August 2013 - ffmpeg-devel - lists.ffmpeg.org

[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.

3 9

[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.)

5 17

[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

2 3

[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.

6 13

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

22 Dec '14

fixes ticket #1937

4 10

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

6 20

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; }

9 46

drop entire frame when RTP packets are lost
by Martin Carroll 12 Jun '14

12 Jun '14

[Please forgive me if this issue has already been discussed -- I am new to ffmpeg-devel. I could not, however, find any discussion on this topic on this list in the past year, so here goes.] In the current design of ffplay/ffmpeg, when an RTP packet is lost, the containing -- and incomplete -- encoded frame is still passed along to the decoder. I have implemented a patch that has the read thread keep track of how many packets are missed, and when any are missed, the entire containing frame is dropped (that is, never makes it out of the read thread). I would be happy to send this patch upstream, or discuss it further if anyone likes. (The patch touches six files and about 25 lines of code.) Martin Carroll

4 20

[PATCH]Refuse to mux invalid transport streams
by Carl Eugen Hoyos 08 Jun '14

08 Jun '14

Hi! FFmpeg currently allows to mux transport streams that cannot be decoded, attached patch allows to report an error in such a case. Please comment, Carl Eugen

3 8

[PATCH 0/5] Scalar product, take 2
by Christophe Gisquet 06 Jun '14

06 Jun '14

Following the previous comments, I have opted for padding of the buffers in order to use the DSP functions. The troublesome cases for wma lossless are still untested. Patches 3 and 4 still try to address them, 3 being more secure than 4 (which is the shape of the best solution in my opinion). TAK decoder was modified in the same fashion. Christophe Gisquet (5): ra144: use scalarproduct_int16 wma lossless: reuse scalarproduct_and_madd_int16 wma lossless: switching back to C dsp function wma lossless: pad coeff buffer with 0 takdec: pad filter coeff buffer for DSP functions libavcodec/dsputil.c | 4 +-- libavcodec/dsputil.h | 3 +++ libavcodec/ra144.c | 14 ++++------ libavcodec/ra144.h | 6 ++++- libavcodec/ra144dec.c | 3 +++ libavcodec/ra144enc.c | 6 +++-- libavcodec/takdec.c | 33 +++++++----------------- libavcodec/wmalosslessdec.c | 63 ++++++++++++++++++++++++--------------------- 8 files changed, 65 insertions(+), 67 deletions(-) -- 1.8.0.msysgit.0

3 26

[PATCH] tests: add overlay rgb and yuv444 tests
by Stefano Sabatini 04 Jun '14

04 Jun '14

Also rename overlay test to overlay_yuv420 accordingly. --- tests/fate/avfilter.mak | 4 +++- tests/lavfi-regression.sh | 4 +++- tests/ref/lavfi/overlay | 1 - tests/ref/lavfi/overlay_rgb | 1 + tests/ref/lavfi/overlay_yuv420 | 1 + tests/ref/lavfi/overlay_yuv444 | 1 + 6 files changed, 9 insertions(+), 3 deletions(-) delete mode 100644 tests/ref/lavfi/overlay create mode 100644 tests/ref/lavfi/overlay_rgb create mode 100644 tests/ref/lavfi/overlay_yuv420 create mode 100644 tests/ref/lavfi/overlay_yuv444 diff --git a/tests/fate/avfilter.mak b/tests/fate/avfilter.mak index 15c8595..8ac8d41 100644 --- a/tests/fate/avfilter.mak +++ b/tests/fate/avfilter.mak @@ -14,7 +14,9 @@ FATE_LAVFI = fate-lavfi-alphaextract_rgb \ fate-lavfi-il \ fate-lavfi-life \ fate-lavfi-null \ - fate-lavfi-overlay \ + fate-lavfi-overlay_rgb \ + fate-lavfi-overlay_yuv420 \ + fate-lavfi-overlay_yuv444 \ fate-lavfi-pad \ fate-lavfi-pixfmts_copy \ fate-lavfi-pixfmts_crop \ diff --git a/tests/lavfi-regression.sh b/tests/lavfi-regression.sh index d6aba4c..9335124 100755 --- a/tests/lavfi-regression.sh +++ b/tests/lavfi-regression.sh @@ -47,7 +47,9 @@ do_lavfi "fade" "fade=in:5:15,fade=out:30:15" do_lavfi "hue" "hue=s=sin(2*PI*t)+1" do_lavfi "idet" "idet" do_lavfi "null" "null" -do_lavfi "overlay" "split[m],scale=88:72,pad=96:80:4:4[o2];[m]fifo[o1],[o1][o2]overlay=240:16" +do_lavfi "overlay_rgb" "split[m],scale=88:72,pad=96:80:4:4[o2];[m]fifo[o1],[o1][o2]overlay=240:16:format=rgb" +do_lavfi "overlay_yuv420" "split[m],scale=88:72,pad=96:80:4:4[o2];[m]fifo[o1],[o1][o2]overlay=240:16:format=yuv420" +do_lavfi "overlay_yuv444" "split[m],scale=88:72,pad=96:80:4:4[o2];[m]fifo[o1],[o1][o2]overlay=240:16:format=yuv444" do_lavfi "pad" "pad=iw*1.5:ih*1.5:iw*0.3:ih*0.2" do_lavfi "pp" "pp=be/hb/vb/tn/l5/al" do_lavfi "pp2" "pp=be/fq:16/h1/v1/lb" diff --git a/tests/ref/lavfi/overlay b/tests/ref/lavfi/overlay deleted file mode 100644 index ca70370..0000000 --- a/tests/ref/lavfi/overlay +++ /dev/null @@ -1 +0,0 @@ -overlay 4e85d551db9e56e8faddc1fc70fd6a00 diff --git a/tests/ref/lavfi/overlay_rgb b/tests/ref/lavfi/overlay_rgb new file mode 100644 index 0000000..ce4e30b --- /dev/null +++ b/tests/ref/lavfi/overlay_rgb @@ -0,0 +1 @@ +overlay_rgb abea47d9e61d7eab40c4734294519524 diff --git a/tests/ref/lavfi/overlay_yuv420 b/tests/ref/lavfi/overlay_yuv420 new file mode 100644 index 0000000..3d12251 --- /dev/null +++ b/tests/ref/lavfi/overlay_yuv420 @@ -0,0 +1 @@ +overlay_yuv420 4e85d551db9e56e8faddc1fc70fd6a00 diff --git a/tests/ref/lavfi/overlay_yuv444 b/tests/ref/lavfi/overlay_yuv444 new file mode 100644 index 0000000..b83e605 --- /dev/null +++ b/tests/ref/lavfi/overlay_yuv444 @@ -0,0 +1 @@ +overlay_yuv444 16215d0b4054a44bbe7f6c46bd97d235 -- 1.7.9.5

3 5

[PATCH] Fix nonsense non-mod16 AMV flipping code.
by Reimar Döffinger 02 Jun '14

02 Jun '14

It is obviously nonsense since it produces wrong results or even crashes (crashes should be encode-only though). Fixes trac issue #1092. Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> --- libavcodec/mjpegdec.c | 3 +-- libavcodec/mjpegenc.c | 2 +- libavcodec/mpegvideo_enc.c | 5 ----- 3 files changed, 2 insertions(+), 8 deletions(-) diff --git a/libavcodec/mjpegdec.c b/libavcodec/mjpegdec.c index a841384..fe54b45 100644 --- a/libavcodec/mjpegdec.c +++ b/libavcodec/mjpegdec.c @@ -972,8 +972,7 @@ static int mjpeg_decode_scan(MJpegDecodeContext *s, int nb_components, int Ah, s->coefs_finished[c] |= 1; if (s->flipped) { // picture should be flipped upside-down for this codec - int offset = (linesize[c] * (s->v_scount[i] * - (8 * s->mb_height - ((s->height / s->v_max) & 7)) - 1)); + int offset = linesize[c] * (s->v_scount[c] * s->height / s->v_max - 1); data[c] += offset; reference_data[c] += offset; linesize[c] *= -1; diff --git a/libavcodec/mjpegenc.c b/libavcodec/mjpegenc.c index 9326d4d..4788c47 100644 --- a/libavcodec/mjpegenc.c +++ b/libavcodec/mjpegenc.c @@ -466,7 +466,7 @@ static int amv_encode_picture(AVCodecContext *avctx, AVPacket *pkt, //picture should be flipped upside-down for(i=0; i < 3; i++) { - pic.data[i] += (pic.linesize[i] * (s->mjpeg_vsample[i] * (8 * s->mb_height -((s->height/V_MAX)&7)) - 1 )); + pic.data[i] += pic.linesize[i] * (s->mjpeg_vsample[i] * s->height / V_MAX - 1); pic.linesize[i] *= -1; } return ff_MPV_encode_picture(avctx, pkt, &pic, got_packet); diff --git a/libavcodec/mpegvideo_enc.c b/libavcodec/mpegvideo_enc.c index 6d89cfe..bb4be28 100644 --- a/libavcodec/mpegvideo_enc.c +++ b/libavcodec/mpegvideo_enc.c @@ -1012,11 +1012,6 @@ static int load_input_picture(MpegEncContext *s, AVFrame *pic_arg) uint8_t *src = pic_arg->data[i]; uint8_t *dst = pic->data[i]; - if(s->codec_id == CODEC_ID_AMV){ - av_assert0(!(s->avctx->flags & CODEC_FLAG_EMU_EDGE)); - h= ((s->height+15)/16*16)>>v_shift; - } - if (!s->avctx->rc_buffer_size) dst += INPLACE_OFFSET; -- 1.7.10

3 13

[PATCH 0/3] sbrdsp: SSE2 functions
by Christophe Gisquet 15 May '14

15 May '14

Drop SSE-only functions as much as possible to avoid any clutter. pre_shuffle and apply_noise were therefore affected. Christophe Gisquet (3): x86: sbrdsp: implement SSE2 qmf_pre_shuffle x86: sbrdsp: implement SSE qmf_deint_neg x86: sbrdsp: implement SSE2 hf_apply_noise libavcodec/aacsbrdata.h | 6 +- libavcodec/x86/sbrdsp.asm | 150 +++++++++++++++++++++++++++++++++++++++++++ libavcodec/x86/sbrdsp_init.c | 20 ++++++ 3 files changed, 175 insertions(+), 1 deletion(-) -- 1.8.0.msysgit.0

2 14

[PATCH] rtpdec: make the NTP time values unsigned.
by flameeyes＠gmail.com 09 May '14

09 May '14

As per RFC3550, section 4, the NTP time is provided as 64-bit unsigned integer, so follow the same logic here. --- libavformat/rtpdec.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h index 86af2b9..87fd419 100644 --- a/libavformat/rtpdec.h +++ b/libavformat/rtpdec.h @@ -148,8 +148,8 @@ struct RTPDemuxContext { RTPStatistics statistics; ///< Statistics for this stream (used by RTCP receiver reports) /* rtcp sender statistics receive */ - int64_t last_rtcp_ntp_time; // TODO: move into statistics - int64_t first_rtcp_ntp_time; // TODO: move into statistics + uint64_t last_rtcp_ntp_time; // TODO: move into statistics + uint64_t first_rtcp_ntp_time; // TODO: move into statistics uint32_t last_rtcp_timestamp; // TODO: move into statistics /* rtcp sender statistics */ -- 1.7.1.1

4 3

[PATCH 1/4] av_d2q: Add a special case for |value| > MAX and |value| < 1/MAX
by Michael Niedermayer 12 Apr '14

12 Apr '14

Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> --- libavutil/rational.c | 2 ++ tests/ref/fate/parseutils | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/libavutil/rational.c b/libavutil/rational.c index 912abc3..62473b4 100644 --- a/libavutil/rational.c +++ b/libavutil/rational.c @@ -115,6 +115,8 @@ AVRational av_d2q(double d, int max) exponent = FFMAX( (int)(log(fabs(d) + 1e-20)/LOG2), 0); den = 1LL << (61 - exponent); av_reduce(&a.num, &a.den, llrint(d * den), den, max); + if ((!a.num || !a.den) && d && max>0 && max<INT_MAX) + av_reduce(&a.num, &a.den, llrint(d * den), den, INT_MAX); return a; } diff --git a/tests/ref/fate/parseutils b/tests/ref/fate/parseutils index 5fa3bf4..1482452 100644 --- a/tests/ref/fate/parseutils +++ b/tests/ref/fate/parseutils @@ -21,7 +21,7 @@ Testing av_parse_video_rate() '.23' -> 23/100 OK '-.23' -> -23/100 ERROR '-0.234' -> -117/500 ERROR -'-0.0000001' -> 0/1 ERROR +'-0.0000001' -> -1/10000000 ERROR ' 21332.2324 ' -> 917286/43 OK ' -21332.2324 ' -> -917286/43 ERROR -- 1.7.9.5

3 9

trac for dshow "todos"
by Roger Pack 07 Mar '14

07 Mar '14

I was contemplating creating a list of "known lacking features" and bugs and wishlist items for dshow. My first thought had been to create it as a wiki page. Would that be ok or should I file "everything" as trac items assigned to myself, even if they might never be implemented, any opinions here? Thank you. -roger-

2 2

[PATCH]Support qclp in mov before stsd version 1, v2
by Carl Eugen Hoyos 21 Feb '14

21 Feb '14

Hi! Attached patch fixes the following sample as reported by Compn on irc: http://web.archive.org/web/20071027002243/http://www.americaone.org/video/i… As the original patch, it does not break samples with variable frame size like h263.mov Contrary to the original patch, this patch does not break stsd version 1 samples with bitrate < FULL. Please comment, Carl Eugen

3 14

ffmpeg configure for ios - using asm options causes MPx (mp1, mp2, mp3) codecs sound problems
by Murat Sudan 30 Jan '14

30 Jan '14

I've made many tests to get good performance for video decoding with ffmpeg on low cpu idevices such as iPad 1 and iPhone 3Gs, and see that using asm option when configuring ffmpeg for cross compiling for ios is very very important for performance.. But when I configure with --enable-asm, the audio stream which using mpx(mp1,mp2,mp) codec sounds squeaky. If I configure with --disable-asm option, than mpx(mp1,mp2,mp) codec sounds well but then the performance is %50 percent of the previous one built with the --enable-asm option... I've also tested these with the ffmpeg versions that are 0.8, 0.9, 0.10, 0.11, 1.0,1.1,1.2 All of them produces same results... Can anyone have the same problem with me ? Also, any idea how can I fix this issue ? please advice me .. (I can not find any patch to fix this issue on the net or in ffmpeg forums.) -mg

3 3

[PATCH 1/2] lavf: set codec time base after finding streams.
by Nicolas George 27 Jan '14

27 Jan '14

For most formats, it is done before reading packets, but for formats without global headers, new streams are added afterwards. Ensure that a time base is set even if no decoder was found. Useful for parsers. Signed-off-by: Nicolas George <nicolas.george(a)normalesup.org> --- libavformat/utils.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/libavformat/utils.c b/libavformat/utils.c index 0f13304..903ada3 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -2947,6 +2947,14 @@ int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options) compute_chapters_end(ic); + for(i = 0; i < ic->nb_streams; i++) { + st = ic->streams[i]; + if ((st->codec->codec_type == AVMEDIA_TYPE_VIDEO || + st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) && + !st->codec->time_base.num) + st->codec->time_base = st->time_base; + } + find_stream_info_err: for (i=0; i < ic->nb_streams; i++) { if (ic->streams[i]->codec) -- 1.7.10.4

5 8

[PATCH] img2enc: add option to use strftime() for filename
by Ramiro Polla 08 Jan '14

08 Jan '14

Hello, Attached patch adds option to use strftime() for img2enc filename, as such: ffmpeg -f video4linux2 -r 1 -re -i /dev/video0 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg" Ramiro

3 4

[PATCH] lavfi/volume: add dynamic expression evaluation
by Stefano Sabatini 25 Dec '13

25 Dec '13

TODO: update docs, bump micro --- libavfilter/af_volume.c | 120 ++++++++++++++++++++++++++++++++++++----------- libavfilter/af_volume.h | 20 ++++++++ 2 files changed, 113 insertions(+), 27 deletions(-) diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c index 5ffa1fe..87898c8 100644 --- a/libavfilter/af_volume.c +++ b/libavfilter/af_volume.c @@ -29,6 +29,7 @@ #include "libavutil/eval.h" #include "libavutil/float_dsp.h" #include "libavutil/opt.h" +#include "libavutil/time.h" #include "audio.h" #include "avfilter.h" #include "formats.h" @@ -39,13 +40,28 @@ static const char *precision_str[] = { "fixed", "float", "double" }; +static const char *const var_names[] = { + "N", ///< frame number (starting at zero) + "NB_CONSUMED_SAMPLES", ///< number of samples consumed by the filter (only audio) + "NB_SAMPLES", ///< number of samples in the current frame (only audio) + "POS", ///< original position in the file of the frame + "PTS", ///< original pts in the file of the frame + "RTC_STARTT", ///< start wallclock (RTC) time in micro seconds + "SAMPLE_RATE", ///< sample rate (only audio) + "STARTPTS", ///< PTS at start of stream + "STARTT", ///< time at start of stream + "T", ///< original time in the file of the frame + "TB", ///< timebase + NULL +}; + #define OFFSET(x) offsetof(VolumeContext, x) #define A AV_OPT_FLAG_AUDIO_PARAM #define F AV_OPT_FLAG_FILTERING_PARAM static const AVOption volume_options[] = { - { "volume", "set volume adjustment", - OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A|F }, + { "volume", "set volume adjustment expression", + OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F }, { "precision", "select mathematical precision", OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" }, { "fixed", "select 8-bit fixed-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A|F, "precision" }, @@ -68,21 +84,25 @@ static av_cold int init(AVFilterContext *ctx, const char *args) if ((ret = av_opt_set_from_string(vol, args, shorthand, "=", ":")) < 0) return ret; - if (vol->precision == PRECISION_FIXED) { - vol->volume_i = (int)(vol->volume * 256 + 0.5); - vol->volume = vol->volume_i / 256.0; - 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 { - av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n", - vol->volume, 20.0*log(vol->volume)/M_LN10, - precision_str[vol->precision]); + if ((ret = av_expr_parse(&vol->volume_pexpr,vol->volume_expr, + var_names, NULL, NULL, NULL, NULL, 0, ctx)) < 0) { + av_log(ctx, AV_LOG_ERROR, "Error while parsing expression '%s'\n", vol->volume_expr); + return ret; } - av_opt_free(vol); + vol->var_values[VAR_N ] = 0.0; + vol->var_values[VAR_STARTPTS] = vol->var_values[VAR_STARTT] = NAN; + return ret; } +static av_cold void uninit(AVFilterContext *ctx) +{ + VolumeContext *vol = ctx->priv; + av_expr_free(vol->volume_pexpr); + av_opt_free(vol); +} + static int query_formats(AVFilterContext *ctx) { VolumeContext *vol = ctx->priv; @@ -182,18 +202,6 @@ static void volume_init(VolumeContext *vol) vol->samples_align = 1; switch (av_get_packed_sample_fmt(vol->sample_fmt)) { - case AV_SAMPLE_FMT_U8: - if (vol->volume_i < 0x1000000) - 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) - vol->scale_samples = scale_samples_s16_small; - else - vol->scale_samples = scale_samples_s16; - break; case AV_SAMPLE_FMT_S32: vol->scale_samples = scale_samples_s32; break; @@ -221,20 +229,73 @@ static int config_output(AVFilterLink *outlink) vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout); vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1; - volume_init(vol); + vol->var_values[VAR_TB] = av_q2d(inlink->time_base); + vol->var_values[VAR_RTC_STARTT] = av_gettime(); + vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate; + + av_log(inlink->src, AV_LOG_VERBOSE, "TB:%f SAMPLE_RATE:%f RTC_STARTT:%f\n", + vol->var_values[VAR_TB], + vol->var_values[VAR_SAMPLE_RATE], + vol->var_values[VAR_RTC_STARTT]); + volume_init(vol); return 0; } +#define D2TS(d) (isnan(d) ? AV_NOPTS_VALUE : (int64_t)(d)) +#define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)) +#define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb)) + static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf) { + AVFilterContext *ctx = inlink->dst; VolumeContext *vol = inlink->dst->priv; AVFilterLink *outlink = inlink->dst->outputs[0]; int nb_samples = buf->audio->nb_samples; AVFilterBufferRef *out_buf; - if (vol->volume == 1.0 || vol->volume_i == 256) - return ff_filter_frame(outlink, buf); + if (isnan(vol->var_values[VAR_STARTPTS])) { + vol->var_values[VAR_STARTPTS] = TS2D(buf->pts); + vol->var_values[VAR_STARTT ] = TS2T(buf->pts, inlink->time_base); + } + vol->var_values[VAR_PTS] = TS2D(buf->pts); + vol->var_values[VAR_T ] = TS2T(buf->pts, inlink->time_base); + vol->var_values[VAR_POS] = buf->pos == -1 ? NAN : buf->pos; + vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL); + if (isnan(vol->volume)) { + av_log(ctx, AV_LOG_WARNING, "Invalid value NaN for volume, setting to 0\n"); + vol->volume = 0; + } + + if (vol->precision == PRECISION_FIXED) { + vol->volume_i = (int)(vol->volume * 256 + 0.5); + vol->volume = vol->volume_i / 256.0; + av_log(ctx, AV_LOG_DEBUG, "volume_i:%d/256 volume:%f volume_db:%1.2f\n", + vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10); + } else { + av_log(ctx, AV_LOG_DEBUG, "volume:%f volume_db:%1.2f\n", + vol->volume, 20.0*log(vol->volume)/M_LN10); + } + + if (vol->volume == 1.0 || vol->volume_i == 256) { + out_buf = buf; + goto end; + } + + switch (av_get_packed_sample_fmt(vol->sample_fmt)) { + case AV_SAMPLE_FMT_U8: + if (vol->volume_i < 0x1000000) + 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) + vol->scale_samples = scale_samples_s16_small; + else + vol->scale_samples = scale_samples_s16; + break; + } /* do volume scaling in-place if input buffer is writable */ if (buf->perms & AV_PERM_WRITE) { @@ -278,6 +339,10 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf) if (buf != out_buf) avfilter_unref_buffer(buf); +end: + vol->var_values[VAR_N] += 1.0; + vol->var_values[VAR_NB_CONSUMED_SAMPLES] += buf->audio->nb_samples; + return ff_filter_frame(outlink, out_buf); } @@ -305,6 +370,7 @@ AVFilter avfilter_af_volume = { .query_formats = query_formats, .priv_size = sizeof(VolumeContext), .init = init, + .uninit = uninit, .inputs = avfilter_af_volume_inputs, .outputs = avfilter_af_volume_outputs, .priv_class = &volume_class, diff --git a/libavfilter/af_volume.h b/libavfilter/af_volume.h index bd7932e..38e43a8 100644 --- a/libavfilter/af_volume.h +++ b/libavfilter/af_volume.h @@ -25,6 +25,7 @@ #define AVFILTER_AF_VOLUME_H #include "libavutil/common.h" +#include "libavutil/eval.h" #include "libavutil/float_dsp.h" #include "libavutil/opt.h" #include "libavutil/samplefmt.h" @@ -35,10 +36,29 @@ enum PrecisionType { PRECISION_DOUBLE, }; +enum VolumeVarName { + VAR_N, + VAR_NB_CONSUMED_SAMPLES, + VAR_NB_SAMPLES, + VAR_POS, + VAR_PTS, + VAR_RTC_STARTT, + VAR_SAMPLE_RATE, + VAR_STARTPTS, + VAR_STARTT, + VAR_T, + VAR_TB, + VAR_VARS_NB +}; + typedef struct VolumeContext { const AVClass *class; AVFloatDSPContext fdsp; enum PrecisionType precision; + const char *volume_expr; + AVExpr *volume_pexpr; + double var_values[VAR_VARS_NB]; + double volume; int volume_i; int channels; -- 1.7.9.5

5 14

[PATCH] dshow: allow for more codecs take 2
by Roger Pack 13 Dec '13

13 Dec '13

Ok after the previous discussion http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2012-October/131942.html it appears the first patch attached (0001 -- re-attached here and modified to apply cleaning to git master) is ok. Also attached is a second patch for better error logging messages. Thank you for your consideration in their regard. -roger-

5 26

[PATCH 0/5] Windows Resource File support
by jamal 26 Nov '13

26 Nov '13

This patchset adds support for Windows Resource files in Windows shared libraries. I thought about doing it for the executables as well, but I'd rather wait until i can get an icon for them. The first four commits add some missing version macros and header dependencies. With this the macros across all libraries will be consistent. This was done because they are needed by the rc files. I implemented this in all libraries except for libavresample. It's not enabled by default and the ffmpeg project isn't maintaining it. Regards jamal (5): avcodec: Add missing include to version.h postproc: Create version.h header swresample: Create version.h header Add missing version macros to libraries Add Windows Resource File support for shared libraries Makefile | 3 +- common.mak | 7 +++++- configure | 15 +++++++++-- libavcodec/Makefile | 3 ++ libavcodec/avcodecres.rc | 49 ++++++++++++++++++++++++++++++++++++++++ libavcodec/version.h | 2 + libavdevice/Makefile | 3 ++ libavdevice/avdeviceres.rc | 49 ++++++++++++++++++++++++++++++++++++++++ libavdevice/version.h | 2 + libavfilter/Makefile | 3 ++ libavfilter/avfilterres.rc | 49 ++++++++++++++++++++++++++++++++++++++++ libavfilter/version.h | 2 + libavformat/Makefile | 3 ++ libavformat/avformatres.rc | 49 ++++++++++++++++++++++++++++++++++++++++ libavutil/Makefile | 3 ++ libavutil/avutilres.rc | 49 ++++++++++++++++++++++++++++++++++++++++ libpostproc/Makefile | 6 ++++- libpostproc/postprocess.h | 18 +-------------- libpostproc/postprocessres.rc | 49 ++++++++++++++++++++++++++++++++++++++++ libpostproc/version.h | 47 ++++++++++++++++++++++++++++++++++++++ library.mak | 2 +- libswresample/Makefile | 6 ++++- libswresample/swresample.h | 8 +----- libswresample/swresampleres.rc | 49 ++++++++++++++++++++++++++++++++++++++++ libswresample/version.h | 45 ++++++++++++++++++++++++++++++++++++ libswscale/Makefile | 3 ++ libswscale/swscaleres.rc | 49 ++++++++++++++++++++++++++++++++++++++++ 27 files changed, 541 insertions(+), 32 deletions(-) create mode 100644 libavcodec/avcodecres.rc create mode 100644 libavdevice/avdeviceres.rc create mode 100644 libavfilter/avfilterres.rc create mode 100644 libavformat/avformatres.rc create mode 100644 libavutil/avutilres.rc create mode 100644 libpostproc/postprocessres.rc create mode 100644 libpostproc/version.h create mode 100644 libswresample/swresampleres.rc create mode 100644 libswresample/version.h create mode 100644 libswscale/swscaleres.rc -- 1.7.8.6

6 21

[PATCH] avutil: expand the size of LINE_SZ to 2048.
by Stephen Hutchinson 20 Nov '13

20 Nov '13

It's possible to make FFMPEG_CONFIGURATION exceed the 1024 limit, causing a cosmetic issue when invoking the ff* binaries. 2048 should be more than enough to fit it comfortably. --- libavutil/log.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavutil/log.c b/libavutil/log.c index 29ad2d0..9ad6875 100644 --- a/libavutil/log.c +++ b/libavutil/log.c @@ -39,7 +39,7 @@ #include "internal.h" #include "log.h" -#define LINE_SZ 1024 +#define LINE_SZ 2048 static int av_log_level = AV_LOG_INFO; static int flags; -- 1.8.1.2

5 20