
Recherche avancée
Médias (1)
-
The Great Big Beautiful Tomorrow
28 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Texte
Autres articles (12)
-
Organiser par catégorie
17 mai 2013, parDans MédiaSPIP, une rubrique a 2 noms : catégorie et rubrique.
Les différents documents stockés dans MédiaSPIP peuvent être rangés dans différentes catégories. On peut créer une catégorie en cliquant sur "publier une catégorie" dans le menu publier en haut à droite ( après authentification ). Une catégorie peut être rangée dans une autre catégorie aussi ce qui fait qu’on peut construire une arborescence de catégories.
Lors de la publication prochaine d’un document, la nouvelle catégorie créée sera proposée (...) -
Les thèmes de MediaSpip
4 juin 20133 thèmes sont proposés à l’origine par MédiaSPIP. L’utilisateur MédiaSPIP peut rajouter des thèmes selon ses besoins.
Thèmes MediaSPIP
3 thèmes ont été développés au départ pour MediaSPIP : * SPIPeo : thème par défaut de MédiaSPIP. Il met en avant la présentation du site et les documents média les plus récents ( le type de tri peut être modifié - titre, popularité, date) . * Arscenic : il s’agit du thème utilisé sur le site officiel du projet, constitué notamment d’un bandeau rouge en début de page. La structure (...) -
Selection of projects using MediaSPIP
2 mai 2011, parThe examples below are representative elements of MediaSPIP specific uses for specific projects.
MediaSPIP farm @ Infini
The non profit organizationInfini develops hospitality activities, internet access point, training, realizing innovative projects in the field of information and communication technologies and Communication, and hosting of websites. It plays a unique and prominent role in the Brest (France) area, at the national level, among the half-dozen such association. Its members (...)
Sur d’autres sites (3213)
-
Video has no audio when played using HTML5
4 mai 2013, par vertigoelectricI have a media site where I can upload video files. After upload, they are converted to MP4 using ffmpeg and then ffmpeg2theora converts that to OGV.
I have recently discovered that some videos that I upload do not play any sound when using the HTML5 player in Firefox/Chrome, but do have sound when using the fallback Flash player in IE.
I've downloaded and checked the processed files and they both still have sound, so the issue seems to be related to the HTML5 player rather than loss of audio during processing.
I double-checked my ffmpeg command line and realized I didn't specify any audio options :
ffmpeg2 -i "inputfile" -vcodec libx264 -profile:v baseline -level 3 "outputfile" 2>&1'
Since only some of the files didn't play audio while some did, this led me to believe that the audio track was being copied directly during the processing and if the original file didn't have audio supported by HTML5, then it output file would have the same result.
I realized this means I have to force re-encoding of the audio stream, so I tried a few things but had no success with any.
I tried 'libfaac' as many online posts suggested, but this resulted in an 'unknown codec' error. Further reading led me to find that 'libfaac' was no longer included with ffmpeg builds. I must have gotten mine after that.
I tried the "Native AAC" codec by simply using the following options :
ffmpeg2 -i "inputfile" -vcodec libx264 -profile:v baseline -level 3 -acodec aac "outputfile" 2>&1'
With this I got the following error : "Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height".
The only codec that didn't give me some sort of error was 'libvo_aacenc'. However, even though the processing completed without error, I still got no audio when played in the HTML5 player.
What am I doing wrong ?
NOTE (just in case this is relevant) : This is currently being run on a Windows host using the ffmpeg executable, but we will soon be moving to a Linux host.
-
Grab partial .mp4 video from onlline source
28 juillet 2016, par FlemingjpI want to be able to grab a partial mp4 file from an online source so I can produce quick thumbnails.
My idea was to pull a patial mp4 file from a http request using the header
Range: 'bytes=1000-'
However when I parse the data it doesn’t have the mp4 container therefore is corrupt. In Firefox it handles this easily as seeking to an unloaded part of the video is very quick.
How can I pull partial mp4 clips ? For example from here I could get a clip that is 0:28-0:30
-
Sending raw h264 video and aac audio frames to an RTMP server using ffmpeg
15 août 2016, par codeimpalerI am receiving raw h264 and aac audio frames from an even driven source. I am trying to send these frames to an rtmp server.
I started working from the ffmpeg example muxing.c which successfully sends a custom stream to the rtmp server. I figure I just need to replace their frame data with my own.I found this suggestion online. I have tried How to pack raw h264 stream to flv container and send over rtmp using ffmpeg (not command)
and
How to publish selfmade stream with ffmpeg and c++ to rtmp server ?
and a few other suggestions but none have worked for me.
I have tried to directly memcpy my byte buffer but my code keeps failing
at ret = avcodec_encode_video2(c, &pkt, frame, &got_packet).
Specifically, I get an invalid access error.
For a little more context, anytime I receive a frame (which is event driven), void RTMPWriter::WriteVideoFrame(...) is called. Assume the constructor has already been called before the first frame is received.
I am not that familiar with ffmpeg and there could be several things wrong with the code. Any input will be really appreciated.#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
RTMPWriter::RTMPWriter()
: seenKeyFrame(false),
video_st({ 0 }),
audio_st({ 0 }),
have_video(0),
have_audio(0)
{
const char *filename;
AVCodec *audio_codec = NULL, *video_codec = NULL;
int ret;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
int i;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
avformat_network_init();
String^ StreamURL = "StreamURL";
String^ out_uri = safe_cast(ApplicationData::Current->LocalSettings->Values->Lookup(StreamURL));
std::wstring out_uriW(out_uri->Begin());
std::string out_uriA(out_uriW.begin(), out_uriW.end());
filename = out_uriA.c_str();
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, "flv", filename);
if (!oc)
{
OutputDebugString(L"Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
{
OutputDebugString(L"Could not allocate using MPEG.\n");
}
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
{
open_video(oc, video_codec, &video_st, opt);
}
if (have_audio)
{
open_audio(oc, audio_codec, &audio_st, opt);
}
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE))
{
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0)
{
OutputDebugString(L"Could not open ");
OutputDebugString(out_uri->Data());
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
if (ret < 0)
{
OutputDebugString(L"Error occurred when writing stream header \n");
}
}
void RTMPWriter::WriteVideoFrame(
boolean isKeyFrame,
boolean hasDiscontinuity,
UINT64 frameId,
UINT32 videoBufferLength,
BYTE *videoBytes)
{
int ret;
AVCodecContext *c;
AVFrame* frame;
int got_packet = 0;
AVPacket pkt = { 0 };
c = video_st.enc;
frame = get_video_frame(videoBufferLength, videoBytes);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
OutputDebugString(L"Error encoding video frame: \n")
}
if (got_packet)
{
ret = write_frame(oc, &c->time_base, video_st.st, &pkt);
}
else {
ret = 0;
}
if (ret < 0) {
OutputDebugString(L"Error while writing video frame: %s\n");
}
}
AVFrame * RTMPWriter::get_video_frame(
UINT32 videoBufferLength,
BYTE *videoBytes)
{
AVCodecContext *c = video_st.enc;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
if (!video_st.sws_ctx) {
video_st.sws_ctx = sws_getContext(c->width, c->height,
AV_PIX_FMT_YUV420P,
c->width, c->height,
c->pix_fmt,
SCALE_FLAGS, NULL, NULL, NULL);
if (!video_st.sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(video_st.tmp_frame, video_st.next_pts, c->width, c->height, videoBufferLength, videoBytes);
sws_scale(video_st.sws_ctx,
(const uint8_t * const *)video_st.tmp_frame->data, video_st.tmp_frame->linesize,
0, c->height, video_st.frame->data, video_st.frame->linesize);
}
else {
fill_yuv_image(video_st.frame, video_st.next_pts, c->width, c->height, videoBufferLength, videoBytes);
}
video_st.frame->pts = video_st.next_pts++;
return video_st.frame;
}
/* Prepare a dummy image. */
void RTMPWriter::fill_yuv_image(
AVFrame *pict,
int frame_index,
int width,
int height,
UINT32 videoBufferLength,
BYTE *videoBytes)
{
//int x, y, i, ret;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(pict);
if (ret < 0)
{
OutputDebugString(L"Unable to make piture writable");
}
memcpy(pict->data, videoBytes, videoBufferLength);
//i = frame_index;
///* Y */
//for (y = 0; y < height; y++)
// for (x = 0; x < width; x++)
// pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
///* Cb and Cr */
//for (y = 0; y < height / 2; y++) {
// for (x = 0; x < width / 2; x++) {
// pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
// pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
// }
//}
}
void RTMPWriter::WriteAudioFrame()
{
}
/* Add an output stream. */
void RTMPWriter::add_stream(
OutputStream *ost,
AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
OutputDebugString(L"Could not find encoder for '%s'\n");
//avcodec_get_name(codec_id));
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
if (!ost->st) {
OutputDebugString(L"Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams - 1;
c = avcodec_alloc_context3(*codec);
if (!c) {
OutputDebugString(L"Could not alloc an encoding context\n");
exit(1);
}
ost->enc = c;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = /*(AVRational)*/{ 1, c->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* 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. */
ost->st->time_base = /*(AVRational)*/{ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B-frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
AVFrame * RTMPWriter::alloc_audio_frame(
enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
if (!frame) {
OutputDebugString(L"Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
OutputDebugString(L"Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
void RTMPWriter::open_audio(
AVFormatContext *oc,
AVCodec *codec,
OutputStream *ost,
AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
OutputDebugString(L"Could not open audio codec: %s\n");// , av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
OutputDebugString(L"Could not copy the stream parameters\n");
exit(1);
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
OutputDebugString(L"Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int(ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int(ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int(ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
OutputDebugString(L"Failed to initialize the resampling context\n");
exit(1);
}
}
int RTMPWriter::write_frame(
AVFormatContext *fmt_ctx,
const AVRational *time_base,
AVStream *st,
AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
/* Write the compressed frame to the media file. */
//log_packet(fmt_ctx, pkt);
OutputDebugString(L"Actually sending video frame: %s\n");
return av_interleaved_write_frame(fmt_ctx, pkt);
}
AVFrame *RTMPWriter::alloc_picture(
enum AVPixelFormat pix_fmt,
int width,
int height)
{
AVFrame *picture;
int ret;
picture = av_frame_alloc();
if (!picture)
return NULL;
picture->format = pix_fmt;
picture->width = width;
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return picture;
}
void RTMPWriter::open_video(
AVFormatContext *oc,
AVCodec *codec,
OutputStream *ost,
AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->enc;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
OutputDebugString(L"Could not open video codec: %s\n");// , av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
OutputDebugString(L"Could not allocate video frame\n");
exit(1);
}
/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
ost->tmp_frame = NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
OutputDebugString(L"Could not allocate temporary picture\n");
exit(1);
}
}
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
OutputDebugString(L"Could not copy the stream parameters\n");
exit(1);
}
}
void RTMPWriter::close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_free_context(&ost->enc);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
RTMPWriter::~RTMPWriter()
{
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
/* free the stream */
avformat_free_context(oc);
}