September 2013 - ffmpeg-devel

[PATCH 1/2] img2: added support for %t output pattern
by Yuval Adam 05 Dec '16

05 Dec '16

From: Yuval Adam <yuv.adm(a)gmail.com> The image2 muxer now supports timestamps in output filenames. When used in an output patterm '%t' will be replaced with the frames timestamp in hours, minutes and seconds (hh:mm:ss). Signed-off-by: Yuval Adam <yuv.adm(a)gmail.com> --- doc/muxers.texi | 5 +++++ libavformat/avformat.h | 6 ++++-- libavformat/img2.c | 13 ++++++++----- libavformat/utils.c | 32 +++++++++++++++++++++++++------- 4 files changed, 42 insertions(+), 14 deletions(-) diff --git a/doc/muxers.texi b/doc/muxers.texi index 60a72d6..72212fe 100644 --- a/doc/muxers.texi +++ b/doc/muxers.texi @@ -109,6 +109,11 @@ If the pattern contains "%d" or "%0@var{N}d", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential. +The pattern may also contain "%t", this string specifies the timestamp +of the frame in hours, minutes and seconds. Note that if two frames are written at the +same second, they will be overwritten, yielding a maximum output of 1 fps. +The "%t" and "%d" patterns may not be used simultaneously. + The pattern may contain a suffix which is used to automatically determine the format of the image files to write. diff --git a/libavformat/avformat.h b/libavformat/avformat.h index 2ed6b56..a9ab0cc 100644 --- a/libavformat/avformat.h +++ b/libavformat/avformat.h @@ -2067,7 +2067,8 @@ attribute_deprecated int find_info_tag(char *arg, int arg_size, const char *tag1 #endif /** - * Return in 'buf' the path with '%d' replaced by a number. + * Return in 'buf' the path with '%d' replaced by the frame number, + * and '%t' replaced by the frame timestamp. * * Also handles the '%0nd' format where 'n' is the total number * of digits and '%%'. @@ -2076,10 +2077,11 @@ attribute_deprecated int find_info_tag(char *arg, int arg_size, const char *tag1 * @param buf_size destination buffer size * @param path numbered sequence string * @param number frame number + * @param ts frame timestamp in seconds * @return 0 if OK, -1 on format error */ int av_get_frame_filename(char *buf, int buf_size, - const char *path, int number); + const char *path, int number, int ts); /** * Check whether filename actually is a numbered sequence generator. diff --git a/libavformat/img2.c b/libavformat/img2.c index bc35591..8012e5b 100644 --- a/libavformat/img2.c +++ b/libavformat/img2.c @@ -142,7 +142,7 @@ static int find_image_range(int *pfirst_index, int *plast_index, /* find the first image */ for(first_index = 0; first_index < 5; first_index++) { - if (av_get_frame_filename(buf, sizeof(buf), path, first_index) < 0){ + if (av_get_frame_filename(buf, sizeof(buf), path, first_index, 0) < 0){ *pfirst_index = *plast_index = 1; if (avio_check(buf, AVIO_FLAG_READ) > 0) @@ -165,7 +165,7 @@ static int find_image_range(int *pfirst_index, int *plast_index, else range1 = 2 * range; if (av_get_frame_filename(buf, sizeof(buf), path, - last_index + range1) < 0) + last_index + range1, 0) < 0) goto fail; if (avio_check(buf, AVIO_FLAG_READ) <= 0) break; @@ -310,7 +310,7 @@ static int read_packet(AVFormatContext *s1, AVPacket *pkt) if (s->img_number > s->img_last) return AVERROR_EOF; if (av_get_frame_filename(filename, sizeof(filename), - s->path, s->img_number)<0 && s->img_number > 1) + s->path, s->img_number, 0)<0 && s->img_number > 1) return AVERROR(EIO); for(i=0; i<3; i++){ if (avio_open2(&f[i], filename, AVIO_FLAG_READ, @@ -389,12 +389,15 @@ static int write_packet(AVFormatContext *s, AVPacket *pkt) VideoData *img = s->priv_data; AVIOContext *pb[3]; char filename[1024]; - AVCodecContext *codec= s->streams[ pkt->stream_index ]->codec; + AVStream *stream = s->streams[ pkt->stream_index ]; + AVCodecContext *codec = stream->codec; int i; + int ts = av_rescale_q(pkt->pts, stream->time_base, AV_TIME_BASE_Q) / AV_TIME_BASE; + if (!img->is_pipe) { if (av_get_frame_filename(filename, sizeof(filename), - img->path, img->img_number) < 0 && img->img_number>1 && !img->updatefirst) { + img->path, img->img_number, ts) < 0 && img->img_number>1 && !img->updatefirst) { av_log(s, AV_LOG_ERROR, "Could not get frame filename number %d from pattern '%s'\n", img->img_number, img->path); diff --git a/libavformat/utils.c b/libavformat/utils.c index f0df759..6db2db0 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -319,7 +319,7 @@ int av_append_packet(AVIOContext *s, AVPacket *pkt, int size) int av_filename_number_test(const char *filename) { char buf[1024]; - return filename && (av_get_frame_filename(buf, sizeof(buf), filename, 1)>=0); + return filename && (av_get_frame_filename(buf, sizeof(buf), filename, 1, 0)>=0); } AVInputFormat *av_probe_input_format3(AVProbeData *pd, int is_opened, int *score_ret) @@ -3774,15 +3774,16 @@ int find_info_tag(char *arg, int arg_size, const char *tag1, const char *info) #endif int av_get_frame_filename(char *buf, int buf_size, - const char *path, int number) + const char *path, int number, int ts) { const char *p; char *q, buf1[20], c; - int nd, len, percentd_found; + int nd, len, percent_found; + int hours, mins, secs; q = buf; p = path; - percentd_found = 0; + percent_found = 0; for(;;) { c = *p++; if (c == '\0') @@ -3800,9 +3801,9 @@ int av_get_frame_filename(char *buf, int buf_size, case '%': goto addchar; case 'd': - if (percentd_found) + if (percent_found) goto fail; - percentd_found = 1; + percent_found = 1; snprintf(buf1, sizeof(buf1), "%0*d", nd, number); len = strlen(buf1); if ((q - buf + len) > buf_size - 1) @@ -3810,6 +3811,23 @@ int av_get_frame_filename(char *buf, int buf_size, memcpy(q, buf1, len); q += len; break; + case 't': + if (percent_found) + goto fail; + percent_found = 1; + secs = ts % 60; + ts /= 60; + mins = ts % 60; + ts /= 60; + hours = ts; + snprintf(buf1, sizeof(buf1), + "%02d:%02d:%02d", hours, mins, secs); + len = strlen(buf1); + if ((q - buf + len) > buf_size - 1) + goto fail; + memcpy(q, buf1, len); + q += len; + break; default: goto fail; } @@ -3819,7 +3837,7 @@ int av_get_frame_filename(char *buf, int buf_size, *q++ = c; } } - if (!percentd_found) + if (!percent_found) goto fail; *q = '\0'; return 0; -- 1.7.2.2

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Re: [FFmpeg-devel] [PATCH] Flush last frame in WMA lossless decoder
by Mashiat Sarker Shakkhar 10 Apr '16

10 Apr '16

On 5/2/2012 11:56 AM, Michael Niedermayer wrote: > On Wed, May 02, 2012 at 03:42:41AM +0600, Mashiat Sarker Shakkhar wrote: >> On 5/2/2012 2:49 AM, Michael Niedermayer wrote: >>> On Sun, Apr 29, 2012 at 10:43:47PM +0200, Jakub Stachowski wrote: >>>> Hello >>>> >>>> Here is cleaned up version of my patches to flush last frames and >>>> skip right number of samples in WMA lossless decoder. This causes >>>> ffmpeg to output the same number of samples as wmal2pcm.exe which I >>>> use as reference. [...] Hi The patch looks OK to the extent of my apprehension. And for the record, I have tested this patch and it fixes luckynight.wma. Patch has cosmetic issues, but that's another matter. Shakkhar P.S. Jakub, why did you use git send-email to send the patch? I tried and failed to review your patch on Gmane.

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[PATCH] Support Ctrl+Break in ffmpeg.exe on Windows as if it was Ctrl+C
by Joe Wreschnig 30 Jun '15

30 Jun '15

On Windows, the Ctrl+Break key combination usually does what Ctrl+C does. It is more common for processes to send other processes Ctrl+Break rather than Ctrl+C, because sending Ctrl+C / SIGINT doesn't work if you started a child in a new process group. This patch makes ffmpeg.exe register what Windows calls a console control event handler, which allows it to intercept Ctrl+Break. It hands it off directly to the usual SIGINT/SIGTERM handler. The same function also processes closing the console window, mapping it to SIGTERM. Obviously, this is only enabled if compiling for a platform where SetConsoleCtrlHandler is available (i.e. modern Windows). (I've also submitted this as a ticket at https://ffmpeg.org/trac/ffmpeg/ticket/1463 but it was recommended I post it here also.)

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jpeg2000: unsupported marker 0xFF5F at pos 0x73
by Wolfgang Woehl 12 Jun '15

12 Jun '15

See fdfcde74_excerpt_000000-000023 (on ftp://upload.ffmpeg.org/MPlayer/incoming/ as suggested by ffplay). The dir contains a MXF (DCI specs) with 24 4K frames and a 1/4-sized proxy video of that segment. The source MXF is real-world and currently in distribution. It was apparently generated with a Clipster system and asdcp-info says http://pastebin.com/raw.php?i=FFnHrW6K about it. ffplay (3e0c78b as of today) says http://pastebin.com/raw.php?i=df5sJDth when trying to play the excerpt MXF and fails to generate frames -- essentially the same as for the source MXF. Yes, the console output is that garbled. Wolfgang

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[PATCH]Accept startcode 001 when remuxing H264 into mpegts
by Carl Eugen Hoyos 26 Feb '15

26 Feb '15

Hi! Attached patch fixes remuxing of h264 into mpeg-ts for some wtv input files. Please review, Carl Eugen

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[PATCH 1/2] lavfi: remove mp=ilpack filter
by Paul B Mahol 27 Jan '15

27 Jan '15

The filter does not work, asserts here. Signed-off-by: Paul B Mahol <onemda(a)gmail.com> --- doc/filters.texi | 1 - libavfilter/Makefile | 1 - libavfilter/libmpcodecs/vf_ilpack.c | 458 ------------------------------------ libavfilter/vf_mp.c | 2 - 4 files changed, 462 deletions(-) delete mode 100644 libavfilter/libmpcodecs/vf_ilpack.c Boring stuff removed.

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[PATCH] avfilter: port qp filter from libmpcodecs
by Paul B Mahol 14 Jan '15

14 Jan '15

Signed-off-by: Paul B Mahol <onemda(a)gmail.com> --- doc/filters.texi | 32 ++++++++++ libavfilter/Makefile | 1 + libavfilter/allfilters.c | 1 + libavfilter/vf_qp.c | 154 +++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 188 insertions(+) create mode 100644 libavfilter/vf_qp.c diff --git a/doc/filters.texi b/doc/filters.texi index 915f310..e7cac2d 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -6236,6 +6236,38 @@ The main purpose of setting @option{mp} to a chroma plane is to reduce CPU load and make pullup usable in realtime on slow machines. @end table +@section qp + +Quantization Parameter (QP) change filter. + +The filter accepts the following option: + +@table @option +@item qp +Set expression for quantization parameter. +@end table + +The expression is evaluated through the eval API and can contain, among others, +the following constants: + +@table @option +@item known +1 if index is not 129, 0 otherwise. + +@item qp +Sequentional index starting from -129 to 128. +@end table + +@subsection Examples + +@itemize +@item +Some equatition like: +@example +qp=2+2*sin(PI*qp) +@end example +@end itemize + @section removelogo Suppress a TV station logo, using an image file to determine which diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 198bf4c..5dc8389 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -174,6 +174,7 @@ OBJS-$(CONFIG_PIXDESCTEST_FILTER) += vf_pixdesctest.o OBJS-$(CONFIG_PP_FILTER) += vf_pp.o OBJS-$(CONFIG_PSNR_FILTER) += vf_psnr.o dualinput.o OBJS-$(CONFIG_PULLUP_FILTER) += vf_pullup.o +OBJS-$(CONFIG_QP_FILTER) += vf_qp.o OBJS-$(CONFIG_REMOVELOGO_FILTER) += bbox.o lswsutils.o lavfutils.o vf_removelogo.o OBJS-$(CONFIG_ROTATE_FILTER) += vf_rotate.o OBJS-$(CONFIG_SEPARATEFIELDS_FILTER) += vf_separatefields.o diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 2825304..69e6cab 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -169,6 +169,7 @@ void avfilter_register_all(void) REGISTER_FILTER(PP, pp, vf); REGISTER_FILTER(PSNR, psnr, vf); REGISTER_FILTER(PULLUP, pullup, vf); + REGISTER_FILTER(QP, qp, vf); REGISTER_FILTER(REMOVELOGO, removelogo, vf); REGISTER_FILTER(ROTATE, rotate, vf); REGISTER_FILTER(SAB, sab, vf); diff --git a/libavfilter/vf_qp.c b/libavfilter/vf_qp.c new file mode 100644 index 0000000..b5b122e --- /dev/null +++ b/libavfilter/vf_qp.c @@ -0,0 +1,154 @@ +/* + * Copyright (C) 2004 Michael Niedermayer <michaelni(a)gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <math.h> +#include "libavutil/eval.h" +#include "libavutil/imgutils.h" +#include "libavutil/pixdesc.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "formats.h" +#include "internal.h" +#include "video.h" + +typedef struct QPContext { + const AVClass *class; + char *qp_expr_str; + int8_t lut[257]; + int8_t *qp; + int h, qstride; +} QPContext; + +#define OFFSET(x) offsetof(QPContext, x) +#define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM + +static const AVOption qp_options[] = { + { "qp", "set qp expression", OFFSET(qp_expr_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, FLAGS }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(qp); + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + QPContext *s = ctx->priv; + int i; + + if (!s->qp_expr_str) + return 0; + + s->h = (inlink->h + 15) >> 4; + s->qstride = (inlink->w + 15) >> 4; + av_reallocp_array(&s->qp, s->qstride * s->h, sizeof(*s->qp)); + if (!s->qp) + return AVERROR(ENOMEM); + + for (i = -129; i < 128; i++) { + double var_values[] = { i != -129, i, 0 }; + static const char *var_names[] = { "known", "qp", NULL }; + double temp_val; + int ret; + + ret = av_expr_parse_and_eval(&temp_val, s->qp_expr_str, + var_names, var_values, + NULL, NULL, NULL, NULL, 0, 0, ctx); + if (ret < 0) + return ret; + + s->lut[i + 129] = lrintf(temp_val); + } + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + QPContext *s = ctx->priv; + AVFrame *out; + + if (!s->qp_expr_str || ctx->is_disabled) + return ff_filter_frame(outlink, in); + + out = av_frame_clone(in); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + + out->qscale_table = s->qp; + out->qstride = s->qstride; + + if (in->qscale_table) { + int y, x; + + for (y = 0; y < s->h; y++) + for (x = 0; x < out->qstride; x++) + out->qscale_table[x + out->qstride * y] = s->lut[129 + + ((int8_t)in->qscale_table[x + in->qstride * y])]; + } else { + int y, x, qp = s->lut[0]; + + for (y = 0; y < s->h; y++) + for (x = 0; x < out->qstride; x++) + out->qscale_table[x + out->qstride * y] = qp; + } + + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + QPContext *s = ctx->priv; + + av_freep(&s->qp); +} + +static const AVFilterPad qp_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_VIDEO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad qp_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_VIDEO, + }, + { NULL } +}; + +AVFilter avfilter_vf_qp = { + .name = "qp", + .description = NULL_IF_CONFIG_SMALL("Change Quantization Parameter."), + .priv_size = sizeof(QPContext), + .uninit = uninit, + .inputs = qp_inputs, + .outputs = qp_outputs, + .priv_class = &qp_class, + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, +}; -- 1.7.11.2

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[PATCH] cdxl: fix duration
by Piotr Bandurski 22 Dec '14

22 Dec '14

fixes ticket #1937

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Use ffmpeg as an Android stagefright plugin
by Chih-Wei Huang 20 Nov '14

20 Nov '14

Hello ffmpeg list, I'm planning to integrate the latest ffmpeg as a stagefright OMXPlugin with Android-x86 4.3. I found the stagefright-plugins developed by Michael Chen is a good start. (repo: https://github.com/omxcodec/stagefright-plugins.git) But it's based on android 4.0 (ICS). My plan is: * Update stagefright-plugins to work with stagefright of android 4.3. * Make it to be a libstagefrighthw.so plugin. * Create Android.mk to build ffmpeg libs. Before I go ahead, I'd like to know if anyone is doing a similar project to avoid duplicate work. Thanks a lot! -- Chih-Wei Android-x86 project http://www.android-x86.org

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One pass volume normalization (ebur128)
by Jan Ehrhardt 15 Jul '14

15 Jul '14

I am once again proposing a patch for one pass volume normalization base on ebur128, as I see this still did not make it into FFMpeg 2.0. My patch is heaveily based on Clément Bsch's proposal in http://permalink.gmane.org/gmane.comp.video.ffmpeg.devel/159978 We have been using this patch now for more than 4 months and 1800+ videos of approximately 1 hour have been transcoded with it. Part of our FFMpeg commandline reads as -filter_complex \ "[0:v]setpts=PTS-STARTPTS[v0];[0:a]asetpts=PTS-STARTPTS,ebur128=metadata=1,volume=metadata=lavfi.r128.I,ebur128[a0]" \ -map [v0] -map [a0] It uses the already present ebur128 meta injection to adjust the volume on the fly. What would be the objection to move this into the FFMpeg core, so I do not have to patch my FFMpeg every time I compile a new one? I applied the patch below to FFMpeg Release/v.2.0. Jan diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c index a2ac1e2..6372bb2 100644 --- a/libavfilter/af_volume.c +++ b/libavfilter/af_volume.c @@ -51,18 +51,24 @@ static const AVOption volume_options[] = { { "fixed", "select 8-bit fixed-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A|F, "precision" }, { "float", "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A|F, "precision" }, { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" }, + { "metadata", "set the metadata key for loudness normalization", OFFSET(metadata), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = A|F }, { NULL }, }; AVFILTER_DEFINE_CLASS(volume); +static void set_fixed_volume(VolumeContext *vol, double volume) +{ + vol->volume_i = (int)(volume * 256 + 0.5); + vol->volume = vol->volume_i / 256.0; +} + static av_cold int init(AVFilterContext *ctx) { VolumeContext *vol = ctx->priv; if (vol->precision == PRECISION_FIXED) { - vol->volume_i = (int)(vol->volume * 256 + 0.5); - vol->volume = vol->volume_i / 256.0; + set_fixed_volume(vol, vol->volume); av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n", vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10); } else { @@ -171,13 +177,13 @@ static av_cold void volume_init(VolumeContext *vol) switch (av_get_packed_sample_fmt(vol->sample_fmt)) { case AV_SAMPLE_FMT_U8: - if (vol->volume_i < 0x1000000) + if (vol->volume_i < 0x1000000 && !vol->metadata) vol->scale_samples = scale_samples_u8_small; else vol->scale_samples = scale_samples_u8; break; case AV_SAMPLE_FMT_S16: - if (vol->volume_i < 0x10000) + if (vol->volume_i < 0x10000 && !vol->metadata) vol->scale_samples = scale_samples_s16_small; else vol->scale_samples = scale_samples_s16; @@ -216,11 +222,30 @@ static int config_output(AVFilterLink *outlink) static int filter_frame(AVFilterLink *inlink, AVFrame *buf) { - VolumeContext *vol = inlink->dst->priv; - AVFilterLink *outlink = inlink->dst->outputs[0]; + AVFilterContext *ctx = inlink->dst; + VolumeContext *vol = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; int nb_samples = buf->nb_samples; AVFrame *out_buf; + if (vol->metadata) { + double loudness, new_volume, timestamp, mx; + AVDictionaryEntry *e; + mx = 20; + timestamp = (float)(1.0 * buf->pts / outlink->sample_rate); + mx = fmin(mx, timestamp); + e = av_dict_get(buf->metadata, vol->metadata, NULL, 0); + if (e) { + loudness = av_strtod(e->value, NULL); + if (loudness > -69) { + new_volume = fmax(-mx,fmin(mx,(-23 - loudness))); + av_log(NULL, AV_LOG_VERBOSE, "loudness=%f => %f => volume=%f\n", + loudness, new_volume, pow(10, new_volume / 20)); + set_fixed_volume(vol, pow(10, new_volume / 20)); + } + } + } + if (vol->volume == 1.0 || vol->volume_i == 256) return ff_filter_frame(outlink, buf); @@ -269,6 +294,12 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf) return ff_filter_frame(outlink, out_buf); } +static av_cold void uninit(AVFilterContext *ctx) +{ + VolumeContext *vol = ctx->priv; + av_opt_free(vol); +} + static const AVFilterPad avfilter_af_volume_inputs[] = { { .name = "default", @@ -294,6 +325,7 @@ AVFilter avfilter_af_volume = { .priv_size = sizeof(VolumeContext), .priv_class = &volume_class, .init = init, + .uninit = uninit, .inputs = avfilter_af_volume_inputs, .outputs = avfilter_af_volume_outputs, .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, diff --git a/libavfilter/af_volume.h b/libavfilter/af_volume.h index bd7932e..4deca9c 100644 --- a/libavfilter/af_volume.h +++ b/libavfilter/af_volume.h @@ -48,6 +48,7 @@ typedef struct VolumeContext { void (*scale_samples)(uint8_t *dst, const uint8_t *src, int nb_samples, int volume); int samples_align; + char *metadata; } VolumeContext; void ff_volume_init_x86(VolumeContext *vol); diff --git a/libavfilter/f_ebur128.c b/libavfilter/f_ebur128.c index 88d37e8..f4ce6d9 100644 --- a/libavfilter/f_ebur128.c +++ b/libavfilter/f_ebur128.c @@ -410,7 +410,7 @@ static av_cold int init(AVFilterContext *ctx) if (ebur128->loglevel != AV_LOG_INFO && ebur128->loglevel != AV_LOG_VERBOSE) { - if (ebur128->do_video || ebur128->metadata) + if (ebur128->do_video) ebur128->loglevel = AV_LOG_VERBOSE; else ebur128->loglevel = AV_LOG_INFO; @@ -689,7 +689,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) SET_META("LRA.high", ebur128->lra_high); } - av_log(ctx, ebur128->loglevel, "t: %-10s " LOG_FMT "\n", + av_log(ctx, ebur128->metadata || !ebur128->do_video ? AV_LOG_VERBOSE : ebur128->loglevel, "t: %-10s " LOG_FMT "\n", av_ts2timestr(pts, &outlink->time_base), loudness_400, loudness_3000, ebur128->integrated_loudness, ebur128->loudness_range); diff --git a/libavfilter/x86/af_volume_init.c b/libavfilter/x86/af_volume_init.c index 81d605f..fab5a03 100644 --- a/libavfilter/x86/af_volume_init.c +++ b/libavfilter/x86/af_volume_init.c @@ -39,7 +39,7 @@ av_cold void ff_volume_init_x86(VolumeContext *vol) enum AVSampleFormat sample_fmt = av_get_packed_sample_fmt(vol->sample_fmt); if (sample_fmt == AV_SAMPLE_FMT_S16) { - if (EXTERNAL_SSE2(mm_flags) && vol->volume_i < 32768) { + if (EXTERNAL_SSE2(mm_flags) && vol->volume_i < 32768 && !vol->metadata) { vol->scale_samples = ff_scale_samples_s16_sse2; vol->samples_align = 8; }

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