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  • ffmpeg : Trying to access Ebur128Context->integrated_loudness but unsuccessful

    12 avril 2019, par Sourabh Jain

    [FFMPEG] Trying to access Ebur128Context->integrated_loudness but unsuccessful

    I am trying to run ebur128Filter on audio file . similar to be doing
    [http://ffmpeg.org/doxygen/2.6/f__ebur128_8c_source.html#l00135]

    ffmpeg -i sample.wav -filter_complex ebur128=peak=true -f null -

    result of which is :

    [Parsed_ebur128_0 @ 0x7f9d38403ec0] Summary:
    

    
Integrated loudness:
    
I: -15.5 LUFS
    
Threshold: -25.6 LUFS
    

    
Loudness range:
    
LRA: 1.5 LU
    
Threshold: -35.5 LUFS
    
LRA low: -16.3 LUFS
    
LRA high: -14.8 LUFS
    

    
True peak:
    
Peak: -0.4 dBFS 
    /*
    
 * Copyright (c) 2010 Nicolas George
    
 * Copyright (c) 2011 Stefano Sabatini
    
 * Copyright (c) 2012 Clément Bœsch
    
 *
    
 * Permission is hereby granted, free of charge, to any person obtaining a copy
    
 * of this software and associated documentation files (the "Software"), to deal
    
 * in the Software without restriction, including without limitation the rights
    
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
    
 * copies of the Software, and to permit persons to whom the Software is
    
 * furnished to do so, subject to the following conditions:
    
 *
    
 * The above copyright notice and this permission notice shall be included in
    
 * all copies or substantial portions of the Software.
    
 *
    
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
    
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
    
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
    
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
    
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
    
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
    
 * THE SOFTWARE.
    
 */
    

    
/**
    
 * @file
    
 * API example for audio decoding and filtering
    
 * @example filtering_audio.c
    
 */
    

    
#include 
    

    
#include <libavcodec></libavcodec>avcodec.h>
    
#include <libavformat></libavformat>avformat.h>
    
#include <libavfilter></libavfilter>buffersink.h>
    
#include <libavfilter></libavfilter>buffersrc.h>
    
#include <libavutil></libavutil>opt.h>
    

    
#define MAX_CHANNELS 63
    

    

    

    
static const char *filter_descr = "ebur128=peak=true";
    

    
static AVFormatContext *fmt_ctx;
    
static AVCodecContext *dec_ctx;
    
AVFilterContext *buffersink_ctx;
    
AVFilterContext *buffersrc_ctx;
    
AVFilterGraph *filter_graph;
    
static int audio_stream_index = -1;
    

    
struct rect { int x, y, w, h; };
    

    

    
struct hist_entry {
    
    int count;                      ///&lt; how many times the corresponding value occurred
    
    double energy;                  ///&lt; E = 10^((L + 0.691) / 10)
    
    double loudness;                ///&lt; L = -0.691 + 10 * log10(E)
    
};
    

    

    
struct integrator {
    
    double *cache[MAX_CHANNELS];    ///&lt; window of filtered samples (N ms)
    
    int cache_pos;                  ///&lt; focus on the last added bin in the cache array
    
    double sum[MAX_CHANNELS];       ///&lt; sum of the last N ms filtered samples (cache content)
    
    int filled;                     ///&lt; 1 if the cache is completely filled, 0 otherwise
    
    double rel_threshold;           ///&lt; relative threshold
    
    double sum_kept_powers;         ///&lt; sum of the powers (weighted sums) above absolute threshold
    
    int nb_kept_powers;             ///&lt; number of sum above absolute threshold
    
    struct hist_entry *histogram;   ///&lt; histogram of the powers, used to compute LRA and I
    
};
    

    
typedef struct EBUR128Context {
    
    const AVClass *class;           ///&lt; AVClass context for log and options purpose
    

    
    /* peak metering */
    
    int peak_mode;                  ///&lt; enabled peak modes
    
    double *true_peaks;             ///&lt; true peaks per channel
    
    double *sample_peaks;           ///&lt; sample peaks per channel
    
    double *true_peaks_per_frame;   ///&lt; true peaks in a frame per channel
    
#if CONFIG_SWRESAMPLE
    
    SwrContext *swr_ctx;            ///&lt; over-sampling context for true peak metering
    
    double *swr_buf;                ///&lt; resampled audio data for true peak metering
    
    int swr_linesize;
    
#endif
    

    
    /* video  */
    
    int do_video;                   ///&lt; 1 if video output enabled, 0 otherwise
    
    int w, h;                       ///&lt; size of the video output
    
    struct rect text;               ///&lt; rectangle for the LU legend on the left
    
    struct rect graph;              ///&lt; rectangle for the main graph in the center
    
    struct rect gauge;              ///&lt; rectangle for the gauge on the right
    
    AVFrame *outpicref;             ///&lt; output picture reference, updated regularly
    
    int meter;                      ///&lt; select a EBU mode between +9 and +18
    
    int scale_range;                ///&lt; the range of LU values according to the meter
    
    int y_zero_lu;                  ///&lt; the y value (pixel position) for 0 LU
    
    int y_opt_max;                  ///&lt; the y value (pixel position) for 1 LU
    
    int y_opt_min;                  ///&lt; the y value (pixel position) for -1 LU
    
    int *y_line_ref;                ///&lt; y reference values for drawing the LU lines in the graph and the gauge
    

    
    /* audio */
    
    int nb_channels;                ///&lt; number of channels in the input
    
    double *ch_weighting;           ///&lt; channel weighting mapping
    
    int sample_count;               ///&lt; sample count used for refresh frequency, reset at refresh
    

    
    /* Filter caches.
    
     * The mult by 3 in the following is for X[i], X[i-1] and X[i-2] */
    
    double x[MAX_CHANNELS * 3];     ///&lt; 3 input samples cache for each channel
    
    double y[MAX_CHANNELS * 3];     ///&lt; 3 pre-filter samples cache for each channel
    
    double z[MAX_CHANNELS * 3];     ///&lt; 3 RLB-filter samples cache for each channel
    

    
#define I400_BINS  (48000 * 4 / 10)
    
#define I3000_BINS (48000 * 3)
    
    struct integrator i400;         ///&lt; 400ms integrator, used for Momentary loudness  (M), and Integrated loudness (I)
    
    struct integrator i3000;        ///&lt;    3s integrator, used for Short term loudness (S), and Loudness Range      (LRA)
    

    
    /* I and LRA specific */
    
    double integrated_loudness;     ///&lt; integrated loudness in LUFS (I)
    
    double loudness_range;          ///&lt; loudness range in LU (LRA)
    
    double lra_low, lra_high;       ///&lt; low and high LRA values
    

    
    /* misc */
    
    int loglevel;                   ///&lt; log level for frame logging
    
    int metadata;                   ///&lt; whether or not to inject loudness results in frames
    
    int dual_mono;                  ///&lt; whether or not to treat single channel input files as dual-mono
    
    double pan_law;                 ///&lt; pan law value used to calculate dual-mono measurements
    
    int target;                     ///&lt; target level in LUFS used to set relative zero LU in visualization
    
    int gauge_type;                 ///&lt; whether gauge shows momentary or short
    
    int scale;                      ///&lt; display scale type of statistics
    
} EBUR128Context;
    

    
void dump_ebur128_context(void *priv);
    

    
static int open_input_file(const char *filename)
    
{
    
    int ret;
    
    AVCodec *dec;
    

    
    if ((ret = avformat_open_input(&amp;fmt_ctx, filename, NULL, NULL)) &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
    
        return ret;
    
    }
    

    
    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
    
        return ret;
    
    }
    

    
    /* select the audio stream */
    
    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &amp;dec, 0);
    
    if (ret &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
    
        return ret;
    
    }
    
    audio_stream_index = ret;
    

    
    /* create decoding context */
    
    dec_ctx = avcodec_alloc_context3(dec);
    
    if (!dec_ctx)
    
        return AVERROR(ENOMEM);
    
    avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
    

    
    /* init the audio decoder */
    
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
    
        return ret;
    
    }
    

    
    return 0;
    
}
    

    
static int init_filters(const char *filters_descr)
    
{
    
    char args[512];
    
    int ret = 0;
    
    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[] = { AV_SAMPLE_FMT_S16, -1 };
    
    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
    
    static const int out_sample_rates[] = { 8000, -1 };
    
    const AVFilterLink *outlink;
    
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
    

    
    filter_graph = avfilter_graph_alloc();
    
    if (!outputs || !inputs || !filter_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,
    
             time_base.num, time_base.den, dec_ctx->sample_rate,
    
             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
    
    ret = avfilter_graph_create_filter(&amp;buffersrc_ctx, abuffersrc, "in",
    
                                       args, NULL, filter_graph);
    
    if (ret &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
    
        goto end;
    
    }
    

    
    /* buffer audio sink: to terminate the filter chain. */
    
    ret = avfilter_graph_create_filter(&amp;buffersink_ctx, abuffersink, "out",
    
                                       NULL, NULL, filter_graph);
    
    if (ret &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
    
        goto end;
    
    }
    

    
    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
    
                              AV_OPT_SEARCH_CHILDREN);
    
    if (ret &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
    
        goto end;
    
    }
    

    
    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
    
                              AV_OPT_SEARCH_CHILDREN);
    
    if (ret &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
    
        goto end;
    
    }
    

    
    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
    
                              AV_OPT_SEARCH_CHILDREN);
    
    if (ret &lt; 0) {
    
        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
    
        goto end;
    
    }
    

    
    /*
    
     * Set the endpoints for the filter graph. The filter_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 = buffersrc_ctx;
    
    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 = buffersink_ctx;
    
    inputs->pad_idx    = 0;
    
    inputs->next       = NULL;
    

    
    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
    
                                        &amp;inputs, &amp;outputs, NULL)) &lt; 0)
    
        goto end;
    

    
    if ((ret = avfilter_graph_config(filter_graph, NULL)) &lt; 0)
    
        goto end;
    

    
    /* Print summary of the sink buffer
    
     * Note: args buffer is reused to store channel layout string */
    
    outlink = buffersink_ctx->inputs[0];
    
    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
    
    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
    
           (int)outlink->sample_rate,
    
           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
    
           args);
    

    
end:
    
    avfilter_inout_free(&amp;inputs);
    
    avfilter_inout_free(&amp;outputs);
    

    
    return ret;
    
}
    

    
static void print_frame(const AVFrame *frame)
    
{
    
//    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
    
//    const uint16_t *p     = (uint16_t*)frame->data[0];
    
//    const uint16_t *p_end = p + n;
    
//
    
//    while (p &lt; p_end) {
    
//        fputc(*p    &amp; 0xff, stdout);
    
//        fputc(*p>>8 &amp; 0xff, stdout);
    
//        p++;
    
//    }
    
//    fflush(stdout);
    
}
    

    
int main(int argc, char **argv)
    
{
    
    av_log_set_level(AV_LOG_DEBUG);
    
    int ret;
    
    AVPacket packet;
    
    AVFrame *frame = av_frame_alloc();
    
    AVFrame *filt_frame = av_frame_alloc();
    

    
    if (!frame || !filt_frame) {
    
        perror("Could not allocate frame");
    
        exit(1);
    
    }
    

    

    
    if ((ret = open_input_file(argv[1])) &lt; 0)
    
        goto end;
    
    if ((ret = init_filters(filter_descr)) &lt; 0)
    
        goto end;
    

    
    /* read all packets */
    
    while (1) {
    
        if ((ret = av_read_frame(fmt_ctx, &amp;packet)) &lt; 0)
    
            break;
    

    
        if (packet.stream_index == audio_stream_index) {
    
            ret = avcodec_send_packet(dec_ctx, &amp;packet);
    
            if (ret &lt; 0) {
    
                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
    
                break;
    
            }
    

    
            while (ret >= 0) {
    
                ret = avcodec_receive_frame(dec_ctx, frame);
    
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
    
                    break;
    
                } else if (ret &lt; 0) {
    
                    av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
    
                    goto end;
    
                }
    

    
                if (ret >= 0) {
    
                    /* push the audio data from decoded frame into the filtergraph */
    
                    if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) &lt; 0) {
    
                        av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
    
                        break;
    
                    }
    

    
                    /* pull filtered audio from the filtergraph */
    
                    while (1) {
    
                        ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
    
                        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
    
                            break;
    
                        if (ret &lt; 0)
    
                            goto end;
    
                        print_frame(filt_frame);
    
                        av_frame_unref(filt_frame);
    
                    }
    
                    av_frame_unref(frame);
    
                }
    
            }
    
        }
    
        av_packet_unref(&amp;packet);
    
    }
    
    if(filter_graph->nb_filters){
    
    av_log(filter_graph, AV_LOG_INFO, "hello : %d \n",
    
                filter_graph->nb_filters);
    
    int i;
    
    for (int i = 0; i &lt; filter_graph->nb_filters; i++){
    
        av_log(filter_graph, AV_LOG_INFO, "name : %s \n",
    
                        filter_graph->filters[i]->name);
    
    }
    
    }
    

    
    av_log(filter_graph, AV_LOG_INFO, "name : %s \n",
    
                            filter_graph->filters[2]->name);
    
    void* priv = filter_graph->filters[2]->priv;
    

    
    dump_ebur128_context(&amp;priv);
    

    
end:
    

    

    
    avfilter_graph_free(&amp;filter_graph);
    
    avcodec_free_context(&amp;dec_ctx);
    
    avformat_close_input(&amp;fmt_ctx);
    
    av_frame_free(&amp;frame);
    
    av_frame_free(&amp;filt_frame);
    

    
    if (ret &lt; 0 &amp;&amp; ret != AVERROR_EOF) {
    
        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
    
        exit(1);
    
    }
    

    
    exit(0);
    
}
    

    
void dump_ebur128_context(void *priv){
    
    EBUR128Context *ebur128 = priv;
    

    
    av_log(ebur128, AV_LOG_INFO, "integrated_loudness : %5.1f \n",
    
                            ebur128->integrated_loudness);
    
    av_log(ebur128, AV_LOG_INFO, "lra_low : %5.1f \n",
    
                                ebur128->lra_low);
    
    av_log(ebur128, AV_LOG_INFO, "lra_high : %5.1f \n",
    
                                ebur128->lra_high);
    

    

    
}
    program fails while accessing integrated loudness in dump_ebur128_context.

    can someone guide me about , how I should proceed in here.

  • lavc : AV-prefix all codec flags

    29 juin 2015, par Vittorio Giovara
    lavc : AV-prefix all codec flags

    Convert doxygen to multiline and express bitfields more simply.
    

    Signed-off-by : Vittorio Giovara <vittorio.giovara@gmail.com>
    • [DBH] avconv.c
    • [DBH] avconv_opt.c
    • [DBH] avplay.c
    • [DBH] doc/examples/avcodec.c
    • [DBH] doc/examples/output.c
    • [DBH] doc/examples/transcode_aac.c
    • [DBH] libavcodec/4xm.c
    • [DBH] libavcodec/aacdec.c
    • [DBH] libavcodec/aacenc.c
    • [DBH] libavcodec/aacpsy.c
    • [DBH] libavcodec/ac3dec.c
    • [DBH] libavcodec/ac3enc.c
    • [DBH] libavcodec/ac3enc_float.c
    • [DBH] libavcodec/asvdec.c
    • [DBH] libavcodec/asvenc.c
    • [DBH] libavcodec/atrac1.c
    • [DBH] libavcodec/atrac3.c
    • [DBH] libavcodec/atrac3plusdec.c
    • [DBH] libavcodec/avcodec.h
    • [DBH] libavcodec/dcadec.c
    • [DBH] libavcodec/dnxhddata.c
    • [DBH] libavcodec/dnxhddec.c
    • [DBH] libavcodec/dnxhdenc.c
    • [DBH] libavcodec/dpxenc.c
    • [DBH] libavcodec/dump_extradata_bsf.c
    • [DBH] libavcodec/dvenc.c
    • [DBH] libavcodec/eamad.c
    • [DBH] libavcodec/eatgq.c
    • [DBH] libavcodec/eatqi.c
    • [DBH] libavcodec/ffv1enc.c
    • [DBH] libavcodec/h263.h
    • [DBH] libavcodec/h263dec.c
    • [DBH] libavcodec/h264.c
    • [DBH] libavcodec/h264_loopfilter.c
    • [DBH] libavcodec/h264_mb.c
    • [DBH] libavcodec/h264_mb_template.c
    • [DBH] libavcodec/h264_ps.c
    • [DBH] libavcodec/h264_slice.c
    • [DBH] libavcodec/hevc_ps.c
    • [DBH] libavcodec/huffyuvdec.c
    • [DBH] libavcodec/huffyuvenc.c
    • [DBH] libavcodec/imc.c
    • [DBH] libavcodec/ituh263enc.c
    • [DBH] libavcodec/jpeg2000dec.c
    • [DBH] libavcodec/libfaac.c
    • [DBH] libavcodec/libfdk-aacenc.c
    • [DBH] libavcodec/libmp3lame.c
    • [DBH] libavcodec/libopenh264enc.c
    • [DBH] libavcodec/libschroedingerenc.c
    • [DBH] libavcodec/libspeexenc.c
    • [DBH] libavcodec/libtheoraenc.c
    • [DBH] libavcodec/libtwolame.c
    • [DBH] libavcodec/libvo-aacenc.c
    • [DBH] libavcodec/libvorbis.c
    • [DBH] libavcodec/libvpxenc.c
    • [DBH] libavcodec/libx264.c
    • [DBH] libavcodec/libx265.c
    • [DBH] libavcodec/libxavs.c
    • [DBH] libavcodec/libxvid.c
    • [DBH] libavcodec/mdec.c
    • [DBH] libavcodec/mjpegenc_common.c
    • [DBH] libavcodec/motion_est.c
    • [DBH] libavcodec/mpeg12dec.c
    • [DBH] libavcodec/mpeg12enc.c
    • [DBH] libavcodec/mpeg4videodec.c
    • [DBH] libavcodec/mpeg4videoenc.c
    • [DBH] libavcodec/mpegaudiodec_template.c
    • [DBH] libavcodec/mpegvideo.c
    • [DBH] libavcodec/mpegvideo_enc.c
    • [DBH] libavcodec/mpegvideo_motion.c
    • [DBH] libavcodec/mpegvideo_xvmc.c
    • [DBH] libavcodec/nellymoserdec.c
    • [DBH] libavcodec/nellymoserenc.c
    • [DBH] libavcodec/nvenc.c
    • [DBH] libavcodec/on2avc.c
    • [DBH] libavcodec/options_table.h
    • [DBH] libavcodec/opus_celt.c
    • [DBH] libavcodec/parser.c
    • [DBH] libavcodec/pngenc.c
    • [DBH] libavcodec/proresenc.c
    • [DBH] libavcodec/pthread.c
    • [DBH] libavcodec/qsvenc.c
    • [DBH] libavcodec/ra288.c
    • [DBH] libavcodec/ratecontrol.c
    • [DBH] libavcodec/remove_extradata_bsf.c
    • [DBH] libavcodec/svq1dec.c
    • [DBH] libavcodec/svq3.c
    • [DBH] libavcodec/tiffenc.c
    • [DBH] libavcodec/twinvq.c
    • [DBH] libavcodec/utils.c
    • [DBH] libavcodec/vble.c
    • [DBH] libavcodec/vc1_block.c
    • [DBH] libavcodec/vc1_loopfilter.c
    • [DBH] libavcodec/vc1_mc.c
    • [DBH] libavcodec/vc1dec.c
    • [DBH] libavcodec/vorbisdec.c
    • [DBH] libavcodec/vorbisenc.c
    • [DBH] libavcodec/vp3.c
    • [DBH] libavcodec/wma.c
    • [DBH] libavcodec/wmaprodec.c
    • [DBH] libavcodec/wmv2.c
    • [DBH] libavcodec/x86/hpeldsp_init.c
    • [DBH] libavcodec/x86/me_cmp_init.c
    • [DBH] libavcodec/x86/mpegvideo.c
    • [DBH] libavcodec/x86/mpegvideoencdsp_init.c
    • [DBH] lib

  • Introducing Updates to the Funnels Feature

    29 mai 2024, par Erin

    We’ve made improvements to the Funnels feature to be more user-friendly and offer you greater flexibility. 

    &lt;script type=&quot;text/javascript&quot;&gt;<br />
           if ('function' === typeof window.playMatomoVideo){<br />
           window.playMatomoVideo(&quot;FunnelsProductUpdate2024&quot;, &quot;#FunnelsProductUpdate2024&quot;)<br />
           } else {<br />
           document.addEventListener(&quot;DOMContentLoaded&quot;, function() { window.playMatomoVideo(&quot;FunnelsProductUpdate2024&quot;, &quot;#FunnelsProductUpdate2024&quot;); });<br />
           }<br />
      &lt;/script&gt;

    Here’s what’s changing :

    Setting up and managing funnels is now easier than ever 

    Previously, creating funnels was tedious and required going through the Goals feature. But we’ve changed that with the introduction of a separate page to configure funnels. 

    Dedicated Manage Funnels page in Matomo

    Create funnels with greater flexibility—no longer tied to goals 

    Funnels is now a standalone feature, providing you with more flexibility. Before, you could only create a funnel if it was tied to a goal, in other words, the final step in the funnel had to be a goal. What’s more, you also couldn’t use goals for steps in the funnel.  

    Previous configuration requirements of Funnels in Matomo
    Previous configuration requirements of Funnels

    Now, funnels are independent of goals, and goals can serve as steps within the funnel. This means you have the freedom to configure any combination of steps in a funnel : 

    • All steps can be goals 
    • No steps need to be goals 
    • Or some steps can be goals, some steps can be events 
    Goals no longer required in Matomo Funnels

    No matter what your customer journey looks like, funnels now offer the versatility to meet your business’s specific needs. 

    Find friction points faster with intuitive visuals 

    One of the most significant improvements is the visual upgrade of the Funnels feature. The new Funnels graph is now visually in line with industry standards and intuitive. 

    New Funnel Analytics chart in Matomo

    The new visual provides a clearer view of your drop-off and conversion rates so you can instantly find points of friction in your funnel to improve the user experience and overall conversion rate.  

    This visualisation also provides a detailed overview of the number of visitors who enter, exit, skip, or proceed at each step of your funnel by using different coloured bars for visual clarity on each step’s performance. 

    With this update, we’ve also replaced ‘backfilled visits’ with ‘skipped steps’ to avoid misinterpretation of the data. 

    New data table for more granular insights 

    Accompanying this visual improvement is a new data table, allowing for more granular insights, segment comparison, and easy data export.

    We’ve also increased Funnel analysis limits. You can now compare funnel data for 2 date periods and 6 segments (up to 12 compared datasets in total). 

    Sign up for our newsletter to receive all the latest Matomo updates. 

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