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Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
Autres articles (62)
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MediaSPIP Core : La Configuration
9 novembre 2010, parMediaSPIP Core fournit par défaut trois pages différentes de configuration (ces pages utilisent le plugin de configuration CFG pour fonctionner) : une page spécifique à la configuration générale du squelettes ; une page spécifique à la configuration de la page d’accueil du site ; une page spécifique à la configuration des secteurs ;
Il fournit également une page supplémentaire qui n’apparait que lorsque certains plugins sont activés permettant de contrôler l’affichage et les fonctionnalités spécifiques (...) -
MediaSPIP v0.2
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Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
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Sur d’autres sites (11408)
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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);
} -
Revision 32042af14b : Renaming of segment constants. Renamed : MAX_MB_SEGMENTS to MAX_SEGMENTS MB_
23 juillet 2013, par Paul WilkinsChanged Paths :
Modify /vp9/common/vp9_loopfilter.c
Modify /vp9/common/vp9_loopfilter.h
Modify /vp9/common/vp9_onyx.h
Modify /vp9/common/vp9_pred_common.c
Modify /vp9/common/vp9_seg_common.h
Modify /vp9/decoder/vp9_decodemv.c
Modify /vp9/decoder/vp9_decodframe.c
Modify /vp9/encoder/vp9_bitstream.c
Modify /vp9/encoder/vp9_onyx_if.c
Modify /vp9/encoder/vp9_onyx_int.h
Modify /vp9/encoder/vp9_segmentation.c
Renaming of segment constants.Renamed :
MAX_MB_SEGMENTS to MAX_SEGMENTS
MB_SEG_TREE_PROBS to SEG_TREE_PROBSThe minimum unit for segmentation in the segment map
is now 8x8 so it is misleading to use MB_ as macro-block
traditionally refers to a 16x16 region.Change-Id : I0b55a6f0426bb46dd13435fcfa5bae0a30a7fa22
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Encoded images into H264 video are skipped and/or missing ?
25 juillet 2013, par JonaI'm trying to encode images into an H264 MP4 video. The issues I'm having is that some of the images are skipped or at the end of the video simply missing. I need the video to play every single image I encode since it is an animation.
Any help setting the encoder properly would be greatly appreciated !
Encoder settings :
AVCodecContext *c;
...
c->codec_id = AV_CODEC_ID_H264;
c->bit_rate = mOutputWidth*mOutputHeight*4;//400000;
/* Resolution must be a multiple of two. */
c->width = mOutputWidth;
c->height = mOutputHeight;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
c->time_base.den = mFps;
c->time_base.num = 1;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = AV_PIX_FMT_YUV420P;
...
av_dict_set(&pOptions, "preset", "medium", 0);
av_dict_set(&pOptions, "tune", "animation", 0);
/* open the codec */
ret = avcodec_open2(c, codec, &pOptions);
if (ret < 0) {
LOGE("Could not open video codec: %s", av_err2str(ret));
return -1;
}Update 07/24/13 :
I was able to achieve a better video by setting thegop_size=FPSand writing the last video frame repeatedlyFPS+1times seemed to resolve all issues. To me it seems odd to do that but might be something standard in the video encoding world ? Any tips feedback about this ?
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