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The Slip - Artworks
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
Autres articles (15)
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Ajouter notes et légendes aux images
7 février 2011, parPour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
Modification lors de l’ajout d’un média
Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...) -
Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...) -
Les vidéos
21 avril 2011, parComme les documents de type "audio", Mediaspip affiche dans la mesure du possible les vidéos grâce à la balise html5 .
Un des inconvénients de cette balise est qu’elle n’est pas reconnue correctement par certains navigateurs (Internet Explorer pour ne pas le nommer) et que chaque navigateur ne gère en natif que certains formats de vidéos.
Son avantage principal quant à lui est de bénéficier de la prise en charge native de vidéos dans les navigateur et donc de se passer de l’utilisation de Flash et (...)
Sur d’autres sites (4691)
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Progress with rtc.io
12 août 2014, par silviaAt the end of July, I gave a presentation about WebRTC and rtc.io at the WDCNZ Web Dev Conference in beautiful Wellington, NZ.
Putting that talk together reminded me about how far we have come in the last year both with the progress of WebRTC, its standards and browser implementations, as well as with our own small team at NICTA and our rtc.io WebRTC toolbox.
One of the most exciting opportunities is still under-exploited : the data channel. When I talked about the above slide and pointed out Bananabread, PeerCDN, Copay, PubNub and also later WebTorrent, that’s where I really started to get Web Developers excited about WebRTC. They can totally see the shift in paradigm to peer-to-peer applications away from the Server-based architecture of the current Web.
Many were also excited to learn more about rtc.io, our own npm nodules based approach to a JavaScript API for WebRTC.
We believe that the World of JavaScript has reached a critical stage where we can no longer code by copy-and-paste of JavaScript snippets from all over the Web universe. We need a more structured module reuse approach to JavaScript. Node with JavaScript on the back end really only motivated this development. However, we’ve needed it for a long time on the front end, too. One big library (jquery anyone ?) that does everything that anyone could ever need on the front-end isn’t going to work any longer with the amount of functionality that we now expect Web applications to support. Just look at the insane growth of npm compared to other module collections :
Packages per day across popular platforms (Shamelessly copied from : http://blog.nodejitsu.com/npm-innovation-through-modularity/) For those that – like myself – found it difficult to understand how to tap into the sheer power of npm modules as a font end developer, simply use browserify. npm modules are prepared following the CommonJS module definition spec. Browserify works natively with that and “compiles” all the dependencies of a npm modules into a single bundle.js file that you can use on the front end through a script tag as you would in plain HTML. You can learn more about browserify and module definitions and how to use browserify.
For those of you not quite ready to dive in with browserify we have prepared prepared the rtc module, which exposes the most commonly used packages of rtc.io through an “RTC” object from a browserified JavaScript file. You can also directly download the JavaScript file from GitHub.
Using rtc.io rtc JS library So, I hope you enjoy rtc.io and I hope you enjoy my slides and large collection of interesting links inside the deck, and of course : enjoy WebRTC ! Thanks to Damon, JEeff, Cathy, Pete and Nathan – you’re an awesome team !
On a side note, I was really excited to meet the author of browserify, James Halliday (@substack) at WDCNZ, whose talk on “building your own tools” seemed to take me back to the times where everything was done on the command-line. I think James is using Node and the Web in a way that would appeal to a Linux Kernel developer. Fascinating !!
The post Progress with rtc.io first appeared on ginger’s thoughts.
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Youtube Live streaming Audio function not working Android
13 avril 2017, par Muthukumar SubramaniamI am using youtube watch me github project. I recorded the live streaming video from android mobile to youtube. Live streaming videos works perfectly but the audio is not working. Please give your suggestions.
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Java.lang.UnsatisfiedLinkError : No implementation found for int [duplicate]
28 mars 2017, par Muthukumar SubramaniamThis question already has an answer here :
I executed youtube watch me android application project. I just add some classes in my project and build with ndk. I got the error like
java.lang.UnsatisfiedLinkError : No implementation found for int com.ephronsystem.mobilizerapp.Ffmpeg.encodeVideoFrame(byte[]) (tried Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeVideoFrame and Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeVideoFrame___3B).
My code :
package com.ephronsystem.mobilizerapp;
public class Ffmpeg {
static {
System.loadLibrary("ffmpeg");
}
public static native boolean init(int width, int height, int audio_sample_rate, String rtmpUrl);
public static native void shutdown();
// Returns the size of the encoded frame.
public static native int encodeVideoFrame(byte[] yuv_image);
public static native int encodeAudioFrame(short[] audio_data, int length);
}This is ffmpeg-jni.c
#include <android></android>log.h>
#include
#include
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libavutil/opt.h"
#ifdef __cplusplus
extern "C" {
#endif
JNIEXPORT jboolean JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_init(JNIEnv *env, jobject thiz,
jint width, jint height,
jint audio_sample_rate,
jstring rtmp_url);
JNIEXPORT void JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_shutdown(JNIEnv *env,
jobject thiz
);
JNIEXPORT jint JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeVideoFrame(JNIEnv
*env,
jobject thiz,
jbyteArray
yuv_image);
JNIEXPORT jint JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeAudioFrame(JNIEnv *env,
jobject thiz,
jshortArray audio_data,
jint length);
#ifdef __cplusplus
}
#endif
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, "ffmpeg-jni", __VA_ARGS__)
#define URL_WRONLY 2
static AVFormatContext *fmt_context;
static AVStream *video_stream;
static AVStream *audio_stream;
static int pts
= 0;
static int last_audio_pts = 0;
// Buffers for UV format conversion
static unsigned char *u_buf;
static unsigned char *v_buf;
static int enable_audio = 1;
static int64_t audio_samples_written = 0;
static int audio_sample_rate = 0;
// Stupid buffer for audio samples. Not even a proper ring buffer
#define AUDIO_MAX_BUF_SIZE 16384 // 2x what we get from Java
static short audio_buf[AUDIO_MAX_BUF_SIZE];
static int audio_buf_size = 0;
void AudioBuffer_Push(const short *audio, int num_samples) {
if (audio_buf_size >= AUDIO_MAX_BUF_SIZE - num_samples) {
LOGI("AUDIO BUFFER OVERFLOW: %i + %i > %i", audio_buf_size, num_samples,
AUDIO_MAX_BUF_SIZE);
return;
}
for (int i = 0; i < num_samples; i++) {
audio_buf[audio_buf_size++] = audio[i];
}
}
int AudioBuffer_Size() { return audio_buf_size; }
short *AudioBuffer_Get() { return audio_buf; }
void AudioBuffer_Pop(int num_samples) {
if (num_samples > audio_buf_size) {
LOGI("Audio buffer Pop WTF: %i vs %i", num_samples, audio_buf_size);
return;
}
memmove(audio_buf, audio_buf + num_samples, num_samples * sizeof(short));
audio_buf_size -= num_samples;
}
void AudioBuffer_Clear() {
memset(audio_buf, 0, sizeof(audio_buf));
audio_buf_size = 0;
}
static void log_callback(void *ptr, int level, const char *fmt, va_list vl) {
char x[2048];
vsnprintf(x, 2048, fmt, vl);
LOGI(x);
}
JNIEXPORT jboolean JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_init(JNIEnv *env, jobject thiz,
jint width, jint height,
jint audio_sample_rate_param,
jstring rtmp_url) {
avcodec_register_all();
av_register_all();
av_log_set_callback(log_callback);
fmt_context = avformat_alloc_context();
AVOutputFormat *ofmt = av_guess_format("flv", NULL, NULL);
if (ofmt) {
LOGI("av_guess_format returned %s", ofmt->long_name);
} else {
LOGI("av_guess_format fail");
return JNI_FALSE;
}
fmt_context->oformat = ofmt;
LOGI("creating video stream");
video_stream = av_new_stream(fmt_context, 0);
if (enable_audio) {
LOGI("creating audio stream");
audio_stream = av_new_stream(fmt_context, 1);
}
// Open Video Codec.
// ======================
AVCodec *video_codec = avcodec_find_encoder(AV_CODEC_ID_H264);
if (!video_codec) {
LOGI("Did not find the video codec");
return JNI_FALSE; // leak!
} else {
LOGI("Video codec found!");
}
AVCodecContext *video_codec_ctx = video_stream->codec;
video_codec_ctx->codec_id = video_codec->id;
video_codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;
video_codec_ctx->level = 31;
video_codec_ctx->width = width;
video_codec_ctx->height = height;
video_codec_ctx->pix_fmt = PIX_FMT_YUV420P;
video_codec_ctx->rc_max_rate = 0;
video_codec_ctx->rc_buffer_size = 0;
video_codec_ctx->gop_size = 12;
video_codec_ctx->max_b_frames = 0;
video_codec_ctx->slices = 8;
video_codec_ctx->b_frame_strategy = 1;
video_codec_ctx->coder_type = 0;
video_codec_ctx->me_cmp = 1;
video_codec_ctx->me_range = 16;
video_codec_ctx->qmin = 10;
video_codec_ctx->qmax = 51;
video_codec_ctx->keyint_min = 25;
video_codec_ctx->refs = 3;
video_codec_ctx->trellis = 0;
video_codec_ctx->scenechange_threshold = 40;
video_codec_ctx->flags |= CODEC_FLAG_LOOP_FILTER;
video_codec_ctx->me_method = ME_HEX;
video_codec_ctx->me_subpel_quality = 6;
video_codec_ctx->i_quant_factor = 0.71;
video_codec_ctx->qcompress = 0.6;
video_codec_ctx->max_qdiff = 4;
video_codec_ctx->time_base.den = 10;
video_codec_ctx->time_base.num = 1;
video_codec_ctx->bit_rate = 3200 * 1000;
video_codec_ctx->bit_rate_tolerance = 0;
video_codec_ctx->flags2 |= 0x00000100;
fmt_context->bit_rate = 4000 * 1000;
av_opt_set(video_codec_ctx, "partitions", "i8x8,i4x4,p8x8,b8x8", 0);
av_opt_set_int(video_codec_ctx, "direct-pred", 1, 0);
av_opt_set_int(video_codec_ctx, "rc-lookahead", 0, 0);
av_opt_set_int(video_codec_ctx, "fast-pskip", 1, 0);
av_opt_set_int(video_codec_ctx, "mixed-refs", 1, 0);
av_opt_set_int(video_codec_ctx, "8x8dct", 0, 0);
av_opt_set_int(video_codec_ctx, "weightb", 0, 0);
if (fmt_context->oformat->flags & AVFMT_GLOBALHEADER)
video_codec_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
LOGI("Opening video codec");
AVDictionary *vopts = NULL;
av_dict_set(&vopts, "profile", "main", 0);
//av_dict_set(&vopts, "vprofile", "main", 0);
av_dict_set(&vopts, "rc-lookahead", 0, 0);
av_dict_set(&vopts, "tune", "film", 0);
av_dict_set(&vopts, "preset", "ultrafast", 0);
av_opt_set(video_codec_ctx->priv_data, "tune", "film", 0);
av_opt_set(video_codec_ctx->priv_data, "preset", "ultrafast", 0);
av_opt_set(video_codec_ctx->priv_data, "tune", "film", 0);
int open_res = avcodec_open2(video_codec_ctx, video_codec, &vopts);
if (open_res < 0) {
LOGI("Error opening video codec: %i", open_res);
return JNI_FALSE; // leak!
}
// Open Audio Codec.
// ======================
if (enable_audio) {
AudioBuffer_Clear();
audio_sample_rate = audio_sample_rate_param;
AVCodec *audio_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (!audio_codec) {
LOGI("Did not find the audio codec");
return JNI_FALSE; // leak!
} else {
LOGI("Audio codec found!");
}
AVCodecContext *audio_codec_ctx = audio_stream->codec;
audio_codec_ctx->codec_id = audio_codec->id;
audio_codec_ctx->codec_type = AVMEDIA_TYPE_AUDIO;
audio_codec_ctx->bit_rate = 128000;
audio_codec_ctx->bit_rate_tolerance = 16000;
audio_codec_ctx->channels = 1;
audio_codec_ctx->profile = FF_PROFILE_AAC_LOW;
audio_codec_ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
audio_codec_ctx->sample_rate = 44100;
LOGI("Opening audio codec");
AVDictionary *opts = NULL;
av_dict_set(&opts, "strict", "experimental", 0);
open_res = avcodec_open2(audio_codec_ctx, audio_codec, &opts);
LOGI("audio frame size: %i", audio_codec_ctx->frame_size);
if (open_res < 0) {
LOGI("Error opening audio codec: %i", open_res);
return JNI_FALSE; // leak!
}
}
const jbyte *url = (*env)->GetStringUTFChars(env, rtmp_url, NULL);
// Point to an output file
if (!(ofmt->flags & AVFMT_NOFILE)) {
if (avio_open(&fmt_context->pb, url, URL_WRONLY) < 0) {
LOGI("ERROR: Could not open file %s", url);
return JNI_FALSE; // leak!
}
}
(*env)->ReleaseStringUTFChars(env, rtmp_url, url);
LOGI("Writing output header.");
// Write file header
if (avformat_write_header(fmt_context, NULL) != 0) {
LOGI("ERROR: av_write_header failed");
return JNI_FALSE;
}
pts = 0;
last_audio_pts = 0;
audio_samples_written = 0;
// Initialize buffers for UV format conversion
int frame_size = video_codec_ctx->width * video_codec_ctx->height;
u_buf = (unsigned char *) av_malloc(frame_size / 4);
v_buf = (unsigned char *) av_malloc(frame_size / 4);
LOGI("ffmpeg encoding init done");
return JNI_TRUE;
}
JNIEXPORT void JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_shutdown(JNIEnv
*env,
jobject thiz
) {
av_write_trailer(fmt_context);
avio_close(fmt_context
->pb);
avcodec_close(video_stream
->codec);
if (enable_audio) {
avcodec_close(audio_stream
->codec);
}
av_free(fmt_context);
av_free(u_buf);
av_free(v_buf);
fmt_context = NULL;
u_buf = NULL;
v_buf = NULL;
}
JNIEXPORT jint JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeVideoFrame(JNIEnv
*env,
jobject thiz,
jbyteArray
yuv_image) {
int yuv_length = (*env)->GetArrayLength(env, yuv_image);
unsigned char *yuv_data = (*env)->GetByteArrayElements(env, yuv_image, 0);
AVCodecContext *video_codec_ctx = video_stream->codec;
//LOGI("Yuv size: %i w: %i h: %i", yuv_length, video_codec_ctx->width, video_codec_ctx->height);
int frame_size = video_codec_ctx->width * video_codec_ctx->height;
const unsigned char *uv = yuv_data + frame_size;
// Convert YUV from NV12 to I420. Y channel is the same so we don't touch it,
// we just have to deinterleave UV.
for (
int i = 0;
i < frame_size / 4; i++) {
v_buf[i] = uv[i * 2];
u_buf[i] = uv[i * 2 + 1];
}
AVFrame source;
memset(&source, 0, sizeof(AVFrame));
source.data[0] =
yuv_data;
source.data[1] =
u_buf;
source.data[2] =
v_buf;
source.linesize[0] = video_codec_ctx->
width;
source.linesize[1] = video_codec_ctx->width / 2;
source.linesize[2] = video_codec_ctx->width / 2;
// only for bitrate regulation. irrelevant for sync.
source.
pts = pts;
pts++;
int out_length = frame_size + (frame_size / 2);
unsigned char *out = (unsigned char *) av_malloc(out_length);
int compressed_length = avcodec_encode_video(video_codec_ctx, out, out_length, &source);
(*env)->
ReleaseByteArrayElements(env, yuv_image, yuv_data,
0);
// Write to file too
if (compressed_length > 0) {
AVPacket pkt;
av_init_packet(&pkt);
pkt.
pts = last_audio_pts;
if (video_codec_ctx->coded_frame && video_codec_ctx->coded_frame->key_frame) {
pkt.flags |= 0x0001;
}
pkt.
stream_index = video_stream->index;
pkt.
data = out;
pkt.
size = compressed_length;
if (
av_interleaved_write_frame(fmt_context,
&pkt) != 0) {
LOGI("Error writing video frame");
}
} else {
LOGI("??? compressed_length <= 0");
}
last_audio_pts++;
av_free(out);
return
compressed_length;
}
JNIEXPORT jint JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeAudioFrame(JNIEnv
*env,
jobject thiz,
jshortArray
audio_data,
jint length
) {
if (!enable_audio) {
return 0;
}
short *audio = (*env)->GetShortArrayElements(env, audio_data, 0);
//LOGI("java audio buffer size: %i", length);
AVCodecContext *audio_codec_ctx = audio_stream->codec;
unsigned char *out = av_malloc(128000);
AudioBuffer_Push(audio, length
);
int total_compressed = 0;
while (
AudioBuffer_Size()
>= audio_codec_ctx->frame_size) {
AVPacket pkt;
av_init_packet(&pkt);
int compressed_length = avcodec_encode_audio(audio_codec_ctx, out, 128000,
AudioBuffer_Get());
total_compressed +=
compressed_length;
audio_samples_written += audio_codec_ctx->
frame_size;
int new_pts = (audio_samples_written * 1000) / audio_sample_rate;
if (compressed_length > 0) {
pkt.
size = compressed_length;
pkt.
pts = new_pts;
last_audio_pts = new_pts;
//LOGI("audio_samples_written: %i comp_length: %i pts: %i", (int)audio_samples_written, (int)compressed_length, (int)new_pts);
pkt.flags |= 0x0001;
pkt.
stream_index = audio_stream->index;
pkt.
data = out;
if (
av_interleaved_write_frame(fmt_context,
&pkt) != 0) {
LOGI("Error writing audio frame");
}
}
AudioBuffer_Pop(audio_codec_ctx
->frame_size);
}
(*env)->
ReleaseShortArrayElements(env, audio_data, audio,
0);
av_free(out);
return
total_compressed;
}