Recherche avancée
Autres articles (48)
-
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
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 (...) -
Automated installation script of MediaSPIP
25 avril 2011, parTo overcome the difficulties mainly due to the installation of server side software dependencies, an "all-in-one" installation script written in bash was created to facilitate this step on a server with a compatible Linux distribution.
You must have access to your server via SSH and a root account to use it, which will install the dependencies. Contact your provider if you do not have that.
The documentation of the use of this installation script is available here.
The code of this (...)
Sur d’autres sites (9135)
-
AttributeError : module 'librosa' has no attribute 'output'
31 mai 2024, par Aditya KumarI am using librosa 0.6 in anaconda and i have also installed ffmpeg but i am still getting this error


the code is


a = np.exp(spectrum) - 1
 p = 2 * np.pi * np.random.random_sample(spectrum.shape) - np.pi
 for i in range(50):
 S = a * np.exp(1j * p)
 x = librosa.istft(S)
 p = np.angle(librosa.stft(x, N_FFT))
 librosa.output.write_wav(outfile, x, sr)



-
A PHP Error was encountered Severity : Core Warning Message : Module 'ffmpeg' already loaded Filename : Unknown Line Number : 0 Backtrace
10 août 2020, par SumonGetting the following error in live



" 
A PHP Error was encountered

Severity : Core Warning

Message : Module 'ffmpeg' already loaded

Filename : Unknown Line Number : 0

Backtrace :".


But i did not receive this error in local host. I am using codeigniter 3. Need Some help..


-
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.


Syndiquer tout le site
