
Recherche avancée
Médias (91)
-
Spitfire Parade - Crisis
15 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
-
Wired NextMusic
14 mai 2011, par
Mis à jour : Février 2012
Langue : English
Type : Video
-
Video d’abeille en portrait
14 mai 2011, par
Mis à jour : Février 2012
Langue : français
Type : Video
-
Sintel MP4 Surround 5.1 Full
13 mai 2011, par
Mis à jour : Février 2012
Langue : English
Type : Video
-
Carte de Schillerkiez
13 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
-
Publier une image simplement
13 avril 2011, par ,
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (41)
-
(Dés)Activation de fonctionnalités (plugins)
18 février 2011, parPour gérer l’ajout et la suppression de fonctionnalités supplémentaires (ou plugins), MediaSPIP utilise à partir de la version 0.2 SVP.
SVP permet l’activation facile de plugins depuis l’espace de configuration de MediaSPIP.
Pour y accéder, il suffit de se rendre dans l’espace de configuration puis de se rendre sur la page "Gestion des plugins".
MediaSPIP est fourni par défaut avec l’ensemble des plugins dits "compatibles", ils ont été testés et intégrés afin de fonctionner parfaitement avec chaque (...) -
Le plugin : Podcasts.
14 juillet 2010, parLe problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
Types de fichiers supportés dans les flux
Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...) -
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs
Sur d’autres sites (7156)
-
Video created using H263 codec and ffmpeg does not play on android device [closed]
21 mars 2013, par susheel tickooI have created a video using FFmpeg and H263 codec. But when I play the video on an Android device the player is unable to play it. I have used both the extensions .mp4 and .3gp.
void generate(JNIEnv *pEnv, jobject pObj,jobjectArray stringArray,int famerate,int width,int height,jstring videoFilename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
//int framesnum=5;
int i,looper, out_size, size, x, y,encodecbuffsize,j;
__android_log_write(ANDROID_LOG_INFO, "record","************into generate************");
int imagecount= (*pEnv)->GetArrayLength(pEnv, stringArray);
__android_log_write(ANDROID_LOG_INFO, "record","************got magecount************");
int retval=-10;
FILE *f;
AVFrame *picture,*encoded_avframe;
uint8_t *encodedbuffer;
jbyte *raw_record;
char logdatadata[100];
int returnvalue = -1,numBytes =-1;
const char *gVideoFileName = (char *)(*pEnv)->GetStringUTFChars(pEnv, videoFilename, NULL);
__android_log_write(ANDROID_LOG_INFO, "record","************got video file name************");
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(CODEC_ID_H264);
if (!codec) {
__android_log_write(ANDROID_LOG_INFO, "record","codec not found");
exit(1);
}
c= avcodec_alloc_context();
/*c->bit_rate = 400000;
c->width = width;
c->height = height;
c->time_base= (AVRational){1,famerate};
c->gop_size = 12; // emit one intra frame every ten frames
c->max_b_frames=0;
c->pix_fmt = PIX_FMT_YUV420P;
c->codec_type = AVMEDIA_TYPE_VIDEO;
c->codec_id = CODEC_ID_H263;*/
c->bit_rate = 400000;
// resolution must be a multiple of two
c->width = 176;
c->height = 144;
c->pix_fmt = PIX_FMT_YUV420P;
c->qcompress = 0.0;
c->qblur = 0.0;
c->gop_size = 20; //or 1
c->sub_id = 1;
c->workaround_bugs = FF_BUG_AUTODETECT;
//pFFmpeg->c->time_base = (AVRational){1,25};
c->time_base.num = 1;
c->time_base.den = famerate;
c->max_b_frames = 0; //pas de B frame en H263
// c->opaque = opaque;
c->dct_algo = FF_DCT_AUTO;
c->idct_algo = FF_IDCT_AUTO;
//lc->rtp_mode = 0;
c->rtp_payload_size = 1000;
c->rtp_callback = 0; // ffmpeg_rtp_callback;
c->flags |= CODEC_FLAG_QSCALE;
c->mb_decision = FF_MB_DECISION_RD;
c->thread_count = 1;
#define DEFAULT_RATE (16 * 8 * 1024)
c->rc_min_rate = DEFAULT_RATE;
c->rc_max_rate = DEFAULT_RATE;
c->rc_buffer_size = DEFAULT_RATE * 64;
c->bit_rate = DEFAULT_RATE;
sprintf(logdatadata, "------width from c ---- = %d",width);
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
sprintf(logdatadata, "------height from c ---- = %d",height);
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
__android_log_write(ANDROID_LOG_INFO, "record","************Found codec and now opening it************");
/* open it */
retval = avcodec_open(c, codec);
if ( retval < 0)
{
sprintf(logdatadata, "------avcodec_open ---- retval = %d",retval);
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
__android_log_write(ANDROID_LOG_INFO, "record","could not open codec");
exit(1);
}
__android_log_write(ANDROID_LOG_INFO, "record","statement 5");
f = fopen(gVideoFileName, "ab");
if (!f) {
__android_log_write(ANDROID_LOG_INFO, "record","could not open video file");
exit(1);
}
__android_log_write(ANDROID_LOG_INFO, "record", "***************Allocating encodedbuffer*********\n");
encodecbuffsize = avpicture_get_size(PIX_FMT_RGB24, c->width, c->height);
sprintf(logdatadata, "encodecbuffsize = %d",encodecbuffsize);
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
encodedbuffer = malloc(encodecbuffsize);
jclass cls = (*pEnv)->FindClass(pEnv, "com/canvasm/mediclinic/VideoGenerator");
jmethodID mid = (*pEnv)->GetMethodID(pEnv, cls, "videoProgress", "(Ljava/lang/String;)Ljava/lang/String;");
jmethodID mid_delete = (*pEnv)->GetMethodID(pEnv, cls, "deleteTempFile", "(Ljava/lang/String;)Ljava/lang/String;");
if (mid == 0)
return;
__android_log_write(ANDROID_LOG_INFO, "native","got method id");
for(i=0;i<=imagecount;i++) {
jboolean isCp;
int progress = 0;
float temp;
jstring string;
if(i==imagecount)
string = (jstring) (*pEnv)->GetObjectArrayElement(pEnv, stringArray, imagecount-1);
else
string = (jstring) (*pEnv)->GetObjectArrayElement(pEnv, stringArray, i);
const char *rawString = (*pEnv)->GetStringUTFChars(pEnv, string, &isCp);
__android_log_write(ANDROID_LOG_INFO, "record",rawString);
picture = OpenImage(rawString,width,height);
//WriteJPEG(c,picture,i);
// encode video
memset(encodedbuffer,0,encodecbuffsize);
//do{
for(looper=0;looper<5;looper++)
{
memset(encodedbuffer,0,encodecbuffsize);
out_size = avcodec_encode_video(c, encodedbuffer, encodecbuffsize, picture);
sprintf(logdatadata, "avcodec_encode_video ----- out_size = %d \n",out_size );
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
if(out_size>0)
break;
}
__android_log_write(ANDROID_LOG_INFO, "record","*************Start looping for same image*******");
returnvalue = fwrite(encodedbuffer, 1, out_size, f);
sprintf(logdatadata, "fwrite ----- returnvalue = %d \n",returnvalue );
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
__android_log_write(ANDROID_LOG_INFO, "record","*************End looping for same image*******");
// publishing progress
progress = ((i*100)/(imagecount+1))+15;//+1 is for last frame duplicated entry
if(progress<20 )
progress =20;
if(progress>=95 )
progress =95;
sprintf(logdatadata, "%d",progress );
jstring jstrBuf = (*pEnv)->NewStringUTF(pEnv, logdatadata);
(*pEnv)->CallObjectMethod(pEnv, pObj, mid,jstrBuf);
if(i>0)
(*pEnv)->CallObjectMethod(pEnv, pObj, mid_delete,string);
}
/* get the delayed frames */
for(; out_size; i++) {
fflush(stdout);
out_size = avcodec_encode_video(c, encodedbuffer, encodecbuffsize, NULL);
fwrite(encodedbuffer, 20, out_size, f);
}
/* add sequence end code to have a real mpeg file */
encodedbuffer[0] = 0x00;
encodedbuffer[1] = 0x00;
encodedbuffer[2] = 0x01;
encodedbuffer[3] = 0xb7;
fwrite(encodedbuffer, 1, 4, f);
fclose(f);
free(encodedbuffer);
avcodec_close(c);
av_free(c);
__android_log_write(ANDROID_LOG_INFO, "record","Video created ");
// last updation of 100%
sprintf(logdatadata, "%d",100 );
jstring jstrBuf = (*pEnv)->NewStringUTF(pEnv, logdatadata);
(*pEnv)->CallObjectMethod(pEnv, pObj, mid,jstrBuf);
}
AVFrame* OpenImage(const char* imageFileName,int w,int h)
{
AVFrame *pFrame;
AVCodec *pCodec ;
AVFormatContext *pFormatCtx;
AVCodecContext *pCodecCtx;
uint8_t *buffer;
int frameFinished,framesNumber = 0,retval = -1,numBytes=0;
AVPacket packet;
char logdatadata[100];
//__android_log_write(ANDROID_LOG_INFO, "OpenImage",imageFileName);
if(av_open_input_file(&pFormatCtx, imageFileName, NULL, 0, NULL)!=0)
//if(avformat_open_input(&pFormatCtx,imageFileName,NULL,NULL)!=0)
{
__android_log_write(ANDROID_LOG_INFO, "record",
"Can't open image file ");
return NULL;
}
pCodecCtx = pFormatCtx->streams[0]->codec;
pCodecCtx->width = w;
pCodecCtx->height = h;
pCodecCtx->pix_fmt = PIX_FMT_YUV420P;
// Find the decoder for the video stream
pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
if (!pCodec)
{
__android_log_write(ANDROID_LOG_INFO, "record",
"Can't open image file ");
return NULL;
}
pFrame = avcodec_alloc_frame();
numBytes = avpicture_get_size(PIX_FMT_YUV420P, pCodecCtx->width, pCodecCtx->height);
buffer = (uint8_t *) av_malloc(numBytes * sizeof(uint8_t));
sprintf(logdatadata, "numBytes = %d",numBytes);
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
retval = avpicture_fill((AVPicture *) pFrame, buffer, PIX_FMT_YUV420P, pCodecCtx->width, pCodecCtx->height);
// Open codec
if(avcodec_open(pCodecCtx, pCodec)<0)
{
__android_log_write(ANDROID_LOG_INFO, "record","Could not open codec");
return NULL;
}
if (!pFrame)
{
__android_log_write(ANDROID_LOG_INFO, "record","Can't allocate memory for AVFrame\n");
return NULL;
}
int readval = -5;
while (readval = av_read_frame(pFormatCtx, &packet) >= 0)
{
if(packet.stream_index != 0)
continue;
int ret = avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet);
sprintf(logdatadata, "avcodec_decode_video2 ret = %d",ret);
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
if (ret > 0)
{
__android_log_write(ANDROID_LOG_INFO, "record","Frame is decoded\n");
pFrame->quality = 4;
av_free_packet(&packet);
av_close_input_file(pFormatCtx);
return pFrame;
}
else
{
__android_log_write(ANDROID_LOG_INFO, "record","error while decoding frame \n");
}
}
sprintf(logdatadata, "readval = %d",readval);
__android_log_write(ANDROID_LOG_INFO, "record",logdatadata);
}The
generate
method takes a list of strings (path to images) and converts them to video and theOpenImage
method is responsible for convertign a single image toAVFrame
. -
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 -
Why does FFMPEG always make large WebM files ?
2 avril 2013, par Student of HogwartsI'm trying to encode my movies into WebM :
ffmpeg -i input.MOV -codec:v libvpx -quality good -cpu-used 0 -b:v 10k
-qmin 10 -qmax 42 -maxrate 10k -bufsize 20k -threads 8 -vf scale=-1:1080
-codec:a libvorbis -b:a 192k
output.webmI want to encode at a couple of different bit rates (video and audio combined) :
- 2192 kbps
- 1692 kbps
- 1000 kbps
The problem is that no matter which bit rates I enter, I always get a file with a bit rate higher than 1900 kbps. (1914 kbps with the code example above.)
What am I doing wrong ?