I need to encode an mpegts video using the ffmpeg C++ API. The output video shall have two streams: the first one shall be of type AVMEDIA_TYPE_VIDEO; the second one shall be of type AVMEDIA_TYPE_DATA and shall contain a set of KLV data.
I have written my own KLV library to manage the KLV format.
However I'm not able to create "from scratch" a new video by combining the two streams. Following the implementation as in FFMPEG C api h.264 encoding / MPEG2 ts streaming problems I can successfully encode a mpegts video with a single video stream.
However I'm not able to add a new AVMEDIA_TYPE_DATA stream to the output video since, as soon as I add a new data stream using methods like avformat_new_stream(...) the output video is empty: neither the data stream nor the video one are produced and the output file is empty.
Can anyone suggest me a tutorial page or a sample on how to properly add a data stream to my output video in mpegts format?
Thanks a lot!
I was able to get a KLV stream added to a muxed output by starting with the "muxing.c" example that comes with the FFmpeg source, and modifying it as follows.
First, I created the AVStream as follows, where "oc" is the AVFormatContext (muxer) variable:
AVStream *klv_stream = klv_stream = avformat_new_stream(oc, NULL);
klv_stream->codec->codec_type = AVMEDIA_TYPE_DATA;
klv_stream->codec->codec_id = AV_CODEC_ID_TIMED_ID3;
klv_stream->time_base = AVRational{ 1, 30 };
klv_stream->id = oc->nb_streams - 1;
Then, during the encoding/muxing loop:
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = (uint8_t*)GetKlv(pkt.size);
auto res = write_frame(oc, &video_st.st->time_base, klv_stream, &pkt);
free(pkt.data);
(The GetKlv() function returns a malloc()'ed array of binary data that would be replaced by whatever you're using to get your encoded KLV. It sets pkt.size to the length of the data.)
With this modification, and specifying a ".ts" target file, I get a three-stream file that plays just fine in VLC. The KLV stream has a stream_type of 0x15, indicating synchronous KLV.
Note the codec_id value of AV_CODEC_ID_TIMED_ID3. According to the libavformat source file "mpegtsenc.c", a value of AV_CODEC_ID_OPUS should result in stream_type 6, for asynchronous KLV (no accompanying PTS or DTS). This is actually important for my application, but I'm unable to get it to work -- the call to avformat_write_header() throws a division by zero error. If I get that figured out, I'll add an update here.
Related
I'm using ReSpeaker Mic Array v2.0 on my robot, I used the following git repo: https://github.com/furushchev/respeaker_ros.git to capture the audio received by the speaker. I subscribed to it's raw audio ros topic /audio which is just byte array data(http://docs.ros.org/noetic/api/audio_common_msgs/html/msg/AudioData.html)
How can I write the AudioData message's uint8[] data into a wav file in C++? I would like to play the wav file by other means afterwards.
I saw that in ros audio_common library example it uses gstreamer to do the writing, but I'm quite confused after reading the code(https://github.com/ros-drivers/audio_common/blob/master/audio_capture/src/audio_capture.cpp)
Example that you saw is using Gstremaer's alsasrc to capture audio from mic in this line
_source = gst_element_factory_make("alsasrc", "source");
So Gstreamer's pipeline is internally handling/capturing audio byte array and, in case of input parameters dst_type=="filesink" and format=="wave", encoding it with
_filter = gst_element_factory_make("wavenc", "filter");
and creating .wav file with
_sink = gst_element_factory_make("filesink", "sink");
On the other hand, running that code with input parameters dst_type=="appsink" and format=="wave" actually captures audio bytes again but, instead of writing to file, publishes them on ros topic /audio.
If you cannot (from any reason) use this code with input parameters dst_type=="filesink" and format=="wave", I suppose you will need to use Gstreamer's appsrc element and feed it with bytes from your AudioData message. In that case, the rest of Gstreamer pipeline for encoding and writing to file should remain the same as in the example.
I made my own rtmp server using libav and ffmpeg. I receive as input either an flv file or an rtmp streaming "containing" an flv file.
Since I manipulate the flv file and the relative composition time of each frame, I would like to know if there is a way to get this composition time.
I thought that given my AVPacket, I could analyze the raw buffer in order to extract the right information since I know that the flv header is 11 bytes and then in the next 16 bytes I should find the composition time.
But it doesn't work.
This is a rough example of code:
AVPacket pkt;
AVFormatContext *ifmt_ctx
while(true)
{
AVStream *in_stream, *out_stream;
ret = av_read_frame(ifmt_ctx, &pkt);
//get the composite time
}
AVPacket needs to be able to represent the data found in all media formats. Some formats (like mp4 and flv) have a decode_time and a composition_time, other (like transport streams) have a decode_time and a presentation_time. To make it easier for the programmer, AVPacket chose one method to store the information and converts when needed. Luckily its an an easy to convert back:
auto cts = pkt.pts - pkt.dts
I wrote code to create H.264 stream, which has a loop to generate H.264 encoded frame.
while(true) {
...
x264_encoder_encode(encoder, &buffer, &i_buffer, &pic_in, &pic_out);
...
/*TODO: Write one frame in the buffer to a streamable mp4 file*/
}
Every single time, an H.264 encoded frame is generated and stored in the buffer. How can I write it into a streamable mp4 file directly through the buffer?
I spent lots of time searching for the solution. All I can find is to read stream from a file using
avformat_open_input(&fmtCtx, in_filename, 0, 0)
Is there any way to read directly from buffer without a file?
MP4 is actually not streamable. So in other words, you can't do it at all. I ran in that very problem.
The reason why it won't work is because when you open an mp4 file, you have to have all sorts of parameters, which by default get saved at the end of the file. When you create an MP4, you can always forcibly save that info at the start. However, to know what those parameters are, you need all the data. And without those parameters, the software trying to load the mp4 fails very early on. This is true for some other formats such as webm videos and .m4a or .wav for audio.
What you have to do is stream the actual H.264, possibly using RTSP or a format of your own if you're in control of both sides.
i use isamplegrabber sampleCB callback to get audio sample, i can get buffer and buffer length from imediasample and i use avcodec_fill_audio_frame(frame,ost->enc->channels,ost->enc->sample_fmt,(uint8_t *)buffer,length,0) to make an avframe , but this frame does not make any audio in my mux file! i think the length is very smaller than frame_size.
can every one help me please? or give me some example if it is possible.
thank you
this is my samplecb code :
HRESULT AudioSampleGrabberCallBack::SampleCB(double Time, IMediaSample*pSample){
BYTE *pBuffer;
pSample->GetPointer(&pBuffer);
long BufferLen = pSample->GetActualDataLength();
muxer->PutAudioFrame(pBuffer,BufferLen);
}
and this is samplegrabber pin media type :
AM_MEDIA_TYPE pmt2;
ZeroMemory(&pmt2, sizeof(AM_MEDIA_TYPE));
pmt2.majortype = MEDIATYPE_Audio;
pmt2.subtype = FOURCCMap(0x1602);
pmt2.formattype = FORMAT_WaveFormatEx;
hr = pSampleGrabber_audio->SetMediaType(&pmt2);
after that i using ffmpeg muxing example to process frames and i think i need only to change the signal generating part of code :
AVFrame *Muxing::get_audio_frame(OutputStream *ost,BYTE* buffer,long length)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
uint16_t *q = (uint16_t*)frame->data[0];
int buffer_size = av_samples_get_buffer_size(NULL, ost->enc->channels,
ost->enc->frame_size,
ost->enc->sample_fmt, 0);
// uint8_t *sample = (uint8_t *) av_malloc(buffer_size);
av_samples_alloc(&frame->data[0], frame->linesize, ost->enc->channels, ost->enc->frame_size, ost->enc->sample_fmt, 1);
avcodec_fill_audio_frame(frame, ost->enc->channels, ost->enc->sample_fmt,frame->data[0], buffer_size, 1);
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
The code snippets suggest you are getting AAC data using Sample Grabber and you are trying to write that into file using FFmpeg's libavformat. This can work out.
You initialize your sample grabber to get audio data in WAVE_FORMAT_AAC_LATM format. This format is not so wide spread and you are interested in reviewing your filter graph to make sure the upstream connection on the Sample Grabber is such that you expect. There is a chance that somehow there is a weird chain of filter that pretend to produce AAC-LATM and the reality is that the data is invalid (or not even reaching grabber callback). So you need to review the filter graph (see Loading a Graph From an External Process and Understanding Your DirectShow Filter Graph), then step through your callback with debugger to make sure you get the data and it makes sense.
Next thing, you are expected to initialize AVFormatContext, AVStream to indicate that you will be writing data in AAC LATM format. Provided code does not show you are doing it right. The sample you are referring to is using default codecs.
Related reading: Support LATM AAC in MP4 container
Then, you need to make sure that both incoming data and your FFmpeg output setup are in agreement about whether the data has or does not have ADTS headers, the provided code does not shed any light on this.
Furthermore, I am afraid you might be preparing your audio data incorrectly. The sample in question generates raw audio data and applies encoder to produce compressed content using avcodec_encode_audio2. Then a packed with compressed audio is being sent to writing using av_interleaved_write_frame. The way you attached your code snippets to the question makes me thing you are doing it wrong. For starters, you still don't show relevant code which makes me think you have troubles identifying what code is relevant exactly. Then you are dealing with your AAC data as if it was raw PCM audio in get_audio_frame code snippet whereas you are interested in reviewing FFmpeg sample code with the thought in mind that you already have compressed AAC data and sample gets to thins point after return from avcodec_encode_audio2 call. This is where you are supposed to merge your code and the sample.
I'm developing an application to send video over RTP to a client that can play only H.263 (1996) and H263+ (1998).
To do this i've encoded the video using libav following these steps: (this is only part of the code)
av_register_all();
avformat_network_init();
Fmt = av_guess_format("rtp", NULL, NULL);
...
st = add_video_stream(FmtCtx, CODEC_ID_H263);
...
avio_open(&FmtCtx->pb, rtp_url, URL_WRONLY)
To finally enter a loop where i encode the video, the problem is that the stream generated by this program is encoded in H.263-2000 (or H.263++) which the other side cannot undertand, even though i use CODEC_ID_H263 or CODEC_ID_H263P in the initialization the same thing happens.
Is it possible to encode in those old H.263 versions using libav? i havent managed to do it not even using ffmpeg commands. The stream is always h.263-2000 (PT=96)