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Head down (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Sur d’autres sites (9414)
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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.
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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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