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Elephants Dream - Cover of the soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Image
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Valkaama DVD Label
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Langue : English
Type : Image
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Publier une image simplement
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Mis à jour : Février 2012
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Type : Video
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Sur d’autres sites (5195)
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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; ///< 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.can someone guide me about , how I should proceed in here.
-
Corrupt playback of .flv when using JW Player
21 mai 2012, par Adam IngmanssonThis question is also asked on Audio-Video Production
I have some files that when played in jwplayer the playback is corrupt.
The files are encoded to h.264 using FFMpeg and there is other files encoded in the same way that works.At the moment we only use the flash version of the player.
The corrupt playback looks like this :
http://adam.ingmansson.com/public/jwplayer-corrupt-video.png
This problem started showing up after we did an upgrade to FFMpeg, so I haven't ruled out that it could be an encoding error.
the command used to run FFMpeg is :
ffmpeg
-i /home/ftp/1c8f08b7d0d9e7fa4b24066156ad50bc981497a0.mov
-vcodec libx264
-preset ultrafast
-profile baseline
-acodec libfaac
-ab 96k
-crf 19
-vf movie="/home/adam/logo.png [watermark]; [in][watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]"
-y /home/ftp/1c8f08b7d0d9e7fa4b24066156ad50bc981497a0.flvI am in no way an expert in FFMpeg commandline, so feel free to point out any mistakes made.
FFMpeg info :
ffmpeg version git-2012-05-02-2330eb1 Copyright (c) 2000-2012 the FFmpeg developers
built on May 3 2012 08:51:25 with gcc 4.4.3
configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-version3 --enable-x11grab
libavutil 51. 49.100 / 51. 49.100
libavcodec 54. 17.101 / 54. 17.101
libavformat 54. 3.100 / 54. 3.100
libavdevice 53. 4.100 / 53. 4.100
libavfilter 2. 72.103 / 2. 72.103
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 11.100 / 0. 11.100
libpostproc 52. 0.100 / 52. 0.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x24300e0] max_analyze_duration 5000000 reached at 5187000
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/ftp/javarecorder/1c8f08b7d0d9e7fa4b24066156ad50bc981497a0.mov':
Metadata:
major_brand : qt
minor_version : 537199360
compatible_brands: qt
creation_time : 2012-05-16 08:19:41
Duration: 00:13:33.00, start: 0.000000, bitrate: 2164 kb/s
Stream #0:0(eng): Video: qtrle (rle / 0x20656C72), rgb24, 1366x768, 1457 kb/s, 8.43 fps, 1k tbr, 1k tbn, 1k tbc
Metadata:
creation_time : 2012-05-16 08:19:41
handler_name : Apple Alias Data Handler
Stream #0:1(eng): Audio: pcm_s16be (twos / 0x736F7774), 44100 Hz, 1 channels, s16, 705 kb/s
Metadata:
creation_time : 2012-05-16 08:19:41
handler_name : Apple Alias Data Handler
Please use -profile:a or -profile:v, -profile is ambiguous
[buffer @ 0x2446ac0] w:1366 h:768 pixfmt:rgb24 tb:1/1000000 sar:0/1 sws_param:flags=2
[movie @ 0x242f840] seek_point:0 format_name:(null) file_name:/home/adam/logo.png stream_index:0
[overlay @ 0x2442840] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'Parsed_overlay_1'
[overlay @ 0x2442840] auto-inserting filter 'auto-inserted scale 1' between the filter 'Parsed_movie_0' and the filter 'Parsed_overlay_1'
[scale @ 0x24444a0] w:1366 h:768 fmt:rgb24 sar:0/1 -> w:1366 h:768 fmt:yuv420p sar:0/1 flags:0x4
[scale @ 0x2445100] w:80 h:80 fmt:rgba sar:1/1 -> w:80 h:80 fmt:yuva420p sar:1/1 flags:0x4
[overlay @ 0x2442840] main w:1366 h:768 fmt:yuv420p overlay x:1276 y:678 w:80 h:80 fmt:yuva420p
[overlay @ 0x2442840] main_tb:1/1000000 overlay_tb:1/25 -> tb:1/1000000 exact:1
[libx264 @ 0x242d8c0] MB rate (4128000) > level limit (2073600)
[libx264 @ 0x242d8c0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2
[libx264 @ 0x242d8c0] profile Constrained Baseline, level 5.2
[libx264 @ 0x242d8c0] 264 - core 124 r2197 69a0443 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=36 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=19.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
[libfaac @ 0x2443540] channel_layout not specified
Guessed Channel Layout for Input Stream #0.1 : mono
Output #0, flv, to '/home/ftp/javarecorder/1c8f08b7d0d9e7fa4b24066156ad50bc981497a0.flv':
Metadata:
major_brand : qt
minor_version : 537199360
compatible_brands: qt
creation_time : 2012-05-16 08:19:41
encoder : Lavf54.3.100
Stream #0:0(eng): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 1366x768, q=-1--1, 1k tbn, 1k tbc
Metadata:
creation_time : 2012-05-16 08:19:41
handler_name : Apple Alias Data Handler
Stream #0:1(eng): Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, mono, s16, 96 kb/s
Metadata:
creation_time : 2012-05-16 08:19:41
handler_name : Apple Alias Data Handler
Stream mapping:
Stream #0:0 -> #0:0 (qtrle -> libx264)
Stream #0:1 -> #0:1 (pcm_s16be -> libfaac)
Press [q] to stop, [?] for help
Input stream #0:1 frame changed from rate:44100 fmt:s16 ch:1 chl:0x0 to rate:44100 fmt:s16 ch:1 chl:0x4
frame= 6856 fps=105 q=-1.0 Lsize= 36030kB time=00:13:32.83 bitrate= 363.1kbits/s
video:27775kB audio:7540kB global headers:0kB muxing overhead 2.026555%
[libx264 @ 0x242d8c0] frame I:28 Avg QP: 4.61 size:238170
[libx264 @ 0x242d8c0] frame P:6828 Avg QP: 7.31 size: 3189
[libx264 @ 0x242d8c0] mb I I16..4: 100.0% 0.0% 0.0%
[libx264 @ 0x242d8c0] mb P I16..4: 0.9% 0.0% 0.0% P16..4: 2.2% 0.0% 0.0% 0.0% 0.0% skip:96.9%
[libx264 @ 0x242d8c0] coded y,uvDC,uvAC intra: 32.3% 30.0% 29.0% inter: 1.0% 1.1% 1.0%
[libx264 @ 0x242d8c0] i16 v,h,dc,p: 66% 32% 1% 1%
[libx264 @ 0x242d8c0] i8c dc,h,v,p: 62% 23% 14% 1%
[libx264 @ 0x242d8c0] kb/s:279.82EDIT :
A coworker was able to view a "corrupted" file. The only difference between my computer and his is that he have a Mac. Same flash version, same JW Player version. Something is not right here.
-
fftools/ffmpeg : delete the unused code.
18 novembre 2018, par Jun Zhao
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