February 2022 - ffmpeg-cvslog

libavcodec/vaapi_encode: Add new API adaption to vaapi_encode
by Wenbin Chen 28 Feb '22

28 Feb '22

ffmpeg | branch: master | Wenbin Chen <wenbin.chen-at-intel.com(a)ffmpeg.org> | Fri Feb 18 11:07:46 2022 +0800| [e0ff86993052b49a64d434bac345e92fc149f446] | committer: Haihao Xiang libavcodec/vaapi_encode: Add new API adaption to vaapi_encode Add vaSyncBuffer to VAAPI encoder. Old version API vaSyncSurface wait surface to complete. When surface is used for multiple operation, it waits all operations to finish. vaSyncBuffer only wait one channel to finish. Signed-off-by: Wenbin Chen <wenbin.chen(a)intel.com> Signed-off-by: Haihao Xiang <haihao.xiang(a)intel.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e0ff86993052b49a6… --- libavcodec/vaapi_encode.c | 32 +++++++++++++++++++++++++++----- libavcodec/vaapi_encode.h | 3 +++ 2 files changed, 30 insertions(+), 5 deletions(-) diff --git a/libavcodec/vaapi_encode.c b/libavcodec/vaapi_encode.c index 3bf379b1a0..3f8c8ace2a 100644 --- a/libavcodec/vaapi_encode.c +++ b/libavcodec/vaapi_encode.c @@ -150,11 +150,25 @@ static int vaapi_encode_wait(AVCodecContext *avctx, "(input surface %#x).\n", pic->display_order, pic->encode_order, pic->input_surface); - vas = vaSyncSurface(ctx->hwctx->display, pic->input_surface); - if (vas != VA_STATUS_SUCCESS) { - av_log(avctx, AV_LOG_ERROR, "Failed to sync to picture completion: " - "%d (%s).\n", vas, vaErrorStr(vas)); - return AVERROR(EIO); +#if VA_CHECK_VERSION(1, 9, 0) + if (ctx->has_sync_buffer_func) { + vas = vaSyncBuffer(ctx->hwctx->display, + pic->output_buffer, + VA_TIMEOUT_INFINITE); + if (vas != VA_STATUS_SUCCESS) { + av_log(avctx, AV_LOG_ERROR, "Failed to sync to output buffer completion: " + "%d (%s).\n", vas, vaErrorStr(vas)); + return AVERROR(EIO); + } + } else +#endif + { // If vaSyncBuffer is not implemented, try old version API. + vas = vaSyncSurface(ctx->hwctx->display, pic->input_surface); + if (vas != VA_STATUS_SUCCESS) { + av_log(avctx, AV_LOG_ERROR, "Failed to sync to picture completion: " + "%d (%s).\n", vas, vaErrorStr(vas)); + return AVERROR(EIO); + } } // Input is definitely finished with now. @@ -2522,6 +2536,14 @@ av_cold int ff_vaapi_encode_init(AVCodecContext *avctx) } } +#if VA_CHECK_VERSION(1, 9, 0) + // check vaSyncBuffer function + vas = vaSyncBuffer(ctx->hwctx->display, VA_INVALID_ID, 0); + if (vas != VA_STATUS_ERROR_UNIMPLEMENTED) { + ctx->has_sync_buffer_func = 1; + } +#endif + return 0; fail: diff --git a/libavcodec/vaapi_encode.h b/libavcodec/vaapi_encode.h index b41604a883..29d9e9b91c 100644 --- a/libavcodec/vaapi_encode.h +++ b/libavcodec/vaapi_encode.h @@ -345,6 +345,9 @@ typedef struct VAAPIEncodeContext { int roi_warned; AVFrame *frame; + + // Whether the driver support vaSyncBuffer + int has_sync_buffer_func; } VAAPIEncodeContext; enum {

1 0

avfilter/vf_amplify: improve performance
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 23:40:57 2022 +0100| [567cab3bd8b797f46506e0b363d6a12555000056] | committer: Paul B Mahol avfilter/vf_amplify: improve performance > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=567cab3bd8b797f46… --- libavfilter/vf_amplify.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/libavfilter/vf_amplify.c b/libavfilter/vf_amplify.c index 3c393ecfb0..0891d26413 100644 --- a/libavfilter/vf_amplify.c +++ b/libavfilter/vf_amplify.c @@ -96,8 +96,7 @@ typedef struct ThreadData { } ThreadData; #define AMPLIFY_SLICE(type, stype, clip) \ - const stype llimit = s->llimit; \ - const stype hlimit = s->hlimit; \ + const stype limit[2] = { s->llimit, s->hlimit }; \ \ for (int p = 0; p < s->nb_planes; p++) { \ const int slice_start = (s->height[p] * jobnr) / nb_jobs; \ @@ -116,23 +115,19 @@ typedef struct ThreadData { for (int y = slice_start; y < slice_end; y++) { \ for (int x = 0; x < s->linesize[p] / sizeof(type); x++) { \ stype src = *(type *)(in[radius]->data[p] + y * in[radius]->linesize[p] + x * sizeof(type));\ - float diff, avg; \ + float diff, abs_diff, avg; \ stype sum = 0; \ \ for (int i = 0; i < nb_inputs; i++) { \ sum += *(type *)(in[i]->data[p] + y * in[i]->linesize[p] + x * sizeof(type));\ } \ \ - avg = sum / (float)nb_inputs; \ + avg = sum * scale; \ diff = src - avg; \ + abs_diff = fabsf(diff); \ \ - if (fabsf(diff) < threshold && fabsf(diff) > tolerance) { \ - stype amp; \ - if (diff < 0) { \ - amp = -FFMIN(FFABS(diff * factor), llimit); \ - } else { \ - amp = FFMIN(FFABS(diff * factor), hlimit); \ - } \ + if (abs_diff < threshold && abs_diff > tolerance) { \ + float amp = copysignf(fminf(abs_diff * factor, limit[diff >= 0]), diff); \ dst[x] = clip(src + amp, depth); \ } else { \ dst[x] = src; \ @@ -143,8 +138,8 @@ typedef struct ThreadData { } \ } -#define CLIP8(x, depth) av_clip_uint8(x) -#define CLIP16(x, depth) av_clip_uintp2_c(x, depth) +#define CLIP8(x, depth) av_clip_uint8(lrintf(x)) +#define CLIP16(x, depth) av_clip_uintp2_c(lrintf(x), depth) #define NOP(x, depth) (x) static int amplify_frame(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) @@ -157,6 +152,7 @@ static int amplify_frame(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs const int nb_inputs = s->nb_inputs; const float threshold = s->threshold; const float tolerance = s->tolerance; + const float scale = 1.f / nb_inputs; const float factor = s->factor; const int depth = s->depth;

1 0

avfilter/af_dynaudnorm: allocate new frame instead of making it writable
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 19:58:06 2022 +0100| [a2dbd1778848c260fc9ef9c903ebc7a3cb67968e] | committer: Paul B Mahol avfilter/af_dynaudnorm: allocate new frame instead of making it writable Later case does not use frame pool at all. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a2dbd1778848c260f… --- libavfilter/af_dynaudnorm.c | 35 +++++++++++++++++++++++++++-------- 1 file changed, 27 insertions(+), 8 deletions(-) diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 7d779856a8..8015c8429c 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -647,23 +647,42 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame } } -static int analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame) +static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink, AVFrame **frame) { if (s->dc_correction || s->compress_factor > DBL_EPSILON) { int ret; - if ((ret = av_frame_make_writable(frame)) < 0) - return ret; + if (!av_frame_is_writable(*frame)) { + AVFrame *out = ff_get_audio_buffer(outlink, (*frame)->nb_samples); + + if (!out) { + av_frame_free(frame); + return AVERROR(ENOMEM); + } + ret = av_frame_copy_props(out, *frame); + if (ret < 0) { + av_frame_free(&out); + return ret; + } + ret = av_frame_copy(out, *frame); + if (ret < 0) { + av_frame_free(&out); + return ret; + } + + av_frame_free(frame); + *frame = out; + } } if (s->dc_correction) - perform_dc_correction(s, frame); + perform_dc_correction(s, *frame); if (s->compress_factor > DBL_EPSILON) - perform_compression(s, frame); + perform_compression(s, *frame); if (s->channels_coupled) { - const local_gain gain = get_max_local_gain(s, frame, -1); + const local_gain gain = get_max_local_gain(s, *frame, -1); int c; for (c = 0; c < s->channels; c++) @@ -672,7 +691,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame) int c; for (c = 0; c < s->channels; c++) - update_gain_history(s, c, get_max_local_gain(s, frame, c)); + update_gain_history(s, c, get_max_local_gain(s, *frame, c)); } return 0; @@ -740,7 +759,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) return ret; } - ret = analyze_frame(s, in); + ret = analyze_frame(s, outlink, &in); if (ret < 0) return ret; if (!s->eof) {

1 0

avfilter/af_dynaudnorm: allow to filter subset of channels
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 19:37:10 2022 +0100| [a9124a75b0a2bce3c810bc5dfd5c2360764c92dd] | committer: Paul B Mahol avfilter/af_dynaudnorm: allow to filter subset of channels > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a9124a75b0a2bce3c… --- doc/filters.texi | 3 +++ libavfilter/af_dynaudnorm.c | 25 +++++++++++++++++++++---- 2 files changed, 24 insertions(+), 4 deletions(-) diff --git a/doc/filters.texi b/doc/filters.texi index 25a11fcd10..b3ae139613 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -4389,6 +4389,9 @@ If input frame volume is above this value frame will be normalized. Otherwise frame may not be normalized at all. The default value is set to 0, which means all input frames will be normalized. This option is mostly useful if digital noise is not wanted to be amplified. + +@item channels, h +Specify which channels to filter, by default all available channels are filtered. @end table @subsection Commands diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 4935b09d08..7d779856a8 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -27,6 +27,7 @@ #include <float.h> #include "libavutil/avassert.h" +#include "libavutil/channel_layout.h" #include "libavutil/opt.h" #define MIN_FILTER_SIZE 3 @@ -76,6 +77,7 @@ typedef struct DynamicAudioNormalizerContext { int channels; int eof; + uint64_t channels_to_filter; int64_t pts; cqueue **gain_history_original; @@ -110,6 +112,8 @@ static const AVOption dynaudnorm_options[] = { { "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS }, { "threshold", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, { "t", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, + { "channels", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS }, + { "h", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS }, { NULL } }; @@ -514,6 +518,11 @@ static inline double update_value(double new, double old, double aggressiveness) return aggressiveness * new + (1.0 - aggressiveness) * old; } +static inline int bypass_channel(DynamicAudioNormalizerContext *s, AVFrame *frame, int ch) +{ + return !(av_channel_layout_extract_channel(frame->channel_layout, ch) & s->channels_to_filter); +} + static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame) { const double diff = 1.0 / frame->nb_samples; @@ -521,6 +530,7 @@ static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *fra int c, i; for (c = 0; c < s->channels; c++) { + const int bypass = bypass_channel(s, frame, c); double *dst_ptr = (double *)frame->extended_data[c]; double current_average_value = 0.0; double prev_value; @@ -531,7 +541,7 @@ static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *fra prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c]; s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1); - for (i = 0; i < frame->nb_samples; i++) { + for (i = 0; i < frame->nb_samples && !bypass; i++) { dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples); } } @@ -604,6 +614,11 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame for (c = 0; c < s->channels; c++) { double *const dst_ptr = (double *)frame->extended_data[c]; + const int bypass = bypass_channel(s, frame, c); + + if (bypass) + continue; + for (i = 0; i < frame->nb_samples; i++) { const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); @@ -611,19 +626,20 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame } } else { for (c = 0; c < s->channels; c++) { + const int bypass = bypass_channel(s, frame, c); const double standard_deviation = compute_frame_std_dev(s, frame, c); const double current_threshold = setup_compress_thresh(FFMIN(1.0, s->compress_factor * standard_deviation)); - const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c]; double prev_actual_thresh, curr_actual_thresh; double *dst_ptr; + s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0); prev_actual_thresh = setup_compress_thresh(prev_value); curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]); dst_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples; i++) { + for (i = 0; i < frame->nb_samples && !bypass; i++) { const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); } @@ -668,13 +684,14 @@ static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in, int c, i; for (c = 0; c < s->channels; c++) { + const int bypass = bypass_channel(s, frame, c); const double *src_ptr = (const double *)in->extended_data[c]; double *dst_ptr = (double *)frame->extended_data[c]; double current_amplification_factor; cqueue_dequeue(s->gain_history_smoothed[c], &current_amplification_factor); - for (i = 0; i < frame->nb_samples && enabled; i++) { + for (i = 0; i < frame->nb_samples && enabled && !bypass; i++) { const double amplification_factor = fade(s->prev_amplification_factor[c], current_amplification_factor, i, frame->nb_samples);

1 0

avfilter/af_dynaudnorm: make frame writable if it may be changed
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 18:46:03 2022 +0100| [b9f91a7cbcf12564ea5205af0bd462dc3541ce40] | committer: Paul B Mahol avfilter/af_dynaudnorm: make frame writable if it may be changed > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b9f91a7cbcf12564e… --- libavfilter/af_dynaudnorm.c | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 121a3ef5d7..4935b09d08 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -631,15 +631,20 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame } } -static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame) +static int analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame) { - if (s->dc_correction) { - perform_dc_correction(s, frame); + if (s->dc_correction || s->compress_factor > DBL_EPSILON) { + int ret; + + if ((ret = av_frame_make_writable(frame)) < 0) + return ret; } - if (s->compress_factor > DBL_EPSILON) { + if (s->dc_correction) + perform_dc_correction(s, frame); + + if (s->compress_factor > DBL_EPSILON) perform_compression(s, frame); - } if (s->channels_coupled) { const local_gain gain = get_max_local_gain(s, frame, -1); @@ -653,6 +658,8 @@ static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame) for (c = 0; c < s->channels; c++) update_gain_history(s, c, get_max_local_gain(s, frame, c)); } + + return 0; } static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in, @@ -684,7 +691,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFilterContext *ctx = inlink->dst; DynamicAudioNormalizerContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; - int ret = 1; + int ret; while (((s->queue.available >= s->filter_size) || (s->eof && s->queue.available)) && @@ -712,9 +719,13 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) if (out != in) av_frame_free(&in); ret = ff_filter_frame(outlink, out); + if (ret < 0) + return ret; } - analyze_frame(s, in); + ret = analyze_frame(s, in); + if (ret < 0) + return ret; if (!s->eof) { ff_bufqueue_add(ctx, &s->queue, in); cqueue_enqueue(s->is_enabled, !ctx->is_disabled); @@ -722,7 +733,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) av_frame_free(&in); } - return ret; + return 1; } static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,

1 0

avfilter/af_acrusher: add timeline support
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 16:52:48 2022 +0100| [b15c26a22cfad3d8c677bb397c734f88607f07ea] | committer: Paul B Mahol avfilter/af_acrusher: add timeline support > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b15c26a22cfad3d8c… --- libavfilter/af_acrusher.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/libavfilter/af_acrusher.c b/libavfilter/af_acrusher.c index 14d66e88ea..6295646c04 100644 --- a/libavfilter/af_acrusher.c +++ b/libavfilter/af_acrusher.c @@ -248,7 +248,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) double sample = src[c] * level_in; sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in; - dst[c] = bitreduction(s, sample) * level_out; + dst[c] = ctx->is_disabled ? src[c] : bitreduction(s, sample) * level_out; } src += c; dst += c; @@ -342,4 +342,5 @@ const AVFilter ff_af_acrusher = { FILTER_OUTPUTS(avfilter_af_acrusher_outputs), FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL), .process_command = process_command, + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, };

1 0

avfilter/af_vibrato: add timeline support
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 16:29:31 2022 +0100| [8bcb7d49f9d2e7dbb7cbdb84ac4ea9a6e217bda0] | committer: Paul B Mahol avfilter/af_vibrato: add timeline support > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8bcb7d49f9d2e7dbb… --- libavfilter/af_vibrato.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavfilter/af_vibrato.c b/libavfilter/af_vibrato.c index 2cf1364273..fbb59059b0 100644 --- a/libavfilter/af_vibrato.c +++ b/libavfilter/af_vibrato.c @@ -71,7 +71,6 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) av_frame_copy_props(out, in); } - for (n = 0; n < in->nb_samples; n++) { double integer, decimal; decimal = modf(s->depth * s->wave_table[s->wave_table_index], &integer); @@ -175,4 +174,5 @@ const AVFilter ff_af_vibrato = { FILTER_INPUTS(avfilter_af_vibrato_inputs), FILTER_OUTPUTS(avfilter_af_vibrato_outputs), FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP), + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, };

1 0

avfilter/af_tremolo: add timeline support
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 15:27:49 2022 +0100| [57580376f7e467e5468d1ddb47ba3995130f732e] | committer: Paul B Mahol avfilter/af_tremolo: add timeline support > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=57580376f7e467e54… --- libavfilter/af_tremolo.c | 1 + 1 file changed, 1 insertion(+) diff --git a/libavfilter/af_tremolo.c b/libavfilter/af_tremolo.c index b34a7b79f7..61c5066bdc 100644 --- a/libavfilter/af_tremolo.c +++ b/libavfilter/af_tremolo.c @@ -137,4 +137,5 @@ const AVFilter ff_af_tremolo = { FILTER_INPUTS(avfilter_af_tremolo_inputs), FILTER_OUTPUTS(avfilter_af_tremolo_outputs), FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL), + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, };

1 0

avfilter/af_afftdn: add more verbose options aliases
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 12:00:21 2022 +0100| [0d0002cd20b496c96d659b21b7cd462a100ebbc9] | committer: Paul B Mahol avfilter/af_afftdn: add more verbose options aliases > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=0d0002cd20b496c96… --- doc/filters.texi | 32 ++++++++++++++++---------------- libavfilter/af_afftdn.c | 15 +++++++++++++++ 2 files changed, 31 insertions(+), 16 deletions(-) diff --git a/doc/filters.texi b/doc/filters.texi index 0650879cfa..25a11fcd10 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1282,64 +1282,64 @@ Denoise audio samples with FFT. A description of the accepted parameters follows. @table @option -@item nr +@item noise_reduction, nr Set the noise reduction in dB, allowed range is 0.01 to 97. Default value is 12 dB. -@item nf +@item noise_floor, nf Set the noise floor in dB, allowed range is -80 to -20. Default value is -50 dB. -@item nt +@item noise_type, nt Set the noise type. It accepts the following values: @table @option -@item w +@item white, w Select white noise. -@item v +@item vinyl, v Select vinyl noise. -@item s +@item shellac, s Select shellac noise. -@item c +@item custom, c Select custom noise, defined in @code{bn} option. Default value is white noise. @end table -@item bn +@item band_noise, bn Set custom band noise for every one of 15 bands. Bands are separated by ' ' or '|'. -@item rf +@item residual_floor, rf Set the residual floor in dB, allowed range is -80 to -20. Default value is -38 dB. -@item tn +@item track_noise, tn Enable noise tracking. By default is disabled. With this enabled, noise floor is automatically adjusted. -@item tr +@item track_residual, tr Enable residual tracking. By default is disabled. -@item om +@item output_mode, om Set the output mode. It accepts the following values: @table @option -@item i +@item input, i Pass input unchanged. -@item o +@item output, o Pass noise filtered out. -@item n +@item noise, n Pass only noise. -Default value is @var{o}. +Default value is @var{output}. @end table @end table diff --git a/libavfilter/af_afftdn.c b/libavfilter/af_afftdn.c index 628336b18b..2758ae8a8d 100644 --- a/libavfilter/af_afftdn.c +++ b/libavfilter/af_afftdn.c @@ -145,20 +145,35 @@ typedef struct AudioFFTDeNoiseContext { #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM static const AVOption afftdn_options[] = { + { "noise_reduction", "set the noise reduction",OFFSET(noise_reduction), AV_OPT_TYPE_FLOAT,{.dbl = 12}, .01, 97, AFR }, { "nr", "set the noise reduction", OFFSET(noise_reduction), AV_OPT_TYPE_FLOAT, {.dbl = 12}, .01, 97, AFR }, + { "noise_floor", "set the noise floor",OFFSET(noise_floor), AV_OPT_TYPE_FLOAT, {.dbl =-50}, -80,-20, AFR }, { "nf", "set the noise floor", OFFSET(noise_floor), AV_OPT_TYPE_FLOAT, {.dbl =-50}, -80,-20, AFR }, + { "noise_type", "set the noise type", OFFSET(noise_type), AV_OPT_TYPE_INT, {.i64 = WHITE_NOISE}, WHITE_NOISE, NB_NOISE-1, AF, "type" }, { "nt", "set the noise type", OFFSET(noise_type), AV_OPT_TYPE_INT, {.i64 = WHITE_NOISE}, WHITE_NOISE, NB_NOISE-1, AF, "type" }, + { "white", "white noise", 0, AV_OPT_TYPE_CONST, {.i64 = WHITE_NOISE}, 0, 0, AF, "type" }, { "w", "white noise", 0, AV_OPT_TYPE_CONST, {.i64 = WHITE_NOISE}, 0, 0, AF, "type" }, + { "vinyl", "vinyl noise", 0, AV_OPT_TYPE_CONST, {.i64 = VINYL_NOISE}, 0, 0, AF, "type" }, { "v", "vinyl noise", 0, AV_OPT_TYPE_CONST, {.i64 = VINYL_NOISE}, 0, 0, AF, "type" }, + { "shellac", "shellac noise", 0, AV_OPT_TYPE_CONST, {.i64 = SHELLAC_NOISE}, 0, 0, AF, "type" }, { "s", "shellac noise", 0, AV_OPT_TYPE_CONST, {.i64 = SHELLAC_NOISE}, 0, 0, AF, "type" }, + { "custom", "custom noise", 0, AV_OPT_TYPE_CONST, {.i64 = CUSTOM_NOISE}, 0, 0, AF, "type" }, { "c", "custom noise", 0, AV_OPT_TYPE_CONST, {.i64 = CUSTOM_NOISE}, 0, 0, AF, "type" }, + { "band_noise", "set the custom bands noise", OFFSET(band_noise_str), AV_OPT_TYPE_STRING, {.str = 0}, 0, 0, AF }, { "bn", "set the custom bands noise", OFFSET(band_noise_str), AV_OPT_TYPE_STRING, {.str = 0}, 0, 0, AF }, + { "residual_floor", "set the residual floor",OFFSET(residual_floor), AV_OPT_TYPE_FLOAT, {.dbl =-38}, -80,-20, AFR }, { "rf", "set the residual floor", OFFSET(residual_floor), AV_OPT_TYPE_FLOAT, {.dbl =-38}, -80,-20, AFR }, + { "track_noise", "track noise", OFFSET(track_noise), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR }, { "tn", "track noise", OFFSET(track_noise), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR }, + { "track_residuals", "track residual",OFFSET(track_residual), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR }, { "tr", "track residual", OFFSET(track_residual), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR }, + { "output_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64 = OUT_MODE}, 0, NB_MODES-1, AFR, "mode" }, { "om", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64 = OUT_MODE}, 0, NB_MODES-1, AFR, "mode" }, + { "input", "input", 0, AV_OPT_TYPE_CONST, {.i64 = IN_MODE}, 0, 0, AFR, "mode" }, { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64 = IN_MODE}, 0, 0, AFR, "mode" }, + { "output", "output", 0, AV_OPT_TYPE_CONST, {.i64 = OUT_MODE}, 0, 0, AFR, "mode" }, { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64 = OUT_MODE}, 0, 0, AFR, "mode" }, + { "noise", "noise", 0, AV_OPT_TYPE_CONST, {.i64 = NOISE_MODE}, 0, 0, AFR, "mode" }, { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64 = NOISE_MODE}, 0, 0, AFR, "mode" }, { NULL } };

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avfilter/af_afftdn: remove ThreadData struct code
by Paul B Mahol 27 Feb '22

27 Feb '22

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Feb 27 11:52:00 2022 +0100| [592cef61353b1712745a121c68df0c26178db86c] | committer: Paul B Mahol avfilter/af_afftdn: remove ThreadData struct code > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=592cef61353b17127… --- libavfilter/af_afftdn.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/libavfilter/af_afftdn.c b/libavfilter/af_afftdn.c index d7db625c35..628336b18b 100644 --- a/libavfilter/af_afftdn.c +++ b/libavfilter/af_afftdn.c @@ -1083,15 +1083,10 @@ static void set_noise_profile(AudioFFTDeNoiseContext *s, s->noise_floor = new_noise_floor; } -typedef struct ThreadData { - AVFrame *in; -} ThreadData; - static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { AudioFFTDeNoiseContext *s = ctx->priv; - ThreadData *td = arg; - AVFrame *in = td->in; + AVFrame *in = arg; const int start = (in->channels * jobnr) / nb_jobs; const int end = (in->channels * (jobnr+1)) / nb_jobs; @@ -1165,7 +1160,6 @@ static int output_frame(AVFilterLink *inlink, AVFrame *in) const int output_mode = ctx->is_disabled ? IN_MODE : s->output_mode; const int offset = s->window_length - s->sample_advance; AVFrame *out; - ThreadData td; for (int ch = 0; ch < s->channels; ch++) { float *src = (float *)s->winframe->extended_data[ch]; @@ -1220,8 +1214,7 @@ static int output_frame(AVFilterLink *inlink, AVFrame *in) } s->block_count++; - td.in = s->winframe; - ff_filter_execute(ctx, filter_channel, &td, NULL, + ff_filter_execute(ctx, filter_channel, s->winframe, NULL, FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx))); if (av_frame_is_writable(in)) {

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