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-rw-r--r--faad2/src/plugins/mpeg4ip/Makefile.am22
-rw-r--r--faad2/src/plugins/mpeg4ip/README_WIN32.txt11
-rw-r--r--faad2/src/plugins/mpeg4ip/aa_file.cpp130
-rw-r--r--faad2/src/plugins/mpeg4ip/aa_file.h26
-rw-r--r--faad2/src/plugins/mpeg4ip/faad2.cpp388
-rw-r--r--faad2/src/plugins/mpeg4ip/faad2.h92
6 files changed, 669 insertions, 0 deletions
diff --git a/faad2/src/plugins/mpeg4ip/Makefile.am b/faad2/src/plugins/mpeg4ip/Makefile.am
new file mode 100644
index 0000000..8024d2a
--- /dev/null
+++ b/faad2/src/plugins/mpeg4ip/Makefile.am
@@ -0,0 +1,22 @@
+libdir = @MPEG4IP_PLAYER_PLUGIN_DIR@
+
+lib_LTLIBRARIES = faad2_plugin.la
+faad2_plugin_la_LDFLAGS = -module
+faad2_plugin_la_SOURCES = \
+ faad2.cpp \
+ faad2.h \
+ aa_file.cpp \
+ aa_file.h
+
+faad2_plugin_la_LIBADD = \
+ $(top_builddir)/libfaad/libfaad.la \
+ -lm
+
+
+INCLUDES = -I$(top_srcdir)/include
+
+AM_CFLAGS = -D_REENTRANT -fexceptions
+
+AM_CXXFLAGS = -D_REENTRANT -DNOCONTROLS -fexceptions
+
+
diff --git a/faad2/src/plugins/mpeg4ip/README_WIN32.txt b/faad2/src/plugins/mpeg4ip/README_WIN32.txt
new file mode 100644
index 0000000..7fc864a
--- /dev/null
+++ b/faad2/src/plugins/mpeg4ip/README_WIN32.txt
@@ -0,0 +1,11 @@
+Creating mpeg4ip plugin for Windows.
+
+You will need to have mpeg4ip installed. If you install it on the same drive, with the top level directory
+name of /mpeg4ip, you will have to do nothing other than move the faad2_plugin.dll from the Release or
+Debug directory to the same directory as the other mpeg4ip plugins.
+
+If you install it somewhere else, you will have to change the include paths and link paths in the project
+settings for faad2_plugin to the proper directory. Look for /mpeg4ip (or \mpeg4ip) and change all occurances
+of these in the file.
+
+It might be best to hand-edit the faad_plugin.sdp file with wordpad and use the search and replace function. \ No newline at end of file
diff --git a/faad2/src/plugins/mpeg4ip/aa_file.cpp b/faad2/src/plugins/mpeg4ip/aa_file.cpp
new file mode 100644
index 0000000..a0cccb9
--- /dev/null
+++ b/faad2/src/plugins/mpeg4ip/aa_file.cpp
@@ -0,0 +1,130 @@
+/*
+** MPEG4IP plugin for FAAD2
+** Copyright (C) 2003 Bill May wmay@cisco.com
+**
+** This program is free software; you can redistribute it and/or modify
+** it under the terms of the GNU General Public License as published by
+** the Free Software Foundation; either version 2 of the License, or
+** (at your option) any later version.
+**
+** This program 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 General Public License for more details.
+**
+** You should have received a copy of the GNU General Public License
+** along with this program; if not, write to the Free Software
+** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+**
+** $Id: aa_file.cpp,v 1.2 2004/01/05 14:05:12 menno Exp $
+**/
+/*
+ * aa_file.cpp - create media structure for aac files
+ */
+
+#include "faad2.h"
+codec_data_t *aac_file_check (lib_message_func_t message,
+ const char *name,
+ double *max,
+ char *desc[4]
+#ifdef HAVE_PLUGIN_VERSION_0_8
+ , CConfigSet *pConfig
+#endif
+)
+{
+ aac_codec_t *aac;
+ int len = strlen(name);
+ if (strcasecmp(name + len - 4, ".aac") != 0) {
+ return (NULL);
+ }
+
+ aac = MALLOC_STRUCTURE(aac_codec_t);
+ memset(aac, 0, sizeof(*aac));
+ *max = 0;
+
+ aac->m_buffer = (uint8_t *)malloc(MAX_READ_BUFFER);
+ aac->m_buffer_size_max = MAX_READ_BUFFER;
+ aac->m_ifile = fopen(name, FOPEN_READ_BINARY);
+ if (aac->m_ifile == NULL) {
+ free(aac);
+ return NULL;
+ }
+ aac->m_output_frame_size = 1024;
+ aac->m_info = faacDecOpen(); // use defaults here...
+ aac->m_buffer_size = fread(aac->m_buffer,
+ 1,
+ aac->m_buffer_size_max,
+ aac->m_ifile);
+
+ unsigned long freq;
+ unsigned char chans;
+
+ faacDecInit(aac->m_info, (unsigned char *)aac->m_buffer,
+ aac->m_buffer_size, &freq, &chans);
+ // may want to actually decode the first frame...
+ if (freq == 0) {
+ message(LOG_ERR, aaclib, "Couldn't determine AAC frame rate");
+ aac_close((codec_data_t *)aac);
+ return (NULL);
+ }
+ aac->m_freq = freq;
+ aac->m_chans = chans;
+ aac->m_faad_inited = 1;
+ aac->m_framecount = 0;
+ return ((codec_data_t *)aac);
+}
+
+
+int aac_file_next_frame (codec_data_t *your,
+ uint8_t **buffer,
+ uint64_t *ts)
+{
+ aac_codec_t *aac = (aac_codec_t *)your;
+
+ if (aac->m_buffer_on > 0) {
+ memmove(aac->m_buffer,
+ &aac->m_buffer[aac->m_buffer_on],
+ aac->m_buffer_size - aac->m_buffer_on);
+ }
+ aac->m_buffer_size -= aac->m_buffer_on;
+ aac->m_buffer_size += fread(aac->m_buffer + aac->m_buffer_size,
+ 1,
+ aac->m_buffer_size_max - aac->m_buffer_size,
+ aac->m_ifile);
+ aac->m_buffer_on = 0;
+ if (aac->m_buffer_size == 0) return 0;
+
+
+ uint64_t calc;
+ calc = aac->m_framecount * 1024 * M_LLU;
+ calc /= aac->m_freq;
+ *ts = calc;
+ *buffer = aac->m_buffer;
+ aac->m_framecount++;
+ return (aac->m_buffer_size);
+}
+
+void aac_file_used_for_frame (codec_data_t *ifptr,
+ uint32_t bytes)
+{
+ aac_codec_t *aac = (aac_codec_t *)ifptr;
+ aac->m_buffer_on += bytes;
+ if (aac->m_buffer_on > aac->m_buffer_size) aac->m_buffer_on = aac->m_buffer_size;
+}
+
+int aac_file_eof (codec_data_t *ifptr)
+{
+ aac_codec_t *aac = (aac_codec_t *)ifptr;
+ return aac->m_buffer_on == aac->m_buffer_size && feof(aac->m_ifile);
+}
+
+int aac_raw_file_seek_to (codec_data_t *ifptr, uint64_t ts)
+{
+ if (ts != 0) return -1;
+
+ aac_codec_t *aac = (aac_codec_t *)ifptr;
+ rewind(aac->m_ifile);
+ aac->m_buffer_size = aac->m_buffer_on = 0;
+ aac->m_framecount = 0;
+ return 0;
+}
diff --git a/faad2/src/plugins/mpeg4ip/aa_file.h b/faad2/src/plugins/mpeg4ip/aa_file.h
new file mode 100644
index 0000000..dfed15d
--- /dev/null
+++ b/faad2/src/plugins/mpeg4ip/aa_file.h
@@ -0,0 +1,26 @@
+/*
+** MPEG4IP plugin for FAAD2
+** Copyright (C) 2003 Bill May wmay@cisco.com
+**
+** This program is free software; you can redistribute it and/or modify
+** it under the terms of the GNU General Public License as published by
+** the Free Software Foundation; either version 2 of the License, or
+** (at your option) any later version.
+**
+** This program 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 General Public License for more details.
+**
+** You should have received a copy of the GNU General Public License
+** along with this program; if not, write to the Free Software
+** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+**
+** $Id: aa_file.h,v 1.1 2003/08/07 17:21:21 menno Exp $
+**/
+/*
+ * aa_file.h - prototypes for aac files
+ */
+int create_media_for_aac_file (CPlayerSession *pspstr,
+ const char *name,
+ const char **errmsg);
diff --git a/faad2/src/plugins/mpeg4ip/faad2.cpp b/faad2/src/plugins/mpeg4ip/faad2.cpp
new file mode 100644
index 0000000..72ff0c8
--- /dev/null
+++ b/faad2/src/plugins/mpeg4ip/faad2.cpp
@@ -0,0 +1,388 @@
+/*
+** MPEG4IP plugin for FAAD2
+** Copyright (C) 2003 Bill May wmay@cisco.com
+**
+** This program is free software; you can redistribute it and/or modify
+** it under the terms of the GNU General Public License as published by
+** the Free Software Foundation; either version 2 of the License, or
+** (at your option) any later version.
+**
+** This program 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 General Public License for more details.
+**
+** You should have received a copy of the GNU General Public License
+** along with this program; if not, write to the Free Software
+** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+**
+** $Id: faad2.cpp,v 1.3 2004/08/20 08:30:53 menno Exp $
+**/
+#include "faad2.h"
+#include <mpeg4_audio_config.h>
+#include <mpeg4_sdp.h>
+#include <mp4.h>
+#include <SDL/SDL.h>
+
+#define DEBUG_SYNC 2
+
+#ifndef M_LLU
+#define M_LLU M_64
+#define LLU U64
+#endif
+const char *aaclib="faad2";
+
+/*
+ * Create CAACodec class
+ */
+static codec_data_t *aac_codec_create (const char *compressor,
+ int type,
+ int profile,
+ format_list_t *media_fmt,
+ audio_info_t *audio,
+ const uint8_t *userdata,
+ uint32_t userdata_size,
+ audio_vft_t *vft,
+ void *ifptr)
+
+{
+ aac_codec_t *aac;
+
+ aac = (aac_codec_t *)malloc(sizeof(aac_codec_t));
+ memset(aac, 0, sizeof(aac_codec_t));
+
+ aac->m_vft = vft;
+ aac->m_ifptr = ifptr;
+ fmtp_parse_t *fmtp = NULL;
+ // Start setting up FAAC stuff...
+
+ aac->m_resync_with_header = 1;
+ aac->m_record_sync_time = 1;
+
+ aac->m_audio_inited = 0;
+
+ // Use media_fmt to indicate that we're streaming.
+ if (media_fmt != NULL) {
+ // haven't checked for null buffer
+ // This is not necessarilly right - it is, for the most part, but
+ // we should be reading the fmtp statement, and looking at the config.
+ // (like we do below in the userdata section...
+ aac->m_freq = media_fmt->rtpmap->clock_rate;
+ fmtp = parse_fmtp_for_mpeg4(media_fmt->fmt_param, vft->log_msg);
+ if (fmtp != NULL) {
+ userdata = fmtp->config_binary;
+ userdata_size = fmtp->config_binary_len;
+ }
+ }
+
+ aac->m_info = faacDecOpen();
+ unsigned long srate;
+ unsigned char chan;
+ if ((userdata == NULL && fmtp == NULL) ||
+ (faacDecInit2(aac->m_info,
+ (uint8_t *)userdata,
+ userdata_size,
+ &srate,
+ &chan) < 0)) {
+ if (fmtp != NULL) free_fmtp_parse(fmtp);
+ return NULL;
+ }
+
+ mp4AudioSpecificConfig mp4ASC;
+ aac->m_output_frame_size = 1024;
+ if (AudioSpecificConfig((unsigned char *)userdata,
+ userdata_size,
+ &mp4ASC)) {
+ if (mp4ASC.frameLengthFlag) {
+ aac->m_output_frame_size = 960;
+ }
+ }
+ aac->m_freq = srate;
+ aac->m_chans = chan;
+ aac->m_faad_inited = 1;
+ aac->m_msec_per_frame = aac->m_output_frame_size;
+ aac->m_msec_per_frame *= M_LLU;
+ aac->m_msec_per_frame /= aac->m_freq;
+
+ // faad_init_bytestream(&m_info->ld, c_read_byte, c_bookmark, m_bytestream);
+
+ aa_message(LOG_INFO, aaclib, "Setting freq to %d", aac->m_freq);
+#if DUMP_OUTPUT_TO_FILE
+ aac->m_outfile = fopen("temp.raw", "w");
+#endif
+ if (fmtp != NULL) {
+ free_fmtp_parse(fmtp);
+ }
+ return (codec_data_t *)aac;
+}
+
+void aac_close (codec_data_t *ptr)
+{
+ if (ptr == NULL) {
+ return;
+ }
+ aac_codec_t *aac = (aac_codec_t *)ptr;
+ faacDecClose(aac->m_info);
+ aac->m_info = NULL;
+
+#if DUMP_OUTPUT_TO_FILE
+ fclose(aac->m_outfile);
+#endif
+ free(aac);
+}
+
+/*
+ * Handle pause - basically re-init the codec
+ */
+static void aac_do_pause (codec_data_t *ifptr)
+{
+ aac_codec_t *aac = (aac_codec_t *)ifptr;
+ aac->m_resync_with_header = 1;
+ aac->m_record_sync_time = 1;
+ aac->m_audio_inited = 0;
+ aac->m_ignore_first_sample = 0;
+ faacDecPostSeekReset(aac->m_info, 0);
+}
+
+/*
+ * Decode task call for FAAC
+ */
+static int aac_decode (codec_data_t *ptr,
+ uint64_t ts,
+ int from_rtp,
+ int *sync_frame,
+ uint8_t *buffer,
+ uint32_t buflen,
+ void *userdata)
+{
+ aac_codec_t *aac = (aac_codec_t *)ptr;
+ unsigned long bytes_consummed;
+ int bits = -1;
+ // struct timezone tz;
+
+ if (aac->m_record_sync_time) {
+ aac->m_current_frame = 0;
+ aac->m_record_sync_time = 0;
+ aac->m_current_time = ts;
+ aac->m_last_rtp_ts = ts;
+ } else {
+ if (aac->m_last_rtp_ts == ts) {
+ aac->m_current_time += aac->m_msec_per_frame;
+ aac->m_current_frame++;
+ } else {
+ aac->m_last_rtp_ts = ts;
+ aac->m_current_time = ts;
+ aac->m_current_frame = 0;
+ }
+
+ // Note - here m_current_time should pretty much always be >= rtpts.
+ // If we're not, we most likely want to stop and resync. We don't
+ // need to keep decoding - just decode this frame and indicate we
+ // need a resync... That should handle fast forwards... We need
+ // someway to handle reverses - perhaps if we're more than .5 seconds
+ // later...
+ }
+
+ if (aac->m_faad_inited == 0) {
+ /*
+ * If not initialized, do so.
+ */
+ abort();
+ unsigned long freq;
+ unsigned char chans;
+
+ faacDecInit(aac->m_info,
+ (unsigned char *)buffer,
+ buflen,
+ &freq,
+ &chans);
+ aac->m_freq = freq;
+ aac->m_chans = chans;
+ aac->m_faad_inited = 1;
+ }
+
+ uint8_t *buff;
+ unsigned long samples;
+ bytes_consummed = buflen;
+ //aa_message(LOG_DEBUG, aaclib, "decoding %d bits", buflen * 8);
+ faacDecFrameInfo frame_info;
+ buff = (uint8_t *)faacDecDecode(aac->m_info,
+ &frame_info,
+ buffer,
+ buflen);
+ if (buff != NULL) {
+ bytes_consummed = frame_info.bytesconsumed;
+#if 0
+ aa_message(LOG_DEBUG, aaclib, LLU" bytes %d samples %d",
+ ts, bytes_consummed, frame_info.samples);
+#endif
+ if (aac->m_audio_inited != 0) {
+ int tempchans = frame_info.channels;
+ if (tempchans != aac->m_chans) {
+ aa_message(LOG_NOTICE, aaclib, "chupdate - chans from data is %d",
+ tempchans);
+ }
+ } else {
+ int tempchans = frame_info.channels;
+
+ if (tempchans == 0) {
+ aa_message(LOG_ERR, aaclib, "initializing aac, returned channels are 0");
+ aac->m_resync_with_header = 1;
+ aac->m_record_sync_time = 1;
+ return bytes_consummed;
+ }
+ aac->m_chans = tempchans;
+ aac->m_freq = frame_info.samplerate;
+
+ aac->m_vft->audio_configure(aac->m_ifptr,
+ aac->m_freq,
+ aac->m_chans,
+ (audio_format_t)AUDIO_S16SYS,
+ aac->m_output_frame_size);
+ uint8_t *now = aac->m_vft->audio_get_buffer(aac->m_ifptr);
+ aac->m_audio_inited = 1;
+ }
+ /*
+ * good result - give it to audio sync class
+ */
+#if DUMP_OUTPUT_TO_FILE
+ fwrite(buff, aac->m_output_frame_size * 4, 1, aac->m_outfile);
+#endif
+ if (frame_info.samples != 0) {
+ aac->m_vft->audio_load_buffer(aac->m_ifptr,
+ buff,
+ frame_info.samples * 2,
+ aac->m_last_ts,
+ aac->m_resync_with_header);
+ if (aac->m_resync_with_header == 1) {
+ aac->m_resync_with_header = 0;
+#ifdef DEBUG_SYNC
+ aa_message(LOG_DEBUG, aaclib, "Back to good at "LLU, aac->m_current_time);
+#endif
+ }
+ }
+ } else {
+ aa_message(LOG_ERR, aaclib, "error return is %d", frame_info.error);
+ aac->m_resync_with_header = 1;
+#ifdef DEBUG_SYNC
+ aa_message(LOG_ERR, aaclib, "Audio decode problem - at "LLU,
+ aac->m_current_time);
+#endif
+ }
+ aac->m_last_ts = aac->m_current_time;
+ return (bytes_consummed);
+}
+
+static const char *aac_compressors[] = {
+ "aac ",
+ "mp4a",
+ "enca",
+ NULL
+};
+
+static int aac_codec_check (lib_message_func_t message,
+ const char *compressor,
+ int type,
+ int profile,
+ format_list_t *fptr,
+ const uint8_t *userdata,
+ uint32_t userdata_size
+#ifdef HAVE_PLUGIN_VERSION_0_8
+ ,CConfigSet *pConfig
+#endif
+ )
+{
+ fmtp_parse_t *fmtp = NULL;
+ if (compressor != NULL &&
+ strcasecmp(compressor, "MP4 FILE") == 0 &&
+ type != -1) {
+ switch (type) {
+ case MP4_MPEG2_AAC_MAIN_AUDIO_TYPE:
+ case MP4_MPEG2_AAC_LC_AUDIO_TYPE:
+ case MP4_MPEG2_AAC_SSR_AUDIO_TYPE:
+ case MP4_MPEG4_AUDIO_TYPE:
+ break;
+ default:
+ return -1;
+ }
+ }
+ if (fptr != NULL &&
+ fptr->rtpmap != NULL &&
+ fptr->rtpmap->encode_name != NULL) {
+ if (strcasecmp(fptr->rtpmap->encode_name, "mpeg4-generic") != 0) {
+ return -1;
+ }
+ if (userdata == NULL) {
+ fmtp = parse_fmtp_for_mpeg4(fptr->fmt_param, message);
+ if (fmtp != NULL) {
+ userdata = fmtp->config_binary;
+ userdata_size = fmtp->config_binary_len;
+ }
+ }
+ }
+ if (userdata != NULL) {
+ mpeg4_audio_config_t audio_config;
+ decode_mpeg4_audio_config(userdata, userdata_size, &audio_config);
+ message(LOG_DEBUG, "aac", "audio type is %d", audio_config.audio_object_type);
+ if (fmtp != NULL) free_fmtp_parse(fmtp);
+
+ if (audio_object_type_is_aac(&audio_config) == 0) {
+ return -1;
+ }
+#if 0
+ if (audio_config.audio_object_type == 17) {
+ message(LOG_INFO, "aac", "audio type is legal ISMA, but not supported");
+ return -1;
+ }
+#endif
+ return 2;
+ }
+#if 0
+ // I'm not sure I want to be here if we don't have an audio config
+ if (compressor != NULL) {
+ const char **lptr = aac_compressors;
+ while (*lptr != NULL) {
+ if (strcasecmp(*lptr, compressor) == 0) {
+ return 2;
+ }
+ lptr++;
+ }
+ }
+#endif
+ return -1;
+}
+
+#ifndef HAVE_PLUGIN_VERSION_0_8
+AUDIO_CODEC_WITH_RAW_FILE_PLUGIN("faad2",
+ aac_codec_create,
+ aac_do_pause,
+ aac_decode,
+ NULL,
+ aac_close,
+ aac_codec_check,
+ aac_file_check,
+ aac_file_next_frame,
+ aac_file_used_for_frame,
+ aac_raw_file_seek_to,
+ aac_file_eof
+ );
+#else
+AUDIO_CODEC_WITH_RAW_FILE_PLUGIN("faad2",
+ aac_codec_create,
+ aac_do_pause,
+ aac_decode,
+ NULL,
+ aac_close,
+ aac_codec_check,
+ aac_file_check,
+ aac_file_next_frame,
+ aac_file_used_for_frame,
+ aac_raw_file_seek_to,
+ aac_file_eof,
+ NULL,
+ 0
+ );
+#endif
+/* end file aa.cpp */
+
+
diff --git a/faad2/src/plugins/mpeg4ip/faad2.h b/faad2/src/plugins/mpeg4ip/faad2.h
new file mode 100644
index 0000000..b74cdff
--- /dev/null
+++ b/faad2/src/plugins/mpeg4ip/faad2.h
@@ -0,0 +1,92 @@
+/*
+** MPEG4IP plugin for FAAD2
+** Copyright (C) 2003 Bill May wmay@cisco.com
+**
+** This program is free software; you can redistribute it and/or modify
+** it under the terms of the GNU General Public License as published by
+** the Free Software Foundation; either version 2 of the License, or
+** (at your option) any later version.
+**
+** This program 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 General Public License for more details.
+**
+** You should have received a copy of the GNU General Public License
+** along with this program; if not, write to the Free Software
+** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+**
+** $Id: faad2.h,v 1.2 2004/01/05 14:05:12 menno Exp $
+**/
+/*
+ * aa.h - class definition for AAC codec.
+ */
+
+#ifndef __AA_H__
+#define __AA_H__ 1
+#include "faad.h"
+#include "codec_plugin.h"
+
+#ifndef M_LLU
+#define M_LLU M_64
+#define LLU U64
+#endif
+
+typedef struct aac_codec_t {
+ codec_data_t c;
+ audio_vft_t *m_vft;
+ void *m_ifptr;
+ faacDecHandle m_info;
+ int m_object_type;
+ int m_resync_with_header;
+ int m_record_sync_time;
+ uint64_t m_current_time;
+ uint64_t m_last_rtp_ts;
+ uint64_t m_msec_per_frame;
+ uint32_t m_current_frame;
+ int m_audio_inited;
+ int m_faad_inited;
+ int m_freq; // frequency
+ int m_chans; // channels
+ int m_output_frame_size;
+#if DUMP_OUTPUT_TO_FILE
+ FILE *m_outfile;
+#endif
+ FILE *m_ifile;
+ uint8_t *m_buffer;
+ uint32_t m_buffer_size_max;
+ uint32_t m_buffer_size;
+ uint32_t m_buffer_on;
+ uint64_t m_framecount;
+ int m_ignore_first_sample;
+ uint64_t m_last_ts;
+} aac_codec_t;
+
+#define m_vft c.v.audio_vft
+#define m_ifptr c.ifptr
+#define MAX_READ_BUFFER (768 * 8)
+
+#define aa_message aac->m_vft->log_msg
+void aac_close(codec_data_t *ptr);
+extern const char *aaclib;
+
+codec_data_t *aac_file_check(lib_message_func_t message,
+ const char *name,
+ double *max,
+ char *desc[4]
+#ifdef HAVE_PLUGIN_VERSION_0_8
+ , CConfigSet *pConfig
+#endif
+);
+
+int aac_file_next_frame(codec_data_t *ifptr,
+ uint8_t **buffer,
+ uint64_t *ts);
+int aac_file_eof(codec_data_t *ifptr);
+
+void aac_file_used_for_frame(codec_data_t *ifptr,
+ uint32_t bytes);
+
+int aac_raw_file_seek_to(codec_data_t *ifptr,
+ uint64_t ts);
+#endif