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Autres articles (63)
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MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Mise à disposition des fichiers
14 avril 2011, parPar défaut, lors de son initialisation, MediaSPIP ne permet pas aux visiteurs de télécharger les fichiers qu’ils soient originaux ou le résultat de leur transformation ou encodage. Il permet uniquement de les visualiser.
Cependant, il est possible et facile d’autoriser les visiteurs à avoir accès à ces documents et ce sous différentes formes.
Tout cela se passe dans la page de configuration du squelette. Il vous faut aller dans l’espace d’administration du canal, et choisir dans la navigation (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)
Sur d’autres sites (8390)
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Matplotlib : 'module' object has no attribute 'FFMpegWriter' / 'Writer'
23 février 2015, par osnozI’m trying to animate a graph with Matplotlib, something which I’ve done on a previous system. My code, however, seems to fail with my current setup.
Here’s the problem :
Writer = animation.writers['ffmpeg']Traceback (most recent call last) :
File "/Users/oliversanders/Documents/Code/PyCharm/plottools/animationTest.py", line 17, in
Writer = animation.writers[’ffmpeg’]
AttributeError : ’module’ object has no attribute ’writers’Or alternatively :
mywriter = animation.FFMpegWriter(fps=15)Traceback (most recent call last) :
File "/Users/oliversanders/Documents/Code/PyCharm/plottools/animatedPointPlotter.py", line 101, in
mywriter = animation.FFMpegWriter(fps=15)
AttributeError : ’module’ object has no attribute ’FFMpegWriter’I’ve just re-installed matplotlib (1.4.2) and ffmpeg (2.5.3) for good measure. I’ve also deleted all .pyc files from matplotlib’s directory to make sure they aren’t messing anything up.
I’ve looked around but been unable to find a solution. See also :
Using FFmpeg and IPython, What could be wrong in saving the following animation in Python ?.Thanks in advance.
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Module not found, ffmpeg not found
5 juin 2020, par Arsh SuriI have being trying to use the ffmpeg module on anaconda for my recent project.
I am unable to import the package for some reason. I have added/installed ffmpeg to my path environment.
I have tried installing the ffmpeg to the anaconda working file, still it shows



ModuleNotFoundError Traceback (most recent call last)
 in 
----> 1 import ffmpeg



ModuleNotFoundError : No module named 'ffmpeg'



How do i proceed with this,


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Exoplayer with FFmpeg module and filtering crash with aac and alac audio formats
25 juin 2020, par Aleksej OtjanHave a code to play audio with exoplayer and ffmpeg decoder. It works. Then I was needed to add equalizer functionality. I did it with ffmpeg avfilters. But now, it crash at some audio formats(if dont use avfilters it works with this formats).


Decode func :


int decodePacket(AVCodecContext *context, AVPacket *packet,
 uint8_t *outputBuffer, int outputSize) {
 int result = 0;
 // Queue input data.
 result = avcodec_send_packet(context, packet);
 if (result) {
 logError("avcodec_send_packet", result);
 return result == AVERROR_INVALIDDATA ? DECODER_ERROR_INVALID_DATA
 : DECODER_ERROR_OTHER;
 }

 // Dequeue output data until it runs out.
 int outSize = 0;
 if (EQUALIZER != nullptr) {
 LOGE("INIT FILTER GRAPH");
 init_filter_graph(context, EQUALIZER);
 }

 while (true) {
 AVFrame *frame = av_frame_alloc();
 if (!frame) {
 LOGE("Failed to allocate output frame.");
 return -1;
 }
 result = avcodec_receive_frame(context, frame);
 if (result) {
 av_frame_free(&frame);
 if (result == AVERROR(EAGAIN)) {
 break;
 }
 logError("avcodec_receive_frame", result);
 return result;
 }

 // Resample output.
 AVSampleFormat sampleFormat = context->sample_fmt;
 int channelCount = context->channels;
 int channelLayout = context->channel_layout;
 int sampleRate = context->sample_rate;
 int sampleCount = frame->nb_samples;
 int dataSize = av_samples_get_buffer_size(NULL, channelCount, sampleCount,
 sampleFormat, 1);
 SwrContext *resampleContext;
 if (context->opaque) {
 resampleContext = (SwrContext *) context->opaque;
 } else {
 resampleContext = swr_alloc();
 av_opt_set_int(resampleContext, "in_channel_layout", channelLayout, 0);
 av_opt_set_int(resampleContext, "out_channel_layout", channelLayout, 0);
 av_opt_set_int(resampleContext, "in_sample_rate", sampleRate, 0);
 av_opt_set_int(resampleContext, "out_sample_rate", sampleRate, 0);
 av_opt_set_int(resampleContext, "in_sample_fmt", sampleFormat, 0);
 // The output format is always the requested format.
 av_opt_set_int(resampleContext, "out_sample_fmt",
 context->request_sample_fmt, 0);
 result = swr_init(resampleContext);
 if (result < 0) {
 logError("swr_init", result);
 av_frame_free(&frame);
 return -1;
 }
 context->opaque = resampleContext;
 }
 int inSampleSize = av_get_bytes_per_sample(sampleFormat);
 int outSampleSize = av_get_bytes_per_sample(context->request_sample_fmt);
 int outSamples = swr_get_out_samples(resampleContext, sampleCount);
 int bufferOutSize = outSampleSize * channelCount * outSamples;
 if (outSize + bufferOutSize > outputSize) {
 LOGE("Output buffer size (%d) too small for output data (%d).",
 outputSize, outSize + bufferOutSize);
 av_frame_free(&frame);
 return -1;
 }
 if (EQUALIZER != nullptr && graph != nullptr) {
 result = av_buffersrc_add_frame_flags(src, frame,AV_BUFFERSRC_FLAG_KEEP_REF);
 if (result < 0) {
 av_frame_unref(frame);
 LOGE("Error submitting the frame to the filtergraph:");
 return -1;
 }
 // Get all the filtered output that is available.
 result = av_buffersink_get_frame(sink, frame);
 LOGE("ERROR SWR %s", av_err2str(result));
 if (result == AVERROR(EAGAIN) || result == AVERROR_EOF) {
 av_frame_unref(frame);
 break;
 }
 if (result < 0) {
 av_frame_unref(frame);
 return -1;
 }
 result = swr_convert(resampleContext, &outputBuffer, bufferOutSize,
 (const uint8_t **) frame->data, frame->nb_samples);
 }else{
 result = swr_convert(resampleContext, &outputBuffer, bufferOutSize,
 (const uint8_t **) frame->data, frame->nb_samples);
 }

 av_frame_free(&frame);
 if (result < 0) {
 logError("swr_convert", result);
 return result;
 }
 int available = swr_get_out_samples(resampleContext, 0);
 if (available != 0) {
 LOGE("Expected no samples remaining after resampling, but found %d.",
 available);
 return -1;
 }
 outputBuffer += bufferOutSize;
 outSize += bufferOutSize;
 }
 avfilter_graph_free(&graph);
 return outSize;
}


Init graph func :


int init_filter_graph(AVCodecContext *dec_ctx, const char *eq) {
 char args[512];
 int ret = 0;
 graph = avfilter_graph_alloc();
 const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
 const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
 AVFilterInOut *outputs = avfilter_inout_alloc();
 AVFilterInOut *inputs = avfilter_inout_alloc();
 static const enum AVSampleFormat out_sample_fmts[] = {dec_ctx->request_sample_fmt,
 static_cast<const avsampleformat="avsampleformat">(-1)};
 static const int64_t out_channel_layouts[] = {static_cast(dec_ctx->channel_layout),
 -1};
 static const int out_sample_rates[] = {dec_ctx->sample_rate, -1};
 const AVFilterLink *outlink;
 AVRational time_base = dec_ctx->time_base;

 if (!outputs || !inputs || !graph) {
 ret = AVERROR(ENOMEM);
 goto end;
 }

 /* buffer audio source: the decoded frames from the decoder will be inserted here. */
 if (!dec_ctx->channel_layout)
 dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
 snprintf(args, sizeof(args),
 "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%" PRIx64,
 1, dec_ctx->sample_rate, dec_ctx->sample_rate,
 av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
 ret = avfilter_graph_create_filter(&src, abuffersrc, "in",
 args, NULL, graph);

 if (ret < 0) {
 LOGE("Cannot create audio buffer source\n");
 goto end;
 }

 /* buffer audio sink: to terminate the filter chain. */
 ret = avfilter_graph_create_filter(&sink, abuffersink, "out",
 NULL, NULL, graph);
 if (ret < 0) {
 LOGE("Cannot create audio buffer sink\n");
 goto end;
 }

 ret = av_opt_set_int_list(sink, "sample_fmts", out_sample_fmts, -1,
 AV_OPT_SEARCH_CHILDREN);
 if (ret < 0) {
 LOGE("Cannot set output sample format\n");
 goto end;
 }

 ret = av_opt_set_int_list(sink, "channel_layouts", out_channel_layouts, -1,
 AV_OPT_SEARCH_CHILDREN);
 if (ret < 0) {
 LOGE("Cannot set output channel layout\n");
 goto end;
 }

 ret = av_opt_set_int_list(sink, "sample_rates", out_sample_rates, -1,
 AV_OPT_SEARCH_CHILDREN);
 if (ret < 0) {
 LOGE("Cannot set output sample rate\n");
 goto end;
 }

 /*
 * Set the endpoints for the filter graph. The graph will
 * be linked to the graph described by filters_descr.
 */

 /*
 * The buffer source output must be connected to the input pad of
 * the first filter described by filters_descr; since the first
 * filter input label is not specified, it is set to "in" by
 * default.
 */
 outputs->name = av_strdup("in");
 outputs->filter_ctx = src;
 outputs->pad_idx = 0;
 outputs->next = NULL;

 /*
 * The buffer sink input must be connected to the output pad of
 * the last filter described by filters_descr; since the last
 * filter output label is not specified, it is set to "out" by
 * default.
 */
 inputs->name = av_strdup("out");
 inputs->filter_ctx = sink;
 inputs->pad_idx = 0;
 inputs->next = NULL;

 if ((ret = avfilter_graph_parse_ptr(graph, eq,
 &inputs, &outputs, NULL)) < 0) {
 goto end;
 }

 if ((ret = avfilter_graph_config(graph, NULL)) < 0)
 goto end;

 /* Print summary of the sink buffer
 * Note: args buffer is reused to store channel layout string */
 outlink = sink->inputs[0];
 av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
 LOGE("Output: srate:%dHz chlayout:%s\n",
 (int) outlink->sample_rate,
 args);
 end:
 avfilter_inout_free(&inputs);
 avfilter_inout_free(&outputs);
 return ret;
}
</const>

Crash when try to play aac, alac audio at this line :


result = swr_convert(resampleContext, &outputBuffer, bufferOutSize,(const uint8_t **) frame->data, frame->nb_samples);


with


Fatal signal 11 (SIGSEGV), code 1 (SEGV_MAPERR), fault addr 0x0 


but work fine when play mp3, flac. What is wrong ? Thx for help.


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