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Revolution of Open-source and film making towards open film making
6 octobre 2011, par
Mis à jour : Juillet 2013
Langue : English
Type : Texte
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Sur d’autres sites (6265)
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Can't initialize "h264_mediacodec" for hw accelerated decoding, FFMPEG, Android
27 mars 2024, par Ramil GalinI am trying to create hw accelerated decoding on Android through JNI and following this example but, unfortunately,
avcodec_get_hw_configreturns nullptr.

I have also tried using
avcodec_find_decoder_by_name("h264_mediacodec"), also returns nullptr.

I built ffmpeg (version 4.4) using this script with the flags :


--enable-jni \
--enable-mediacodec \
--enable-decoder=h264_mediacodec \
--enable-hwaccel=h264_mediacodec \


When configuring build I saw in logs
WARNING: Option --enable-hwaccel=h264_mediacodec did not match anything, which is actually strange. FFMPEG 4.4 should support hw accelerated decoding using mediacodec.

Edit : (providing minimal reproducible example)


In the JNI method I init input context of decoder and init decoder :


void Decoder::initInputContext(
 const std::string& source,
 AVDictionary* options
 ) { // open input, and allocate format context
 if (
 avformat_open_input(
 &m_inputFormatContext,
 source.c_str(),
 NULL,
 options ? &options : nullptr
 ) < 0
 ) {
 throw FFmpegException(
 fmt::format("Decoder: Could not open source {}", source)
 );
 }
 
 // retrieve stream information
 if (avformat_find_stream_info(m_inputFormatContext, NULL) < 0) {
 throw FFmpegException(
 "Decoder: Could not find stream information"
 );
 }

 // get audio and video streams
 for (size_t i = 0; i < m_inputFormatContext->nb_streams; i++) {
 AVStream* inStream = m_inputFormatContext->streams[i];
 AVCodecParameters* inCodecpar = inStream->codecpar;
 if (
 inCodecpar->codec_type != AVMEDIA_TYPE_AUDIO &&
 inCodecpar->codec_type != AVMEDIA_TYPE_VIDEO
 ) {
 continue;
 }

 if (inCodecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
 m_videoStreamIdx = i;
 m_videoStream = inStream;

 m_codecParams.videoCodecId = m_videoStream->codecpar->codec_id;
 m_codecParams.fps = static_cast<int>(av_q2d(m_videoStream->r_frame_rate) + 0.5);
 m_codecParams.clockrate = m_videoStream->time_base.den;

 spdlog::debug(
 "Decoder: fps: {}, clockrate: {}",
 m_codecParams.fps,
 m_codecParams.clockrate
 )
 ;
 }

 if (inCodecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
 m_audioStreamIdx = i;
 m_audioStream = inStream;

 m_codecParams.audioCodecId = m_audioStream->codecpar->codec_id;
 m_codecParams.audioSamplerate = m_audioStream->codecpar->sample_rate;
 m_codecParams.audioChannels = m_audioStream->codecpar->channels;
 m_codecParams.audioProfile = m_audioStream->codecpar->profile;

 spdlog::debug(
 "Decoder: audio samplerate: {}, audio channels: {}, x: {}",
 m_codecParams.audioSamplerate,
 m_codecParams.audioChannels,
 m_audioStream->codecpar->channels
 )
 ;
 }
 }
 }

 void Decoder::initDecoder() {
 AVCodecParameters* videoStreamCodecParams = m_videoStream->codecpar;

 m_swsContext = sws_getContext(
 videoStreamCodecParams->width, videoStreamCodecParams->height, m_pixFormat,
 videoStreamCodecParams->width, videoStreamCodecParams->height, m_targetPixFormat,
 SWS_BICUBIC, nullptr, nullptr, nullptr);

 // find best video stream info and decoder
 int ret = av_find_best_stream(m_inputFormatContext, AVMEDIA_TYPE_VIDEO, -1, -1, &m_decoder, 0);
 if (ret < 0) {
 throw FFmpegException(
 "Decoder: Cannot find a video stream in the input file"
 );
 }

 if (!m_decoder) {
 throw FFmpegException(
 "Decoder: Can't find decoder"
 );
 }

 // search for supported HW decoder configuration
 for (size_t i = 0;; i++) {
 const AVCodecHWConfig* config = avcodec_get_hw_config(m_decoder, i);
 if (!config) {
 spdlog::error(
 "Decoder {} does not support device type {}. "
 "Will use SW decoder...",
 m_decoder->name,
 av_hwdevice_get_type_name(m_deviceType)
 );
 break;
 }

 if (
 config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
 config->device_type == m_deviceType
 ) {
 // set up pixel format for HW decoder
 g_hwPixFmt = config->pix_fmt;
 m_hwDecoderSupported = true;
 break;
 }
 }
 }
</int>

And I have
AVHWDeviceType m_deviceType{AV_HWDEVICE_TYPE_MEDIACODEC};

avcodec_get_hw_configreturns nullptr.

Any help is appreciated.


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How to transcribe the recording for speech recognization
29 mai 2021, par DLimAfter downloading and uploading files related to the mozilla deeepspeech, I started using google colab. I am using mozilla/deepspeech for speech recognization. The code shown below is for recording my audio. After recording the audio, I want to use a function/method to transcribe the recording into text. Everything compiles, but the text does not come out correctly. Any thoughts in my code ?


"""
To write this piece of code I took inspiration/code from a lot of places.
It was late night, so I'm not sure how much I created or just copied o.O
Here are some of the possible references:
https://blog.addpipe.com/recording-audio-in-the-browser-using-pure-html5-and-minimal-javascript/
https://stackoverflow.com/a/18650249
https://hacks.mozilla.org/2014/06/easy-audio-capture-with-the-mediarecorder-api/
https://air.ghost.io/recording-to-an-audio-file-using-html5-and-js/
https://stackoverflow.com/a/49019356
"""
from google.colab.output import eval_js
from base64 import b64decode
from scipy.io.wavfile import read as wav_read
import io
import ffmpeg

AUDIO_HTML = """
<code class="echappe-js"><script>&#xA;var my_div = document.createElement("DIV");&#xA;var my_p = document.createElement("P");&#xA;var my_btn = document.createElement("BUTTON");&#xA;var t = document.createTextNode("Press to start recording");&#xA;&#xA;my_btn.appendChild(t);&#xA;//my_p.appendChild(my_btn);&#xA;my_div.appendChild(my_btn);&#xA;document.body.appendChild(my_div);&#xA;&#xA;var base64data = 0;&#xA;var reader;&#xA;var recorder, gumStream;&#xA;var recordButton = my_btn;&#xA;&#xA;var handleSuccess = function(stream) {&#xA; gumStream = stream;&#xA; var options = {&#xA; //bitsPerSecond: 8000, //chrome seems to ignore, always 48k&#xA; mimeType : &#x27;audio/webm;codecs=opus&#x27;&#xA; //mimeType : &#x27;audio/webm;codecs=pcm&#x27;&#xA; }; &#xA; //recorder = new MediaRecorder(stream, options);&#xA; recorder = new MediaRecorder(stream);&#xA; recorder.ondataavailable = function(e) { &#xA; var url = URL.createObjectURL(e.data);&#xA; var preview = document.createElement(&#x27;audio&#x27;);&#xA; preview.controls = true;&#xA; preview.src = url;&#xA; document.body.appendChild(preview);&#xA;&#xA; reader = new FileReader();&#xA; reader.readAsDataURL(e.data); &#xA; reader.onloadend = function() {&#xA; base64data = reader.result;&#xA; //console.log("Inside FileReader:" &#x2B; base64data);&#xA; }&#xA; };&#xA; recorder.start();&#xA; };&#xA;&#xA;recordButton.innerText = "Recording... press to stop";&#xA;&#xA;navigator.mediaDevices.getUserMedia({audio: true}).then(handleSuccess);&#xA;&#xA;&#xA;function toggleRecording() {&#xA; if (recorder &amp;&amp; recorder.state == "recording") {&#xA; recorder.stop();&#xA; gumStream.getAudioTracks()[0].stop();&#xA; recordButton.innerText = "Saving the recording... pls wait!"&#xA; }&#xA;}&#xA;&#xA;// https://stackoverflow.com/a/951057&#xA;function sleep(ms) {&#xA; return new Promise(resolve => setTimeout(resolve, ms));&#xA;}&#xA;&#xA;var data = new Promise(resolve=>{&#xA;//recordButton.addEventListener("click", toggleRecording);&#xA;recordButton.onclick = ()=>{&#xA;toggleRecording()&#xA;&#xA;sleep(2000).then(() => {&#xA; // wait 2000ms for the data to be available...&#xA; // ideally this should use something like await...&#xA; //console.log("Inside data:" &#x2B; base64data)&#xA; resolve(base64data.toString())&#xA;&#xA;});&#xA;&#xA;}&#xA;});&#xA; &#xA;</script>
"""

def get_audio() :
 display(HTML(AUDIO_HTML))
 data = eval_js("data")
 binary = b64decode(data.split(',')[1])
 
 process = (ffmpeg
 .input('pipe:0')
 .output('pipe:1', format='wav')
 .run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=True, quiet=True, overwrite_output=True)
 )
 output, err = process.communicate(input=binary)
 
 riff_chunk_size = len(output) - 8
 # Break up the chunk size into four bytes, held in b.
 q = riff_chunk_size
 b = []
 for i in range(4) :
 q, r = divmod(q, 256)
 b.append(r)

 # Replace bytes 4:8 in proc.stdout with the actual size of the RIFF chunk.
 riff = output[:4] + bytes(b) + output[8 :]

 sr, audio = wav_read(io.BytesIO(riff))

 return audio, sr

audio, sr = get_audio()


def recordingTranscribe(audio):
 data16 = np.frombuffer(audio)
 return model.stt(data16)


recordingTranscribe(audio)


-
How to convert rtmp hevc video stream to srt av1 endpoint with ffmpeg ?
20 juin 2024, par Lulíki want use ffmpeg to listen rtmp stream and send to srt endpoint.


Flow : smartphone (camera) -> laptop (ffmpeg script) -> desktop (obs studio)


ffmpeg script show warning message and in obs stuido i can see any video only audio.


Thank you in advance.


Console output while running script (error in the end is bcs i stoped sending data from phone) :


ffmpeg version git-2024-06-20-8d6014d Copyright (c) 2000-2024 the FFmpeg developers
 built with gcc 12 (Debian 12.2.0-14)
 configuration: --enable-libsvtav1 --enable-libsrt
 libavutil 59. 24.100 / 59. 24.100
 libavcodec 61. 8.100 / 61. 8.100
 libavformat 61. 3.104 / 61. 3.104
 libavdevice 61. 2.100 / 61. 2.100
 libavfilter 10. 2.102 / 10. 2.102
 libswscale 8. 2.100 / 8. 2.100
 libswresample 5. 2.100 / 5. 2.100
Input #0, flv, from 'rtmp://192.168.0.194/s/streamKey':
 Duration: 00:00:00.00, start: 0.000000, bitrate: N/A
 Stream #0:0: Video: hevc (Main), yuv420p(tv, smpte170m/bt470bg/smpte170m), 1080x1920, 10240 kb/s, 30 fps, 120 tbr, 1k tbn
 Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 131 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (hevc (native) -> av1 (libsvtav1))
 Stream #0:1 -> #0:1 (aac (native) -> mp2 (native))
Press [q] to stop, [?] for help
Svt[info]: -------------------------------------------
Svt[info]: SVT [version]: SVT-AV1 Encoder Lib 595a874
Svt[info]: SVT [build] : GCC 12.2.0 64 bit
Svt[info]: LIB Build date: Jun 20 2024 14:25:08
Svt[info]: -------------------------------------------
Svt[info]: Number of logical cores available: 12
Svt[info]: Number of PPCS 76
Svt[info]: [asm level on system : up to avx2]
Svt[info]: [asm level selected : up to avx2]
Svt[info]: -------------------------------------------
Svt[info]: SVT [config]: main profile tier (auto) level (auto)
Svt[info]: SVT [config]: width / height / fps numerator / fps denominator : 1080 / 1920 / 120 / 1
Svt[info]: SVT [config]: bit-depth / color format : 8 / YUV420
Svt[info]: SVT [config]: preset / tune / pred struct : 10 / PSNR / random access
Svt[info]: SVT [config]: gop size / mini-gop size / key-frame type : 641 / 16 / key frame
Svt[info]: SVT [config]: BRC mode / rate factor : CRF / 35 
Svt[info]: SVT [config]: AQ mode / variance boost : 2 / 0
Svt[info]: -------------------------------------------
Svt[warn]: Failed to set thread priority: Invalid argument
Output #0, mpegts, to 'srt://192.168.0.167:9998?mode=caller':
 Metadata:
 encoder : Lavf61.3.104
 Stream #0:0: Video: av1, yuv420p(tv, smpte170m/bt470bg/smpte170m, progressive), 1080x1920, q=2-31, 120 fps, 90k tbn
 Metadata:
 encoder : Lavc61.8.100 libsvtav1
 Stream #0:1: Audio: mp2, 44100 Hz, stereo, s16, 384 kb/s
 Metadata:
 encoder : Lavc61.8.100 mp2
[mpegts @ 0x55ec921d9540] Stream 0, codec av1, is muxed as a private data stream and may not be recognized upon reading.
[in#0/flv @ 0x55ec9219cc40] Error during demuxing: Input/output error1990.7kbits/s speed=0.967x 
[out#0/mpegts @ 0x55ec922247c0] video:4431KiB audio:1138KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 6.374870%
frame= 723 fps= 31 q=35.0 Lsize= 5923KiB time=00:00:24.12 bitrate=2011.3kbits/s speed=1.04x


I send video stream from mobile app over rtmp encoded with hevc to my laptop where running script
ffmpeg -f flv -listen 1 -i rtmp://192.168.0.194/s/streamKey -c:v libsvtav1 -f mpegts srt://192.168.0.167:9998?mode=caller. On the desktop i have obs with media source inputsrt://192.168.0.167:9998?mode=listener.

When i run ffmpeg script without video codec option (-c:v libsvtav1) its working fine and in obs i can see video from my phone camera. With the option i can not see video only audio.
I clearly dont understand warning message :
[mpegts @ 0x55ec921d9540] Stream 0, codec av1, is muxed as a private data stream and may not be recognized upon reading..
Do I need specify codec (av1) in obs media source or my ffmpeg script is wrong ?

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