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

22 Dec '14

fixes ticket #1937

4 10

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] 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] 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 1/2] 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

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[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