diff options
| -rw-r--r-- | doc/filters.texi | 11 | ||||
| -rw-r--r-- | libavfilter/af_anlmdn.c | 103 |
2 files changed, 77 insertions, 37 deletions
diff --git a/doc/filters.texi b/doc/filters.texi index 8c9eccb00e..15e4873dcf 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1935,16 +1935,7 @@ Set smooth factor. Default value is @var{11}. Allowed range is from @var{1} to @ @subsection Commands -This filter supports the following commands: -@table @option -@item s -Change denoise strength. Argument is single float number. -Syntax for the command is : "@var{s}" - -@item o -Change output mode. -Syntax for the command is : "i", "o" or "n" string. -@end table +This filter supports the all above options as @ref{commands}. @section anlms Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second audio stream. diff --git a/libavfilter/af_anlmdn.c b/libavfilter/af_anlmdn.c index ea473bdab8..e2661e102f 100644 --- a/libavfilter/af_anlmdn.c +++ b/libavfilter/af_anlmdn.c @@ -72,13 +72,12 @@ enum OutModes { }; #define OFFSET(x) offsetof(AudioNLMeansContext, x) -#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM #define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM static const AVOption anlmdn_options[] = { { "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AFT }, - { "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AF }, - { "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AF }, + { "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT }, + { "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT }, { "o", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, "mode" }, { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AFT, "mode" }, { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AFT, "mode" }, @@ -147,32 +146,73 @@ void ff_anlmdn_init(AudioNLMDNDSPContext *dsp) ff_anlmdn_init_x86(dsp); } -static int config_output(AVFilterLink *outlink) +static int config_filter(AVFilterContext *ctx) { - AVFilterContext *ctx = outlink->src; AudioNLMeansContext *s = ctx->priv; - int ret; + AVFilterLink *outlink = ctx->outputs[0]; + int newK, newS, newH, newN; + AVFrame *new_in, *new_cache; - s->K = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE); - s->S = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE); + newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE); + newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE); - s->eof_left = -1; - s->pts = AV_NOPTS_VALUE; - s->H = s->K * 2 + 1; - s->N = s->H + (s->K + s->S) * 2; + newH = newK * 2 + 1; + newN = newH + (newK + newS) * 2; - av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", s->K, s->S, s->H, s->N); + av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN); - av_frame_free(&s->in); - av_frame_free(&s->cache); - s->in = ff_get_audio_buffer(outlink, s->N); - if (!s->in) + if (!s->cache || s->cache->nb_samples < newS * 2) { + new_cache = ff_get_audio_buffer(outlink, newS * 2); + if (new_cache) { + av_frame_free(&s->cache); + s->cache = new_cache; + } else { + return AVERROR(ENOMEM); + } + } + if (!s->cache) return AVERROR(ENOMEM); - s->cache = ff_get_audio_buffer(outlink, s->S * 2); - if (!s->cache) + s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE; + for (int i = 0; i < WEIGHT_LUT_SIZE; i++) { + float w = -i / s->pdiff_lut_scale; + + s->weight_lut[i] = expf(w); + } + + if (!s->in || s->in->nb_samples < newN) { + new_in = ff_get_audio_buffer(outlink, newN); + if (new_in) { + av_frame_free(&s->in); + s->in = new_in; + } else { + return AVERROR(ENOMEM); + } + } + if (!s->in) return AVERROR(ENOMEM); + s->K = newK; + s->S = newS; + s->H = newH; + s->N = newN; + + return 0; +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AudioNLMeansContext *s = ctx->priv; + int ret; + + s->eof_left = -1; + s->pts = AV_NOPTS_VALUE; + + ret = config_filter(ctx); + if (ret < 0) + return ret; + s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N); if (!s->fifo) return AVERROR(ENOMEM); @@ -181,13 +221,6 @@ static int config_output(AVFilterLink *outlink) if (ret < 0) return ret; - s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE; - for (int i = 0; i < WEIGHT_LUT_SIZE; i++) { - float w = -i / s->pdiff_lut_scale; - - s->weight_lut[i] = expf(w); - } - ff_anlmdn_init(&s->dsp); return 0; @@ -331,6 +364,22 @@ static int request_frame(AVFilterLink *outlink) return ret; } +static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, + char *res, int res_len, int flags) +{ + int ret; + + ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); + if (ret < 0) + return ret; + + ret = config_filter(ctx); + if (ret < 0) + return ret; + + return 0; +} + static av_cold void uninit(AVFilterContext *ctx) { AudioNLMeansContext *s = ctx->priv; @@ -368,7 +417,7 @@ AVFilter ff_af_anlmdn = { .uninit = uninit, .inputs = inputs, .outputs = outputs, - .process_command = ff_filter_process_command, + .process_command = process_command, .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | AVFILTER_FLAG_SLICE_THREADS, }; |