April 2019 - ffmpeg-cvslog - lists.ffmpeg.org

avfilter/af_adeclick: switch to activate
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 22:19:15 2019 +0200| [e6c7d838ea90e569e2b03457a6c6637c6b315c10] | committer: Paul B Mahol avfilter/af_adeclick: switch to activate > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e6c7d838ea90e569e… --- libavfilter/af_adeclick.c | 143 +++++++++++++++++++++++++++------------------- 1 file changed, 83 insertions(+), 60 deletions(-) diff --git a/libavfilter/af_adeclick.c b/libavfilter/af_adeclick.c index bf0b7cb408..50eef74921 100644 --- a/libavfilter/af_adeclick.c +++ b/libavfilter/af_adeclick.c @@ -22,7 +22,9 @@ #include "libavutil/opt.h" #include "avfilter.h" #include "audio.h" +#include "filters.h" #include "formats.h" +#include "internal.h" typedef struct DeclickChannel { double *auxiliary; @@ -73,6 +75,7 @@ typedef struct AudioDeclickContext { uint64_t nb_samples; uint64_t detected_errors; int samples_left; + int eof; AVAudioFifo *fifo; double *window_func_lut; @@ -544,68 +547,57 @@ static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) return 0; } -static int filter_frame(AVFilterLink *inlink, AVFrame *in) +static int filter_frame(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; AVFilterLink *outlink = ctx->outputs[0]; AudioDeclickContext *s = ctx->priv; AVFrame *out = NULL; - int ret = 0; + int ret = 0, j, ch, detected_errors = 0; + ThreadData td; - if (s->pts == AV_NOPTS_VALUE) - s->pts = in->pts; - - ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data, - in->nb_samples); - av_frame_free(&in); - - while (av_audio_fifo_size(s->fifo) >= s->window_size) { - int j, ch, detected_errors = 0; - ThreadData td; - - out = ff_get_audio_buffer(outlink, s->hop_size); - if (!out) - return AVERROR(ENOMEM); + out = ff_get_audio_buffer(outlink, s->hop_size); + if (!out) + return AVERROR(ENOMEM); - ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, - s->window_size); - if (ret < 0) - break; + ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, + s->window_size); + if (ret < 0) + goto fail; - td.out = out; - ret = ctx->internal->execute(ctx, filter_channel, &td, NULL, inlink->channels); - if (ret < 0) - goto fail; + td.out = out; + ret = ctx->internal->execute(ctx, filter_channel, &td, NULL, inlink->channels); + if (ret < 0) + goto fail; - for (ch = 0; ch < s->in->channels; ch++) { - double *is = (double *)s->is->extended_data[ch]; + for (ch = 0; ch < s->in->channels; ch++) { + double *is = (double *)s->is->extended_data[ch]; - for (j = 0; j < s->hop_size; j++) { - if (is[j]) - detected_errors++; - } + for (j = 0; j < s->hop_size; j++) { + if (is[j]) + detected_errors++; } + } - av_audio_fifo_drain(s->fifo, s->hop_size); + av_audio_fifo_drain(s->fifo, s->hop_size); - if (s->samples_left > 0) - out->nb_samples = FFMIN(s->hop_size, s->samples_left); + if (s->samples_left > 0) + out->nb_samples = FFMIN(s->hop_size, s->samples_left); - out->pts = s->pts; - s->pts += s->hop_size; + out->pts = s->pts; + s->pts += s->hop_size; - s->detected_errors += detected_errors; - s->nb_samples += out->nb_samples * inlink->channels; + s->detected_errors += detected_errors; + s->nb_samples += out->nb_samples * inlink->channels; - ret = ff_filter_frame(outlink, out); - if (ret < 0) - break; + ret = ff_filter_frame(outlink, out); + if (ret < 0) + goto fail; - if (s->samples_left > 0) { - s->samples_left -= s->hop_size; - if (s->samples_left <= 0) - av_audio_fifo_drain(s->fifo, av_audio_fifo_size(s->fifo)); - } + if (s->samples_left > 0) { + s->samples_left -= s->hop_size; + if (s->samples_left <= 0) + av_audio_fifo_drain(s->fifo, av_audio_fifo_size(s->fifo)); } fail: @@ -614,27 +606,58 @@ fail: return ret; } -static int request_frame(AVFilterLink *outlink) +static int activate(AVFilterContext *ctx) { - AVFilterContext *ctx = outlink->src; + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; AudioDeclickContext *s = ctx->priv; - int ret = 0; + AVFrame *in; + int ret, status; + int64_t pts; - ret = ff_request_frame(ctx->inputs[0]); + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); - if (ret == AVERROR_EOF && av_audio_fifo_size(s->fifo) > 0) { - if (!s->samples_left) - s->samples_left = av_audio_fifo_size(s->fifo) - s->overlap_skip; + ret = ff_inlink_consume_samples(inlink, s->window_size, s->window_size, &in); + if (ret < 0) + return ret; + if (ret > 0) { + if (s->pts == AV_NOPTS_VALUE) + s->pts = in->pts; + + ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data, + in->nb_samples); + av_frame_free(&in); + if (ret < 0) + return ret; + } + + if (av_audio_fifo_size(s->fifo) >= s->window_size || + s->samples_left > 0) + return filter_frame(inlink); - if (s->samples_left > 0) { - AVFrame *in = ff_get_audio_buffer(outlink, s->window_size - s->samples_left); - if (!in) - return AVERROR(ENOMEM); - ret = filter_frame(ctx->inputs[0], in); + if (av_audio_fifo_size(s->fifo) >= s->window_size) { + ff_filter_set_ready(ctx, 100); + return 0; + } + + if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) { + if (status == AVERROR_EOF) { + s->eof = 1; + s->samples_left = av_audio_fifo_size(s->fifo) - s->overlap_skip; + ff_filter_set_ready(ctx, 100); + return 0; } } - return ret; + if (s->eof && s->samples_left <= 0) { + ff_outlink_set_status(outlink, AVERROR_EOF, s->pts); + return 0; + } + + if (!s->eof) + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; } static av_cold int init(AVFilterContext *ctx) @@ -697,7 +720,6 @@ static const AVFilterPad inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .filter_frame = filter_frame, .config_props = config_input, }, { NULL } @@ -707,7 +729,6 @@ static const AVFilterPad outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .request_frame = request_frame, }, { NULL } }; @@ -719,6 +740,7 @@ AVFilter ff_af_adeclick = { .priv_size = sizeof(AudioDeclickContext), .priv_class = &adeclick_class, .init = init, + .activate = activate, .uninit = uninit, .inputs = inputs, .outputs = outputs, @@ -746,6 +768,7 @@ AVFilter ff_af_adeclip = { .priv_size = sizeof(AudioDeclickContext), .priv_class = &adeclip_class, .init = init, + .activate = activate, .uninit = uninit, .inputs = inputs, .outputs = outputs,

1 0

avfilter/af_rubberband: make use of initial input timestamp
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 21:11:46 2019 +0200| [12a284fa6356dffe55d89f249240e54df9975aa5] | committer: Paul B Mahol avfilter/af_rubberband: make use of initial input timestamp > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=12a284fa6356dffe5… --- libavfilter/af_rubberband.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/libavfilter/af_rubberband.c b/libavfilter/af_rubberband.c index ca2be4dcb9..10471bc94d 100644 --- a/libavfilter/af_rubberband.c +++ b/libavfilter/af_rubberband.c @@ -37,6 +37,7 @@ typedef struct RubberBandContext { smoothing, formant, opitch, channels; int64_t nb_samples_out; int64_t nb_samples_in; + int64_t first_pts; int nb_samples; } RubberBandContext; @@ -124,6 +125,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFrame *out; int ret = 0, nb_samples; + if (s->first_pts == AV_NOPTS_VALUE) + s->first_pts = in->pts; + rubberband_process(s->rbs, (const float *const *)in->data, in->nb_samples, ff_outlink_get_status(inlink)); s->nb_samples_in += in->nb_samples; @@ -134,7 +138,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) av_frame_free(&in); return AVERROR(ENOMEM); } - out->pts = av_rescale_q(s->nb_samples_out, + out->pts = s->first_pts + av_rescale_q(s->nb_samples_out, (AVRational){ 1, outlink->sample_rate }, outlink->time_base); nb_samples = rubberband_retrieve(s->rbs, (float *const *)out->data, nb_samples); @@ -162,6 +166,7 @@ static int config_input(AVFilterLink *inlink) return AVERROR(ENOMEM); s->nb_samples = rubberband_get_samples_required(s->rbs); + s->first_pts = AV_NOPTS_VALUE; return 0; }

1 0

avfilter/avf_showwaves: Add draw mode also to showwavespic filter
by Martin Vobruba 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Martin Vobruba <vobruba.martin(a)gmail.com> | Fri Apr 26 08:17:11 2019 +0200| [093a504414ef7781123f0db49326b0059e32a818] | committer: Paul B Mahol avfilter/avf_showwaves: Add draw mode also to showwavespic filter See commit a8c2d375ca68b7f001564ced14d8ac0757f53a29 > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=093a504414ef77811… --- doc/filters.texi | 14 ++++++++++++++ libavfilter/avf_showwaves.c | 3 +++ 2 files changed, 17 insertions(+) diff --git a/doc/filters.texi b/doc/filters.texi index 7cc3937b40..cd82869849 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -22704,6 +22704,20 @@ Cubic root. @end table Default is linear. + +@item draw +Set the draw mode. + +Available values are: +@table @samp +@item scale +Scale pixel values for each drawn sample. + +@item full +Draw every sample directly. +@end table + +Default value is @code{scale}. @end table @subsection Examples diff --git a/libavfilter/avf_showwaves.c b/libavfilter/avf_showwaves.c index 0e683cf42d..a0d2b25d55 100644 --- a/libavfilter/avf_showwaves.c +++ b/libavfilter/avf_showwaves.c @@ -765,6 +765,9 @@ static const AVOption showwavespic_options[] = { { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=SCALE_LOG}, .flags=FLAGS, .unit="scale"}, { "sqrt", "square root", 0, AV_OPT_TYPE_CONST, {.i64=SCALE_SQRT}, .flags=FLAGS, .unit="scale"}, { "cbrt", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64=SCALE_CBRT}, .flags=FLAGS, .unit="scale"}, + { "draw", "set draw mode", OFFSET(draw_mode), AV_OPT_TYPE_INT, {.i64 = DRAW_SCALE}, 0, DRAW_NB-1, FLAGS, .unit="draw" }, + { "scale", "scale pixel values for each drawn sample", 0, AV_OPT_TYPE_CONST, {.i64=DRAW_SCALE}, .flags=FLAGS, .unit="draw"}, + { "full", "draw every pixel for sample directly", 0, AV_OPT_TYPE_CONST, {.i64=DRAW_FULL}, .flags=FLAGS, .unit="draw"}, { NULL } };

1 0

avfilter/f_ebur128: use correct type for chl
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 16:46:42 2019 +0200| [31990046ac68f218efed846931824adab9d51370] | committer: Paul B Mahol avfilter/f_ebur128: use correct type for chl > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=31990046ac68f218e… --- libavfilter/f_ebur128.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavfilter/f_ebur128.c b/libavfilter/f_ebur128.c index f613d8def2..f25d5f096e 100644 --- a/libavfilter/f_ebur128.c +++ b/libavfilter/f_ebur128.c @@ -420,7 +420,7 @@ static int config_audio_output(AVFilterLink *outlink) for (i = 0; i < nb_channels; i++) { /* channel weighting */ - const uint16_t chl = av_channel_layout_extract_channel(outlink->channel_layout, i); + const uint64_t chl = av_channel_layout_extract_channel(outlink->channel_layout, i); if (chl & (AV_CH_LOW_FREQUENCY|AV_CH_LOW_FREQUENCY_2)) { ebur128->ch_weighting[i] = 0; } else if (chl & BACK_MASK) {

1 0

avfilter/af_dynaudnorm: switch to activate
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 14:53:26 2019 +0200| [d7fead80ad994243b35bf4d43324f3fa72c11e9d] | committer: Paul B Mahol avfilter/af_dynaudnorm: switch to activate > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d7fead80ad994243b… --- libavfilter/af_dynaudnorm.c | 76 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 59 insertions(+), 17 deletions(-) diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 5919304267..6f6541cc49 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -34,6 +34,7 @@ #include "audio.h" #include "avfilter.h" +#include "filters.h" #include "internal.h" typedef struct cqueue { @@ -67,6 +68,8 @@ typedef struct DynamicAudioNormalizerContext { int channels; int delay; + int eof; + int64_t pts; cqueue **gain_history_original; cqueue **gain_history_minimum; @@ -292,10 +295,7 @@ static int config_input(AVFilterLink *inlink) uninit(ctx); - s->frame_len = - inlink->min_samples = - inlink->max_samples = - inlink->partial_buf_size = frame_size(inlink->sample_rate, s->frame_len_msec); + s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec); av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len); s->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0])); @@ -661,7 +661,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFilterContext *ctx = inlink->dst; DynamicAudioNormalizerContext *s = ctx->priv; AVFilterLink *outlink = inlink->dst->outputs[0]; - int ret = 0; + int ret = 1; if (!cqueue_empty(s->gain_history_smoothed[0])) { AVFrame *out = ff_bufqueue_get(&s->queue); @@ -670,6 +670,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) ret = ff_filter_frame(outlink, out); } + av_frame_make_writable(in); analyze_frame(s, in); ff_bufqueue_add(ctx, &s->queue, in); @@ -701,34 +702,75 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink, return filter_frame(inlink, out); } -static int request_frame(AVFilterLink *outlink) +static int flush(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; DynamicAudioNormalizerContext *s = ctx->priv; int ret = 0; - ret = ff_request_frame(ctx->inputs[0]); + if (!cqueue_empty(s->gain_history_smoothed[0])) { + ret = flush_buffer(s, ctx->inputs[0], outlink); + } else if (s->queue.available) { + AVFrame *out = ff_bufqueue_get(&s->queue); + + s->pts = out->pts; + ret = ff_filter_frame(outlink, out); + } - if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay) { - if (!cqueue_empty(s->gain_history_smoothed[0])) { - ret = flush_buffer(s, ctx->inputs[0], outlink); - } else if (s->queue.available) { - AVFrame *out = ff_bufqueue_get(&s->queue); + return ret; +} - ret = ff_filter_frame(outlink, out); +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + DynamicAudioNormalizerContext *s = ctx->priv; + AVFrame *in = NULL; + int ret = 0, status; + int64_t pts; + + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + if (!s->eof) { + ret = ff_inlink_consume_samples(inlink, s->frame_len, s->frame_len, &in); + if (ret < 0) + return ret; + if (ret > 0) { + ret = filter_frame(inlink, in); + if (ret <= 0) + return ret; + } + + if (ff_inlink_queued_samples(inlink) >= s->frame_len) { + ff_filter_set_ready(ctx, 10); + return 0; } } - return ret; + if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) { + if (status == AVERROR_EOF) + s->eof = 1; + } + + if (s->eof && s->delay > 0) + return flush(outlink); + + if (s->eof && s->delay <= 0) { + ff_outlink_set_status(outlink, AVERROR_EOF, s->pts); + return 0; + } + + if (!s->eof) + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; } static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .filter_frame = filter_frame, .config_props = config_input, - .needs_writable = 1, }, { NULL } }; @@ -737,7 +779,6 @@ static const AVFilterPad avfilter_af_dynaudnorm_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .request_frame = request_frame, }, { NULL } }; @@ -749,6 +790,7 @@ AVFilter ff_af_dynaudnorm = { .priv_size = sizeof(DynamicAudioNormalizerContext), .init = init, .uninit = uninit, + .activate = activate, .inputs = avfilter_af_dynaudnorm_inputs, .outputs = avfilter_af_dynaudnorm_outputs, .priv_class = &dynaudnorm_class,

1 0

avfilter/af_rubberband: switch to activate
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 14:22:13 2019 +0200| [a40bcb5c93b576c7158e0388a750b077eb6aa807] | committer: Paul B Mahol avfilter/af_rubberband: switch to activate > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a40bcb5c93b576c71… --- libavfilter/af_rubberband.c | 66 +++++++++++++++------------------------------ 1 file changed, 22 insertions(+), 44 deletions(-) diff --git a/libavfilter/af_rubberband.c b/libavfilter/af_rubberband.c index 767e50f1bb..ca2be4dcb9 100644 --- a/libavfilter/af_rubberband.c +++ b/libavfilter/af_rubberband.c @@ -24,6 +24,7 @@ #include "audio.h" #include "avfilter.h" +#include "filters.h" #include "formats.h" #include "internal.h" @@ -36,7 +37,7 @@ typedef struct RubberBandContext { smoothing, formant, opitch, channels; int64_t nb_samples_out; int64_t nb_samples_in; - int flushed; + int nb_samples; } RubberBandContext; #define OFFSET(x) offsetof(RubberBandContext, x) @@ -123,7 +124,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFrame *out; int ret = 0, nb_samples; - rubberband_process(s->rbs, (const float *const *)in->data, in->nb_samples, 0); + rubberband_process(s->rbs, (const float *const *)in->data, in->nb_samples, ff_outlink_get_status(inlink)); s->nb_samples_in += in->nb_samples; nb_samples = rubberband_available(s->rbs); @@ -160,53 +161,31 @@ static int config_input(AVFilterLink *inlink) if (!s->rbs) return AVERROR(ENOMEM); - inlink->partial_buf_size = - inlink->min_samples = - inlink->max_samples = rubberband_get_samples_required(s->rbs); + s->nb_samples = rubberband_get_samples_required(s->rbs); return 0; } -static int request_frame(AVFilterLink *outlink) +static int activate(AVFilterContext *ctx) { - AVFilterContext *ctx = outlink->src; - RubberBandContext *s = ctx->priv; AVFilterLink *inlink = ctx->inputs[0]; - int ret = 0; - - ret = ff_request_frame(ctx->inputs[0]); - - if (ret == AVERROR_EOF && !s->flushed) { - if (rubberband_available(s->rbs) > 0) { - AVFrame *out = ff_get_audio_buffer(inlink, 1); - int nb_samples; - - if (!out) - return AVERROR(ENOMEM); - - rubberband_process(s->rbs, (const float *const *)out->data, 1, 1); - av_frame_free(&out); - nb_samples = rubberband_available(s->rbs); - - if (nb_samples > 0) { - out = ff_get_audio_buffer(outlink, nb_samples); - if (!out) - return AVERROR(ENOMEM); - out->pts = av_rescale_q(s->nb_samples_out, - (AVRational){ 1, outlink->sample_rate }, - outlink->time_base); - nb_samples = rubberband_retrieve(s->rbs, (float *const *)out->data, nb_samples); - out->nb_samples = nb_samples; - ret = ff_filter_frame(outlink, out); - s->nb_samples_out += nb_samples; - } - } - s->flushed = 1; - av_log(ctx, AV_LOG_DEBUG, "nb_samples_in %"PRId64" nb_samples_out %"PRId64"\n", - s->nb_samples_in, s->nb_samples_out); - } + AVFilterLink *outlink = ctx->outputs[0]; + RubberBandContext *s = ctx->priv; + AVFrame *in = NULL; + int ret; - return ret; + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in); + if (ret < 0) + return ret; + if (ret > 0) + return filter_frame(inlink, in); + + FF_FILTER_FORWARD_STATUS(inlink, outlink); + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; } static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, @@ -246,7 +225,6 @@ static const AVFilterPad rubberband_inputs[] = { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, - .filter_frame = filter_frame, }, { NULL } }; @@ -255,7 +233,6 @@ static const AVFilterPad rubberband_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .request_frame = request_frame, }, { NULL } }; @@ -267,6 +244,7 @@ AVFilter ff_af_rubberband = { .priv_size = sizeof(RubberBandContext), .priv_class = &rubberband_class, .uninit = uninit, + .activate = activate, .inputs = rubberband_inputs, .outputs = rubberband_outputs, .process_command = process_command,

1 0

avfilter/af_rubberband: check if rbs is valid
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 13:47:46 2019 +0200| [a89ec33fd5915afdeee2f094b73fd44b6a577e7b] | committer: Paul B Mahol avfilter/af_rubberband: check if rbs is valid > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a89ec33fd5915afde… --- libavfilter/af_rubberband.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/libavfilter/af_rubberband.c b/libavfilter/af_rubberband.c index ea6f4ff2c9..767e50f1bb 100644 --- a/libavfilter/af_rubberband.c +++ b/libavfilter/af_rubberband.c @@ -157,6 +157,8 @@ static int config_input(AVFilterLink *inlink) if (s->rbs) rubberband_delete(s->rbs); s->rbs = rubberband_new(inlink->sample_rate, inlink->channels, opts, 1. / s->tempo, s->pitch); + if (!s->rbs) + return AVERROR(ENOMEM); inlink->partial_buf_size = inlink->min_samples =

1 0

avfilter/af_sofalizer: switch to activate
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 13:35:04 2019 +0200| [21d1bb00c4d17b61abee40c802f9c39094a7b438] | committer: Paul B Mahol avfilter/af_sofalizer: switch to activate > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=21d1bb00c4d17b61a… --- libavfilter/af_sofalizer.c | 36 ++++++++++++++++++++++++++++++------ 1 file changed, 30 insertions(+), 6 deletions(-) diff --git a/libavfilter/af_sofalizer.c b/libavfilter/af_sofalizer.c index 109cf5695b..ccf3872e77 100644 --- a/libavfilter/af_sofalizer.c +++ b/libavfilter/af_sofalizer.c @@ -35,6 +35,7 @@ #include "libavutil/intmath.h" #include "libavutil/opt.h" #include "avfilter.h" +#include "filters.h" #include "internal.h" #include "audio.h" @@ -81,6 +82,7 @@ typedef struct SOFAlizerContext { int buffer_length; /* is: longest IR plus max. delay in all SOFA files */ /* then choose next power of 2 */ int n_fft; /* number of samples in one FFT block */ + int nb_samples; /* netCDF variables */ int *delay[2]; /* broadband delay for each channel/IR to be convolved */ @@ -603,6 +605,31 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) return ff_filter_frame(outlink, out); } +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + SOFAlizerContext *s = ctx->priv; + AVFrame *in; + int ret; + + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + if (s->nb_samples) + ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in); + else + ret = ff_inlink_consume_frame(inlink, &in); + if (ret < 0) + return ret; + if (ret > 0) + return filter_frame(inlink, in); + + FF_FILTER_FORWARD_STATUS(inlink, outlink); + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; +} + static int query_formats(AVFilterContext *ctx) { struct SOFAlizerContext *s = ctx->priv; @@ -964,11 +991,8 @@ static int config_input(AVFilterLink *inlink) SOFAlizerContext *s = ctx->priv; int ret; - if (s->type == FREQUENCY_DOMAIN) { - inlink->partial_buf_size = - inlink->min_samples = - inlink->max_samples = s->framesize; - } + if (s->type == FREQUENCY_DOMAIN) + s->nb_samples = s->framesize; /* gain -3 dB per channel */ s->gain_lfe = expf((s->gain - 3 * inlink->channels + s->lfe_gain) / 20 * M_LN10); @@ -1047,7 +1071,6 @@ static const AVFilterPad inputs[] = { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, - .filter_frame = filter_frame, }, { NULL } }; @@ -1066,6 +1089,7 @@ AVFilter ff_af_sofalizer = { .priv_size = sizeof(SOFAlizerContext), .priv_class = &sofalizer_class, .init = init, + .activate = activate, .uninit = uninit, .query_formats = query_formats, .inputs = inputs,

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avfilter/avf_avectorscope: switch to activate
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 12:52:50 2019 +0200| [21b1f08ea2ec785f47d7d90955b2eaa28ea345ee] | committer: Paul B Mahol avfilter/avf_avectorscope: switch to activate > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=21b1f08ea2ec785f4… --- libavfilter/avf_avectorscope.c | 32 ++++++++++++++++++++++++++------ 1 file changed, 26 insertions(+), 6 deletions(-) diff --git a/libavfilter/avf_avectorscope.c b/libavfilter/avf_avectorscope.c index af9f1da1ec..d9ec151331 100644 --- a/libavfilter/avf_avectorscope.c +++ b/libavfilter/avf_avectorscope.c @@ -28,6 +28,7 @@ #include "libavutil/opt.h" #include "libavutil/parseutils.h" #include "avfilter.h" +#include "filters.h" #include "formats.h" #include "audio.h" #include "video.h" @@ -69,6 +70,7 @@ typedef struct AudioVectorScopeContext { int mirror; unsigned prev_x, prev_y; AVRational frame_rate; + int nb_samples; } AudioVectorScopeContext; #define OFFSET(x) offsetof(AudioVectorScopeContext, x) @@ -208,12 +210,8 @@ static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; AudioVectorScopeContext *s = ctx->priv; - int nb_samples; - nb_samples = FFMAX(1024, ((double)inlink->sample_rate / av_q2d(s->frame_rate)) + 0.5); - inlink->partial_buf_size = - inlink->min_samples = - inlink->max_samples = nb_samples; + s->nb_samples = FFMAX(1, ((double)inlink->sample_rate / av_q2d(s->frame_rate)) + 0.5); return 0; } @@ -365,6 +363,28 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) return ff_filter_frame(outlink, av_frame_clone(s->outpicref)); } +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + AudioVectorScopeContext *s = ctx->priv; + AVFrame *in; + int ret; + + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in); + if (ret < 0) + return ret; + if (ret > 0) + return filter_frame(inlink, in); + + FF_FILTER_FORWARD_STATUS(inlink, outlink); + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; +} + static av_cold void uninit(AVFilterContext *ctx) { AudioVectorScopeContext *s = ctx->priv; @@ -377,7 +397,6 @@ static const AVFilterPad audiovectorscope_inputs[] = { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, - .filter_frame = filter_frame, }, { NULL } }; @@ -397,6 +416,7 @@ AVFilter ff_avf_avectorscope = { .uninit = uninit, .query_formats = query_formats, .priv_size = sizeof(AudioVectorScopeContext), + .activate = activate, .inputs = audiovectorscope_inputs, .outputs = audiovectorscope_outputs, .priv_class = &avectorscope_class,

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avfilter/avf_ahistogram: switch to activate
by Paul B Mahol 29 Apr '19

29 Apr '19

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Mon Apr 29 12:43:18 2019 +0200| [77aacdb03683a9fc7e9f4255b92421e62a0866fc] | committer: Paul B Mahol avfilter/avf_ahistogram: switch to activate > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=77aacdb03683a9fc7… --- libavfilter/avf_ahistogram.c | 33 ++++++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 7 deletions(-) diff --git a/libavfilter/avf_ahistogram.c b/libavfilter/avf_ahistogram.c index 6daafb0746..7795f3913c 100644 --- a/libavfilter/avf_ahistogram.c +++ b/libavfilter/avf_ahistogram.c @@ -22,6 +22,7 @@ #include "libavutil/opt.h" #include "libavutil/parseutils.h" #include "avfilter.h" +#include "filters.h" #include "formats.h" #include "audio.h" #include "video.h" @@ -54,6 +55,7 @@ typedef struct AudioHistogramContext { float *combine_buffer; AVFrame *in[101]; int first; + int nb_samples; } AudioHistogramContext; #define OFFSET(x) offsetof(AudioHistogramContext, x) @@ -117,13 +119,8 @@ static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; AudioHistogramContext *s = ctx->priv; - int nb_samples; - - nb_samples = FFMAX(1024, ((double)inlink->sample_rate / av_q2d(s->frame_rate)) + 0.5); - inlink->partial_buf_size = - inlink->min_samples = - inlink->max_samples = nb_samples; + s->nb_samples = FFMAX(1, ((double)inlink->sample_rate / av_q2d(s->frame_rate)) + 0.5); s->dchannels = s->dmode == SINGLE ? 1 : inlink->channels; s->shistogram = av_calloc(s->w, s->dchannels * sizeof(*s->shistogram)); if (!s->shistogram) @@ -369,6 +366,28 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) return ff_filter_frame(outlink, av_frame_clone(s->out)); } +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + AudioHistogramContext *s = ctx->priv; + AVFrame *in; + int ret; + + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in); + if (ret < 0) + return ret; + if (ret > 0) + return filter_frame(inlink, in); + + FF_FILTER_FORWARD_STATUS(inlink, outlink); + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; +} + static av_cold void uninit(AVFilterContext *ctx) { AudioHistogramContext *s = ctx->priv; @@ -387,7 +406,6 @@ static const AVFilterPad ahistogram_inputs[] = { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, - .filter_frame = filter_frame, }, { NULL } }; @@ -407,6 +425,7 @@ AVFilter ff_avf_ahistogram = { .uninit = uninit, .query_formats = query_formats, .priv_size = sizeof(AudioHistogramContext), + .activate = activate, .inputs = ahistogram_inputs, .outputs = ahistogram_outputs, .priv_class = &ahistogram_class,

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