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La conservation du net art au musée. Les stratégies à l’œuvre
26 mai 2011
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (63)
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La file d’attente de SPIPmotion
28 novembre 2010, parUne file d’attente stockée dans la base de donnée
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Cette nouvelle table est constituée des champs suivants : id_spipmotion_attente, l’identifiant numérique unique de la tâche à traiter ; id_document, l’identifiant numérique du document original à encoder ; id_objet l’identifiant unique de l’objet auquel le document encodé devra être attaché automatiquement ; objet, le type d’objet auquel (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
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Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...) -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
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Sur d’autres sites (8881)
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Why am I getting blips when encoding a sound file using Java JNA ?
21 mars 2014, par yonranI have implemented a hello world libavcodec using JNA to generate a wav file containing a pure 440Hz sine wave. But when I actually run the program the wav file contains annoying clicks and blips (compare to pure sin wav created from the C program). How am I calling
avcodec_encode_audio2wrong ?Here is my Java code. All the sources are also at github in case you want to try to compile it.
import java.io.IOException;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
import java.nio.IntBuffer;
import java.util.Objects;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.TargetDataLine;
public class Sin {
/**
* Abstract class that allows you to put the initialization and cleanup
* code at the same place instead of separated by the big try block.
*/
public static abstract class SharedPtr<t> implements AutoCloseable {
public T ptr;
public SharedPtr(T ptr) {
this.ptr = ptr;
}
/**
* Abstract override forces method to throw no checked exceptions.
* Subclasses will call a C function that throws no exceptions.
*/
@Override public abstract void close();
}
/**
* @param args
* @throws IOException
* @throws LineUnavailableException
*/
public static void main(String[] args) throws IOException, LineUnavailableException {
final AvcodecLibrary avcodec = AvcodecLibrary.INSTANCE;
final AvformatLibrary avformat = AvformatLibrary.INSTANCE;
final AvutilLibrary avutil = AvutilLibrary.INSTANCE;
avcodec.avcodec_register_all();
avformat.av_register_all();
AVOutputFormat.ByReference format = null;
String format_name = "wav", file_url = "file:sinjava.wav";
for (AVOutputFormat.ByReference formatIter = avformat.av_oformat_next(null); formatIter != null; formatIter = avformat.av_oformat_next(formatIter)) {
formatIter.setAutoWrite(false);
String iterName = formatIter.name;
if (format_name.equals(iterName)) {
format = formatIter;
break;
}
}
Objects.requireNonNull(format);
System.out.format("Found format %s%n", format_name);
AVCodec codec = avcodec.avcodec_find_encoder(format.audio_codec); // one of AvcodecLibrary.CodecID
Objects.requireNonNull(codec);
codec.setAutoWrite(false);
try (
SharedPtr<avformatcontext> fmtCtxPtr = new SharedPtr<avformatcontext>(avformat.avformat_alloc_context()) {@Override public void close(){if (null!=ptr) avformat.avformat_free_context(ptr);}};
) {
AVFormatContext fmtCtx = Objects.requireNonNull(fmtCtxPtr.ptr);
fmtCtx.setAutoWrite(false);
fmtCtx.setAutoRead(false);
fmtCtx.oformat = format; fmtCtx.writeField("oformat");
AVStream st = avformat.avformat_new_stream(fmtCtx, codec);
if (null == st)
throw new IllegalStateException();
AVCodecContext c = st.codec;
if (null == c)
throw new IllegalStateException();
st.setAutoWrite(false);
fmtCtx.readField("nb_streams");
st.id = fmtCtx.nb_streams - 1; st.writeField("id");
assert st.id >= 0;
System.out.format("New stream: id=%d%n", st.id);
if (0 != (format.flags & AvformatLibrary.AVFMT_GLOBALHEADER)) {
c.flags |= AvcodecLibrary.CODEC_FLAG_GLOBAL_HEADER;
}
c.writeField("flags");
c.bit_rate = 64000; c.writeField("bit_rate");
int bestSampleRate;
if (null == codec.supported_samplerates) {
bestSampleRate = 44100;
} else {
bestSampleRate = 0;
for (int offset = 0, sample_rate = codec.supported_samplerates.getInt(offset); sample_rate != 0; codec.supported_samplerates.getInt(++offset)) {
bestSampleRate = Math.max(bestSampleRate, sample_rate);
}
assert bestSampleRate > 0;
}
c.sample_rate = bestSampleRate; c.writeField("sample_rate");
c.channel_layout = AvutilLibrary.AV_CH_LAYOUT_STEREO; c.writeField("channel_layout");
c.channels = avutil.av_get_channel_layout_nb_channels(c.channel_layout); c.writeField("channels");
assert 2 == c.channels;
c.sample_fmt = AvutilLibrary.AVSampleFormat.AV_SAMPLE_FMT_S16; c.writeField("sample_fmt");
c.time_base.num = 1;
c.time_base.den = bestSampleRate;
c.writeField("time_base");
c.setAutoWrite(false);
AudioFormat javaSoundFormat = new AudioFormat(bestSampleRate, Short.SIZE, c.channels, true, ByteOrder.nativeOrder() == ByteOrder.BIG_ENDIAN);
DataLine.Info javaDataLineInfo = new DataLine.Info(TargetDataLine.class, javaSoundFormat);
if (! AudioSystem.isLineSupported(javaDataLineInfo))
throw new IllegalStateException();
int err;
if ((err = avcodec.avcodec_open(c, codec)) < 0) {
throw new IllegalStateException();
}
assert c.channels != 0;
AVIOContext.ByReference[] ioCtxReference = new AVIOContext.ByReference[1];
if (0 != (err = avformat.avio_open(ioCtxReference, file_url, AvformatLibrary.AVIO_FLAG_WRITE))) {
throw new IllegalStateException("averror " + err);
}
try (
SharedPtr ioCtxPtr = new SharedPtr(ioCtxReference[0]) {@Override public void close(){if (null!=ptr) avutil.av_free(ptr.getPointer());}}
) {
AVIOContext.ByReference ioCtx = Objects.requireNonNull(ioCtxPtr.ptr);
fmtCtx.pb = ioCtx; fmtCtx.writeField("pb");
int averr = avformat.avformat_write_header(fmtCtx, null);
if (averr < 0) {
throw new IllegalStateException("" + averr);
}
st.read(); // it is modified by avformat_write_header
System.out.format("Wrote header. fmtCtx->nb_streams=%d, st->time_base=%d/%d; st->avg_frame_rate=%d/%d%n", fmtCtx.nb_streams, st.time_base.num, st.time_base.den, st.avg_frame_rate.num, st.avg_frame_rate.den);
avformat.avio_flush(ioCtx);
int frame_size = c.frame_size != 0 ? c.frame_size : 4096;
int expectedBufferSize = frame_size * c.channels * (Short.SIZE/8);
boolean supports_small_last_frame = c.frame_size == 0 ? true : 0 != (codec.capabilities & AvcodecLibrary.CODEC_CAP_SMALL_LAST_FRAME);
int bufferSize = avutil.av_samples_get_buffer_size((IntBuffer)null, c.channels, frame_size, c.sample_fmt, 1);
assert bufferSize == expectedBufferSize: String.format("expected %d; got %d", expectedBufferSize, bufferSize);
ByteBuffer samples = ByteBuffer.allocate(expectedBufferSize);
samples.order(ByteOrder.nativeOrder());
int audio_time = 0; // unit: (c.time_base) s = (1/c.sample_rate) s
int audio_sample_count = supports_small_last_frame ?
3 * c.sample_rate :
3 * c.sample_rate / frame_size * frame_size;
while (audio_time < audio_sample_count) {
int frame_audio_time = audio_time;
samples.clear();
int nb_samples_in_frame = 0;
// encode a single tone sound
for (; samples.hasRemaining() && audio_time < audio_sample_count; nb_samples_in_frame++, audio_time++) {
double x = 2*Math.PI*440/c.sample_rate * audio_time;
double y = 10000 * Math.sin(x);
samples.putShort((short) y);
samples.putShort((short) y);
}
samples.flip();
try (
SharedPtr<avframe> framePtr = new SharedPtr<avframe>(avcodec.avcodec_alloc_frame()) {@Override public void close() {if (null!=ptr) avutil.av_free(ptr.getPointer());}};
) {
AVFrame frame = Objects.requireNonNull(framePtr.ptr);
frame.setAutoRead(false); // will be an in param
frame.setAutoWrite(false);
frame.nb_samples = nb_samples_in_frame; frame.writeField("nb_samples"); // actually unused during encoding
// Presentation time, in AVStream.time_base units.
frame.pts = avutil.av_rescale_q(frame_audio_time, c.time_base, st.time_base); // i * codec_time_base / st_time_base
frame.writeField("pts");
assert c.channels > 0;
int bytesPerSample = avutil.av_get_bytes_per_sample(c.sample_fmt);
assert bytesPerSample > 0;
if (0 != (err = avcodec.avcodec_fill_audio_frame(frame, c.channels, c.sample_fmt, samples, samples.capacity(), 1))) {
throw new IllegalStateException(""+err);
}
AVPacket packet = new AVPacket(); // one of the few structs from ffmpeg with guaranteed size
avcodec.av_init_packet(packet);
packet.size = 0;
packet.data = null;
packet.stream_index = st.index; packet.writeField("stream_index");
// encode the samples
IntBuffer gotPacket = IntBuffer.allocate(1);
if (0 != (err = avcodec.avcodec_encode_audio2(c, packet, frame, gotPacket))) {
throw new IllegalStateException("" + err);
} else if (0 != gotPacket.get()) {
packet.read();
averr = avformat.av_write_frame(fmtCtx, packet);
if (averr < 0)
throw new IllegalStateException("" + averr);
}
System.out.format("encoded frame: codec time = %d; pts=%d = av_rescale_q(%d,%d/%d,%d/%d) (%.02fs) contains %d samples (%.02fs); got_packet=%d; packet.size=%d%n",
frame_audio_time,
frame.pts,
frame_audio_time, st.codec.time_base.num,st.codec.time_base.den,st.time_base.num,st.time_base.den,
1.*frame_audio_time/c.sample_rate, frame.nb_samples, 1.*frame.nb_samples/c.sample_rate, gotPacket.array()[0], packet.size);
}
}
if (0 != (err = avformat.av_write_trailer(fmtCtx))) {
throw new IllegalStateException();
}
avformat.avio_flush(ioCtx);
}
}
System.out.println("Done writing");
}
}
</avframe></avframe></avformatcontext></avformatcontext></t>I also rewrote it in C, and the C version works fine without any blips. But I can’t figure out how I am using the library differently ; all the library function calls should be identical !
//! gcc --std=c99 sin.c $(pkg-config --cflags --libs libavutil libavformat libavcodec) -o sin
// sudo apt-get install libswscale-dev
#include
#include
#include
#include
#include <libavutil></libavutil>opt.h>
#include <libavutil></libavutil>mathematics.h>
#include <libavformat></libavformat>avformat.h>
#include <libswscale></libswscale>swscale.h>
#include <libavcodec></libavcodec>avcodec.h>
int main(int argc, char *argv[]) {
const char *format_name = "wav", *file_url = "file:sin.wav";
avcodec_register_all();
av_register_all();
AVOutputFormat *format = NULL;
for (AVOutputFormat *formatIter = av_oformat_next(NULL); formatIter != NULL; formatIter = av_oformat_next(formatIter)) {
int hasEncoder = NULL != avcodec_find_encoder(formatIter->audio_codec);
if (0 == strcmp(format_name, formatIter->name)) {
format = formatIter;
break;
}
}
printf("Found format %s\n", format->name);
AVCodec *codec = avcodec_find_encoder(format->audio_codec);
if (! codec) {
fprintf(stderr, "Could not find codec %d\n", format->audio_codec);
exit(1);
}
AVFormatContext *fmtCtx = avformat_alloc_context();
if (! fmtCtx) {
fprintf(stderr, "error allocating AVFormatContext\n");
exit(1);
}
fmtCtx->oformat = format;
AVStream *st = avformat_new_stream(fmtCtx, codec);
if (! st) {
fprintf(stderr, "error allocating AVStream\n");
exit(1);
}
if (fmtCtx->nb_streams != 1) {
fprintf(stderr, "avformat_new_stream should have incremented nb_streams, but it's still %d\n", fmtCtx->nb_streams);
exit(1);
}
AVCodecContext *c = st->codec;
if (! c) {
fprintf(stderr, "avformat_new_stream should have allocated a AVCodecContext for my stream\n");
exit(1);
}
st->id = fmtCtx->nb_streams - 1;
printf("Created stream %d\n", st->id);
if (0 != (format->flags & AVFMT_GLOBALHEADER)) {
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
c->bit_rate = 64000;
int bestSampleRate;
if (NULL == codec->supported_samplerates) {
bestSampleRate = 44100;
printf("Setting sample rate: %d\n", bestSampleRate);
} else {
bestSampleRate = 0;
for (const int *sample_rate_iter = codec->supported_samplerates; *sample_rate_iter != 0; sample_rate_iter++) {
if (*sample_rate_iter >= bestSampleRate)
bestSampleRate = *sample_rate_iter;
}
printf("Using best supported sample rate: %d\n", bestSampleRate);
}
c->sample_rate = bestSampleRate;
c->channel_layout = AV_CH_LAYOUT_STEREO;
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->time_base.num = 1;
c->time_base.den = c->sample_rate;
if (c->channels != 2) {
fprintf(stderr, "av_get_channel_layout_nb_channels returned %d instead of 2\n", c->channels);
exit(1);
}
c->sample_fmt = AV_SAMPLE_FMT_S16;
int averr;
if ((averr = avcodec_open2(c, codec, NULL)) < 0) {
fprintf(stderr, "avcodec_open2 returned error %d\n", averr);
exit(1);
}
AVIOContext *ioCtx = NULL;
if (0 != (averr = avio_open(&ioCtx, file_url, AVIO_FLAG_WRITE))) {
fprintf(stderr, "avio_open returned error %d\n", averr);
exit(1);
}
if (ioCtx == NULL) {
fprintf(stderr, "AVIOContext should have been set by avio_open\n");
exit(1);
}
fmtCtx->pb = ioCtx;
if (0 != (averr = avformat_write_header(fmtCtx, NULL))) {
fprintf(stderr, "avformat_write_header returned error %d\n", averr);
exit(1);
}
printf("Wrote header. fmtCtx->nb_streams=%d, st->time_base=%d/%d; st->avg_frame_rate=%d/%d\n", fmtCtx->nb_streams, st->time_base.num, st->time_base.den, st->avg_frame_rate.num, st->avg_frame_rate.den);
int align = 1;
int sample_size = av_get_bytes_per_sample(c->sample_fmt);
if (sample_size != sizeof(int16_t)) {
fprintf(stderr, "expected sample size=%zu but got %d\n", sizeof(int16_t), sample_size);
exit(1);
}
int frame_size = c->frame_size != 0 ? c->frame_size : 4096;
int bufferSize = av_samples_get_buffer_size(NULL, c->channels, frame_size, c->sample_fmt, align);
int expectedBufferSize = frame_size * c->channels * sample_size;
int supports_small_last_frame = c->frame_size == 0 ? 1 : 0 != (codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME);
if (bufferSize != expectedBufferSize) {
fprintf(stderr, "expected buffer size=%d but got %d\n", expectedBufferSize, bufferSize);
exit(1);
}
int16_t *samples = (int16_t*)malloc(bufferSize);
uint32_t audio_time = 0; // unit: (1/c->sample_rate) s
uint32_t audio_sample_count = supports_small_last_frame ?
3 * c->sample_rate :
3 * c->sample_rate / frame_size * frame_size;
while (audio_time < audio_sample_count) {
uint32_t frame_audio_time = audio_time; // unit: (1/c->sample_rate) s
AVFrame *frame = avcodec_alloc_frame();
if (frame == NULL) {
fprintf(stderr, "avcodec_alloc_frame failed\n");
exit(1);
}
for (uint32_t i = 0; i != frame_size && audio_time < audio_sample_count; i++, audio_time++) {
samples[2*i] = samples[2*i + 1] = 10000 * sin(2*M_PI*440/c->sample_rate * audio_time);
frame->nb_samples = i+1; // actually unused during encoding
}
// frame->format = c->sample_fmt; // unused during encoding
frame->pts = av_rescale_q(frame_audio_time, c->time_base, st->time_base);
if (0 != (averr = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (const uint8_t*)samples, bufferSize, align))) {
fprintf(stderr, "avcodec_fill_audio_frame returned error %d\n", averr);
exit(1);
}
AVPacket packet;
av_init_packet(&packet);
packet.data = NULL;
packet.size = 0;
int got_packet;
if (0 != (averr = avcodec_encode_audio2(c, &packet, frame, &got_packet))) {
fprintf(stderr, "avcodec_encode_audio2 returned error %d\n", averr);
exit(1);
}
if (got_packet) {
packet.stream_index = st->index;
if (0 < (averr = av_write_frame(fmtCtx, &packet))) {
fprintf(stderr, "av_write_frame returned error %d\n", averr);
exit(1);
} else if (averr == 1) {
// end of stream wanted.
}
}
printf("encoded frame: codec time = %u; format pts=%ld = av_rescale_q(%u,%d/%d,%d/%d) (%.02fs) contains %d samples (%.02fs); got_packet=%d; packet.size=%d\n",
frame_audio_time,
frame->pts,
frame_audio_time, c->time_base.num, c->time_base.den, st->time_base.num, st->time_base.den,
1.*frame_audio_time/c->sample_rate, frame->nb_samples, 1.*frame->nb_samples/c->sample_rate, got_packet, packet.size);
av_free(frame);
}
free(samples);
cleanupFile:
if (0 != (averr = av_write_trailer(fmtCtx))) {
fprintf(stderr, "av_write_trailer returned error %d\n", averr);
exit(1);
}
avio_flush(ioCtx);
avio_close(ioCtx);
avformat_free_context(fmtCtx);
} -
How do I use ffmpeg with Python by passing File Objects (instead of locations to files on disk)
1er mai 2012, par Lyle PrattI'm trying to use ffmpeg with Python's subprocess module to convert some audio files. I grab the audio files from a URL and would like to just be able to pass the Python File Objects to ffmpeg, instead of first saving them to disk. It would also be very nice if I could just get back a file stream instead of having ffmpeg save the output to a file.
For reference, this is what I'm doing now :
tmp = "/dev/shm"
audio_wav_file = requests.get(audio_url)
## ## ##
## This is what I don't want to have to do ##
wavfile = open(tmp+filename, 'wrb')
wavfile.write(audio_wav_file.content)
wavfile.close()
## ## ##
conversion = subprocess.Popen('ffmpeg -i "'+tmp+filename+'" -y "'+tmp+filename_noext+'.flac" 2>&1', shell = True, stdout = subprocess.PIPE).stdout.read()Does anyone know how to do this ?
Thanks !
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Does 'keyint infinite' improve overall compression with crf and mb-tree ?
23 avril 2012, par HitomiTenshiAfter reading what intra-refresh does, I kinda got confused. I'm encoding for YouTube and YouTube will re-encode the video-file anyways, so I thought "why not removing all these heavy IDR-frames ?" I just want to compress my video as hard as possible, enabling everything that could help me getting a lower filesize, but still maintaining a high quality.
I used to test around stuff with lossless QP encoding, but the only thing I could max out was the merange. Here is a paste of my x264 settings : Pastebin.
I want to achieve highest compression while maintaining visually lossless quality. (using crf values around 10 - 13, and merange 32)
Could anyone give me advice on how to compress my video super hard (without touching the crf value !). I also want to know if it's true, that keyint infinite reduces overall compression.
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