ref: 1111ed43e82eaea2b93f4e4e99914807fe772776
dir: /plugins/mpeg4ip/faad2.cpp/
/*
** 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 */