First problem is with audio rescaling. I'm trying to redo doc/examples/transcode_aac.c so that it also resamples from 41100 to 48000, it contained a warning that it can't do it.
Using doc/examples/resampling_audio.c as a reference, I saw that before doing swr_convert, I need to find the number of audio samples at the output with the code like this:
int dst_nb_samples = av_rescale_rnd( input_frame->nb_samples + swr_get_delay(resampler_context, 41100),
48000, 41100, AV_ROUND_UP);
Problem is, when I just set int dst_nb_samples = input_frame->nb_samples (which is 1024), it encodes and plays normally, but when I do that av_rescale_rnd thing (which results in 1196), audio is slowed down and distorted, like there are skips in the audio.
Second problem is with trying to mux webm with opus audio.
When I set AVStream->time_base to 1/48000, and increase AVFrame->pts by 960, the resulted file is played in the player as a file that is much bigger. 17 seconds audio shows as 16m11s audio, but it plays normally.
When I increase pts by 20, it displays normally, but has a lot of [libopus # 00ffa660] Queue input is backward in time messages during the encoding. Same for pts 30, still has those messages.
Should I try time_scale 1/1000? webm always have timecodes in milliseconds, and opus have packet size of 20ms (960 samples at 48000 Hz).
Search for pts += 20;
Here is the whole file, all modification I did are marked with //MINE: http://www.mediafire.com/file/jlgo7x4hiz7bw64/transcode_aac.c
Here is the file I tested it on http://www.mediafire.com/file/zdy0zarlqw3qn6s/480P_600K_71149981_soundonly.mkv
The easiest way to achieve that is by using swr_convert_frame which take a frame and resample it to a completely different one.
You can read more about it here: https://ffmpeg.org/doxygen/3.2/swresample_8h_source.html
dst_nb_samples can be calculated as this:
dst_nb_samples = 48000.0 / audio_stream->codec->sample_rate * inputAudioFrame->nb_samples;
Yours probably correct too, I didn't check, but this one I used before, confirm with yours but the number you gave check out. So real problem is probably somewhere else. Try to supply 960 samples in sync with video frames, to do this you need to store audio frames to an additional liner buffer. See if problem fixes.
And/or:
2ndly my experiences says audio pts increase as number of samples per frame (i.e. 960 for 50fps video for 48000hz (48000/50)), not by ms. If you supply 1196 samples, use pts += 1196 (if not used additional buffer I mentioned above). This is different then video frame pts. Hope that helps.
You are definitely in right path. I'll examine the source code if I have time. Anyway hope that helps.
Related
I'm new to FFmpeg. When learn it with the nice repo(https://github.com/leandromoreira/ffmpeg-libav-tutorial)，in the hello_world example I find avcodec_receive_frame dosen't return the first I frame until it gets the third packet, as following screenshot shows:
I'm wondering why additional packets are needed to receive an I frame.
Most modern video codecs are using I/P/B frames which brings the decoding time stamp (DTS) and presentation time stamp (PTS). So, what hello_world does with ffmpeg's lib is the following:-
Demuxing (av_read_frame)
Demuxes packets based on file format (mp4/avi/mkv etc.) until you have a packet for the stream that you want (eg. video) - (We might could say NAL units as an example here - not sure)
Feeds the decoder with the packet (avcodec_send_packet)
Starts the decoding process until it has enough packets to give you the first frame (decodes based on DTS)
Checks whether a frame is ready to be presented (avcodec_receive_frame)
Asks the decoder if it has a frame to be presented after feeding it. It might not be ready and you need to re-feed it or even it might give you more than 1 frames at once. (Frames comes out based on PTS)
I have a set of bare MP3 files. Bare as in I removed all tags (no ID3, no Xing, no Info) from those files.
Just before sending one of these files to the client, I want to add an Info tag. All of my files are CBR so we will use an Info tag (no Xing).
Right now I get the first 4 bytes of the existing MP3 to get the Version (MPEG-1, Layer III), Bitrate, Frequency, Stereo Mode, etc. and thus determine the size of one frame. I create the tag that way, reusing these 4 bytes for the Info tag and determining the size of the frame.
For those wondering, these 4 bytes may look like this:
FF FB 78 04
To me it felt like you are expected to use the exact same first 4 bytes in the Info tag as found in the other audio frames of the MP3, but when using ffmpeg, they stick an Info tag with a hard coded header (wrong bitrate, wrong frequency, etc.)
My question is: Is ffmpeg really doing it right? (LAME doesn't do that) Could I do the same, skipping the load of the first 4 bytes and still have the greater majority of the players out there play my files as expected?
Note: since I read these 4 bytes over the network, it would definitely save a lot of time and some bandwidth to not have to load these 4 bytes on a HEAD request. Resources I could use for the GET requests instead...
The reason for the difference is that with certain configurations, the size of a frame is less than 192 bytes. In that case, the full Info/Xing tag will not fit (and from what I can see, the four optional fields are always included, so an Info/Xing tag is always full even if not required to be).
So, for example, if you have a single channel with 44.1kHz data at 32kbps, the MP3 frame is 117 or 118 bytes. This is less than what is necessary to save the Info/Xing tag.
What LAME does in that situation is forfeit the Info/Xing tag. It's not going to be seen anywhere in the file.
On the other hand, what FFMPEG does is create a frame with a higher bitrate. So instead of 32kbps, it will try with 48kbps and then 64kbps. Once it finds a configuration which offers a frame large enough to support the Info/Xing tag, it stops. (I have not looked at the code, so how FFMPEG really finds a large enough frame, I do not know, but on my end I just incremented the bitrate index field by one until frame size >= 192 and it works).
You can replicate the feat by first creating (or converting) a WAVE file at 44.1kHz using a 32kbps bitrate then try to convert it to MP3 using ffmpeg and see that the Info/Xing tag has a different bitrate.
I'm trying to make a simple CLI program that parses a SRT subtitle file and creates a new one, editing the timestamps to fit the desired framerate.
Eg I have a one-hour video track that runs at 25.0fps, with proper subtitles.
When encoding the same video at 23.976fps, the output video is a few seconds shorter (3 seconds approximately)
I've tried applying the following cross product to each time value in my srt file :
timestamp = timestamp * outputfps / inputfps
This produces captions that are approx. 3 minutes earlier compared to the input SRT (for the last captions - for the first ones the delay is obviously lesser), where the maximum delay should be 3 seconds, according to the new video file length.
This is all new for me and it seems obvious that something's wrong with the way I convert these timestamps. Could you please highlight my mistake?
Edit : According to j_random_hacker clever answer, the video should have the same duration at 25 than at 12 fps, which is easily verified. Seems like the 3 seconds offset I have is there no matter what the output framerate is - I guess there's some sort of trimming happening back there.
The main question remains : how does one convert a subtitle track so it doesn't go out of sync as the video file plays? (see my own comment below if this is unclear)
I'm using FFmpegFrameRecorder to create mp4(H264) video from camera preview. My recorder configuration is as follows.
recorder = new FFmpegFrameRecorder(filePath, width, height);
recorder.setVideoCodec(avcodec.AV_CODEC_ID_H264);
recorder.setFormat("mp4");
recorder.setFrameRate(VIDEO_FPS);
recorder.setVideoBitrate(16384);
recorder.setPixelFormat(avutil.AV_PIX_FMT_YUV420P);
For the rest I follows closely to the sample code RecordActivity.java and was able to verify that
recorder.record(yuvIplimage)
gets called 20 (or more) times, which should create an mp4 with 20 frames. However, the resulting mp4 files after open up only has 2 frames (two first frame of the preview)! I have no idea what have caused such behavior. Any help would be greatly appreciate. Thank you.
Long Le
I figured it out: the issue was because I didn't know what I was doing. I was new to javacv, and I was assuming, based on this stackoverflow entry, that the number of frames in the resulting video should be equal to the number of record() calls. However, this is not the case with video encoding, especially with H264. I figured this out by trying with MPEG4 encoding and there's definitely more than 2 frames. H264 seems to require a minimum number of input frames and hence is not suitable for short (<1 minute) video clips generation (which is my application). One solution is to switch to MPEG4 encoding. However, most browser that does play .mp4 files does not support MPEG4 encoding. Another the solution is to use H264 with minimize compression, by adding the following configuration
recorder.setVideoQuality(0); // maximum quality, replace recorder.setVideoBitrate(16384);
recorder.setVideoOption("preset", "veryfast"); // or ultrafast or fast, etc.
Problem:
omxplayer's source code calls the ffmpeg av_seek_frame() method using the AVSEEK_FLAG_BACKWARD flag. Although not 100% sure, I believe this seeks to the closest i-frame. Instead, I want to seek to exact locations, so I modified the source code such that the av_seek_frame() method now uses the AVSEEK_FLAG_ANY flag. Now, when the movie loads, I get a grey screen, generally for 1 second, during which I can hear the audio. I have tried this on multiple computers (I am actually synchronizing them, therefore, at the same time too) so it is not a n isolated incident. My guess is that seeking to non i-frames is computationally more expensive, resulting in the initial grey screen.
Question: How, using ffmpeg, can I instruct the audio to wait until the video is ready before proceeding.
Actually, AVSEEK_FLAG_BACKWARD indicates that you want to find closest keyframe having a smaller timestamp than the one you are seeking.
By using AVSEEK_FLAG_ANY, you get the frame that corresponds exactly to the timestamp you asked for. But this frame might not be a keyframe, which means that it cannot be fully decoded. That explains your "grey screen", that appears until the next keyframe is reached.
The solution would therefore be to seek backward using AVSEEK_FLAG_BACKWARD and, from this keyframe, read the next frames (e.g. using av_read_frame()) until you get to the one corresponding to your timestamp. At this point, your frame would be fully decoded, and would not appear as a "grey screen" anymore.
NOTE: It appears that, for some reason, av_seek_frame() using AVSEEK_FLAG_BACKWARD returns the next keyframe when the frame that I am seeking is the one directly before this keyframe. Otherwise it returns the previous keyframe (which is what I want). My solution is to change the timestamp I give to av_seek_frame() to ensure that it will return the keyframe before the frame I am seeking.
Completing JonesV answer with some code:
void seekFrame(unsigned frameIndex)
{
// Seek is done on packet dts
int64_t target_dts_usecs = (int64_t)round(frameIndex
* (double)m_video_stream->r_frame_rate.den
/ m_video_stream->r_frame_rate.num * AV_TIME_BASE);
// Remove first dts: when non zero seek should be more accurate
auto first_dts_usecs = (int64_t)round(m_video_stream->first_dts
* (double)m_video_stream->time_base.num
/ m_video_stream->time_base.den * AV_TIME_BASE);
target_dts_usecs += first_dts_usecs;
int rv = av_seek_frame(
m_format_ctx, -1, target_dts_usecs, AVSEEK_FLAG_BACKWARD);
if (rv < 0)
throw exception("Failed to seek");
avcodec_flush_buffers(m_codec_ctx);
}
Then you can begin decoding checking AVPacket.dts against original target dts, computed on AVStream.time_base. As soon as you reached the target dts, the next decoded frame should be the desired frame.