mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-11-24 03:59:43 +00:00
cleanup
fix streamcopy Originally committed as revision 3026 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer
parent
bf17ff617b
commit
068f2a22db
@ -40,8 +40,6 @@ static int flv_probe(AVProbeData *p)
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return 0;
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}
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#define FRAME_RATE_UNKNOWN 31415 //yes i know its beatifull
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static int flv_read_header(AVFormatContext *s,
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AVFormatParameters *ap)
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{
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@ -55,25 +53,6 @@ static int flv_read_header(AVFormatContext *s,
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url_fskip(&s->pb, 4);
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flags = get_byte(&s->pb);
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if ((flags & 1)) {
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st = av_new_stream(s, 0);
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if (!st)
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return AVERROR_NOMEM;
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st->codec.codec_type = CODEC_TYPE_VIDEO;
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st->codec.codec_id = CODEC_ID_FLV1;
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st->codec.frame_rate= FRAME_RATE_UNKNOWN;
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// st->codec.frame_rate= ap->frame_rate;
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// st->codec.frame_rate_base= ap->frame_rate_base;
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}
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if ((flags & 4)) {
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st = av_new_stream(s, 1);
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if (!st)
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return AVERROR_NOMEM;
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st->codec.codec_type = CODEC_TYPE_AUDIO;
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st->codec.codec_id = CODEC_ID_MP3;
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}
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offset = get_be32(&s->pb);
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url_fseek(&s->pb, offset, SEEK_SET);
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@ -82,10 +61,10 @@ static int flv_read_header(AVFormatContext *s,
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static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int ret, i, type, size, pts, flags;
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int ret, i, type, size, pts, flags, is_audio;
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AVStream *st;
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redo:
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for(;;){
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url_fskip(&s->pb, 4); /* size of previous packet */
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type = get_byte(&s->pb);
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size = get_be24(&s->pb);
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@ -95,43 +74,68 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
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return -EIO;
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url_fskip(&s->pb, 4); /* reserved */
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flags = 0;
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if(size == 0)
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continue;
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if (type == 8) {
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is_audio=1;
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flags = get_byte(&s->pb);
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size--;
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if ((flags >> 4) != 2) { /* 0: uncompressed 1: ADPCM 2: mp3 5: Nellymoser 8kHz mono 6: Nellymoser*/
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goto skip;
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}
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} else if (type == 9) {
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is_audio=0;
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flags = get_byte(&s->pb);
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size--;
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if ((flags & 0xF) != 2) { /* 2: only format */
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goto skip;
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}
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} else {
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skip:
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/* skip packet */
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av_log(s, AV_LOG_ERROR, "skipping flv packet: type %d, size %d, flags %d\n", type, size, flags);
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url_fskip(&s->pb, size);
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goto redo;
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continue;
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}
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/* now find stream */
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for(i=0;i<s->nb_streams;i++) {
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st = s->streams[i];
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if (st->id == ((type == 9) ? 0 : 1))
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goto found;
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if (st->id == is_audio)
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break;
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}
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goto skip;
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found:
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if(type == 8 && st->codec.sample_rate == 0){
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st->codec.channels = (flags&1)+1;
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st->codec.sample_rate = (44100<<((flags>>2)&3))>>3;
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if(i == s->nb_streams){
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st = av_new_stream(s, is_audio);
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if (!st)
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return AVERROR_NOMEM;
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st->codec.frame_rate_base= 0;
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}
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break;
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}
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//guess the frame rate
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if(type==9 && st->codec.frame_rate == FRAME_RATE_UNKNOWN && pts){
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st->codec.frame_rate_base=1;
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st->codec.frame_rate= (1000 + pts/2)/pts;
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if(is_audio){
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if(st->codec.sample_rate == 0){
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st->codec.codec_type = CODEC_TYPE_AUDIO;
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st->codec.channels = (flags&1)+1;
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if((flags >> 4) == 5)
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st->codec.sample_rate= 8000;
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else
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st->codec.sample_rate = (44100<<((flags>>2)&3))>>3;
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switch(flags >> 4){/* 0: uncompressed 1: ADPCM 2: mp3 5: Nellymoser 8kHz mono 6: Nellymoser*/
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case 2: st->codec.codec_id = CODEC_ID_MP3; break;
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default:
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st->codec.codec_tag= (flags >> 4);
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}
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}
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}else{
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if(st->codec.frame_rate_base == 0){
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st->codec.codec_type = CODEC_TYPE_VIDEO;
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//guess the frame rate
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if(pts){
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st->codec.frame_rate_base=1;
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st->codec.frame_rate= (1000 + pts/2)/pts;
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}
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switch(flags & 0xF){
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case 2: st->codec.codec_id = CODEC_ID_FLV1; break;
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default:
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st->codec.codec_tag= flags & 0xF;
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}
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}
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}
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if (av_new_packet(pkt, size) < 0)
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@ -18,6 +18,9 @@
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*/
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#include "avformat.h"
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#undef NDEBUG
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#include <assert.h>
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#define VIDEO_FIFO_SIZE 512
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typedef struct FLVFrame {
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@ -32,26 +35,14 @@ typedef struct FLVFrame {
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typedef struct FLVContext {
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int hasAudio;
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int hasVideo;
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#ifdef CONFIG_MP3LAME
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int audioTime;
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int audioInPos;
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int audioOutPos;
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int audioSize;
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// int audioRate;
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int initDelay;
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int soundDelay;
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uint8_t *audioFifo;
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int64_t sampleCount;
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#endif // CONFIG_MP3LAME
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int64_t frameCount;
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FLVFrame *frames;
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} FLVContext;
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#ifdef CONFIG_MP3LAME
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#define AUDIO_FIFO_SIZE 65536
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static const int sSampleRates[3][4] = {
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{44100, 48000, 32000, 0},
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{22050, 24000, 16000, 0},
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@ -138,16 +129,7 @@ static int flv_write_header(AVFormatContext *s)
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flv->hasAudio = 0;
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flv->hasVideo = 0;
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#ifdef CONFIG_MP3LAME
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flv->audioTime = -1;
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flv->audioFifo = av_malloc(AUDIO_FIFO_SIZE);
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flv->audioInPos = 0;
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flv->audioOutPos = 0;
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flv->audioSize = 0;
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// flv->audioRate = 44100;
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flv->initDelay = -1;
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flv->soundDelay = 0;
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#endif // CONFIG_MP3LAME
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flv->frames = 0;
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@ -248,19 +230,22 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
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ByteIOContext *pb = &s->pb;
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AVCodecContext *enc = &s->streams[stream_index]->codec;
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FLVContext *flv = s->priv_data;
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FLVFrame *frame = av_malloc(sizeof(FLVFrame));
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frame->next = 0;
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frame->size = size;
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frame->data = av_malloc(size);
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frame->timestamp = timestamp;
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memcpy(frame->data,buf,size);
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// av_log(s, AV_LOG_DEBUG, "type:%d pts: %lld size:%d\n", enc->codec_type, timestamp, size);
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if (enc->codec_type == CODEC_TYPE_VIDEO) {
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FLVFrame *frame = av_malloc(sizeof(FLVFrame));
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frame->next = 0;
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frame->type = 9;
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frame->flags = 2; // choose h263
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frame->flags |= enc->coded_frame->key_frame ? 0x10 : 0x20; // add keyframe indicator
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frame->timestamp = timestamp;
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//frame->timestamp = ( ( flv->frameCount * (int64_t)FRAME_RATE_BASE * (int64_t)1000 ) / (int64_t)enc->frame_rate );
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//printf("%08x %f %f\n",frame->timestamp,(double)enc->frame_rate/(double)FRAME_RATE_BASE,1000*(double)FRAME_RATE_BASE/(double)enc->frame_rate);
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frame->size = size;
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frame->data = av_malloc(size);
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memcpy(frame->data,buf,size);
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flv->hasVideo = 1;
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InsertSorted(flv,frame);
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@ -268,105 +253,81 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
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flv->frameCount ++;
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}
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else if (enc->codec_type == CODEC_TYPE_AUDIO) {
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#ifdef CONFIG_MP3LAME
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if (enc->codec_id == CODEC_ID_MP3 ) {
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int c=0;
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int soundFormat = 0x02;
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for (;c<size;c++) {
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flv->audioFifo[(flv->audioOutPos+c)%AUDIO_FIFO_SIZE] = buf[c];
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}
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flv->audioSize += size;
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flv->audioOutPos += size;
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flv->audioOutPos %= AUDIO_FIFO_SIZE;
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switch (enc->sample_rate) {
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case 44100:
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soundFormat |= 0x0C;
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break;
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case 22050:
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soundFormat |= 0x08;
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break;
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case 11025:
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soundFormat |= 0x04;
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break;
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case 8000: //nellymoser only
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case 5512: //not mp3
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soundFormat |= 0x00;
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break;
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default:
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assert(0);
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}
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if ( flv->initDelay == -1 ) {
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flv->initDelay = timestamp;
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}
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if (enc->channels > 1) {
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soundFormat |= 0x01;
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}
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switch(enc->codec_id){
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case CODEC_ID_MP3:
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soundFormat |= 0x20;
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break;
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case 0:
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soundFormat |= enc->codec_tag<<4;
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break;
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default:
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assert(0);
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}
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assert(size);
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if ( flv->initDelay == -1 ) {
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flv->initDelay = timestamp;
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}
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frame->type = 8;
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frame->flags = soundFormat;
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// if ( flv->audioTime == -1 ) {
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flv->audioTime = timestamp;
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// flv->audioTime = ( ( ( flv->sampleCount - enc->delay ) * 8000 ) / flv->audioRate ) - flv->initDelay - 250;
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// if ( flv->audioTime < 0 ) {
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// flv->audioTime = 0;
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// }
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// }
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}
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for ( ; flv->audioSize >= 4 ; ) {
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#ifdef CONFIG_MP3LAME
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if (enc->codec_id == CODEC_ID_MP3 ) {
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int mp3FrameSize = 0;
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int mp3SampleRate = 0;
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int mp3IsMono = 0;
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int mp3SamplesPerFrame = 0;
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int c=0;
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/* copy out mp3 header from ring buffer */
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uint8_t header[4];
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for (c=0; c<4; c++) {
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header[c] = flv->audioFifo[(flv->audioInPos+c) % AUDIO_FIFO_SIZE];
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}
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if(!mp3info(buf,&mp3FrameSize,&mp3SamplesPerFrame,&mp3SampleRate,&mp3IsMono))
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assert(0);
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assert ( size == mp3FrameSize );
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assert(enc->sample_rate == mp3SampleRate);
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// assert(enc->frame_size == mp3SamplesPerFrame);
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//av_log(NULL, AV_LOG_DEBUG, "sizes: %d %d\n", enc->frame_size, mp3SamplesPerFrame);
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if ( mp3info(header,&mp3FrameSize,&mp3SamplesPerFrame,&mp3SampleRate,&mp3IsMono) ) {
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if ( flv->audioSize >= mp3FrameSize ) {
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int soundFormat = 0x22;
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int c=0;
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FLVFrame *frame = av_malloc(sizeof(FLVFrame));
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// flv->audioRate = mp3SampleRate;
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switch (mp3SampleRate) {
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case 44100:
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soundFormat |= 0x0C;
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break;
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case 22050:
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soundFormat |= 0x08;
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break;
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case 11025:
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soundFormat |= 0x04;
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break;
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}
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if ( !mp3IsMono ) {
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soundFormat |= 0x01;
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}
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frame->next = 0;
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frame->type = 8;
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frame->flags = soundFormat;
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frame->timestamp = flv->audioTime;
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frame->timestamp = (1000*flv->sampleCount + mp3SampleRate/2)/(mp3SampleRate);
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frame->size = mp3FrameSize;
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frame->data = av_malloc(mp3FrameSize);
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for (;c<mp3FrameSize;c++) {
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frame->data[c] = flv->audioFifo[(flv->audioInPos+c)%AUDIO_FIFO_SIZE];
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}
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flv->audioInPos += mp3FrameSize;
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flv->audioSize -= mp3FrameSize;
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flv->audioInPos %= AUDIO_FIFO_SIZE;
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flv->sampleCount += mp3SamplesPerFrame;
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flv->audioTime += 1000*mp3SamplesPerFrame/mp3SampleRate;
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// We got audio! Make sure we set this to the global flags on closure
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flv->hasAudio = 1;
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InsertSorted(flv,frame);
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// av_log(NULL,AV_LOG_DEBUG, "insert sound\n");
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continue;
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}
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// av_log(NULL,AV_LOG_DEBUG, "insuficent data\n");
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break;
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}
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av_log(NULL,AV_LOG_DEBUG, "head trashed\n");
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flv->audioInPos ++;
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flv->audioSize --;
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flv->audioInPos %= AUDIO_FIFO_SIZE;
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// no audio in here!
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flv->audioTime = -1;
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frame->timestamp = (1000*flv->sampleCount + enc->sample_rate/2)/(enc->sample_rate);
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flv->sampleCount += mp3SamplesPerFrame;
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}
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#endif
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}
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// We got audio! Make sure we set this to the global flags on closure
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flv->hasAudio = 1;
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InsertSorted(flv,frame);
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}else
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assert(0);
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Dump(flv,pb,128);
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put_flush_packet(pb);
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return 0;
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