source: trunk/ff_private.c @ 72

/*****************************************************************************
*
*  Avi Import Component Private Functions
*
*  Copyright(C) 2006 Christoph Naegeli <chn1@mac.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
*
****************************************************************************/

#include "ff_private.h"
#include "avcodec.h"
#include "Codecprintf.h"

#include <CoreServices/CoreServices.h>
#include <AudioToolbox/AudioToolbox.h>

#define BSWAP(a) ( (((a)&0xff) << 24) | (((a)&0xff00) << 8) | (((a)&0xff0000) >> 8) | (((a) >> 24) & 0xff) )

/* This routine checks if the system requirements are fullfilled */
ComponentResult check_system()
{
        ComponentResult result;
        long systemVersion;
        
        result = Gestalt(gestaltSystemVersion, &systemVersion);
        require_noerr(result,bail);
        
        /* Make sure we have at least 10.4 installed...*/
        if(systemVersion < 0x00001040)
                result = -1;
        
bail:
                return result;
} /* check_system() */

/* This routine does register the ffmpeg parsers which would normally
 * be registred through the normal initialization process */
void register_parsers()
{
        /* Do we need more parsers here? */
    av_register_codec_parser(&mpegaudio_parser);
        av_register_codec_parser(&ac3_parser);
} /* register_parsers() */


/* This function prepares the target Track to receivve the movie data,
 * it is called if QT has asked an import operation which should just
 * load this track. After success, *outmap points to a valid stream maping
 * Return values:
 *        0: ok
 *      < 0: couldn't find a matching track
 */
int prepare_track(AVFormatContext *ic, NCStream **out_map, Track targetTrack, Handle dataRef, OSType dataRefType)
{
        int j;
        AVStream *st;
        AVStream *outstr = NULL;
        Media media;
        NCStream *map = NULL;
        
        /* If the track has already a media, return an err */
        media = GetTrackMedia(targetTrack);
        if(media) goto err;
        
        /* Search the AVFormatContext for a video stream */
        for(j = 0; j < ic->nb_streams && !outstr; j++) {
                st = ic->streams[j];
                if(st->codec->codec_type == CODEC_TYPE_VIDEO)
                        outstr = st;
        }
        /* Search the AVFormatContext for an audio stream (no video stream exists) */
        for(j = 0; j < ic->nb_streams && !outstr; j++) {
                st = ic->streams[j];
                if(st->codec->codec_type == CODEC_TYPE_AUDIO)
                        outstr = st;
        }
        /* Still no stream, then err */
        if(!outstr) goto err;
        
        /* prepare the stream map & initialize*/
        map = av_mallocz(sizeof(NCStream));
        map->index = st->index;
        map->str = outstr;
        
        if(st->codec->codec_type == CODEC_TYPE_VIDEO)
                initialize_video_map(map, targetTrack, dataRef, dataRefType);
        else if(st->codec->codec_type == CODEC_TYPE_AUDIO)
                initialize_audio_map(map, targetTrack, dataRef, dataRefType);
        
        /* return the map */
        *out_map = map;
        
        return 0;
err:
                if(map)
                        av_free(map);
        return -1;
} /* prepare_track() */

/* Initializes the map & targetTrack to receive video data */
void initialize_video_map(NCStream *map, Track targetTrack, Handle dataRef, OSType dataRefType)
{
        Media media;
        ImageDescriptionHandle imgHdl;
        AVCodecContext *codec;
        double num,den;
        
        codec = map->str->codec;
        
        map->base = map->str->time_base;
        if(map->base.den > 100000) {
                /* if that's the case, then we probably ran out of timevalues!
                * a timescale of 100000 allows a movie duration of 5-6 hours
                * so I think this will be enough */
                den = map->base.den;
                num = map->base.num;
                
                /* we use the following approach ##.### frames per second.
                        * afterwards rounding */
                den = den / num * 1000.0;
                map->base.num = 1000;
                map->base.den = round(den);
        }
        
        media = NewTrackMedia(targetTrack, VideoMediaType, map->base.den, dataRef, dataRefType);
        map->media = media;
        
        /* Create the Image Description Handle */
        imgHdl = (ImageDescriptionHandle)NewHandleClear(sizeof(ImageDescription));
        (*imgHdl)->idSize = sizeof(ImageDescription);
        
        if (codec->codec_tag)
                (*imgHdl)->cType = BSWAP(codec->codec_tag);
        else
                // need to lookup the fourcc from the codec_id
                (*imgHdl)->cType = map_video_codec_to_mov_tag(codec->codec_id);
        Codecprintf(NULL, "fourcc: %c%c%c%c\n",0xff & (*imgHdl)->cType,0xff & (*imgHdl)->cType>>8,0xff & (*imgHdl)->cType>>16,0xff & (*imgHdl)->cType>>24);
        
        (*imgHdl)->temporalQuality = codecMaxQuality;
        (*imgHdl)->spatialQuality = codecMaxQuality;
        (*imgHdl)->width = codec->width;
        (*imgHdl)->height = codec->height;
        (*imgHdl)->hRes = 72 << 16;
        (*imgHdl)->vRes = 72 << 16;
        (*imgHdl)->depth = codec->bits_per_sample;
        (*imgHdl)->clutID = -1; // no color lookup table...
        
        map->sampleHdl = (SampleDescriptionHandle)imgHdl;
        
} /* initialize_video_map() */

/* Initializes the map & targetTrack to receive audio data */
void initialize_audio_map(NCStream *map, Track targetTrack, Handle dataRef, OSType dataRefType)
{
        Media media;
        SoundDescriptionHandle sndHdl;
        AudioStreamBasicDescription asbd;
        AVCodecContext *codec;
        UInt32 ioSize;
        OSStatus err;
        
        uint8_t *cookie = NULL;
        int cookieSize = 0;
        
        codec = map->str->codec;
        map->base = map->str->time_base;
        
        media = NewTrackMedia(targetTrack, SoundMediaType, codec->sample_rate, dataRef, dataRefType);
        
        map->media = media;
        
        memset(&asbd,0,sizeof(asbd));
        map_avi_to_mov_tag(codec->codec_id, &asbd);
        if(asbd.mFormatID == 0) /* no know codec, use the ms tag */
                asbd.mFormatID = 'ms\0\0' + codec->codec_tag; /* the number is stored in the last byte => big endian */
        
        /* Ask the AudioToolbox about vbr of the codec */
        ioSize = sizeof(UInt32);
        AudioFormatGetProperty(kAudioFormatProperty_FormatIsVBR, sizeof(AudioStreamBasicDescription), &asbd, &ioSize, &map->vbr);
        
        /* Set some fields of the AudioStreamBasicDescription. Then ask the AudioToolbox
                * to fill as much as possible before creating the SoundDescriptionHandle */
        asbd.mSampleRate = codec->sample_rate;
        asbd.mChannelsPerFrame = codec->channels;
        if(!map->vbr) /* This works for all the tested codecs. but is there any better way? */
                asbd.mBytesPerPacket = codec->block_align; /* this is tested for alaw/mulaw/msadpcm */
        asbd.mFramesPerPacket = codec->frame_size; /* works for mp3, all other codecs this is 0 anyway */
        asbd.mBitsPerChannel = codec->bits_per_sample;
        
        // this probably isn't quite right; FLV doesn't set frame_size or block_align, 
        // but we need > 0 frames per packet or Apple's mp3 decoder won't work
        if (asbd.mBytesPerPacket == 0 && asbd.mFramesPerPacket == 0)
                asbd.mFramesPerPacket = 1;
        
        /* ask the toolbox about more information */
        ioSize = sizeof(AudioStreamBasicDescription);
        AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &ioSize, &asbd);
        
        /* If we have vbr audio, the media scale most likely has to be set to the time_base denumerator */
        if(map->vbr) {
                /* if we have mFramesPerPacket, set mBytesPerPacket to 0 as this can cause
                * errors if set incorrectly. But in vbr, we just need the mFramesPerPacket
                * value */
                if(asbd.mFramesPerPacket)
                        asbd.mBytesPerPacket = 0;
                SetMediaTimeScale(media, map->str->time_base.den);
        }
        
        /* FIXME. in the MSADPCM codec, we get a wrong mFramesPerPacket entry because
                * of the difference in the sample_rate and the time_base denumerator. So we
                * recalculate here the mFramesPerPacket entry */
        if(asbd.mBytesPerPacket) {
                /* For calculation, lets assume a packet duration of 1, use ioSize as tmp storage */
                ioSize = map->str->time_base.num * codec->sample_rate / map->str->time_base.den;
                /* downscale to correct bytes_per_packet */
                asbd.mFramesPerPacket = ioSize * asbd.mBytesPerPacket / codec->block_align;
        }
        
        /* here use a version1, because version2 will fail! (no idea why)
                * and as we are using version1, we may not have more than 2 channels.
                * perhaps we should go to version2 some day. */
        if(asbd.mChannelsPerFrame > 2)
                asbd.mChannelsPerFrame = 2;
        err = QTSoundDescriptionCreate(&asbd, NULL, 0, NULL, 0, kQTSoundDescriptionKind_Movie_LowestPossibleVersion, &sndHdl);
        if(err) fprintf(stderr, "AVI IMPORTER: Error creating the sound description\n");
        
        /* Create the magic cookie */
        cookie = create_cookie(codec, &cookieSize, asbd.mFormatID);
        if(cookie) {
                err = QTSoundDescriptionSetProperty(sndHdl, kQTPropertyClass_SoundDescription, kQTSoundDescriptionPropertyID_MagicCookie,
                                                                                        cookieSize, cookie);
                if(err) fprintf(stderr, "AVI IMPORTER: Error appending the magic cookie to the sound description\n");
                av_free(cookie);
        }
        
        map->sampleHdl = (SampleDescriptionHandle)sndHdl;
        map->asbd = asbd;
} /* initialize_audio_map() */

OSType map_video_codec_to_mov_tag(enum CodecID codec_id)
{
        switch(codec_id) {
                case CODEC_ID_FLV1:
                        return 'FLV1';
                case CODEC_ID_VP6F:
                        return 'VP6F';
                case CODEC_ID_FLASHSV:
                        return 'FSV1';
        }
        return 0;
}

/* maps the codec_id tag of libavformat to a constant the AudioToolbox can work with */
void map_avi_to_mov_tag(enum CodecID codec_id, AudioStreamBasicDescription *asbd)
{
        switch(codec_id) {
                case CODEC_ID_MP2:
                        asbd->mFormatID = kAudioFormatMPEGLayer2;
                        break;
                case CODEC_ID_MP3:
                        asbd->mFormatID = kAudioFormatMPEGLayer3;
                        break;
                        /* currently we use ms four_char_code for ac3 */
                        /* case CODEC_ID_AC3:
                        asbd->mFormatID = kAudioFormatAC3;
                        break; */
                case CODEC_ID_PCM_S16LE:
                        asbd->mFormatID = kAudioFormatLinearPCM;
                        asbd->mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
                        break;
                case CODEC_ID_PCM_U8:
                        asbd->mFormatID = kAudioFormatLinearPCM;
                        asbd->mFormatFlags = kLinearPCMFormatFlagIsBigEndian;
                        break;
                case CODEC_ID_PCM_ALAW:
                        asbd->mFormatID = kAudioFormatALaw;
                        break;
                case CODEC_ID_PCM_MULAW:
                        asbd->mFormatID = kAudioFormatULaw;
                        break;
                case CODEC_ID_ADPCM_MS:
                        asbd->mFormatID = kMicrosoftADPCMFormat;
                        break;
                case CODEC_ID_AAC:
                case CODEC_ID_MPEG4AAC:
                        asbd->mFormatID = kAudioFormatMPEG4AAC;
                        break;
                case CODEC_ID_VORBIS:
                        asbd->mFormatID = 'OggV';
                        break;
                default:
                        break;
        }
} /* map_avi_to_mov_tag() */

/* This function creates a magic cookie basec on the codec parameter and formatID
 * Return value: a pointer to a magic cookie which has to be av_free()'d
 * in cookieSize, the size of the magic cookie is returned to the caller */
uint8_t *create_cookie(AVCodecContext *codec, int *cookieSize, UInt32 formatID)
{
        uint8_t *result = NULL;
        uint8_t *ptr;
        AudioFormatAtom formatAtom;
        AudioTerminatorAtom termAtom;
        long waveSize;
        uint8_t *waveAtom = NULL;
        int size = 0;
        
        /* Do we need an endia atom, too? */
        
        /* initialize the user Atom
                * 8 bytes                       for the atom size & atom type
                * 18 bytes                      for the already extracted part, see wav.c in the ffmpeg project
                * extradata_size        for the data still stored in the AVCodecContext structure */
        waveSize = 18 + codec->extradata_size + 8;
        waveAtom = av_malloc(waveSize);
        
        /* now construct the wave atom */
        /* QT Atoms are big endian, I think but the WAVE data should be little endian */
        ptr = write_int32(waveAtom, EndianS32_NtoB(waveSize));
        ptr = write_int32(ptr, formatID);
        ptr = write_int16(ptr, EndianS16_NtoL(codec->codec_tag));
        ptr = write_int16(ptr, EndianS16_NtoL(codec->channels));
        ptr = write_int32(ptr, EndianS32_NtoL(codec->sample_rate));
        ptr = write_int32(ptr, EndianS32_NtoL(codec->bit_rate / 8));
        ptr = write_int16(ptr, EndianS16_NtoL(codec->block_align));
        ptr = write_int16(ptr, EndianS16_NtoL(codec->bits_per_sample));
        ptr = write_int16(ptr, EndianS16_NtoL(codec->extradata_size));
        /* now the remaining stuff */
        ptr = write_data(ptr, codec->extradata, codec->extradata_size);
        
        /* Calculate the size of the cookie */
        size  = sizeof(formatAtom) + sizeof(termAtom) + waveSize;
        
        /* Audio Format Atom */
        formatAtom.size = sizeof(AudioFormatAtom);
        formatAtom.atomType = kAudioFormatAtomType;
        formatAtom.format = formatID;
        
        /* Terminator Atom */
        termAtom.atomType = kAudioTerminatorAtomType;
        termAtom.size = sizeof(AudioTerminatorAtom);
        
        result = av_malloc(size);
        
        /* use ptr to write to the result */
        ptr = result;
        
        /* format atom */
        ptr = write_data(ptr, (uint8_t*)&formatAtom, sizeof(formatAtom));
        ptr = write_data(ptr, waveAtom, waveSize);
        ptr = write_data(ptr, (uint8_t*)&termAtom, sizeof(termAtom));
        
bail:
                *cookieSize = size;
        if(waveAtom)
                av_free(waveAtom);
        return result;
} /* create_cookie() */

/* write the int32_t data to target & then return a pointer which points after that data */
uint8_t *write_int32(uint8_t *target, int32_t data)
{
        return write_data(target, (uint8_t*)&data, sizeof(data));
} /* write_int32() */

/* write the int16_t data to target & then return a pointer which points after that data */
uint8_t *write_int16(uint8_t *target, int16_t data)
{
        return write_data(target, (uint8_t*)&data, sizeof(data));
} /* write_int16() */

/* write the data to the target adress & then return a pointer which points after the written data */
uint8_t *write_data(uint8_t *target, uint8_t* data, int32_t data_size)
{
        if(data_size > 0)
                memcpy(target, data, data_size);
        return (target + data_size);
} /* write_data() */


/* Add the meta data that lavf exposes to the movie */
void add_metadata(AVFormatContext *ic, Movie theMovie)
{
    QTMetaDataRef movie_metadata;
    OSType key, err;
    
    err = QTCopyMovieMetaData(theMovie, &movie_metadata);
    if (err) return;
    
    key = kQTMetaDataCommonKeyDisplayName;
    QTMetaDataAddItem(movie_metadata, kQTMetaDataStorageFormatQuickTime, kQTMetaDataKeyFormatCommon, 
                      (UInt8 *)&key, sizeof(key), (UInt8 *)ic->title, strlen(ic->title), kQTMetaDataTypeUTF8, NULL);

    key = kQTMetaDataCommonKeyAuthor;
    QTMetaDataAddItem(movie_metadata, kQTMetaDataStorageFormatQuickTime, kQTMetaDataKeyFormatCommon, 
                      (UInt8 *)&key, sizeof(key), (UInt8 *)ic->author, strlen(ic->author), kQTMetaDataTypeUTF8, NULL);

    key = kQTMetaDataCommonKeyCopyright;
    QTMetaDataAddItem(movie_metadata, kQTMetaDataStorageFormatQuickTime, kQTMetaDataKeyFormatCommon, 
                      (UInt8 *)&key, sizeof(key), (UInt8 *)ic->copyright, strlen(ic->copyright), kQTMetaDataTypeUTF8, NULL);

    key = kQTMetaDataCommonKeyComment;
    QTMetaDataAddItem(movie_metadata, kQTMetaDataStorageFormatQuickTime, kQTMetaDataKeyFormatCommon, 
                      (UInt8 *)&key, sizeof(key), (UInt8 *)ic->comment, strlen(ic->comment), kQTMetaDataTypeUTF8, NULL);

    key = kQTMetaDataCommonKeyComment;
    QTMetaDataAddItem(movie_metadata, kQTMetaDataStorageFormatQuickTime, kQTMetaDataKeyFormatCommon, 
                      (UInt8 *)&key, sizeof(key), (UInt8 *)ic->album, strlen(ic->album), kQTMetaDataTypeUTF8, NULL);

    key = kQTMetaDataCommonKeyGenre;
    QTMetaDataAddItem(movie_metadata, kQTMetaDataStorageFormatQuickTime, kQTMetaDataKeyFormatCommon, 
                      (UInt8 *)&key, sizeof(key), (UInt8 *)ic->genre, strlen(ic->genre), kQTMetaDataTypeUTF8, NULL);
    QTMetaDataRelease(movie_metadata);
}

/* This function prepares the movie to receivve the movie data,
 * After success, *out_map points to a valid stream maping
 * Return values:
 *        0: ok
 */
int prepare_movie(AVFormatContext *ic, NCStream **out_map, Movie theMovie, Handle dataRef, OSType dataRefType)
{
        int j;
        AVStream *st;
        NCStream *map;
        Track track;
        Track first_audio_track = NULL;
        
        /* make the stream map structure */
        map = av_mallocz(ic->nb_streams * sizeof(NCStream));
        
        for(j = 0; j < ic->nb_streams; j++) {
                
                st = ic->streams[j];
                map[j].index = st->index;
                map[j].str = st;
                
                if(st->codec->codec_type == CODEC_TYPE_VIDEO) {
                        track = NewMovieTrack(theMovie, st->codec->width << 16, st->codec->height << 16, kNoVolume);
                        initialize_video_map(&map[j], track, dataRef, dataRefType);
                } else {
                        track = NewMovieTrack(theMovie, 0, 0, kFullVolume);
                        initialize_audio_map(&map[j], track, dataRef, dataRefType);
                        
                        if (first_audio_track == NULL)
                                first_audio_track = track;
                        else
                                SetTrackAlternate(track, first_audio_track);
                }
        }
        
    add_metadata(ic, theMovie);
    
        *out_map = map;
        
        return 0;
} /* prepare_movie() */

/* This function imports the avi represented by the AVFormatContext to the movie media represented
 * in the map function. The aviheader_offset is used to calculate the packet offset from the
 * beginning of the file */
void import_avi(AVFormatContext *ic, NCStream *map, int64_t aviheader_offset)
{
        int j, k, l;
        NCStream *ncstr;
        AVStream *stream;
        AVCodecContext *codec;
        SampleReference64Record sampleRec;
        int64_t offset,duration;
        OSStatus err;
        short flags;
        short hadIndex = 0;
        
        /* process each stream in ic */
        for(j = 0; j < ic->nb_streams; j++) {
                
                ncstr = &map[j];
                stream = ncstr->str;
                codec = stream->codec;
                
                /* no stream we can read */
                if(!ncstr->media)
                        continue;
                
                /* now parse the index entries */
                for(k = 0; k < stream->nb_index_entries; k++) {
                        
                        hadIndex = 1;
                        
                        /* file offset */
                        offset = aviheader_offset + stream->index_entries[k].pos;
                        
                        /* flags */
                        flags = 0;
                        if((stream->index_entries[k].flags & AVINDEX_KEYFRAME) == 0)
                                flags |= mediaSampleNotSync;
                        
                        /* set as many fields in sampleRec as possible */
                        memset(&sampleRec, 0, sizeof(sampleRec));
                        sampleRec.dataOffset.hi = offset >> 32;
                        sampleRec.dataOffset.lo = (uint32_t)offset;
                        sampleRec.dataSize = stream->index_entries[k].size;
                        sampleRec.sampleFlags = flags;
                        
                        /* some samples have a data_size of zero. if that's the case, ignore them
                                * they seem to be used to stretch the frame duration & are already handled
                                * by the previous pkt */
                        if(sampleRec.dataSize <= 0)
                                continue;
                        
                        /* switch for the remaining fields */
                        if(codec->codec_type == CODEC_TYPE_VIDEO) {
                                
                                /* Calculate the frame duration */
                                duration = 1;
                                for(l = k+1; l < stream->nb_index_entries; l++) {
                                        if(stream->index_entries[l].size > 0)
                                                break;
                                        duration++;
                                }
                                
                                sampleRec.durationPerSample = map->base.num * duration;
                                sampleRec.numberOfSamples = 1;
                        }
                        else if(codec->codec_type == CODEC_TYPE_AUDIO) {
                                
                                /* FIXME: check if that's really the right thing to do here */
                                if(ncstr->vbr) {
                                        if(codec->frame_size == ncstr->base.num) {
                                                sampleRec.durationPerSample = codec->frame_size;
                                                sampleRec.numberOfSamples = 1;
                                        } else if (ncstr->asbd.mFormatID == kAudioFormatMPEG4AAC) {
                                                /* AVI-mux GUI, the author of which created this hack in the first place,
                                                * seems to special-case getting an AAC audio sample's duration this way */
                                                sampleRec.durationPerSample = ic->streams[j]->time_base.num;
                                                sampleRec.numberOfSamples = 1;
                                        } else {
                                                /* This seems to work. Although I have no idea why.
                                                * Perhaps the stream's timebase is adjusted to
                                                * let that work. as the timebase has strange values...*/
                                                sampleRec.durationPerSample = sampleRec.dataSize;
                                                sampleRec.numberOfSamples = 1;
                                        }
                                } else {
                                        sampleRec.durationPerSample = 1;
                                        sampleRec.numberOfSamples = (stream->index_entries[k].size * ncstr->asbd.mFramesPerPacket) / ncstr->asbd.mBytesPerPacket;
                                }
                        }
                        
                        /* Add the sample to the media */
                        err = AddMediaSampleReferences64(ncstr->media, ncstr->sampleHdl, 1, &sampleRec, NULL);
                }
        }
        if(hadIndex == 0)
        {
                AVPacket pkt;
                while(av_read_frame(ic, &pkt) == noErr)
                {
                        ncstr = &map[pkt.stream_index];
                        stream = ncstr->str;
                        codec = stream->codec;
                        
                        flags = 0;
                        if((pkt.flags & PKT_FLAG_KEY) == 0)
                                flags |= mediaSampleNotSync;
                        
                        memset(&sampleRec, 0, sizeof(sampleRec));
                        sampleRec.dataOffset.hi = pkt.pos >> 32;
                        sampleRec.dataOffset.lo = (uint32_t) pkt.pos;
                        sampleRec.dataSize = pkt.size;
                        sampleRec.sampleFlags = flags;
                        
                        if(sampleRec.dataSize <= 0)
                                continue;
                        
                        if (codec->codec_type == CODEC_TYPE_AUDIO && !ncstr->vbr)
                                sampleRec.numberOfSamples = (pkt.size * ncstr->asbd.mFramesPerPacket) / ncstr->asbd.mBytesPerPacket;
                        else
                                sampleRec.numberOfSamples = 1;
                                
                        // we have a sample waiting to be added; calculate the duration and add it
                        if (ncstr->lastSample.numberOfSamples > 0) {
                                ncstr->lastSample.durationPerSample = (pkt.pts - ncstr->lastpts) * ncstr->base.num;
                                AddMediaSampleReferences64(ncstr->media, ncstr->sampleHdl, 1, &ncstr->lastSample, NULL);
                        }
                        
                        if (pkt.duration == 0) {
                                // no duration, we'll have to wait for the next packet to calculate it
                                // keep the duration of the last sample, so we can use it if it's the last frame
                                sampleRec.durationPerSample = ncstr->lastSample.durationPerSample;
                                ncstr->lastSample = sampleRec;
                                ncstr->lastpts = pkt.pts;
                                
                        } else {
                                ncstr->lastSample.numberOfSamples = 0;
                                if (codec->codec_type == CODEC_TYPE_AUDIO && !ncstr->vbr)
                                        sampleRec.durationPerSample = 1;
                                else
                                        sampleRec.durationPerSample = pkt.duration * ncstr->base.num;
                                AddMediaSampleReferences64(ncstr->media, ncstr->sampleHdl, 1, &sampleRec, NULL);
                        }
#if 0
                        if(codec->codec_type == CODEC_TYPE_VIDEO)
                        {
                                if(pkt.duration == 0)
                                        sampleRec.durationPerSample = map->base.num;
                                else
                                        sampleRec.durationPerSample = map->base.num * pkt.duration;
                                sampleRec.numberOfSamples = 1;
                        }
                        else if (codec->codec_type == CODEC_TYPE_AUDIO)
                        {
                                if(ncstr->vbr) {
                                        if(codec->frame_size == ncstr->base.num) {
                                                sampleRec.durationPerSample = codec->frame_size;
                                                sampleRec.numberOfSamples = 1;
                                        } else if (ncstr->asbd.mFormatID == kAudioFormatMPEG4AAC) {
                                                /* AVI-mux GUI, the author of which created this hack in the first place,
                                                * seems to special-case getting an AAC audio sample's duration this way */
                                                sampleRec.durationPerSample = ic->streams[pkt.stream_index]->time_base.num;
                                                sampleRec.numberOfSamples = 1;
                                        } else {
                                                /* This seems to work. Although I have no idea why.
                                                * Perhaps the stream's timebase is adjusted to
                                                * let that work. as the timebase has strange values...*/
                                                sampleRec.durationPerSample = sampleRec.dataSize;
                                                sampleRec.numberOfSamples = 1;
                                        }
                                } else {
                                        sampleRec.durationPerSample = 1;
                                        sampleRec.numberOfSamples = (pkt.size * ncstr->asbd.mFramesPerPacket) / ncstr->asbd.mBytesPerPacket;
                                }
                        }
                        err = AddMediaSampleReferences64(ncstr->media, ncstr->sampleHdl, 1, &sampleRec, NULL);
#endif
                        //Need to do something like this when the libavformat doesn't give us a position
/*                      Handle dataIn = NewHandle(pkt.size);
                        HLock(dataIn);
                        memcpy(*dataIn, pkt.data, pkt.size);
                        HUnlock(dataIn);
                        err = AddMediaSample(ncstr->media, dataIn, 0, pkt.size, 1, ncstr->sampleHdl, pkt.duration, sampleRec.sampleFlags, NULL);*/
                        av_free_packet(&pkt);
                }
                // import the last frames
                for (j = 0; j < ic->nb_streams; j++) {
                        ncstr = &map[j];
                        if (ncstr->lastSample.numberOfSamples > 0)
                                AddMediaSampleReferences64(ncstr->media, ncstr->sampleHdl, 1, &ncstr->lastSample, NULL);
                }
        }
} /* import_avi() */