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Publier sur MédiaSpip
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Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
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Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
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Trying to sync audio/visual using FFMpeg and openAL
22 août 2013, par user1379811hI have been studying dranger ffmpeg tutorial which explains how to sync audio and visual once you have the frames displayed and audio playing which is where im at.
Unfortunately, the tutorial is out of date (Stephen Dranger explaained that himself to me) and also uses sdl which im not doing - this is for Blackberry 10 application.
I just cannot make the video frames display at the correct speed (they are just playing very fast) and I have been trying for over a week now - seriously !
I have 3 threads happening - one to read from stream into audio and video queues and then 2 threads for audio and video.
If somebody could explain whats happening after scanning my relevent code you would be a lifesaver.
The delay (what I pass to usleep(testDelay) seems to be going up (incrementing) which doesn't seem right to me.
count = 1;
MyApp* inst = worker->app;//(VideoUploadFacebook*)arg;
qDebug() << "\n start loadstream";
w = new QWaitCondition();
w2 = new QWaitCondition();
context = avformat_alloc_context();
inst->threadStarted = true;
cout << "start of decoding thread";
cout.flush();
av_register_all();
avcodec_register_all();
avformat_network_init();
av_log_set_callback(&log_callback);
AVInputFormat *pFormat;
//const char device[] = "/dev/video0";
const char formatName[] = "mp4";
cout << "2start of decoding thread";
cout.flush();
if (!(pFormat = av_find_input_format(formatName))) {
printf("can't find input format %s\n", formatName);
//return void*;
}
//open rtsp
if(avformat_open_input(&context, inst->capturedUrl.data(), pFormat,NULL) != 0){
// return ;
cout << "error opening of decoding thread: " << inst->capturedUrl.data();
cout.flush();
}
cout << "3start of decoding thread";
cout.flush();
// av_dump_format(context, 0, inst->capturedUrl.data(), 0);
/* if(avformat_find_stream_info(context,NULL) < 0){
return EXIT_FAILURE;
}
*/
//search video stream
for(int i =0;inb_streams;i++){
if(context->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
inst->video_stream_index = i;
}
cout << "3z start of decoding thread";
cout.flush();
AVFormatContext* oc = avformat_alloc_context();
av_read_play(context);//play RTSP
AVDictionary *optionsDict = NULL;
ccontext = context->streams[inst->video_stream_index]->codec;
inst->audioc = context->streams[1]->codec;
cout << "4start of decoding thread";
cout.flush();
codec = avcodec_find_decoder(ccontext->codec_id);
ccontext->pix_fmt = PIX_FMT_YUV420P;
AVCodec* audio_codec = avcodec_find_decoder(inst->audioc->codec_id);
inst->packet = new AVPacket();
if (!audio_codec) {
cout << "audio codec not found\n"; //fflush( stdout );
exit(1);
}
if (avcodec_open2(inst->audioc, audio_codec, NULL) < 0) {
cout << "could not open codec\n"; //fflush( stdout );
exit(1);
}
if (avcodec_open2(ccontext, codec, &optionsDict) < 0) exit(1);
cout << "5start of decoding thread";
cout.flush();
inst->pic = avcodec_alloc_frame();
av_init_packet(inst->packet);
while(av_read_frame(context,inst->packet) >= 0 && &inst->keepGoing)
{
if(inst->packet->stream_index == 0){//packet is video
int check = 0;
// av_init_packet(inst->packet);
int result = avcodec_decode_video2(ccontext, inst->pic, &check, inst->packet);
if(check)
break;
}
}
inst->originalVideoWidth = inst->pic->width;
inst->originalVideoHeight = inst->pic->height;
float aspect = (float)inst->originalVideoHeight / (float)inst->originalVideoWidth;
inst->newVideoWidth = inst->originalVideoWidth;
int newHeight = (int)(inst->newVideoWidth * aspect);
inst->newVideoHeight = newHeight;//(int)inst->originalVideoHeight / inst->originalVideoWidth * inst->newVideoWidth;// = new height
int size = avpicture_get_size(PIX_FMT_YUV420P, inst->originalVideoWidth, inst->originalVideoHeight);
uint8_t* picture_buf = (uint8_t*)(av_malloc(size));
avpicture_fill((AVPicture *) inst->pic, picture_buf, PIX_FMT_YUV420P, inst->originalVideoWidth, inst->originalVideoHeight);
picrgb = avcodec_alloc_frame();
int size2 = avpicture_get_size(PIX_FMT_YUV420P, inst->newVideoWidth, inst->newVideoHeight);
uint8_t* picture_buf2 = (uint8_t*)(av_malloc(size2));
avpicture_fill((AVPicture *) picrgb, picture_buf2, PIX_FMT_YUV420P, inst->newVideoWidth, inst->newVideoHeight);
if(ccontext->pix_fmt != PIX_FMT_YUV420P)
{
std::cout << "fmt != 420!!!: " << ccontext->pix_fmt << std::endl;//
// return (EXIT_SUCCESS);//-1;
}
if (inst->createForeignWindow(inst->myForeignWindow->windowGroup(),
"HelloForeignWindowAppIDqq", 0,
0, inst->newVideoWidth,
inst->newVideoHeight)) {
} else {
qDebug() << "The ForeginWindow was not properly initialized";
}
inst->keepGoing = true;
inst->img_convert_ctx = sws_getContext(inst->originalVideoWidth, inst->originalVideoHeight, PIX_FMT_YUV420P, inst->newVideoWidth, inst->newVideoHeight,
PIX_FMT_YUV420P, SWS_BILINEAR, NULL, NULL, NULL);
is = (VideoState*)av_mallocz(sizeof(VideoState));
if (!is)
return NULL;
is->audioStream = 1;
is->audio_st = context->streams[1];
is->audio_buf_size = 0;
is->audio_buf_index = 0;
is->videoStream = 0;
is->video_st = context->streams[0];
is->frame_timer = (double)av_gettime() / 1000000.0;
is->frame_last_delay = 40e-3;
is->av_sync_type = DEFAULT_AV_SYNC_TYPE;
//av_strlcpy(is->filename, filename, sizeof(is->filename));
is->iformat = pFormat;
is->ytop = 0;
is->xleft = 0;
/* start video display */
is->pictq_mutex = new QMutex();
is->pictq_cond = new QWaitCondition();
is->subpq_mutex = new QMutex();
is->subpq_cond = new QWaitCondition();
is->video_current_pts_time = av_gettime();
packet_queue_init(&audioq);
packet_queue_init(&videoq);
is->audioq = audioq;
is->videoq = videoq;
AVPacket* packet2 = new AVPacket();
ccontext->get_buffer = our_get_buffer;
ccontext->release_buffer = our_release_buffer;
av_init_packet(packet2);
while(inst->keepGoing)
{
if(av_read_frame(context,packet2) < 0 && keepGoing)
{
printf("bufferframe Could not read a frame from stream.\n");
fflush( stdout );
}else {
if(packet2->stream_index == 0) {
packet_queue_put(&videoq, packet2);
} else if(packet2->stream_index == 1) {
packet_queue_put(&audioq, packet2);
} else {
av_free_packet(packet2);
}
if(!videoThreadStarted)
{
videoThreadStarted = true;
QThread* thread = new QThread;
videoThread = new VideoStreamWorker(this);
// Give QThread ownership of Worker Object
videoThread->moveToThread(thread);
connect(videoThread, SIGNAL(error(QString)), this, SLOT(errorHandler(QString)));
QObject::connect(videoThread, SIGNAL(refreshNeeded()), this, SLOT(refreshNeededSlot()));
connect(thread, SIGNAL(started()), videoThread, SLOT(doWork()));
connect(videoThread, SIGNAL(finished()), thread, SLOT(quit()));
connect(videoThread, SIGNAL(finished()), videoThread, SLOT(deleteLater()));
connect(thread, SIGNAL(finished()), thread, SLOT(deleteLater()));
thread->start();
}
if(!audioThreadStarted)
{
audioThreadStarted = true;
QThread* thread = new QThread;
AudioStreamWorker* videoThread = new AudioStreamWorker(this);
// Give QThread ownership of Worker Object
videoThread->moveToThread(thread);
// Connect videoThread error signal to this errorHandler SLOT.
connect(videoThread, SIGNAL(error(QString)), this, SLOT(errorHandler(QString)));
// Connects the thread’s started() signal to the process() slot in the videoThread, causing it to start.
connect(thread, SIGNAL(started()), videoThread, SLOT(doWork()));
connect(videoThread, SIGNAL(finished()), thread, SLOT(quit()));
connect(videoThread, SIGNAL(finished()), videoThread, SLOT(deleteLater()));
// Make sure the thread object is deleted after execution has finished.
connect(thread, SIGNAL(finished()), thread, SLOT(deleteLater()));
thread->start();
}
}
} //finished main loop
int MyApp::video_thread() {
//VideoState *is = (VideoState *)arg;
AVPacket pkt1, *packet = &pkt1;
int len1, frameFinished;
double pts;
pic = avcodec_alloc_frame();
for(;;) {
if(packet_queue_get(&videoq, packet, 1) < 0) {
// means we quit getting packets
break;
}
pts = 0;
global_video_pkt_pts2 = packet->pts;
// Decode video frame
len1 = avcodec_decode_video2(ccontext, pic, &frameFinished, packet);
if(packet->dts == AV_NOPTS_VALUE
&& pic->opaque && *(uint64_t*)pic->opaque != AV_NOPTS_VALUE) {
pts = *(uint64_t *)pic->opaque;
} else if(packet->dts != AV_NOPTS_VALUE) {
pts = packet->dts;
} else {
pts = 0;
}
pts *= av_q2d(is->video_st->time_base);
// Did we get a video frame?
if(frameFinished) {
pts = synchronize_video(is, pic, pts);
actualPts = pts;
refreshSlot();
}
av_free_packet(packet);
}
av_free(pic);
return 0;
}
int MyApp::audio_thread() {
//VideoState *is = (VideoState *)arg;
AVPacket pkt1, *packet = &pkt1;
int len1, frameFinished;
ALuint source;
ALenum format = 0;
// ALuint frequency;
ALenum alError;
ALint val2;
ALuint buffers[NUM_BUFFERS];
int dataSize;
ALCcontext *aContext;
ALCdevice *device;
if (!alutInit(NULL, NULL)) {
// printf(stderr, "init alut error\n");
}
device = alcOpenDevice(NULL);
if (device == NULL) {
// printf(stderr, "device error\n");
}
//Create a context
aContext = alcCreateContext(device, NULL);
alcMakeContextCurrent(aContext);
if(!(aContext)) {
printf("Could not create the OpenAL context!\n");
return 0;
}
alListener3f(AL_POSITION, 0.0f, 0.0f, 0.0f);
//ALenum alError;
if(alGetError() != AL_NO_ERROR) {
cout << "could not create buffers";
cout.flush();
fflush( stdout );
return 0;
}
alGenBuffers(NUM_BUFFERS, buffers);
alGenSources(1, &source);
if(alGetError() != AL_NO_ERROR) {
cout << "after Could not create buffers or the source.\n";
cout.flush( );
return 0;
}
int i;
int indexOfPacket;
double pts;
//double pts;
int n;
for(i = 0; i < NUM_BUFFERS; i++)
{
if(packet_queue_get(&audioq, packet, 1) < 0) {
// means we quit getting packets
break;
}
cout << "streamindex=audio \n";
cout.flush( );
//printf("before decode audio\n");
//fflush( stdout );
// AVPacket *packet = new AVPacket();//malloc(sizeof(AVPacket*));
AVFrame *decodedFrame = NULL;
int gotFrame = 0;
// AVFrame* decodedFrame;
if(!decodedFrame) {
if(!(decodedFrame = avcodec_alloc_frame())) {
cout << "Run out of memory, stop the streaming...\n";
fflush( stdout );
cout.flush();
return -2;
}
} else {
avcodec_get_frame_defaults(decodedFrame);
}
int len = avcodec_decode_audio4(audioc, decodedFrame, &gotFrame, packet);
if(len < 0) {
cout << "Error while decoding.\n";
cout.flush( );
return -3;
}
if(len < 0) {
/* if error, skip frame */
is->audio_pkt_size = 0;
//break;
}
is->audio_pkt_data += len;
is->audio_pkt_size -= len;
pts = is->audio_clock;
// *pts_ptr = pts;
n = 2 * is->audio_st->codec->channels;
is->audio_clock += (double)packet->size/
(double)(n * is->audio_st->codec->sample_rate);
if(gotFrame) {
cout << "got audio frame.\n";
cout.flush( );
// We have a buffer ready, send it
dataSize = av_samples_get_buffer_size(NULL, audioc->channels,
decodedFrame->nb_samples, audioc->sample_fmt, 1);
if(!format) {
if(audioc->sample_fmt == AV_SAMPLE_FMT_U8 ||
audioc->sample_fmt == AV_SAMPLE_FMT_U8P) {
if(audioc->channels == 1) {
format = AL_FORMAT_MONO8;
} else if(audioc->channels == 2) {
format = AL_FORMAT_STEREO8;
}
} else if(audioc->sample_fmt == AV_SAMPLE_FMT_S16 ||
audioc->sample_fmt == AV_SAMPLE_FMT_S16P) {
if(audioc->channels == 1) {
format = AL_FORMAT_MONO16;
} else if(audioc->channels == 2) {
format = AL_FORMAT_STEREO16;
}
}
if(!format) {
cout << "OpenAL can't open this format of sound.\n";
cout.flush( );
return -4;
}
}
printf("albufferdata audio b4.\n");
fflush( stdout );
alBufferData(buffers[i], format, *decodedFrame->data, dataSize, decodedFrame->sample_rate);
cout << "after albufferdata all buffers \n";
cout.flush( );
av_free_packet(packet);
//=av_free(packet);
av_free(decodedFrame);
if((alError = alGetError()) != AL_NO_ERROR) {
printf("Error while buffering.\n");
printAlError(alError);
return -6;
}
}
}
cout << "before quoe buffers \n";
cout.flush();
alSourceQueueBuffers(source, NUM_BUFFERS, buffers);
cout << "before play.\n";
cout.flush();
alSourcePlay(source);
cout << "after play.\n";
cout.flush();
if((alError = alGetError()) != AL_NO_ERROR) {
cout << "error strating stream.\n";
cout.flush();
printAlError(alError);
return 0;
}
// AVPacket *pkt = &is->audio_pkt;
while(keepGoing)
{
while(packet_queue_get(&audioq, packet, 1) >= 0) {
// means we quit getting packets
do {
alGetSourcei(source, AL_BUFFERS_PROCESSED, &val2);
usleep(SLEEP_BUFFERING);
} while(val2 <= 0);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error gettingsource :(\n");
return 1;
}
while(val2--)
{
ALuint buffer;
alSourceUnqueueBuffers(source, 1, &buffer);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error unqueue buffers :(\n");
// return 1;
}
AVFrame *decodedFrame = NULL;
int gotFrame = 0;
// AVFrame* decodedFrame;
if(!decodedFrame) {
if(!(decodedFrame = avcodec_alloc_frame())) {
cout << "Run out of memory, stop the streaming...\n";
//fflush( stdout );
cout.flush();
return -2;
}
} else {
avcodec_get_frame_defaults(decodedFrame);
}
int len = avcodec_decode_audio4(audioc, decodedFrame, &gotFrame, packet);
if(len < 0) {
cout << "Error while decoding.\n";
cout.flush( );
is->audio_pkt_size = 0;
return -3;
}
is->audio_pkt_data += len;
is->audio_pkt_size -= len;
if(packet->size <= 0) {
/* No data yet, get more frames */
//continue;
}
if(gotFrame) {
pts = is->audio_clock;
len = synchronize_audio(is, (int16_t *)is->audio_buf,
packet->size, pts);
is->audio_buf_size = packet->size;
pts = is->audio_clock;
// *pts_ptr = pts;
n = 2 * is->audio_st->codec->channels;
is->audio_clock += (double)packet->size /
(double)(n * is->audio_st->codec->sample_rate);
if(packet->pts != AV_NOPTS_VALUE) {
is->audio_clock = av_q2d(is->audio_st->time_base)*packet->pts;
}
len = av_samples_get_buffer_size(NULL, audioc->channels,
decodedFrame->nb_samples, audioc->sample_fmt, 1);
alBufferData(buffer, format, *decodedFrame->data, len, decodedFrame->sample_rate);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error buffering :(\n");
return 1;
}
alSourceQueueBuffers(source, 1, &buffer);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error queueing buffers :(\n");
return 1;
}
}
}
alGetSourcei(source, AL_SOURCE_STATE, &val2);
if(val2 != AL_PLAYING)
alSourcePlay(source);
}
//pic = avcodec_alloc_frame();
}
qDebug() << "end audiothread";
return 1;
}
void MyApp::refreshSlot()
{
if(true)
{
printf("got frame %d, %d\n", pic->width, ccontext->width);
fflush( stdout );
sws_scale(img_convert_ctx, (const uint8_t **)pic->data, pic->linesize,
0, originalVideoHeight, &picrgb->data[0], &picrgb->linesize[0]);
printf("rescaled frame %d, %d\n", newVideoWidth, newVideoHeight);
fflush( stdout );
//av_free_packet(packet);
//av_init_packet(packet);
qDebug() << "waking audio as video finished";
////mutex.unlock();
//mutex2.lock();
doingVideoFrame = false;
//doingAudioFrame = false;
////mutex2.unlock();
//mutex2.unlock();
//w2->wakeAll();
//w->wakeAll();
qDebug() << "now woke audio";
//pic = picrgb;
uint8_t *srcy = picrgb->data[0];
uint8_t *srcu = picrgb->data[1];
uint8_t *srcv = picrgb->data[2];
printf("got src yuv frame %d\n", &srcy);
fflush( stdout );
unsigned char *ptr = NULL;
screen_get_buffer_property_pv(mScreenPixelBuffer, SCREEN_PROPERTY_POINTER, (void**) &ptr);
unsigned char *y = ptr;
unsigned char *u = y + (newVideoHeight * mStride) ;
unsigned char *v = u + (newVideoHeight * mStride) / 4;
int i = 0;
printf("got buffer picrgbwidth= %d \n", newVideoWidth);
fflush( stdout );
for ( i = 0; i < newVideoHeight; i++)
{
int doff = i * mStride;
int soff = i * picrgb->linesize[0];
memcpy(&y[doff], &srcy[soff], newVideoWidth);
}
for ( i = 0; i < newVideoHeight / 2; i++)
{
int doff = i * mStride / 2;
int soff = i * picrgb->linesize[1];
memcpy(&u[doff], &srcu[soff], newVideoWidth / 2);
}
for ( i = 0; i < newVideoHeight / 2; i++)
{
int doff = i * mStride / 2;
int soff = i * picrgb->linesize[2];
memcpy(&v[doff], &srcv[soff], newVideoWidth / 2);
}
printf("before posttoscreen \n");
fflush( stdout );
video_refresh_timer();
qDebug() << "end refreshslot";
}
else
{
}
}
void MyApp::refreshNeededSlot2()
{
printf("blitting to buffer");
fflush(stdout);
screen_buffer_t screen_buffer;
screen_get_window_property_pv(mScreenWindow, SCREEN_PROPERTY_RENDER_BUFFERS, (void**) &screen_buffer);
int attribs[] = { SCREEN_BLIT_SOURCE_WIDTH, newVideoWidth, SCREEN_BLIT_SOURCE_HEIGHT, newVideoHeight, SCREEN_BLIT_END };
int res2 = screen_blit(mScreenCtx, screen_buffer, mScreenPixelBuffer, attribs);
printf("dirty rectangles");
fflush(stdout);
int dirty_rects[] = { 0, 0, newVideoWidth, newVideoHeight };
screen_post_window(mScreenWindow, screen_buffer, 1, dirty_rects, 0);
printf("done screneposdtwindow");
fflush(stdout);
}
void MyApp::video_refresh_timer() {
testDelay = 0;
// VideoState *is = ( VideoState* )userdata;
VideoPicture *vp;
//double pts = 0 ;
double actual_delay, delay, sync_threshold, ref_clock, diff;
if(is->video_st) {
if(false)////is->pictq_size == 0)
{
testDelay = 1;
schedule_refresh(is, 1);
} else {
// vp = &is->pictq[is->pictq_rindex];
delay = actualPts - is->frame_last_pts; /* the pts from last time */
if(delay <= 0 || delay >= 1.0) {
/* if incorrect delay, use previous one */
delay = is->frame_last_delay;
}
/* save for next time */
is->frame_last_delay = delay;
is->frame_last_pts = actualPts;
is->video_current_pts = actualPts;
is->video_current_pts_time = av_gettime();
/* update delay to sync to audio */
ref_clock = get_audio_clock(is);
diff = actualPts - ref_clock;
/* Skip or repeat the frame. Take delay into account
FFPlay still doesn't "know if this is the best guess." */
sync_threshold = (delay > AV_SYNC_THRESHOLD) ? delay : AV_SYNC_THRESHOLD;
if(fabs(diff) < AV_NOSYNC_THRESHOLD) {
if(diff <= -sync_threshold) {
delay = 0;
} else if(diff >= sync_threshold) {
delay = 2 * delay;
}
}
is->frame_timer += delay;
/* computer the REAL delay */
actual_delay = is->frame_timer - (av_gettime() / 1000000.0);
if(actual_delay < 0.010) {
/* Really it should skip the picture instead */
actual_delay = 0.010;
}
testDelay = (int)(actual_delay * 1000 + 0.5);
schedule_refresh(is, (int)(actual_delay * 1000 + 0.5));
/* show the picture! */
//video_display(is);
// SDL_CondSignal(is->pictq_cond);
// SDL_UnlockMutex(is->pictq_mutex);
}
} else {
testDelay = 100;
schedule_refresh(is, 100);
}
}
void MyApp::schedule_refresh(VideoState *is, int delay) {
qDebug() << "start schedule refresh timer" << delay;
typeOfEvent = FF_REFRESH_EVENT2;
w->wakeAll();
// SDL_AddTimer(delay,
}I am currently waiting on data in a loop in the following way
QMutex mutex;
mutex.lock();
while(keepGoing)
{
qDebug() << "MAINTHREAD" << testDelay;
w->wait(&mutex);
mutex.unlock();
qDebug() << "MAINTHREAD past wait";
if(!keepGoing)
{
break;
}
if(testDelay > 0 && typeOfEvent == FF_REFRESH_EVENT2)
{
usleep(testDelay);
refreshNeededSlot2();
}
else if(testDelay > 0 && typeOfEvent == FF_QUIT_EVENT2)
{
keepGoing = false;
exit(0);
break;
// usleep(testDelay);
// refreshNeededSlot2();
}
qDebug() << "MAINTHREADend";
mutex.lock();
}
mutex.unlock();Please let me know if I need to provide any more relevent code. I'm sorry my code is untidy - I still learning c++ and have been modifying this code for over a week now as previously mentioned.
Just added a sample of output I'm seeing from print outs I do to console - I can't get my head around it (it's almost too complicated for my level of expertise) but when you see the frames being played and audio playing it's very difficult to give up especially when it took me a couple of weeks to get to this stage.
Please someone give me a hand if they spot the problem.
MAINTHREAD past wait
pts after syncvideo= 1073394046
got frame 640, 640
start video_refresh_timer
actualpts = 1.66833
frame lastpts = 1.63497
start schedule refresh timer need to delay for 123pts after syncvideo= 1073429033
got frame 640, 640
MAINTHREAD loop delay before refresh = 123
start video_refresh_timer
actualpts = 1.7017
frame lastpts = 1.66833
start schedule refresh timer need to delay for 115MAINTHREAD past wait
pts after syncvideo= 1073464021
got frame 640, 640
start video_refresh_timer
actualpts = 1.73507
frame lastpts = 1.7017
start schedule refresh timer need to delay for 140MAINTHREAD loop delay before refresh = 140
pts after syncvideo= 1073499008
got frame 640, 640
start video_refresh_timer
actualpts = 1.76843
frame lastpts = 1.73507
start schedule refresh timer need to delay for 163MAINTHREAD past wait
pts after syncvideo= 1073533996
got frame 640, 640
start video_refresh_timer
actualpts = 1.8018
frame lastpts = 1.76843
start schedule refresh timer need to delay for 188MAINTHREAD loop delay before refresh = 188
pts after syncvideo= 1073568983
got frame 640, 640
start video_refresh_timer
actualpts = 1.83517
frame lastpts = 1.8018
start schedule refresh timer need to delay for 246MAINTHREAD past wait
pts after syncvideo= 1073603971
got frame 640, 640
start video_refresh_timer
actualpts = 1.86853
frame lastpts = 1.83517
start schedule refresh timer need to delay for 299MAINTHREAD loop delay before refresh = 299
pts after syncvideo= 1073638958
got frame 640, 640
start video_refresh_timer
actualpts = 1.9019
frame lastpts = 1.86853
start schedule refresh timer need to delay for 358MAINTHREAD past wait
pts after syncvideo= 1073673946
got frame 640, 640
start video_refresh_timer
actualpts = 1.93527
frame lastpts = 1.9019
start schedule refresh timer need to delay for 416MAINTHREAD loop delay before refresh = 416
pts after syncvideo= 1073708933
got frame 640, 640
start video_refresh_timer
actualpts = 1.96863
frame lastpts = 1.93527
start schedule refresh timer need to delay for 474MAINTHREAD past wait
pts after syncvideo= 1073742872
got frame 640, 640
MAINTHREAD loop delay before refresh = 474
start video_refresh_timer
actualpts = 2.002
frame lastpts = 1.96863
start schedule refresh timer need to delay for 518MAINTHREAD past wait
pts after syncvideo= 1073760366
got frame 640, 640
start video_refresh_timer
actualpts = 2.03537
frame lastpts = 2.002
start schedule refresh timer need to delay for 575 -
Linux Media Player Survey Circa 2001
2 septembre 2010, par Multimedia Mike — GeneralHere’s a document I scavenged from my archives. It was dated September 1, 2001 and I now publish it 9 years later. It serves as sort of a time capsule for the state of media player programs at the time. Looking back on this list, I can’t understand why I couldn’t find MPlayer while I was conducting this survey, especially since MPlayer is the project I eventually started to work for a few months after writing this piece.
For a little context, I had been studying multimedia concepts and tech for a year and was itching to get my hands dirty with practical multimedia coding. But I wanted to tackle what I perceived as unsolved problems– like playback of proprietary codecs. I didn’t want to have to build a new media playback framework just to start working on my problems. So I surveyed the players available to see which ones I could plug into and use as a testbed for implementing new decoders.
Regarding Real Player, I wrote : “We’re trying to move away from the proprietary, closed-source “solutions”. Heh. Was I really an insufferable open source idealist back in the day ?
Anyway, here’s the text with some Where are they now ? commentary [in brackets] :
Towards an All-Inclusive Media Playing Solution for Linux
I don’t feel that the media playing solutions for Linux set their sights high enough, even though they do tend to be quite ambitious.
I want to create a media player for Linux that can open a file, figure out what type of file it is (AVI, MOV, etc.), determine the compression algorithms used to encode the audio and video chunks inside (MPEG, Cinepak, Sorenson, etc.) and replay the file using the best audio, video, and CPU facilities available on the computer.
Video and audio playback is a solved problem on Linux ; I don’t wish to solve that problem again. The problem that isn’t solved is reliance on proprietary multimedia solutions through some kind of WINE-like layer in order to decode compressed multimedia files.
Survey of Linux solutions for decoding proprietary multimedia
updated 2001-09-01AVI Player for XMMS
This is based on Avifile. All the same advantages and limitations apply.
[Top Google hit is a Freshmeat page that doesn’t indicate activity since 2001-2002.]Avifile
This player does a great job at taking apart AVI and ASF files and then feeding the compressed chunks of multimedia data through to the binary Win32 decoders.The program is written in C++ and I’m not very good at interpreting that kind of code. But I’m learning all over again. Examining the object hierarchy, it appears that the designers had the foresight to include native support for decoders that are compiled into the program from source code. However, closer examination reveals that there is support for ONE source decoder and that’s the “decoder” for uncompressed data. Still, I tried to manipulate this routine to accept and decode data from other codecs but no dice. It’s really confounding. The program always crashes when I feed non-uncompressed data through the source decoder.
[Lives at http://avifile.sourceforge.net/ ; not updated since 2006.]Real Player
There’s not much to do with this since it is closed source and proprietary. Even though there is a plugin architecture, that’s not satisfactory. We’re trying to move away from the proprietary, closed-source “solutions”.
[Still kickin’ with version 11.]XAnim
This is a well-established Unix media player. To his credit, the author does as well as he can with the resources he has. In other words, he supports the non-proprietary video codecs well, and even has support for some proprietary video codecs through binary-only decoders.The source code is extremely difficult to work with as the author chose to use the X coding format which I’ve never seen used anywhere else except for X header files. The infrastructure for extending the program and supporting other codecs and file formats is there, I suppose, but I would have to wrap my head around the coding style. Maybe I can learn to work past that. The other thing that bothers me about this program is the decoding approach : It seems that each video decoder includes routines to decompress the multimedia data into every conceivable RGB and YUV output format. This seems backwards to me ; it seems better to have one decoder function that decodes the data into its native format it was compressed from (e.g., YV12 for MPEG data) and then pass that data to another layer of the program that’s in charge of presenting the data and possibly converting it if necessary. This layer would encompass highly-optimized software conversion routines including special CPU-specific instructions (e.g., MMX and SSE) and eliminate the need to place those routines in lots of other routines. But I’m getting ahead of myself.
[This one was pretty much dead before I made this survey, the most recent update being in 1999. Still, we owe it much respect as the granddaddy of Unix multimedia playback programs.]Xine
This seems like a promising program. It was originally designed to play MPEGs from DVDs. It can also play MPEG files on a hard drive and utilizes the Xv extensions for hardware YUV playback. It’s also supposed to play AVI files using the same technique as Avifile but I have never, ever gotten it to work. If an AVI file has both video and sound, the binary video decoder can’t decode any frames. If the AVI file has video and no sound, the program gets confused and crashes, as far as I can tell.Still, it’s promising, and I’ve been trying to work around these crashes. It doesn’t yet have the type of modularization I’d like to see. Right now, it tailored to suit MPEG playback and AVI playback is an afterthought. Still, it appears to have a generalized interface for dropping in new file demultiplexers.
I tried to extend the program for supporting source decoders by rewriting w32codec.c from scratch. I’m not having a smooth time of it so far. I’m able to perform some manipulations on the output window. However, I can’t get the program to deal with an RGB image format. It has trouble allocating an RGB surface with XvShmCreateImage(). This isn’t suprising, per my limited knowledge of X which is that Xv applies to YUV images, but it could also apply to RGB images as well. Anyway, the program should be able to fall back on regular RGB pixmaps if that Xv call fails.
Right now, this program is looking the most promising. It will take some work to extend the underlying infrastructure, but it seems doable since I know C quite well and can understand the flow of this program, as opposed to Avifile and its C++. The C code also compiles about 10 times faster.
[My home project for many years after a brief flirtation with MPlayer. It is still alive ; its latest release was just a month ago.]XMovie
This library is a Quicktime movie player. I haven’t looked at it too extensively yet, but I do remember looking at it at one point and reading the documentation that said it doesn’t support key frames. Still, I should examine it again since they released a new version recently.
[Heroine Virtual still puts out some software but XMovie has not been updated since 2005.]XMPS
This program compiles for me, but doesn’t do much else. It can play an MP3 file. I have been able to get MPEG movies to play through it, but it refuses to show the full video frame, constricting it to a small window (obviously a bug).
[This project is hosted on SourceForge and is listed with a registration date of 2003, well after this survey was made. So the project obviously lived elsewhere in 2001. Meanwhile, it doesn’t look like any files ever made it to SF for hosting.]XTheater
I can’t even get this program to compile. It’s supposed to be an MPEG player based on SMPEG. As such, it probably doesn’t hold much promise for being easily extended into a general media player.
[Last updated in 2002.]GMerlin
I can’t get this to compile yet. I have a bug report in to the dev group.
[Updated consistently in the last 9 years. Last update was in February of this year. I can’t find any record of my bug report, though.] -
Video streaming error : Uncaught (in promise) NotSupportedError : Failed to load because no supported source was found
18 septembre 2024, par AizenHere is my problem : I have one video src 1080p (on the frontend). On the frontend, I send this video-route to the backend :


const req = async()=>{try{const res = await axios.get('/catalog/item',{params:{SeriesName:seriesName}});return {data:res.data};}catch(err){console.log(err);return false;}}const fetchedData = await req();-On the backend i return seriesName.Now i can make a full path,what the video is,and where it is,code:


const videoUrl = 'C:/Users/arMori/Desktop/RedditClone/reddit/public/videos';console.log('IT VideoURL',videoUrl);


const selectedFile = `${videoUrl}/${fetchedData.data.VideoSource}/${seriesName}-1080p.mp4`
console.log(`ITS'S SELECTED FILE: ${selectedFile}`);


Ok, I have my src 1080p, now is the time to send it to the backend :


const response = await axios.post('/videoFormat', {videoUrl:selectedFile})console.log('Это консоль лог путей: ',response.data);const videoPaths = response.data;


Backend takes it and FFMpqg makes two types of resolution,720p and 480p,save it to the temp storage on backend, and then returns two routes where these videos stores


async videoUpload(videoUrl:string){try{const tempDir = C:/Users/arMori/Desktop/RedditClone/reddit_back/src/video/temp;const inputFile = videoUrl;console.log('VIDEOURL: ',videoUrl);


const outputFiles = [];
 
 await this.createDirectories(tempDir); 
 outputFiles.push(await this.convertVideo(inputFile, '1280x720', '720p.mp4'));
 outputFiles.push(await this.convertVideo(inputFile, '854x480', '480p.mp4'));
 console.log('OUTUPT FILES SERVICE: ',outputFiles);
 
 return outputFiles;

 }catch(err){
 console.error('VideoFormatterService Error: ',err);
 
 }
}

private convertVideo(inputPath:string,resolution:string,outputFileName:string):Promise<string>{
 const temp = `C:/Users/arMori/Desktop/RedditClone/reddit_back/src/video/temp`;
 return new Promise(async(resolve,reject)=>{
 const height = resolution.split('x')[1];
 console.log('HIEGHT: ',height);
 
 const outputDir = `C:/Users/arMori/Desktop/RedditClone/reddit_back/src/video/temp/${height}p`;
 const outputPath = join(outputDir, outputFileName);
 const isExists = await fs.access(outputPath).then(() => true).catch(() => false);
 if(isExists){ 
 console.log(`File already exists: ${outputPath}`);
 return resolve(outputPath)
 };
 ffmpeg(inputPath)
 .size(`${resolution}`)
 .videoCodec('libx264') // Кодек H.264
 .audioCodec('aac') 
 .output(outputPath)
 .on('end',()=>resolve(outputPath))
 .on('error',(err)=>reject(err))
 .run()
 
 })
}

private async createDirectories(temp:string){
 try{
 const dir720p = `${temp}/720p`;
 const dir480p = `${temp}/480p`;
 const dir720pExists = await fs.access(dir720p).then(() => true).catch(() => false);
 const dir480pExists = await fs.access(dir480p).then(() => true).catch(() => false);
 if(dir720pExists && dir480pExists){
 console.log('FILES ALIVE');
 return;
 }
 if (!dir720pExists) {
 await fs.mkdir(dir720p, { recursive: true });
 console.log('Папка 720p создана');
 }
 
 if (!dir480pExists) {
 await fs.mkdir(dir480p, { recursive: true });
 console.log('Папка 480p создана');
 }
 } catch (err) {
 console.error('Ошибка при создании директорий:', err);
 }
}
</string>

Continue frontentd code :


let videoPath;

if (quality === '720p') {
 videoPath = videoPaths[0];
} else if (quality === '480p') {
 videoPath = videoPaths[1];
}

if (!videoPath) {
 console.error('Video path not found!');
 return;
}

// Получаем видео по его пути
console.log('VIDEOPATH LOG: ',videoPath);
 
const videoRes = await axios.get('/videoFormat/getVideo', { 
 params: { path: videoPath } ,
 headers: { Range: 'bytes=0-' },
 responseType: 'blob'
 });
 console.log('Video fetched: ', videoRes);
 const videoBlob = new Blob([videoRes.data], { type: 'video/mp4' });
 const videoURL = URL.createObjectURL(videoBlob);
 return videoURL;
 /* console.log('Видео успешно загружено:', response.data); */
 } catch (error) {
 console.error('Ошибка при загрузке видео:', error);
 }
}


Here I just choose one of the route and make a new GET request (VideoRes), now in the controller in the backend, I'm trying to do a video streaming :


@Public()
 @Get('/getVideo')
 async getVideo(@Query('path') videoPath:string,@Req() req:Request,@Res() res:Response){
 try {
 console.log('PATH ARGUMENT: ',videoPath);
 console.log('VIDEOPATH IN SERVICE: ',videoPath);
 const videoSize = (await fs.stat(videoPath)).size;
 const CHUNK_SIZE = 10 ** 6;
 const range = req.headers['range'] as string | undefined;
 if (!range) {
 return new ForbiddenException('Range не найденно');
 }
 const start = Number(range.replace(/\D/g,""));
 const end = Math.min(start + CHUNK_SIZE,videoSize - 1);

 const contentLength = end - start + 1;
 const videoStream = fsSync.createReadStream(videoPath, { start, end });
 const headers = {
 'Content-Range':`bytes ${start}-${end}/${videoSize}`,
 'Accept-Ranges':'bytes',
 'Content-Length':contentLength,
 'Content-Type':'video/mp4'
 }
 
 res.writeHead(206,headers);

 // Передаем поток в ответ
 videoStream.pipe(res);
 

 // Если возникнет ошибка при стриминге, логируем ошибку
 videoStream.on('error', (error) => {
 console.error('Ошибка при чтении видео:', error);
 res.status(500).send('Ошибка при чтении видео');
 });
 } catch (error) {
 console.error('Ошибка при обработке запросов:', error);
 return res.status(400).json({ message: 'Ошибка при обработке getVideo запросов' });
 }
 }


Send to the frontend


res.writeHead(206,headers);


In the frontend, I make blob url for video src and return it


const videoBlob = new Blob([videoRes.data], { type: 'video/mp4' });const videoURL = URL.createObjectURL(videoBlob);return videoURL;


And assign src to the video :


useVideo(seriesName,quality).then(src => {
 if (src) {
 console.log('ITS VIDEOLOGISC GOIDA!');
 if(!playRef.current) return;
 
 const oldURL = playRef.current.src;
 if (oldURL && oldURL.startsWith('blob:')) {
 URL.revokeObjectURL(oldURL);
 }
 playRef.current.pause();
 playRef.current.src = '';
 setQuality(quality);
 console.log('SRC: ',src);
 
 playRef.current.src = src;
 playRef.current.load();
 console.log('ITS VIDEOURL GOIDA!');
 togglePlayPause();
 }
 })
 .catch(err => console.error('Failed to fetch video', err));


But the problem is :




Vinland-Saga:1 Uncaught (in promise) NotSupportedError : Failed to load because no supported source was found




And I don't know why...


I tried everything, but I don't understand why src is incorrect..


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