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ffmpeg.wasm-core
Commit 068f2a22 authored by Michael Niedermayer's avatar Michael Niedermayer
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cleanup 

fix streamcopy

Originally committed as revision 3026 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent bf17ff61
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Showing with 116 additions and 151 deletions
libavformat/flvdec.c
View file @ 068f2a22
......@@ -40,8 +40,6 @@ static int flv_probe(AVProbeData *p)
return 0;
}
#define FRAME_RATE_UNKNOWN 31415 //yes i know its beatifull
static int flv_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
......@@ -55,25 +53,6 @@ static int flv_read_header(AVFormatContext *s,
url_fskip(&s->pb, 4);
flags = get_byte(&s->pb);
if ((flags & 1)) {
st = av_new_stream(s, 0);
if (!st)
return AVERROR_NOMEM;
st->codec.codec_type = CODEC_TYPE_VIDEO;
st->codec.codec_id = CODEC_ID_FLV1;
st->codec.frame_rate= FRAME_RATE_UNKNOWN;
// st->codec.frame_rate= ap->frame_rate;
// st->codec.frame_rate_base= ap->frame_rate_base;
}
if ((flags & 4)) {
st = av_new_stream(s, 1);
if (!st)
return AVERROR_NOMEM;
st->codec.codec_type = CODEC_TYPE_AUDIO;
st->codec.codec_id = CODEC_ID_MP3;
}
offset = get_be32(&s->pb);
url_fseek(&s->pb, offset, SEEK_SET);
......@@ -82,10 +61,10 @@ static int flv_read_header(AVFormatContext *s,
static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, i, type, size, pts, flags;
int ret, i, type, size, pts, flags, is_audio;
AVStream *st;
redo:
for(;;){
url_fskip(&s->pb, 4); /* size of previous packet */
type = get_byte(&s->pb);
size = get_be24(&s->pb);
......@@ -95,43 +74,68 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
return -EIO;
url_fskip(&s->pb, 4); /* reserved */
flags = 0;
if(size == 0)
continue;
if (type == 8) {
is_audio=1;
flags = get_byte(&s->pb);
size--;
if ((flags >> 4) != 2) { /* 0: uncompressed 1: ADPCM 2: mp3 5: Nellymoser 8kHz mono 6: Nellymoser*/
goto skip;
}
} else if (type == 9) {
is_audio=0;
flags = get_byte(&s->pb);
size--;
if ((flags & 0xF) != 2) { /* 2: only format */
goto skip;
}
} else {
skip:
/* skip packet */
av_log(s, AV_LOG_ERROR, "skipping flv packet: type %d, size %d, flags %d\n", type, size, flags);
url_fskip(&s->pb, size);
goto redo;
continue;
}
/* now find stream */
for(i=0;i<s->nb_streams;i++) {
st = s->streams[i];
if (st->id == ((type == 9) ? 0 : 1))
goto found;
if (st->id == is_audio)
break;
}
goto skip;
found:
if(type == 8 && st->codec.sample_rate == 0){
st->codec.channels = (flags&1)+1;
st->codec.sample_rate = (44100<<((flags>>2)&3))>>3;
if(i == s->nb_streams){
st = av_new_stream(s, is_audio);
if (!st)
return AVERROR_NOMEM;
st->codec.frame_rate_base= 0;
}
//guess the frame rate
if(type==9 && st->codec.frame_rate == FRAME_RATE_UNKNOWN && pts){
st->codec.frame_rate_base=1;
st->codec.frame_rate= (1000 + pts/2)/pts;
break;
}
if(is_audio){
if(st->codec.sample_rate == 0){
st->codec.codec_type = CODEC_TYPE_AUDIO;
st->codec.channels = (flags&1)+1;
if((flags >> 4) == 5)
st->codec.sample_rate= 8000;
else
st->codec.sample_rate = (44100<<((flags>>2)&3))>>3;
switch(flags >> 4){/* 0: uncompressed 1: ADPCM 2: mp3 5: Nellymoser 8kHz mono 6: Nellymoser*/
case 2: st->codec.codec_id = CODEC_ID_MP3; break;
default:
st->codec.codec_tag= (flags >> 4);
}
}
}else{
if(st->codec.frame_rate_base == 0){
st->codec.codec_type = CODEC_TYPE_VIDEO;
//guess the frame rate
if(pts){
st->codec.frame_rate_base=1;
st->codec.frame_rate= (1000 + pts/2)/pts;
}
switch(flags & 0xF){
case 2: st->codec.codec_id = CODEC_ID_FLV1; break;
default:
st->codec.codec_tag= flags & 0xF;
}
}
}
if (av_new_packet(pkt, size) < 0)
......
libavformat/flvenc.c
View file @ 068f2a22
......@@ -18,6 +18,9 @@
*/
#include "avformat.h"
#undef NDEBUG
#include <assert.h>
#define VIDEO_FIFO_SIZE 512
typedef struct FLVFrame {
......@@ -32,26 +35,14 @@ typedef struct FLVFrame {
typedef struct FLVContext {
int hasAudio;
int hasVideo;
#ifdef CONFIG_MP3LAME
int audioTime;
int audioInPos;
int audioOutPos;
int audioSize;
// int audioRate;
int initDelay;
int soundDelay;
uint8_t *audioFifo;
int64_t sampleCount;
#endif // CONFIG_MP3LAME
int64_t frameCount;
FLVFrame *frames;
} FLVContext;
#ifdef CONFIG_MP3LAME
#define AUDIO_FIFO_SIZE 65536
static const int sSampleRates[3][4] = {
{44100, 48000, 32000, 0},
{22050, 24000, 16000, 0},
......@@ -138,16 +129,7 @@ static int flv_write_header(AVFormatContext *s)
flv->hasAudio = 0;
flv->hasVideo = 0;
#ifdef CONFIG_MP3LAME
flv->audioTime = -1;
flv->audioFifo = av_malloc(AUDIO_FIFO_SIZE);
flv->audioInPos = 0;
flv->audioOutPos = 0;
flv->audioSize = 0;
// flv->audioRate = 44100;
flv->initDelay = -1;
flv->soundDelay = 0;
#endif // CONFIG_MP3LAME
flv->frames = 0;
......@@ -248,19 +230,22 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
ByteIOContext *pb = &s->pb;
AVCodecContext *enc = &s->streams[stream_index]->codec;
FLVContext *flv = s->priv_data;
FLVFrame *frame = av_malloc(sizeof(FLVFrame));
frame->next = 0;
frame->size = size;
frame->data = av_malloc(size);
frame->timestamp = timestamp;
memcpy(frame->data,buf,size);
// av_log(s, AV_LOG_DEBUG, "type:%d pts: %lld size:%d\n", enc->codec_type, timestamp, size);
if (enc->codec_type == CODEC_TYPE_VIDEO) {
FLVFrame *frame = av_malloc(sizeof(FLVFrame));
frame->next = 0;
frame->type = 9;
frame->flags = 2; // choose h263
frame->flags |= enc->coded_frame->key_frame ? 0x10 : 0x20; // add keyframe indicator
frame->timestamp = timestamp;
//frame->timestamp = ( ( flv->frameCount * (int64_t)FRAME_RATE_BASE * (int64_t)1000 ) / (int64_t)enc->frame_rate );
//printf("%08x %f %f\n",frame->timestamp,(double)enc->frame_rate/(double)FRAME_RATE_BASE,1000*(double)FRAME_RATE_BASE/(double)enc->frame_rate);
frame->size = size;
frame->data = av_malloc(size);
memcpy(frame->data,buf,size);
flv->hasVideo = 1;
InsertSorted(flv,frame);
......@@ -268,105 +253,81 @@ static int flv_write_packet(AVFormatContext *s, int stream_index,
flv->frameCount ++;
}
else if (enc->codec_type == CODEC_TYPE_AUDIO) {
#ifdef CONFIG_MP3LAME
if (enc->codec_id == CODEC_ID_MP3 ) {
int c=0;
int soundFormat = 0x02;
for (;c<size;c++) {
flv->audioFifo[(flv->audioOutPos+c)%AUDIO_FIFO_SIZE] = buf[c];
}
flv->audioSize += size;
flv->audioOutPos += size;
flv->audioOutPos %= AUDIO_FIFO_SIZE;
switch (enc->sample_rate) {
case 44100:
soundFormat |= 0x0C;
break;
case 22050:
soundFormat |= 0x08;
break;
case 11025:
soundFormat |= 0x04;
break;
case 8000: //nellymoser only
case 5512: //not mp3
soundFormat |= 0x00;
break;
default:
assert(0);
}
if ( flv->initDelay == -1 ) {
flv->initDelay = timestamp;
}
if (enc->channels > 1) {
soundFormat |= 0x01;
}
switch(enc->codec_id){
case CODEC_ID_MP3:
soundFormat |= 0x20;
break;
case 0:
soundFormat |= enc->codec_tag<<4;
break;
default:
assert(0);
}
assert(size);
if ( flv->initDelay == -1 ) {
flv->initDelay = timestamp;
}
frame->type = 8;
frame->flags = soundFormat;
// if ( flv->audioTime == -1 ) {
flv->audioTime = timestamp;
// flv->audioTime = ( ( ( flv->sampleCount - enc->delay ) * 8000 ) / flv->audioRate ) - flv->initDelay - 250;
// if ( flv->audioTime < 0 ) {
// flv->audioTime = 0;
// }
// }
}
for ( ; flv->audioSize >= 4 ; ) {
#ifdef CONFIG_MP3LAME
if (enc->codec_id == CODEC_ID_MP3 ) {
int mp3FrameSize = 0;
int mp3SampleRate = 0;
int mp3IsMono = 0;
int mp3SamplesPerFrame = 0;
int c=0;
/* copy out mp3 header from ring buffer */
uint8_t header[4];
for (c=0; c<4; c++) {
header[c] = flv->audioFifo[(flv->audioInPos+c) % AUDIO_FIFO_SIZE];
}
if ( mp3info(header,&mp3FrameSize,&mp3SamplesPerFrame,&mp3SampleRate,&mp3IsMono) ) {
if ( flv->audioSize >= mp3FrameSize ) {
int soundFormat = 0x22;
int c=0;
FLVFrame *frame = av_malloc(sizeof(FLVFrame));
// flv->audioRate = mp3SampleRate;
switch (mp3SampleRate) {
case 44100:
soundFormat |= 0x0C;
break;
case 22050:
soundFormat |= 0x08;
break;
case 11025:
soundFormat |= 0x04;
break;
}
if ( !mp3IsMono ) {
soundFormat |= 0x01;
}
frame->next = 0;
frame->type = 8;
frame->flags = soundFormat;
frame->timestamp = flv->audioTime;
frame->timestamp = (1000*flv->sampleCount + mp3SampleRate/2)/(mp3SampleRate);
frame->size = mp3FrameSize;
frame->data = av_malloc(mp3FrameSize);
for (;c<mp3FrameSize;c++) {
frame->data[c] = flv->audioFifo[(flv->audioInPos+c)%AUDIO_FIFO_SIZE];
}
flv->audioInPos += mp3FrameSize;
flv->audioSize -= mp3FrameSize;
flv->audioInPos %= AUDIO_FIFO_SIZE;
flv->sampleCount += mp3SamplesPerFrame;
flv->audioTime += 1000*mp3SamplesPerFrame/mp3SampleRate;
// We got audio! Make sure we set this to the global flags on closure
flv->hasAudio = 1;
InsertSorted(flv,frame);
// av_log(NULL,AV_LOG_DEBUG, "insert sound\n");
continue;
}
// av_log(NULL,AV_LOG_DEBUG, "insuficent data\n");
break;
}
av_log(NULL,AV_LOG_DEBUG, "head trashed\n");
flv->audioInPos ++;
flv->audioSize --;
flv->audioInPos %= AUDIO_FIFO_SIZE;
// no audio in here!
flv->audioTime = -1;
if(!mp3info(buf,&mp3FrameSize,&mp3SamplesPerFrame,&mp3SampleRate,&mp3IsMono))
assert(0);
assert ( size == mp3FrameSize );
assert(enc->sample_rate == mp3SampleRate);
// assert(enc->frame_size == mp3SamplesPerFrame);
//av_log(NULL, AV_LOG_DEBUG, "sizes: %d %d\n", enc->frame_size, mp3SamplesPerFrame);
frame->timestamp = (1000*flv->sampleCount + enc->sample_rate/2)/(enc->sample_rate);
flv->sampleCount += mp3SamplesPerFrame;
}
#endif
}
// We got audio! Make sure we set this to the global flags on closure
flv->hasAudio = 1;
InsertSorted(flv,frame);
}else
assert(0);
Dump(flv,pb,128);
put_flush_packet(pb);
return 0;
......
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