
+ const int n2 = n >> 1;
+ const int n4 = n >> 2;
+ const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
+ AAC_RENAME(ff_aac_eld_window_512);
+
+ // Inverse transform, mapped to the conventional IMDCT by
+ // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
+ // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
+ // International Conference on Audio, Language and Image Processing, ICALIP 2008.
+ // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
+ for (i = 0; i < n2; i+=2) {
+ INTFLOAT temp;
+ temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
+ temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
+ }
+#if !USE_FIXED
+ if (n == 480)
+ ac->mdct480->imdct_half(ac->mdct480, buf, in, 1);
+ else
+#endif
+ ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+
+#if USE_FIXED
+ for (i = 0; i < 1024; i++)
+ buf[i] = (buf[i] + 1) >> 1;
+#endif /* USE_FIXED */
+
+ for (i = 0; i < n; i+=2) {
+ buf[i] = -buf[i];
+ }
+ // Like with the regular IMDCT at this point we still have the middle half
+ // of a transform but with even symmetry on the left and odd symmetry on
+ // the right
+
+ // window overlapping
+ // The spec says to use samples [0..511] but the reference decoder uses
+ // samples [128..639].
+ for (i = n4; i < n2; i ++) {
+ out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
+ AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
+ AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
+ AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
+ }
+ for (i = 0; i < n2; i ++) {
+ out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
+ AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
+ AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
+ AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
+ }
+ for (i = 0; i < n4; i ++) {
+ out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
+ AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
+ AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
+ }
+
+ // buffer update
+ memmove(saved + n, saved, 2 * n * sizeof(*saved));
+ memcpy( saved, buf, n * sizeof(*saved));
+}
+
+/**
+ * channel coupling transformation interface
+ *
+ * @param apply_coupling_method pointer to (in)dependent coupling function
+ */
+static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
+ enum RawDataBlockType type, int elem_id,
+ enum CouplingPoint coupling_point,
+ void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
+{
+ int i, c;
+
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *cce = ac->che[TYPE_CCE][i];
+ int index = 0;
+
+ if (cce && cce->coup.coupling_point == coupling_point) {
+ ChannelCoupling *coup = &cce->coup;
+
+ for (c = 0; c <= coup->num_coupled; c++) {
+ if (coup->type[c] == type && coup->id_select[c] == elem_id) {
+ if (coup->ch_select[c] != 1) {
+ apply_coupling_method(ac, &cc->ch[0], cce, index);
+ if (coup->ch_select[c] != 0)
+ index++;
+ }
+ if (coup->ch_select[c] != 2)
+ apply_coupling_method(ac, &cc->ch[1], cce, index++);
+ } else
+ index += 1 + (coup->ch_select[c] == 3);
+ }
+ }
+ }
+}
+
+/**
+ * Convert spectral data to samples, applying all supported tools as appropriate.
+ */
+static void spectral_to_sample(AACContext *ac, int samples)
+{
+ int i, type;
+ void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
+ switch (ac->oc[1].m4ac.object_type) {
+ case AOT_ER_AAC_LD:
+ imdct_and_window = imdct_and_windowing_ld;
+ break;
+ case AOT_ER_AAC_ELD:
+ imdct_and_window = imdct_and_windowing_eld;
+ break;
+ default:
+ if (ac->oc[1].m4ac.frame_length_short)
+ imdct_and_window = imdct_and_windowing_960;
+ else
+ imdct_and_window = ac->imdct_and_windowing;
+ }
+ for (type = 3; type >= 0; type--) {
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *che = ac->che[type][i];
+ if (che && che->present) {
+ if (type <= TYPE_CPE)
+ apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(apply_dependent_coupling));
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
+ if (che->ch[0].ics.predictor_present) {
+ if (che->ch[0].ics.ltp.present)
+ ac->apply_ltp(ac, &che->ch[0]);
+ if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
+ ac->apply_ltp(ac, &che->ch[1]);
+ }
+ }
+ if (che->ch[0].tns.present)
+ ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
+ if (che->ch[1].tns.present)
+ ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
+ if (type <= TYPE_CPE)
+ apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, AAC_RENAME(apply_dependent_coupling));
+ if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
+ imdct_and_window(ac, &che->ch[0]);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
+ ac->update_ltp(ac, &che->ch[0]);
+ if (type == TYPE_CPE) {
+ imdct_and_window(ac, &che->ch[1]);
+ if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
+ ac->update_ltp(ac, &che->ch[1]);
+ }
+ if (ac->oc[1].m4ac.sbr > 0) {
+ AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
+ }
+ }
+ if (type <= TYPE_CCE)
+ apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));
+
+#if USE_FIXED
+ {
+ int j;
+ /* preparation for resampler */
+ for(j = 0; j<samples; j++){
+ che->ch[0].ret[j] = (int32_t)av_clip64((int64_t)che->ch[0].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
+ if(type == TYPE_CPE)
+ che->ch[1].ret[j] = (int32_t)av_clip64((int64_t)che->ch[1].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
+ }
+ }
+#endif /* USE_FIXED */
+ che->present = 0;
+ } else if (che) {
+ av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
+ }
+ }
+ }
+}
+
+static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
+{
+ int size;
+ AACADTSHeaderInfo hdr_info;
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int layout_map_tags, ret;
+
- size = avpriv_aac_parse_header(gb, &hdr_info);
++ size = ff_adts_header_parse(gb, &hdr_info);
+ if (size > 0) {
+ if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
+ // This is 2 for "VLB " audio in NSV files.
+ // See samples/nsv/vlb_audio.
+ avpriv_report_missing_feature(ac->avctx,
+ "More than one AAC RDB per ADTS frame");
+ ac->warned_num_aac_frames = 1;
+ }
+ push_output_configuration(ac);
+ if (hdr_info.chan_config) {
+ ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
+ if ((ret = set_default_channel_config(ac->avctx,
+ layout_map,
+ &layout_map_tags,
+ hdr_info.chan_config)) < 0)
+ return ret;
+ if ((ret = output_configure(ac, layout_map, layout_map_tags,
+ FFMAX(ac->oc[1].status,
+ OC_TRIAL_FRAME), 0)) < 0)
+ return ret;
+ } else {
+ ac->oc[1].m4ac.chan_config = 0;
+ /**
+ * dual mono frames in Japanese DTV can have chan_config 0
+ * WITHOUT specifying PCE.
+ * thus, set dual mono as default.
+ */
+ if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
+ layout_map_tags = 2;
+ layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
+ layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
+ layout_map[0][1] = 0;
+ layout_map[1][1] = 1;
+ if (output_configure(ac, layout_map, layout_map_tags,
+ OC_TRIAL_FRAME, 0))
+ return -7;
+ }
+ }
+ ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
+ ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
+ ac->oc[1].m4ac.object_type = hdr_info.object_type;
+ ac->oc[1].m4ac.frame_length_short = 0;
+ if (ac->oc[0].status != OC_LOCKED ||
+ ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
+ ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
+ ac->oc[1].m4ac.sbr = -1;
+ ac->oc[1].m4ac.ps = -1;
+ }
+ if (!hdr_info.crc_absent)
+ skip_bits(gb, 16);
+ }
+ return size;
+}
+
+static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, GetBitContext *gb)
+{
+ AACContext *ac = avctx->priv_data;
+ const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
+ ChannelElement *che;
+ int err, i;
+ int samples = m4ac->frame_length_short ? 960 : 1024;
+ int chan_config = m4ac->chan_config;
+ int aot = m4ac->object_type;
+
+ if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
+ samples >>= 1;
+
+ ac->frame = data;
+
+ if ((err = frame_configure_elements(avctx)) < 0)
+ return err;
+
+ // The FF_PROFILE_AAC_* defines are all object_type - 1
+ // This may lead to an undefined profile being signaled
+ ac->avctx->profile = aot - 1;
+
+ ac->tags_mapped = 0;
+
+ if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
+ avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
+ chan_config);
+ return AVERROR_INVALIDDATA;
+ }
+ for (i = 0; i < tags_per_config[chan_config]; i++) {
+ const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
+ const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
+ if (!(che=get_che(ac, elem_type, elem_id))) {
+ av_log(ac->avctx, AV_LOG_ERROR,
+ "channel element %d.%d is not allocated\n",
+ elem_type, elem_id);
+ return AVERROR_INVALIDDATA;
+ }
+ che->present = 1;
+ if (aot != AOT_ER_AAC_ELD)
+ skip_bits(gb, 4);
+ switch (elem_type) {
+ case TYPE_SCE:
+ err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ break;
+ case TYPE_CPE:
+ err = decode_cpe(ac, gb, che);
+ break;
+ case TYPE_LFE:
+ err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ break;
+ }
+ if (err < 0)
+ return err;
+ }
+
+ spectral_to_sample(ac, samples);
+
+ if (!ac->frame->data[0] && samples) {
+ av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ ac->frame->nb_samples = samples;
+ ac->frame->sample_rate = avctx->sample_rate;
+ *got_frame_ptr = 1;
+
+ skip_bits_long(gb, get_bits_left(gb));
+ return 0;
+}
+
+static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
+{
+ AACContext *ac = avctx->priv_data;
+ ChannelElement *che = NULL, *che_prev = NULL;
+ enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
+ int err, elem_id;
+ int samples = 0, multiplier, audio_found = 0, pce_found = 0;
+ int is_dmono, sce_count = 0;
+ int payload_alignment;
+
+ ac->frame = data;
+
+ if (show_bits(gb, 12) == 0xfff) {
+ if ((err = parse_adts_frame_header(ac, gb)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+ goto fail;
+ }
+ if (ac->oc[1].m4ac.sampling_index > 12) {
+ av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ }
+
+ if ((err = frame_configure_elements(avctx)) < 0)
+ goto fail;
+
+ // The FF_PROFILE_AAC_* defines are all object_type - 1
+ // This may lead to an undefined profile being signaled
+ ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
+
+ payload_alignment = get_bits_count(gb);
+ ac->tags_mapped = 0;
+ // parse
+ while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
+ elem_id = get_bits(gb, 4);
+
+ if (avctx->debug & FF_DEBUG_STARTCODE)
+ av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
+
+ if (!avctx->channels && elem_type != TYPE_PCE) {
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+
+ if (elem_type < TYPE_DSE) {
+ if (!(che=get_che(ac, elem_type, elem_id))) {
+ av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
+ elem_type, elem_id);
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;
+ che->present = 1;
+ }
+
+ switch (elem_type) {
+
+ case TYPE_SCE:
+ err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
+ sce_count++;
+ break;
+
+ case TYPE_CPE:
+ err = decode_cpe(ac, gb, che);
+ audio_found = 1;
+ break;
+
+ case TYPE_CCE:
+ err = decode_cce(ac, gb, che);
+ break;
+
+ case TYPE_LFE:
+ err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
+ break;
+
+ case TYPE_DSE:
+ err = skip_data_stream_element(ac, gb);
+ break;
+
+ case TYPE_PCE: {
+ uint8_t layout_map[MAX_ELEM_ID*4][3];
+ int tags;
+
+ int pushed = push_output_configuration(ac);
+ if (pce_found && !pushed) {
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+
+ tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
+ payload_alignment);
+ if (tags < 0) {
+ err = tags;
+ break;
+ }
+ if (pce_found) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Not evaluating a further program_config_element as this construct is dubious at best.\n");
+ pop_output_configuration(ac);
+ } else {
+ err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
+ if (!err)
+ ac->oc[1].m4ac.chan_config = 0;
+ pce_found = 1;
+ }
+ break;
+ }
+
+ case TYPE_FIL:
+ if (elem_id == 15)
+ elem_id += get_bits(gb, 8) - 1;
+ if (get_bits_left(gb) < 8 * elem_id) {
+ av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ while (elem_id > 0)
+ elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
+ err = 0; /* FIXME */
+ break;
+
+ default:
+ err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
+ break;
+ }
+
+ if (elem_type < TYPE_DSE) {
+ che_prev = che;
+ che_prev_type = elem_type;
+ }
+
+ if (err)
+ goto fail;
+
+ if (get_bits_left(gb) < 3) {
+ av_log(avctx, AV_LOG_ERROR, overread_err);
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+ }
+
+ if (!avctx->channels) {
+ *got_frame_ptr = 0;
+ return 0;
+ }
+
+ multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
+ samples <<= multiplier;
+
+ spectral_to_sample(ac, samples);
+
+ if (ac->oc[1].status && audio_found) {
+ avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
+ avctx->frame_size = samples;
+ ac->oc[1].status = OC_LOCKED;
+ }
+
+ if (multiplier)
+ avctx->internal->skip_samples_multiplier = 2;
+
+ if (!ac->frame->data[0] && samples) {
+ av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
+ err = AVERROR_INVALIDDATA;
+ goto fail;
+ }
+
+ if (samples) {
+ ac->frame->nb_samples = samples;
+ ac->frame->sample_rate = avctx->sample_rate;
+ } else
+ av_frame_unref(ac->frame);
+ *got_frame_ptr = !!samples;
+
+ /* for dual-mono audio (SCE + SCE) */
+ is_dmono = ac->dmono_mode && sce_count == 2 &&
+ ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
+ if (is_dmono) {
+ if (ac->dmono_mode == 1)
+ ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
+ else if (ac->dmono_mode == 2)
+ ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
+ }
+
+ return 0;
+fail:
+ pop_output_configuration(ac);
+ return err;
+}
+
+static int aac_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ AACContext *ac = avctx->priv_data;
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ GetBitContext gb;
+ int buf_consumed;
+ int buf_offset;
+ int err;
+ int new_extradata_size;
+ const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_NEW_EXTRADATA,
+ &new_extradata_size);
+ int jp_dualmono_size;
+ const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_JP_DUALMONO,
+ &jp_dualmono_size);
+
+ if (new_extradata && 0) {
+ av_free(avctx->extradata);
+ avctx->extradata = av_mallocz(new_extradata_size +
+ AV_INPUT_BUFFER_PADDING_SIZE);
+ if (!avctx->extradata)
+ return AVERROR(ENOMEM);
+ avctx->extradata_size = new_extradata_size;
+ memcpy(avctx->extradata, new_extradata, new_extradata_size);
+ push_output_configuration(ac);
+ if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+ avctx->extradata,
+ avctx->extradata_size*8LL, 1) < 0) {
+ pop_output_configuration(ac);
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ ac->dmono_mode = 0;
+ if (jp_dualmono && jp_dualmono_size > 0)
+ ac->dmono_mode = 1 + *jp_dualmono;
+ if (ac->force_dmono_mode >= 0)
+ ac->dmono_mode = ac->force_dmono_mode;
+
+ if (INT_MAX / 8 <= buf_size)
+ return AVERROR_INVALIDDATA;
+
+ if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
+ return err;
+
+ switch (ac->oc[1].m4ac.object_type) {
+ case AOT_ER_AAC_LC:
+ case AOT_ER_AAC_LTP:
+ case AOT_ER_AAC_LD:
+ case AOT_ER_AAC_ELD:
+ err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
+ break;
+ default:
+ err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
+ }
+ if (err < 0)
+ return err;
+
+ buf_consumed = (get_bits_count(&gb) + 7) >> 3;
+ for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
+ if (buf[buf_offset])
+ break;
+
+ return buf_size > buf_offset ? buf_consumed : buf_size;
+}
+
+static av_cold int aac_decode_close(AVCodecContext *avctx)
+{
+ AACContext *ac = avctx->priv_data;
+ int i, type;
+
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ for (type = 0; type < 4; type++) {
+ if (ac->che[type][i])
+ AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
+ av_freep(&ac->che[type][i]);
+ }
+ }
+
+ ff_mdct_end(&ac->mdct);
+ ff_mdct_end(&ac->mdct_small);
+ ff_mdct_end(&ac->mdct_ld);
+ ff_mdct_end(&ac->mdct_ltp);
+#if !USE_FIXED
+ ff_mdct15_uninit(&ac->mdct120);
+ ff_mdct15_uninit(&ac->mdct480);
+ ff_mdct15_uninit(&ac->mdct960);
+#endif
+ av_freep(&ac->fdsp);
+ return 0;
+}
+
+static void aacdec_init(AACContext *c)
+{
+ c->imdct_and_windowing = imdct_and_windowing;
+ c->apply_ltp = apply_ltp;
+ c->apply_tns = apply_tns;
+ c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
+ c->update_ltp = update_ltp;
+#if USE_FIXED
+ c->vector_pow43 = vector_pow43;
+ c->subband_scale = subband_scale;
+#endif
+
+#if !USE_FIXED
+ if(ARCH_MIPS)
+ ff_aacdec_init_mips(c);
+#endif /* !USE_FIXED */
+}
+/**
+ * AVOptions for Japanese DTV specific extensions (ADTS only)
+ */
+#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption options[] = {
+ {"dual_mono_mode", "Select the channel to decode for dual mono",
+ offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
+ AACDEC_FLAGS, "dual_mono_mode"},
+
+ {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+ {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
+
+ {NULL},
+};
+
+static const AVClass aac_decoder_class = {
+ .class_name = "AAC decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
diff --cc libavcodec/adts_parser.c
index 0000000000,5821e6fb79..5c9f8ff6f2
mode 000000,100644..100644
--- a/libavcodec/adts_parser.c
+++ b/libavcodec/adts_parser.c
@@@ -1,0 -1,44 +1,44 @@@
+ /*
- * 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
+ */
+
+ #include "config.h"
+
+ #include <stddef.h>
+ #include <stdint.h>
+
+ #include "adts_header.h"
+ #include "adts_parser.h"
+
+ int av_adts_header_parse(const uint8_t *buf, uint32_t *samples, uint8_t *frames)
+ {
+ #if CONFIG_ADTS_HEADER
+ GetBitContext gb;
+ AACADTSHeaderInfo hdr;
+ int err = init_get_bits8(&gb, buf, AV_AAC_ADTS_HEADER_SIZE);
+ if (err < 0)
+ return err;
+ err = ff_adts_header_parse(&gb, &hdr);
+ if (err < 0)
+ return err;
+ *samples = hdr.samples;
+ *frames = hdr.num_aac_frames;
+ return 0;
+ #else
+ return AVERROR(ENOSYS);
+ #endif
+ }
diff --cc libavcodec/adts_parser.h
index 0000000000,1a3328f10e..f85becd131
mode 000000,100644..100644
--- a/libavcodec/adts_parser.h
+++ b/libavcodec/adts_parser.h
@@@ -1,0 -1,37 +1,37 @@@
+ /*
- * 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
+ */
+
+ #ifndef AVCODEC_ADTS_PARSER_H
+ #define AVCODEC_ADTS_PARSER_H
+
+ #include <stddef.h>
+ #include <stdint.h>
+
+ #define AV_AAC_ADTS_HEADER_SIZE 7
+
+ /**
+ * Extract the number of samples and frames from AAC data.
+ * @param[in] buf pointer to AAC data buffer
+ * @param[out] samples Pointer to where number of samples is written
+ * @param[out] frames Pointer to where number of frames is written
+ * @return Returns 0 on success, error code on failure.
+ */
+ int av_adts_header_parse(const uint8_t *buf, uint32_t *samples,
+ uint8_t *frames);
+
+ #endif /* AVCODEC_ADTS_PARSER_H */
diff --cc libavformat/spdifdec.c
index f7288376f6,38692c27e6..21bfce4226
--- a/libavformat/spdifdec.c
+++ b/libavformat/spdifdec.c
@@@ -132,7 -131,7 +136,7 @@@ int ff_spdif_probe(const uint8_t *p_buf
} else
consecutive_codes = 0;
- if (buf + 4 + AAC_ADTS_HEADER_SIZE > p_buf + buf_size)
- if (buf + 4 + AV_AAC_ADTS_HEADER_SIZE > p->buf + p->buf_size)
++ if (buf + 4 + AV_AAC_ADTS_HEADER_SIZE > p_buf + buf_size)
break;
/* continue probing to find more sync codes */
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