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The Slip - Artworks
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
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Podcasting Legal guide
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Mis à jour : Mai 2011
Langue : English
Type : Texte
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Creativecommons informational flyer
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Mis à jour : Juillet 2013
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Autres articles (78)
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Personnaliser les catégories
21 juin 2013, parFormulaire de création d’une catégorie
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Dans le cas d’un document de type catégorie, les champs proposés par défaut sont : Texte
On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire.
Dans le cas d’un document de type média, les champs non affichés par défaut sont : Descriptif rapide
Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...) -
Gestion de la ferme
2 mars 2010, parLa ferme est gérée dans son ensemble par des "super admins".
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Dans un premier temps il utilise le plugin "Gestion de mutualisation" -
MediaSPIP Player : les contrôles
26 mai 2010, parLes contrôles à la souris du lecteur
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Sur d’autres sites (10116)
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Can't playback huge video which was uploaded to google storage. I Get Error : Unable to retrieve manifest /stream.m3u8 -Error : Could not retrieve file
18 juillet 2022, par Dmytro PetskovychI upload file to google storage using "@ffmpeg-installer/ffmpeg" and @google-cloud/storage in my node.js App.
Step 1. file uploading to fs is in child processes - one process for each type of resolution (totaly six).
step 2. encription (converting to stream)
step 3. upload to google storage


This way is working fine only with small file. When i upload large file I Get server Error : Unable to retrieve manifest /stream.m3u8 or Unable to retrieve .


It seems that not all splits are uploaded to the cloud. but i checked they are there.


I am currently using "Upload a directory to a bucket" in order to send the video from the client to the Google Cloud Storage bucket.


when I upload video, actually I upload six videos, one for each type resolution


for example when I upload video with duration one hour it split on chunk and totally I get more three thousands files.


not all of this files are accessible when i try playback video. that's why I can't play video in all types of resolution and get error like


maybe someone had the similar problem and helps fix it
.
i have no idea why i get that behavior only with huge file


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Video files recorded in Google Chrome have stuttering audio
4 juin 2018, par maxpajBackground
I’m developing a platform where users can record videos of themselves or their screen and send them as video messages to customers / clients.
I have limited users to only using my application with Google Chrome and I’m using the MediaRecorder API to record the video data from the users screen or webcamera. The codecs that are used for recording are VP8/OPUS (WEBM container).
I need the videos to run in as many browsers as possible, so I’m using a 3rd party service to transcode videos from whatever format I’m getting from the users to a H.265/AAC MP4 container (caniuse MPEG-4/H.264).
Issue
Lately I’ve seen that some videos recorded on Mac OSX machines have the video and audio out of sync or that the video and audio stutters, depending on which player I’m using. I call these video files corrupt, for lack of a better word. Playing a corrupt file in Google Chrome renders smooth playing audio. Playing the video in VLC on my Windows machine renders stuttering audio.
When I run the corrupt video files through the transcoding service I get video files with stuttering audio, no matter which player I’m using.
This is an unwanted result and pretty much unacceptable since the audio needs to be smooth in order for the recipient of a video to not be bothered with the quality.
Debugging
According to the transcoding service support, this happens because of their mechanisms that try to sync up the audio and video from the corrupt file :
Inspecting our encoding logs, I’ve noticed the following kind of
warnings :[2018-05-16 14:08:38.009] [pcm_s16le @ 0x1d608c0] pcm_encode_frame :
filling in for 5856 missing samples (122 ms) before pts 40800 to
correct sync ! [2018-05-16 14:08:38.009] [pcm_s16le @ 0x1d608c0]
pcm_encode_frame : dropping 2880 samples (60 ms) at pts 43392 to help
correct sync to -3168 samples (-66 ms) !The problem here comes from the way that the audio in the original
source file is encoded.-
you should ensure that the audio is not out of sync (audio timestamps
are correct) in your source file before submitting the jobRunning a corrupt file through ffmpeg on my own machine, re-encoding with the same codecs, produces the same kind of stuttering video. The logs produce an alarming amount of errors. Here is a sample of the log output :
[libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 194.8kbits/s dup=0 drop=5 speed=0.31x
[webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15434, current: 15394; changing to 15434. This may result in incorrect timestamps in the output file.
[webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15434, current: 15414; changing to 15434. This may result in incorrect timestamps in the output file.
[libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 193.3kbits/s dup=0 drop=5 speed=0.309x
[webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15539, current: 15499; changing to 15539. This may result in incorrect timestamps in the output file.
[webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15539, current: 15519; changing to 15539. This may result in incorrect timestamps in the output file.
[libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 192.0kbits/s dup=0 drop=5 speed=0.308x
[webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15667, current: 15627; changing to 15667. This may result in incorrect timestamps in the output file.
[webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15667, current: 15647; changing to 15667. This may result in incorrect timestamps in the output file.
[libopus @ 0000029938e24d80] Queue input is backward in timeI tried running the same inputs through another transcoding service and those outputs worked a lot better - video was still stuttering but the audio played smoothly, which is more important to the use case of my application.
To my knowledge, this have so far only occurred for users on Mac OSX machines.
Questions
-
Is there anything I can do to have the files work better ? Or is this entirely a consequence of how encoding of video and audio in Google Chrome works ?
-
One step in the right direction would be to just be able to detect when the video is corrupt. How can I do that ?
-
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C++ ffmpeg - export to wav error : Invalid PCM packet, data has size 2 but at least a size of 4 was expected
9 septembre 2024, par Chris PC++ code :


AudioSegment AudioSegment::from_file(const std::string& file_path, const std::string& format, const std::string& codec,
 const std::map& parameters, int start_second, int duration) {

 avformat_network_init();
 av_log_set_level(AV_LOG_ERROR); // Adjust logging level as needed

 AVFormatContext* format_ctx = nullptr;
 if (avformat_open_input(&format_ctx, file_path.c_str(), nullptr, nullptr) != 0) {
 std::cerr << "Error: Could not open audio file." << std::endl;
 return AudioSegment(); // Return an empty AudioSegment on failure
 }

 if (avformat_find_stream_info(format_ctx, nullptr) < 0) {
 std::cerr << "Error: Could not find stream information." << std::endl;
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 int audio_stream_index = -1;
 for (unsigned int i = 0; i < format_ctx->nb_streams; i++) {
 if (format_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
 audio_stream_index = i;
 break;
 }
 }

 if (audio_stream_index == -1) {
 std::cerr << "Error: Could not find audio stream." << std::endl;
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 AVCodecParameters* codec_par = format_ctx->streams[audio_stream_index]->codecpar;
 const AVCodec* my_codec = avcodec_find_decoder(codec_par->codec_id);
 AVCodecContext* codec_ctx = avcodec_alloc_context3(my_codec);

 if (!codec_ctx) {
 std::cerr << "Error: Could not allocate codec context." << std::endl;
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 if (avcodec_parameters_to_context(codec_ctx, codec_par) < 0) {
 std::cerr << "Error: Could not initialize codec context." << std::endl;
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 if (avcodec_open2(codec_ctx, my_codec, nullptr) < 0) {
 std::cerr << "Error: Could not open codec." << std::endl;
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 SwrContext* swr_ctx = swr_alloc();
 if (!swr_ctx) {
 std::cerr << "Error: Could not allocate SwrContext." << std::endl;
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }
 codec_ctx->sample_rate = 44100;
 // Set up resampling context to convert to S16 format with 2 bytes per sample
 av_opt_set_chlayout(swr_ctx, "in_chlayout", &codec_ctx->ch_layout, 0);
 av_opt_set_int(swr_ctx, "in_sample_rate", codec_ctx->sample_rate, 0);
 av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", codec_ctx->sample_fmt, 0);

 AVChannelLayout dst_ch_layout;
 av_channel_layout_copy(&dst_ch_layout, &codec_ctx->ch_layout);
 av_channel_layout_uninit(&dst_ch_layout);
 av_channel_layout_default(&dst_ch_layout, 2);

 av_opt_set_chlayout(swr_ctx, "out_chlayout", &dst_ch_layout, 0);
 av_opt_set_int(swr_ctx, "out_sample_rate", codec_ctx->sample_rate, 0); // Match input sample rate
 av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); // Force S16 format

 if (swr_init(swr_ctx) < 0) {
 std::cerr << "Error: Failed to initialize the resampling context" << std::endl;
 swr_free(&swr_ctx);
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 AVPacket packet;
 AVFrame* frame = av_frame_alloc();
 if (!frame) {
 std::cerr << "Error: Could not allocate frame." << std::endl;
 swr_free(&swr_ctx);
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 std::vector<char> output;
 while (av_read_frame(format_ctx, &packet) >= 0) {
 if (packet.stream_index == audio_stream_index) {
 if (avcodec_send_packet(codec_ctx, &packet) == 0) {
 while (avcodec_receive_frame(codec_ctx, frame) == 0) {
 if (frame->pts != AV_NOPTS_VALUE) {
 frame->pts = av_rescale_q(frame->pts, codec_ctx->time_base, format_ctx->streams[audio_stream_index]->time_base);
 }

 uint8_t* output_buffer;
 int output_samples = av_rescale_rnd(
 swr_get_delay(swr_ctx, codec_ctx->sample_rate) + frame->nb_samples,
 codec_ctx->sample_rate, codec_ctx->sample_rate, AV_ROUND_UP);

 int output_buffer_size = av_samples_get_buffer_size(
 nullptr, 2, output_samples, AV_SAMPLE_FMT_S16, 1);

 output_buffer = (uint8_t*)av_malloc(output_buffer_size);

 if (output_buffer) {
 memset(output_buffer, 0, output_buffer_size); // Zero padding to avoid random noise
 int converted_samples = swr_convert(swr_ctx, &output_buffer, output_samples,
 (const uint8_t**)frame->extended_data, frame->nb_samples);

 if (converted_samples >= 0) {
 output.insert(output.end(), output_buffer, output_buffer + output_buffer_size);
 }
 else {
 std::cerr << "Error: Failed to convert audio samples." << std::endl;
 }
 // Make sure output_buffer is valid before freeing
 if (output_buffer != nullptr) {
 av_free(output_buffer);
 output_buffer = nullptr; // Prevent double-free
 }
 }
 else {
 std::cerr << "Error: Could not allocate output buffer." << std::endl;
 }
 }
 }
 else {
 std::cerr << "Error: Failed to send packet to codec context." << std::endl;
 }
 }
 av_packet_unref(&packet);
 }

 int frame_width = av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * 2; // Use 2 bytes per sample and 2 channels

 std::map metadata = {
 {"sample_width", 2}, // S16 format has 2 bytes per sample
 {"frame_rate", codec_ctx->sample_rate}, // Use the input sample rate
 {"channels", 2}, // Assuming stereo output
 {"frame_width", frame_width}
 };

 av_frame_free(&frame);
 swr_free(&swr_ctx);
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);

 return AudioSegment(static_cast<const>(output.data()), output.size(), metadata);
}

std::ofstream AudioSegment::export_segment_to_wav_file(const std::string& out_f) {
 std::cout << this->get_channels() << std::endl;
 av_log_set_level(AV_LOG_ERROR);
 AVCodecContext* codec_ctx = nullptr;
 AVFormatContext* format_ctx = nullptr;
 AVStream* stream = nullptr;
 AVFrame* frame = nullptr;
 AVPacket* pkt = nullptr;
 int ret;

 // Initialize format context for WAV
 if (avformat_alloc_output_context2(&format_ctx, nullptr, "wav", out_f.c_str()) < 0) {
 throw std::runtime_error("Could not allocate format context.");
 }

 // Find encoder for PCM
 const AVCodec* codec_ptr = avcodec_find_encoder(AV_CODEC_ID_PCM_S16LE);
 if (!codec_ptr) {
 throw std::runtime_error("PCM encoder not found.");
 }

 // Add stream
 stream = avformat_new_stream(format_ctx, codec_ptr);
 if (!stream) {
 throw std::runtime_error("Failed to create new stream.");
 }

 // Allocate codec context
 codec_ctx = avcodec_alloc_context3(codec_ptr);
 if (!codec_ctx) {
 throw std::runtime_error("Could not allocate audio codec context.");
 }

 // Set codec parameters for PCM
 codec_ctx->bit_rate = 128000; // Bitrate
 codec_ctx->sample_rate = this->get_frame_rate(); // Use correct sample rate
 codec_ctx->ch_layout.nb_channels = this->get_channels(); // Set the correct channel count

 // Set the channel layout: stereo or mono
 if (this->get_channels() == 2) {
 av_channel_layout_default(&codec_ctx->ch_layout, 2); // Stereo layout
 }
 else {
 av_channel_layout_default(&codec_ctx->ch_layout, 1); // Mono layout
 }

 codec_ctx->sample_fmt = AV_SAMPLE_FMT_S16; // PCM 16-bit format

 // Open codec
 if (avcodec_open2(codec_ctx, codec_ptr, nullptr) < 0) {
 throw std::runtime_error("Could not open codec.");
 }

 // Set codec parameters to the stream
 if (avcodec_parameters_from_context(stream->codecpar, codec_ctx) < 0) {
 throw std::runtime_error("Could not initialize stream codec parameters.");
 }

 // Open output file
 std::ofstream out_file(out_f, std::ios::binary);
 if (!out_file) {
 throw std::runtime_error("Failed to open output file.");
 }

 if (!(format_ctx->oformat->flags & AVFMT_NOFILE)) {
 if (avio_open(&format_ctx->pb, out_f.c_str(), AVIO_FLAG_WRITE) < 0) {
 throw std::runtime_error("Could not open output file.");
 }
 }

 // Write file header
 if (avformat_write_header(format_ctx, nullptr) < 0) {
 throw std::runtime_error("Error occurred when writing file header.");
 }

 // Initialize packet
 pkt = av_packet_alloc();
 if (!pkt) {
 throw std::runtime_error("Could not allocate AVPacket.");
 }

 // Initialize frame
 frame = av_frame_alloc();
 if (!frame) {
 throw std::runtime_error("Could not allocate AVFrame.");
 }

 // Set the frame properties
 frame->format = codec_ctx->sample_fmt;
 frame->ch_layout = codec_ctx->ch_layout;

 // Number of audio samples available in the data buffer
 int total_samples = data_.size() / (av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels);
 int samples_read = 0;

 // Set the number of samples per frame dynamically based on the input data
 while (samples_read < total_samples) {
 // Determine how many samples to read in this iteration (don't exceed the total sample count)
 int num_samples = std::min(codec_ctx->frame_size, total_samples - samples_read);
 if (num_samples == 0) {
 num_samples = 1024;
 codec_ctx->frame_size = 1024;
 }
 // Ensure num_samples is not zero
 if (num_samples <= 0) {
 throw std::runtime_error("Invalid number of samples in frame.");
 }

 // Set the number of samples in the frame
 frame->nb_samples = num_samples;

 // Allocate the frame buffer based on the number of samples
 ret = av_frame_get_buffer(frame, 0);
 if (ret < 0) {
 std::cerr << "Error allocating frame buffer: " << ret << std::endl;
 throw std::runtime_error("Could not allocate audio data buffers.");
 }

 // Copy the audio data into the frame's buffer (interleaving if necessary)
 /*if (codec_ctx->ch_layout.nb_channels == 2) {
 // If stereo, interleave planar data into packed format
 for (int i = 0; i < num_samples; ++i) {
 ((int16_t*)frame->data[0])[2 * i] = ((int16_t*)data_.data())[i]; // Left channel
 ((int16_t*)frame->data[0])[2 * i + 1] = ((int16_t*)data_.data())[total_samples + i]; // Right channel
 }
 }
 else {
 // For mono or packed data, directly copy the samples
 std::memcpy(frame->data[0], data_.data() + samples_read * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels,
 num_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels);
 }
 */
 std::memcpy(frame->data[0], data_.data() + samples_read * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels,
 num_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels);

 // Send the frame for encoding
 ret = avcodec_send_frame(codec_ctx, frame);
 if (ret < 0) {
 std::cerr << "Error sending frame for encoding: " << ret << std::endl;
 throw std::runtime_error("Error sending frame for encoding.");
 }

 // Receive and write encoded packets
 while (ret >= 0) {
 ret = avcodec_receive_packet(codec_ctx, pkt);
 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
 break;
 }
 else if (ret < 0) {
 throw std::runtime_error("Error during encoding.");
 }

 out_file.write(reinterpret_cast(pkt->data), pkt->size);
 av_packet_unref(pkt);
 }

 samples_read += num_samples;
 }

 // Flush the encoder
 if (avcodec_send_frame(codec_ctx, nullptr) < 0) {
 throw std::runtime_error("Error flushing the encoder.");
 }

 while (avcodec_receive_packet(codec_ctx, pkt) >= 0) {
 out_file.write(reinterpret_cast(pkt->data), pkt->size);
 av_packet_unref(pkt);
 }

 // Write file trailer
 av_write_trailer(format_ctx);

 // Cleanup
 av_frame_free(&frame);
 av_packet_free(&pkt);
 avcodec_free_context(&codec_ctx);

 if (!(format_ctx->oformat->flags & AVFMT_NOFILE)) {
 avio_closep(&format_ctx->pb);
 }
 avformat_free_context(format_ctx);

 out_file.close();
 return out_file;
}

</const></char>

Run code :


#include "audio_segment.h"
#include "effects.h"
#include "playback.h"
#include "cppaudioop.h"
#include "exceptions.h"
#include "generators.h"
#include "silence.h"
#include "utils.h"

#include <iostream>
#include <filesystem>

using namespace cppdub;

int main() {
 try {
 // Load the source audio file
 AudioSegment seg_1 = AudioSegment::from_file("../data/test10.mp3");
 std::string out_file_name = "ah-ah-ah.wav";

 // Export the audio segment to a new file with specified settings
 //seg_1.export_segment(out_file_name, "mp3");
 seg_1.export_segment_to_wav_file(out_file_name);


 // Optionally play the audio segment to verify
 // play(seg_1);

 // Load the exported audio file
 AudioSegment seg_2 = AudioSegment::from_file(out_file_name);

 // Play segments
 //play(seg_1);
 play(seg_2);
 }
 catch (const std::exception& e) {
 std::cerr << "An error occurred: " << e.what() << std::endl;
 }

 return 0;
}
</filesystem></iostream>

Error in second call of from_file function :


[pcm_s16le @ 000002d82ca5bfc0] Invalid PCM packet, data has size 2 but at least a size of 4 was expected


The process continue, i call hear the seg_2 with play(seg_2) call, but i can't directly play seg_2 export wav file (from windows explorer).


I had a guess that error may be because packed vs plannar formats missmatch but i am not quit sure. Maybe a swr_convert is necessary.


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