[FFMPEG] Попытка получить доступ к Ebur128Context-> integrated_loudness, но безуспешно
Я пытаюсь запустить ebur128Filter для аудиофайла.аналогично выполнению*
[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 <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <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; ///< how many times the corresponding value occurred
double energy; ///< E = 10^((L + 0.691) / 10)
double loudness; ///< L = -0.691 + 10 * log10(E)
};
struct integrator {
double *cache[MAX_CHANNELS]; ///< window of filtered samples (N ms)
int cache_pos; ///< focus on the last added bin in the cache array
double sum[MAX_CHANNELS]; ///< sum of the last N ms filtered samples (cache content)
int filled; ///< 1 if the cache is completely filled, 0 otherwise
double rel_threshold; ///< relative threshold
double sum_kept_powers; ///< sum of the powers (weighted sums) above absolute threshold
int nb_kept_powers; ///< number of sum above absolute threshold
struct hist_entry *histogram; ///< histogram of the powers, used to compute LRA and I
};
typedef struct EBUR128Context {
const AVClass *class; ///< AVClass context for log and options purpose
/* peak metering */
int peak_mode; ///< enabled peak modes
double *true_peaks; ///< true peaks per channel
double *sample_peaks; ///< sample peaks per channel
double *true_peaks_per_frame; ///< true peaks in a frame per channel
#if CONFIG_SWRESAMPLE
SwrContext *swr_ctx; ///< over-sampling context for true peak metering
double *swr_buf; ///< resampled audio data for true peak metering
int swr_linesize;
#endif
/* video */
int do_video; ///< 1 if video output enabled, 0 otherwise
int w, h; ///< size of the video output
struct rect text; ///< rectangle for the LU legend on the left
struct rect graph; ///< rectangle for the main graph in the center
struct rect gauge; ///< rectangle for the gauge on the right
AVFrame *outpicref; ///< output picture reference, updated regularly
int meter; ///< select a EBU mode between +9 and +18
int scale_range; ///< the range of LU values according to the meter
int y_zero_lu; ///< the y value (pixel position) for 0 LU
int y_opt_max; ///< the y value (pixel position) for 1 LU
int y_opt_min; ///< the y value (pixel position) for -1 LU
int *y_line_ref; ///< y reference values for drawing the LU lines in the graph and the gauge
/* audio */
int nb_channels; ///< number of channels in the input
double *ch_weighting; ///< channel weighting mapping
int sample_count; ///< 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]; ///< 3 input samples cache for each channel
double y[MAX_CHANNELS * 3]; ///< 3 pre-filter samples cache for each channel
double z[MAX_CHANNELS * 3]; ///< 3 RLB-filter samples cache for each channel
#define I400_BINS (48000 * 4 / 10)
#define I3000_BINS (48000 * 3)
struct integrator i400; ///< 400ms integrator, used for Momentary loudness (M), and Integrated loudness (I)
struct integrator i3000; ///< 3s integrator, used for Short term loudness (S), and Loudness Range (LRA)
/* I and LRA specific */
double integrated_loudness; ///< integrated loudness in LUFS (I)
double loudness_range; ///< loudness range in LU (LRA)
double lra_low, lra_high; ///< low and high LRA values
/* misc */
int loglevel; ///< log level for frame logging
int metadata; ///< whether or not to inject loudness results in frames
int dual_mono; ///< whether or not to treat single channel input files as dual-mono
double pan_law; ///< pan law value used to calculate dual-mono measurements
int target; ///< target level in LUFS used to set relative zero LU in visualization
int gauge_type; ///< whether gauge shows momentary or short
int scale; ///< 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(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 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, &dec, 0);
if (ret < 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)) < 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(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 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(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 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 < 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 < 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 < 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,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 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(&inputs);
avfilter_inout_free(&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 < p_end) {
// fputc(*p & 0xff, stdout);
// fputc(*p>>8 & 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])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == audio_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
if (ret < 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 < 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) < 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 < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
}
av_packet_unref(&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 < 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(&priv);
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && 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.
Может кто-нибудь подсказать мне, как мне поступить здесь.