What I want to do is to produce a video composed from a single image, repeated for many frames.
I have tried the below code but it is producing a video file of size 0 bytes.
IplImage *image = cvLoadImage("images/img1.jpg", 1);
CvVideoWriter* writer = cvCreateVideoWriter("Video from Images.flv",
CV_FOURCC('D','I','V','X'), fps, size);
for(int counter=0; counter < 300; counter++)
{
/*The below statement writes the frame one by one to the video ...*/
cvWriteFrame(writer, image);
}
You need to call cvReleaseVideoWriter(CvVideoWriter** writer) at the end.
Had you used the C++ API, the destructor would have taken care of this for you.
Related
Trying to write program which uses libav to extract raw pixel data (~BMP) from arbitrary video. Everything goes well except sws_scale() failing to convert AVFrame to RGB24.
I formulated minimal example of it where AVFrame is being created and initialized with 4 different methods found on internet: https://github.com/SlavMFM/libav_bmp_example - all of them fail in different ways. How can I fix it so sws_scale() does the convertion?
First, don't use avcodec_decode_video2. Use avcodec_send_packet and avcodec_receive_frame
second, Don't call av_frame_get_buffer on source Just allocate it with av_frame_alloc, avcodec_receive_frame will set up the rest
Then allocate a destination frame frame like:
AVFrame* frame = av_frame_alloc();
frame->format = whatever;
frame->width = w;
frame->height = h;
av_frame_get_buffer(frame, 32);
I'm trying to use libavcodec library in FFMpeg to decode then re-encode a h264 video.
I have the decoding part working (rendes to an SDL window fine) but when I try to re-encode the frames I get bad data in the re-encoded videos samples.
Here is a cut down code snippet of my encode logic.
EncodeResponse H264Codec::EncodeFrame(AVFrame* pFrame, StreamCodecContainer* pStreamCodecContainer, AVPacket* pPacket)
{
int result = 0;
result = avcodec_send_frame(pStreamCodecContainer->pEncodingCodecContext, pFrame);
if(result < 0)
{
return EncodeResponse::Fail;
}
while (result >= 0)
{
result = avcodec_receive_packet(pStreamCodecContainer->pEncodingCodecContext, pPacket);
// If the encoder needs more frames to create a packed then return and wait for
// method to be called again upon a new frame been present.
// Else check if we have failed to encode for some reason.
// Else a packet has successfully been returned, then write it to the file.
if (result == AVERROR(EAGAIN) || result == AVERROR_EOF)
{
// Higher level logic, dedcodes next frame from source
// video then calls this method again.
return EncodeResponse::SendNextFrame;
}
else if (result < 0)
{
return EncodeResponse::Fail;
}
else
{
// Prepare packet for muxing.
if (pStreamCodecContainer->codecType == AVMEDIA_TYPE_VIDEO)
{
av_packet_rescale_ts(m_pPacket, pStreamCodecContainer->pEncodingCodecContext->time_base,
m_pDecodingFormatContext->streams[pStreamCodecContainer->streamIndex]->time_base);
}
m_pPacket->stream_index = pStreamCodecContainer->streamIndex;
int result = av_interleaved_write_frame(m_pEncodingFormatContext, m_pPacket);
av_packet_unref(m_pPacket);
}
}
return EncodeResponse::EncoderEndOfFile;
}
Strange behaviour I notice is that before I get the first packet from avcodec_receive_packet I have to send 50+ frames to avcodec_send_frame.
I built a debug build of FFMpeg and stepping into the code I notice that AVERROR(EAGAIN) is returned by avcodec_receive_packet because of the following in x264encoder::encode in encoder.c
if( h->frames.i_input <= h->frames.i_delay + 1 - h->i_thread_frames )
{
/* Nothing yet to encode, waiting for filling of buffers */
pic_out->i_type = X264_TYPE_AUTO;
return 0;
}
For some reason my code-context (h) never has any frames. I have spent a long time trying to debug ffmpeg and to determine what I'm doing wrong. But have reached the limit of my video codec knowledge (which is little).
I'm testing this with a video that has no audio to reduce complication.
I have created a cut down version of my application and provided a self contained (with ffmpeg and SDL built dependencies) project. Hopefully this can help anyone-one willing to help me :).
Project Link
https://github.com/maxhap/video-codec
After looking into encoder initialisation I found that I have to set the codec AV_CODEC_FLAG_GLOBAL_HEADER before calling avcodec_open2
pStreamCodecContainer->pEncodingCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
This change led to the re-encoded moov box looking much heathier (used MP4Box.js to parse it). However, the video still does not play correctly, the output video has grey frames at the start when played in VLC and won't play in other players.
I have since tried creating an encoding context via the sample code, rather than using my decoding codec parameters. This led to fixing the bad/data or encoding issue. However, my DTS times are scaling to huge numbers
Here is my new codec init
if (pStreamCodecContainer->codecType == AVMEDIA_TYPE_VIDEO)
{
pStreamCodecContainer->pEncodingCodecContext->height = pStreamCodecContainer->pDecodingCodecContext->height;
pStreamCodecContainer->pEncodingCodecContext->width = pStreamCodecContainer->pDecodingCodecContext->width;
pStreamCodecContainer->pEncodingCodecContext->sample_aspect_ratio = pStreamCodecContainer->pDecodingCodecContext->sample_aspect_ratio;
/* take first format from list of supported formats */
if (pStreamCodecContainer->pEncodingCodec->pix_fmts)
{
pStreamCodecContainer->pEncodingCodecContext->pix_fmt = pStreamCodecContainer->pEncodingCodec->pix_fmts[0];
}
else
{
pStreamCodecContainer->pEncodingCodecContext->pix_fmt = pStreamCodecContainer->pDecodingCodecContext->pix_fmt;
}
/* video time_base can be set to whatever is handy and supported by encoder */
pStreamCodecContainer->pEncodingCodecContext->time_base = av_inv_q(pStreamCodecContainer->pDecodingCodecContext->framerate);
pStreamCodecContainer->pEncodingCodecContext->sample_aspect_ratio = pStreamCodecContainer->pDecodingCodecContext->sample_aspect_ratio;
}
else
{
pStreamCodecContainer->pEncodingCodecContext->channel_layout = pStreamCodecContainer->pDecodingCodecContext->channel_layout;
pStreamCodecContainer->pEncodingCodecContext->channels =
av_get_channel_layout_nb_channels(pStreamCodecContainer->pEncodingCodecContext->channel_layout);
/* take first format from list of supported formats */
pStreamCodecContainer->pEncodingCodecContext->sample_fmt = pStreamCodecContainer->pEncodingCodec->sample_fmts[0];
pStreamCodecContainer->pEncodingCodecContext->time_base = AVRational{ 1, pStreamCodecContainer->pEncodingCodecContext->sample_rate };
}
Any ideas why my DTS time is re-scaling incorrectly?
I managed to fix the DTS scalling by using the time_base value directly from the decoding streams.
So
pStreamCodecContainer->pEncodingCodecContext->time_base = m_pDecodingFormatContext->streams[pStreamCodecContainer->streamIndex]->time_base
Instead of
pStreamCodecContainer->pEncodingCodecContext->time_base = av_inv_q(pStreamCodecContainer->pDecodingCodecContext->framerate);
I will create an answer based on all my finding.
To fix the initial problem of a corrupted moov box I had to add the AV_CODEC_FLAG_GLOBAL_HEADER flag to the encoding codec context before calling avcodec_open2.
encCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
The next issue was badly scaled DTS values in the encoded package, this was causing a side effect of the final mp4 duration being in the hundreds of hours long. To fix this I had to change the encoding codec context timebase to be that of the decoding context streams timebase. This is different than using av_inv_q(framerate) as suggested in the avcodec transcoding example.
encCodecContext->time_base = decCodecFormatContext->streams[streamIndex]->time_base;
I use packet duration to translate from frame index to pts and back, and I'd like to be sure that this is a reliable method of doing so.
Alternatively, is there a better way to translate pts to a frame index and vice versa?
A snippet showing my usage:
bool seekFrame(int64_t frame)
{
if(frame > container.frameCount)
frame = container.frameCount;
// Seek to a frame behind the desired frame because nextFrame() will also increment the frame index
int64_t seek = pts_cache[frame-1]; // pts_cache is an array of all frame pts values
// get the nearest prior keyframe
int preceedingKeyframe = av_index_search_timestamp(container.video_st, seek, AVSEEK_FLAG_BACKWARD);
// here's where I'm worried that packetDuration isn't a reliable method of translating frame index to
// pts value
int64_t nearestKeyframePts = preceedingKeyframe * container.packetDuration;
avcodec_flush_buffers(container.pCodecCtx);
int ret = av_seek_frame(container.pFormatCtx, container.videoStreamIndex, nearestKeyframePts, AVSEEK_FLAG_ANY);
if(ret < 0) return false;
container.lastPts = nearestKeyframePts;
AVFrame *pFrame = NULL;
while(nextFrame(pFrame, NULL) && container.lastPts < seek)
{
;
}
container.currentFrame = frame-1;
av_free(pFrame);
return true;
}
No, not guaranteed. It may work with some codec/container combination where frame-rate is static. avi, h264 raw (annex-b) and yuv4mpeg come to mind. But other containers like flv, mp4, ts, have a PTS/DTS (or CTS) for EVERY frame. The source could be variable frame rate, or frames could have be dropped at some point during processing due to bandwidth. Also some codecs will remove duplicate frames.
So unless you created the file yourself. Do not trust it. There is no guaranteed way to look at a frame and know its 'index' except start at the beginning and count.
Your method, MAY be good enough for most files however.
I'm using ffms2 (aka FFmpegSource) for decoding video frames and display on UI based on wxWidgets.
My player works fine for low resolution video (320*240, 640*480) but for higher resolution (1080) it is very slow. I'm not able to meed the desired frame for high resolution video.
After time analysis I found that FFMS_GetFrame() frame function takes much longer time for high resolution frame.
Here are the results.
1. 320*240 FFMS_GetFrame takes 4-6ms
2. 640*480 FFMS_GetFrame takes >20ms
3. 1080*720 FFMS_GetFrame takes >40
Which means that I'll never meets 30 fps requirement for 1080p frame with FFMS2. But I'm not sure if this is the case.
Please suggest what could be going wrong.
void SetPosition(int64 pos)
{
uint8_t* data_ptr = NULL;
/*check if position is valid*/
if (!m_track || pos < 0 && pos > m_videoProp->NumFrames - 1)
return; // ERR_POS;
wxMilliClock_t start_wx_t = wxGetLocalTimeMillis();
long long start_t = start_wx_t.GetValue();
m_frameId = pos;
if(m_video)
{
m_frameProp = FFMS_GetFrame(m_video, m_frameId, &m_errInfo);
if(!m_frameProp) return;
if(m_frameProp)
{
m_width_ffms2 = m_frameProp->EncodedWidth;
m_height_ffms2 = m_frameProp->EncodedHeight;
}
wxMilliClock_t end_wx_t = wxGetLocalTimeMillis();
long long end_t = end_wx_t.GetValue();
long long diff_t = end_t - start_t;
wxLogDebug(wxString(wxT("Frame Grabe Millisec") + ToString(diff_t)));
//m_frameInfo = FFMS_GetFrameInfo(m_track, FFMS_TYPE_VIDEO);
/* If you want to change the output colorspace or resize the output frame size, now is the time to do it.
IMPORTANT: This step is also required to prevent resolution and colorspace changes midstream. You can
always tell a frame's original properties by examining the Encoded properties in FFMS_Frame. */
/* A -1 terminated list of the acceptable output formats (see pixfmt.h for the list of pixel formats/colorspaces).
To get the name of a given pixel format, strip the leading PIX_FMT_ and convert to lowercase. For example,
PIX_FMT_YUV420P becomes "yuv420p". */
#if 0
int pixfmt[2];
pixfmt[0] = FFMS_GetPixFmt("bgr24");
pixfmt[1] = -1;
#endif
// FFMS_SetOutputFormatV2 returns 0 on success. It Returns non-0 and sets ErrorMsg on failure.
int failure = FFMS_SetOutputFormatV2(m_video, pixfmt, m_width_ffms2, m_height_ffms2, FFMS_RESIZER_BICUBIC, &m_errInfo);
if (failure)
{
//FFMS_DestroyVideoSource(m_video);
//m_video = NULL;
return; //return ERR_POS;
}
data_ptr = m_frameProp->Data[0];
}
else
{
m_width_ffms2 = 320;
m_height_ffms2 = 240;
}
if(data_ptr)
{
memcpy(m_buf, data_ptr, 3*m_height_ffms2 * m_width_ffms2);
}
else
{
memset(m_buf, 0, 3*m_height_ffms2 * m_width_ffms2);
}
}
Slower video decoding with larger frames is totally normal. 1080x720 has about ten times as many pixels as 320x240, so having GetFrame take about ten times as long is not surprising (it's not a strictly linear relationship as there's a lot of other factors that play into decoding speed, but pixel count and time to decode are fairly correlated).
Setting the output format for every frame is unnecessary and is going to be making things a lot slower. Unless you specifically want the output format to change you should call it just once after opening the video, and it'll apply to all frames requested after that.
Imagine I have H.264 AnxB frames coming in from a real-time conversation. What is the best way to encapsulate in MPEG2 transport stream while maintaining the timing information for subsequent playback?
I am using libavcodec and libavformat libraries. When I obtain pointer to object (*pcc) of type AVCodecContext, I set the foll.
pcc->codec_id = CODEC_ID_H264;
pcc->bit_rate = br;
pcc->width = 640;
pcc->height = 480;
pcc->time_base.num = 1;
pcc->time_base.den = fps;
When I receive NAL units, I create a AVPacket and call av_interleaved_write_frame().
AVPacket pkt;
av_init_packet( &pkt );
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = pst->index;
pkt.data = (uint8_t*)p_NALunit;
pkt.size = len;
pkt.dts = AV_NOPTS_VALUE;
pkt.pts = AV_NOPTS_VALUE;
av_interleaved_write_frame( fc, &pkt );
I basically have two questions:
1) For variable framerate, is there a way to not specify the foll.
pcc->time_base.num = 1;
pcc->time_base.den = fps;
and replace it with something to indicate variable framerate?
2) While submitting packets, what "timestamps" should I assign to
pkt.dts and pkt.pts?
Right now, when I play the output using ffplay it is playing at constant framerate (fps) which I use in the above code.
I also would love to know how to accommodate varying spatial resolution. In the stream that I receive, each keyframe is preceded by SPS and PPS. I know whenever the spatial resolution changes.
IS there a way to not have to specify
pcc->width = 640;
pcc->height = 480;
upfront? In other words, indicate that the spatial resolution can change mid-stream.
Thanks a lot,
Eddie
DTS and PTS are measured in a 90 KHz clock. See ISO 13818 part 1 section 2.4.3.6 way down below the syntax table.
As for the variable frame rate, your framework may or may not have a way to generate this (vui_parameters.fixed_frame_rate_flag=0). Whether the playback software handles it is an ENTIRELY different question. Most players assume a fixed frame rate regardless of PTS or DTS. mplayer can't even compute the frame rate correctly for a fixed-rate transport stream generated by ffmpeg.
I think if you're going to change the resolution you need to end the stream (nal_unit_type 10 or 11) and start a new sequence. It can be in the same transport stream (assuming your client's not too simple).