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Médias (16)
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#7 Ambience
16 octobre 2011, par
Mis à jour : Juin 2015
Langue : English
Type : Audio
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#6 Teaser Music
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#5 End Title
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#3 The Safest Place
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#4 Emo Creates
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#2 Typewriter Dance
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
Autres articles (1)
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Déploiements possibles
31 janvier 2010, parDeux types de déploiements sont envisageable dépendant de deux aspects : La méthode d’installation envisagée (en standalone ou en ferme) ; Le nombre d’encodages journaliers et la fréquentation envisagés ;
L’encodage de vidéos est un processus lourd consommant énormément de ressources système (CPU et RAM), il est nécessaire de prendre tout cela en considération. Ce système n’est donc possible que sur un ou plusieurs serveurs dédiés.
Version mono serveur
La version mono serveur consiste à n’utiliser qu’une (...)
Sur d’autres sites (1483)
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ffmpeg c++/cli wrapper for using in c# . AccessViolationException after call dll function by it's pointer
25 juillet 2015, par skynet_vMy target is to write a c++/cli wrap arount ffmpeg library, using by importing ffmpeg functions from dll-modules.
Later I will use this interface in c#.
This is my challenge, don’t ask me why))So i’ve implemented Wrap class, which is listed below :
namespace FFMpegWrapLib
{
public class Wrap
{
private:
public:
//wstring libavcodecDllName = "avcodec-56.dll";
//wstring libavformatDllName = "avformat-56.dll";
//wstring libswscaleDllName = "swscale-3.dll";
//wstring libavutilDllName = "avutil-54.dll";
HMODULE libavcodecDLL;
HMODULE libavformatDLL;
HMODULE libswsscaleDLL;
HMODULE libavutilDLL;
AVFormatContext **pFormatCtx = nullptr;
AVCodecContext *pCodecCtxOrig = nullptr;
AVCodecContext *pCodecCtx = nullptr;
AVCodec **pCodec = nullptr;
AVFrame **pFrame = nullptr;
AVFrame **pFrameRGB = nullptr;
AVPacket *packet = nullptr;
int *frameFinished;
int numBytes;
uint8_t *buffer = nullptr;
struct SwsContext *sws_ctx = nullptr;
void Init();
void AVRegisterAll();
void Release();
bool SaveFrame(const char *pFileName, AVFrame * frame, int w, int h);
bool GetStreamInfo();
int FindVideoStream();
bool OpenInput(const char* file);
AVCodec* FindDecoder();
AVCodecContext* AllocContext3();
bool CopyContext();
bool OpenCodec2();
AVFrame* AllocFrame();
int PictureGetSize();
void* Alloc(size_t size);
int PictureFill(AVPicture *, const uint8_t *, enum AVPixelFormat, int, int);
SwsContext* GetSwsContext(int, int, enum AVPixelFormat, int, int, enum AVPixelFormat, int, SwsFilter *, SwsFilter *, const double *);
int ReadFrame(AVFormatContext *s, AVPacket *pkt);
int DecodeVideo2(AVCodecContext *avctx, AVFrame *picture, int *got_picture_ptr, const AVPacket *avpkt);
int SwsScale(struct SwsContext *c, const uint8_t *const srcSlice[], const int srcStride[], int srcSliceY, int srcSliceH, uint8_t *const dst[], const int dstStride[]);
void PacketFree(AVPacket *pkt);
void BufferFree(void *ptr);
void FrameFree(AVFrame **frame);
int CodecClose(AVCodecContext *);
void CloseInput(AVFormatContext **);
bool SeekFrame(AVFormatContext *s, int stream_index, int64_t timestamp, int flags);
Wrap();
~Wrap();
bool GetVideoFrame(char* str_in_file, char* str_out_img, uint64_t time);
};
public ref class managedWrap
{
public:
managedWrap(){}
~managedWrap(){ delete unmanagedWrap; }
bool GetVideoFrameToFile(char* str_in_file, char* str_out_img, uint64_t time)
{
return unmanagedWrap->GetVideoFrame(str_in_file, str_out_img, time);
}
static Wrap* unmanagedWrap = new Wrap();
};
}So the imports to libavcodec and etc. are succesful.
The problem is in AccessViolationException during calling dll func, for example, in OpenInput (i.e. av_open_input in native ffmpeg library)The OpenInput func code is below :
bool FFMpegWrapLib::Wrap::OpenInput(const char* file)
{
typedef int avformat_open_input(AVFormatContext **, const char *, AVInputFormat *, AVDictionary **);
avformat_open_input* pavformat_open_input = (avformat_open_input *)GetProcAddress(libavformatDLL, "avformat_open_input");
if (pavformat_open_input == nullptr)
{
throw exception("Unable to find avformat_open_input function address in libavformat module");
return false;
}
//pin_ptr<avformatcontext> pinFormatContext = &(new interior_ptr<avformatcontext>(pCodecCtx));
pFormatCtx = new AVFormatContext*;
//*pFormatCtx = new AVFormatContext;
int ret = pavformat_open_input(pFormatCtx, file, NULL, NULL); // here it fails
return ret == 0;
}
</avformatcontext></avformatcontext>So the problem, i think, is that class-fields of Wrap class are in secure memory. And ffmpeg works with native memory, initialising pFormatCtx variable by it’s address.
Can I avoid this, or it is impossible ? -
Mix video and audio to mp4 file with ffmpeg but audio does't keep step with video when playback
28 juillet 2015, par dragonflyI managed to write a program to record video(
h264
/aac
) on android with ffmpeg. The detail is as follows :-
Implement
android.hardware.Camera.PreviewCallback
to capture every frame from camera (yuv image
) and send it to the ffmpeg in the jni layer.@Override
public void onPreviewFrame(byte[] data, Camera camera) {
// Log.d(TAG, "onPreviewFrame");
if (mIsRecording) {
// Log.d(TAG, "handlePreviewFrame");
Parameters param = camera.getParameters();
Size s = param.getPreviewSize();
handlePreviewFrame(data, s.width, s.height, mBufSize);
}
camera.addCallbackBuffer(mPreviewBuffer);
}
private void handlePreviewFrame(byte[] data, int width, int height, int size) {
if (mFormats == ImageFormat.NV21) {
//process the yuv data
}
synchronized (mMuxer) {
//jni api
mMuxer.putAudioVisualData(mYuvData, size, 0);
}
} -
Use
android.media.AudioRecord
to read the pcm data from the microphone, write pcm data to ffmpeg in the jni layer in a loop.while (this.isRecording) {
int ret = audioRecord.read(tempBuffer, 0, 1024);
if (ret == AudioRecord.ERROR_INVALID_OPERATION) {
throw new IllegalStateException(
"read() returned AudioRecord.ERROR_INVALID_OPERATION");
} else if (ret == AudioRecord.ERROR_BAD_VALUE) {
throw new IllegalStateException("read() returned AudioRecord.ERROR_BAD_VALUE");
} else if (ret == AudioRecord.ERROR_INVALID_OPERATION) {
throw new IllegalStateException(
"read() returned AudioRecord.ERROR_INVALID_OPERATION");
}
// 处理数据
handleAudioData(tempBuffer, ret);
}
private void handleAudioData(short[] data, int size)
{
// convert to byte[]
//Log.d("VideoCaptureActivity", "handleAudioData");
ByteBuffer buffer = ByteBuffer.allocate(data.length * 2);
buffer.order(ByteOrder.LITTLE_ENDIAN);
buffer.asShortBuffer().put(data);
buffer.limit(size * 2);
byte[] bytes = buffer.array();
synchronized (muxing) {
Log.d(TAG, "putAudio Data :" + size*2);
muxing.putAudioVisualData(bytes, size * 2, 1);
}
} -
mix audio and video data in the jni layer. I refer to the example : https://ffmpeg.org/doxygen/trunk/muxing_8c-source.html
The problem is that the example demonstrates audio and video encoding from some dummy source data generated on the fly. I need to encode audio from microphone and video from camera.
I think the reason of my failure is that the pts in the expample is not applicable for my situation. my av function code is as follows :
static int write_video_frame(AVFormatContext *oc, OutputStream *ost, char *data,
int size) {
int ret;
AVCodecContext *c;
int got_packet = 0;
c = ost->st->codec;
AVPacket pkt = { 0 };
av_init_packet(&pkt);
if (!video_st.hwcodec) {
if (ost->zoom) {
zoom(oc, ost, data);
} else {
avpicture_fill((AVPicture*) ost->frame, (const uint8_t *) data,
c->pix_fmt, c->width, c->height);
}
av_frame_make_writable(ost->frame);
//ost->frame->pts = ost->next_pts++;
ost->frame->pts = frame_count;
/* encode the image */
//ALOGI("avcodec_encode_video2 start");
ret = avcodec_encode_video2(c, &pkt, ost->frame, &got_packet);
//ALOGI("avcodec_encode_video2 end");
if (ret < 0) {
ALOGE("Error encoding video frame: %s", av_err2str(ret));
return -1;
}
} else {
if (size != 0) {
pkt.data = (uint8_t *) data;
pkt.size = size;
pkt.pts = pkt.dts = ost->next_pts++;
got_packet = 1;
}
}
if (got_packet) {
//ALOGI("video write_frame start");
//pkt.pts = (int) timestamp;
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
//ALOGI("video write_frame end");
if (ret < 0) {
ALOGE("Error while writing video frame: %s", av_err2str(ret));
return -1;
}
}
frame_count++;
return 0;
}
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost, char *data) {
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->st->codec;
if (audio_st.speex_echo_cancellation == 1
&& g_audio_echo_play_queue->start_flag == 1) {
//ALOGI("encode_audio_handler in echo_cancel");
QUEUE_ITEM* item = Get_Queue_Item(g_audio_echo_play_queue);
if (item) {
speex_dsp_echo_play_back((spx_int16_t *) item->data);
//ALOGI("encode_audio_handler echo_play begin speex_echo_play_back");
short *echo_processed = (short *) av_malloc(160 * sizeof(short));
speex_dsp_echo_capture((spx_int16_t *) data, echo_processed);
memcpy(data, (uint8_t *) echo_processed, 160);
av_free(echo_processed);
Free_Queue_Item(item, 1);
}
}
frame = ost->tmp_frame;
//update pts
//frame->pts = ost->next_pts;
//ost->next_pts += frame->nb_samples;
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(
swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
memcpy(frame->data[0], data, frame->nb_samples * 2);
//frame->data[0] = data;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0) {
ALOGE("write_audio_frame av_frame_make_writable ERROR %s",
av_err2str(ret));
return -1;
}
/* convert to destination format */
ret = swr_convert(ost->swr_ctx, ost->frame->data, dst_nb_samples,
(const uint8_t **) frame->data, frame->nb_samples);
if (ret < 0) {
ALOGI("Error while converting %s", av_err2str(ret));
return -1;
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count,
(AVRational ) { 1, c->sample_rate }, c->time_base);
ost->samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
ALOGE("Error encoding audio frame: %s", av_err2str(ret));
return -1;
}
if (got_packet) {
//pkt.pts = (int) timestamp;
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
ALOGE("Error while writing audio frame: %s", av_err2str(ret));
return -1;
}
}
return (frame || got_packet) ? 0 : 1;
}How do I deal with the pts of video and audio stream for my situation ? Who can give me some advice ?
Can I ignore the pts provided by ffmpeg and calculate the pts in the java layer by myself and transmit it to ffmpeg ?
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Flv stream to sockets with ffmpeg, node.js and socket.io
7 août 2015, par Dan-Levi Tømtaso i am having some problems with understanding how to pull this off. I know that streaming video is a topic that is hard, and that there is a lot to take account off, but anyway here we go to start of learning how to stream video.
I am using SocketIoClientDotNet as the node.js client for the c# application.
I am sending byte arrays of the video to node which are creating a temporary file and appending the buffer to it. I have tried to set the source of the video element to that file but it doesnt read it as video and are all black. I have tried to download a copy of the file since it did not work and it turns out vlc cant play it either. Ok to the code :
C#
bool ffWorkerIsWorking = false;
private void btnFFMpeg_Click(object sender, RoutedEventArgs e)
{
BackgroundWorker ffWorker = new BackgroundWorker();
ffWorker.WorkerSupportsCancellation = true;
ffWorker.DoWork += ((ffWorkerObj,ffWorkerEventArgs) =>
{
ffWorkerIsWorking = true;
using (var FFProcess = new Process())
{
var processStartInfo = new ProcessStartInfo
{
FileName = "ffmpeg.exe",
RedirectStandardInput = true,
RedirectStandardOutput = true,
UseShellExecute = false,
CreateNoWindow = true,
Arguments = " -loglevel panic -hide_banner -y -f gdigrab -draw_mouse 1 -i desktop -f flv -"
};
FFProcess.StartInfo = processStartInfo;
FFProcess.Start();
byte[] buffer = new byte[32768];
using (MemoryStream ms = new MemoryStream())
{
while (!FFProcess.HasExited)
{
int read = FFProcess.StandardOutput.BaseStream.Read(buffer, 0, buffer.Length);
if (read <= 0)
break;
ms.Write(buffer, 0, read);
clientSocket.Emit("video", ms.ToArray());
ms.Flush();
if (!ffWorkerIsWorking)
{
ffWorker.CancelAsync();
break;
}
}
}
}
});
ffWorker.RunWorkerAsync();
}JS (server)
var buffer = new Buffer(32768);
var isBuffering = false;
var wstream;
socket.on('video', function(data) {
if(!isBuffering){
wstream = fs.createWriteStream('fooTest.flv');
isBuffering = true;
}
buffer = Buffer.concat([buffer, data]);
fs.appendFile('public/fooTest.flv', buffer, function (err) {
if (err) throw err;
console.log('The "data to append" was appended to file!');
});
});What am i doing wrong here ?