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[79.124.17.100]) by mx.google.com with ESMTP id 93si211142wrh.187.2017.09.06.09.50.36; Wed, 06 Sep 2017 09:50:39 -0700 (PDT) Received-SPF: pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) client-ip=79.124.17.100; Authentication-Results: mx.google.com; dkim=neutral (body hash did not verify) header.i=@gmail.com header.s=20161025 header.b=BwpgWH/i; spf=pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) smtp.mailfrom=ffmpeg-devel-bounces@ffmpeg.org; dmarc=fail (p=NONE sp=NONE dis=NONE) header.from=gmail.com Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id C18AE68A3A2; Wed, 6 Sep 2017 19:50:32 +0300 (EEST) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from mail-wm0-f42.google.com (mail-wm0-f42.google.com [74.125.82.42]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id 09DB4689FAD for ; Wed, 6 Sep 2017 19:50:27 +0300 (EEST) Received: by mail-wm0-f42.google.com with SMTP id u26so33311137wma.0 for ; Wed, 06 Sep 2017 09:50:30 -0700 (PDT) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20161025; h=from:to:subject:date:message-id; bh=t4kSlzaE9OYb+4yKoBUwsYJ9T5LvPzhm9lVWEP067dk=; b=BwpgWH/izEvfSSQdwKO9+FFXEs3KldgVZj6orbcHREJblY4hN02ZZqJdRiA5OhBpgQ KBZZZaasmjRafMA4IPQMQzhxFtxOwyESjBEJfJ5aEtlRaB23/H6I6WbQ03pzYpVldxl3 vunRb7JDBBZMm+Dx263VrkO+vcguh2FTTecHARkBMhhZf/1oz+S1EBsryfV2sGNUxrzG G768KCQVTT4rj8/LBqLW7gA11J06wpvfFM3kcrhn1yf6UyA3cbTe0HMhRPrGHCrM9DVx r8N5mDXFFjzSVv3ppTSJgEVr96l9C0yCUix9DweJ84j3th2i/mO6GYhXSyQuM2+ZJnWg 8+Ag== X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=1e100.net; s=20161025; h=x-gm-message-state:from:to:subject:date:message-id; bh=t4kSlzaE9OYb+4yKoBUwsYJ9T5LvPzhm9lVWEP067dk=; b=YuttxCfRJHDA8MhvHxMqlGFisB3anNeg0vhXOaMULNfnuXHFEtOKHKE1oBSO4/xHqW TMrdQPnOtU6pCoMkjA7e1DmWBJ0ovlZU0nIzD9FLEjR8+O3cmjHqWR2tNs+ymezHYaq+ ymHtC1cLo5w4NDDUV/jKeO5QgZ/cTIiygXq8tfOGKYoV8p+aqKo7akfqKPEcGE9skwgy qyFj4/Vw8YXjj1RcRtl6px8yfDeetl3bD84dcRHV8K15APPR+LW+G3xbEJuPeUvllPFM x9elGIaDYGV8W3C9/DstIx0kS564A4ebw7TDiolqGzA5CTzQcpAJeK/j+TuisyX41LxN wWQA== X-Gm-Message-State: AHPjjUiD+PvwFpxNvG58aKuhd9FFSPggRVTxyAuYccWqeuihfdzhUJFZ Uw8E0BAusZdp2sKe X-Google-Smtp-Source: ADKCNb6ruuAL4zymQ1t1ij5dPQLVUr6yRTn+6FZPw+wUL9y+81gozaVtvZdhLk8NFVdsn0s8d6k6og== X-Received: by 10.28.92.136 with SMTP id q130mr272767wmb.104.1504716628903; Wed, 06 Sep 2017 09:50:28 -0700 (PDT) Received: from localhost.localdomain ([94.250.174.60]) by smtp.gmail.com with ESMTPSA id n6sm1818253wmg.11.2017.09.06.09.50.27 for (version=TLS1_2 cipher=ECDHE-RSA-AES128-GCM-SHA256 bits=128/128); Wed, 06 Sep 2017 09:50:28 -0700 (PDT) From: Paul B Mahol To: ffmpeg-devel@ffmpeg.org Date: Wed, 6 Sep 2017 18:50:15 +0200 Message-Id: <20170906165015.15673-1-onemda@gmail.com> X-Mailer: git-send-email 2.9.3 Subject: [FFmpeg-devel] [PATCH] avfilter: add Haas stereo enhancer X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches MIME-Version: 1.0 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Signed-off-by: Paul B Mahol --- doc/filters.texi | 64 ++++++++++++++ libavfilter/Makefile | 1 + libavfilter/af_haas.c | 226 +++++++++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 292 insertions(+) create mode 100644 libavfilter/af_haas.c diff --git a/doc/filters.texi b/doc/filters.texi index 7790367..c3c54fd 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2770,6 +2770,70 @@ Set delay-line interpolation, @var{linear} or @var{quadratic}. Default is @var{linear}. @end table +@section haas +Apply Haas effect to audio. + +Note that this makes most sense to apply on mono signals. +With this filter applied to mono signals it give some directionality and +streches its stereo image. + +The filter accepts the following options: + +@table @option +@item level_in +Set input level. By default is @var{1}, or 0dB + +@item level_out +Set output level. By default is @var{1}, or 0dB. + +@item side_gain +Set gain applied to side part of signal. By default is @var{1}. + +@item middle_source +Set kind of middle source. Can be one of the following: + +@table @samp +@item left +Pick left channel. + +@item right +Pick right channel. + +@item mid +Pick middle part signal of stereo image. + +@item side +Pick side part signal of stereo image. +@end table + +@item middle_phase +Change middle phase. By default is disabled. + +@item left_delay +Set left channel delay. By default is @var{2.05} milliseconds. + +@item left_balance +Set left channel balance. By default is @var{-1}. + +@item left_gain +Set left channel gain. By default is @var{1}. + +@item left_phase +Change left phase. By default is disabled. + +@item right_delay +Set right channel delay. By defaults is @var{2.12} milliseconds. + +@item right_balance +Set right channel balance. By default is @var{1}. + +@item right_gain +Set right channel gain. By default is @var{1}. + +@item right_phase +Change right phase. By default is enabled. +@end table + @section hdcd Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 1e460ab..4268633 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -91,6 +91,7 @@ OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o OBJS-$(CONFIG_EXTRASTEREO_FILTER) += af_extrastereo.o OBJS-$(CONFIG_FIREQUALIZER_FILTER) += af_firequalizer.o OBJS-$(CONFIG_FLANGER_FILTER) += af_flanger.o generate_wave_table.o +OBJS-$(CONFIG_HAAS_FILTER) += af_haas.o OBJS-$(CONFIG_HDCD_FILTER) += af_hdcd.o OBJS-$(CONFIG_HEADPHONE_FILTER) += af_headphone.o OBJS-$(CONFIG_HIGHPASS_FILTER) += af_biquads.o diff --git a/libavfilter/af_haas.c b/libavfilter/af_haas.c new file mode 100644 index 0000000..a9761bd --- /dev/null +++ b/libavfilter/af_haas.c @@ -0,0 +1,226 @@ +/* + * Copyright (c) 2001-2010 Vladimir Sadovnikov + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavutil/channel_layout.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" + +typedef struct HaasContext { + const AVClass *class; + + int par_m_source; + int par_delay0; + int par_delay1; + int par_phase0; + int par_phase1; + int par_middle_phase; + double par_side_gain; + double par_gain0; + double par_gain1; + double par_balance0; + double par_balance1; + double level_in; + double level_out; + + double *buffer; + int buffer_size; + uint32_t write_ptr; + uint32_t delay[2]; + double balance_l[2]; + double balance_r[2]; + double phase0; + double phase1; +} HaasContext; + +#define OFFSET(x) offsetof(HaasContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption haas_options[] = { + { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, + { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, + { "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, + { "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, "source" }, + { "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "source" }, + { "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "source" }, + { "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "source" }, + { "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "source" }, + { "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, + { "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, 40, A }, + { "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A }, + { "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, + { "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, + { "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, 40, A }, + { "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A }, + { "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, + { "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(haas); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layout = NULL; + int ret; + + if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 || + (ret = ff_set_common_formats (ctx , formats )) < 0 || + (ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 || + (ret = ff_set_common_channel_layouts (ctx , layout )) < 0) + return ret; + + formats = ff_all_samplerates(); + return ff_set_common_samplerates(ctx, formats); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + HaasContext *s = ctx->priv; + size_t min_buf_size = (size_t)(inlink->sample_rate * 10 * 0.001); + size_t new_buf_size = 1; + + while (new_buf_size < min_buf_size) + new_buf_size <<= 1; + + av_freep(&s->buffer); + s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer)); + if (!s->buffer) + return AVERROR(ENOMEM); + + s->buffer_size = new_buf_size; + s->write_ptr = 0; + + s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate); + s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate); + + s->phase0 = s->par_phase0 ? 1.0 : -1.0; + s->phase1 = s->par_phase1 ? 1.0 : -1.0; + + s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0; + s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0; + s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1; + s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1; + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + HaasContext *s = ctx->priv; + const double *src = (const double *)in->data[0]; + const double level_in = s->level_in; + const double level_out = s->level_out; + const uint32_t mask = s->buffer_size - 1; + double *buffer = s->buffer; + AVFrame *out; + double *dst; + int n; + + if (av_frame_is_writable(in)) { + out = in; + } else { + out = ff_get_audio_buffer(inlink, in->nb_samples); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + } + dst = (double *)out->data[0]; + + for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) { + double mid, side[2], side_l, side_r; + uint32_t s0_ptr, s1_ptr; + + switch (s->par_m_source) { + case 0: mid = src[0]; break; + case 1: mid = src[1]; break; + case 2: mid = (src[0] + src[1]) * 0.5; break; + case 3: mid = (src[0] - src[1]) * 0.5; break; + } + + buffer[s->write_ptr] = mid; + + mid *= level_in; + + s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask; + s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask; + + if (s->par_middle_phase) + mid = -mid; + + side[0] = buffer[s0_ptr] * s->par_side_gain; + side[1] = buffer[s1_ptr] * s->par_side_gain; + side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1]; + side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0]; + + dst[0] = (mid + side_l) * level_out; + dst[1] = (mid + side_r) * level_out; + + s->write_ptr = (s->write_ptr + 1) & mask; + } + + if (out != in) + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + HaasContext *s = ctx->priv; + + av_freep(&s->buffer); + s->buffer_size = 0; +} + +static const AVFilterPad inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_haas = { + .name = "haas", + .description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."), + .query_formats = query_formats, + .priv_size = sizeof(HaasContext), + .priv_class = &haas_class, + .uninit = uninit, + .inputs = inputs, + .outputs = outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 9a2cfea..9bbc6d6 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -104,6 +104,7 @@ static void register_all(void) REGISTER_FILTER(EXTRASTEREO, extrastereo, af); REGISTER_FILTER(FIREQUALIZER, firequalizer, af); REGISTER_FILTER(FLANGER, flanger, af); + REGISTER_FILTER(HAAS, haas, af); REGISTER_FILTER(HDCD, hdcd, af); REGISTER_FILTER(HEADPHONE, headphone, af); REGISTER_FILTER(HIGHPASS, highpass, af);