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[86.146.196.130]) by smtp.gmail.com with ESMTPSA id c58sm25150104wrc.9.2017.03.05.18.46.53 (version=TLS1_2 cipher=ECDHE-RSA-AES128-GCM-SHA256 bits=128/128); Sun, 05 Mar 2017 18:46:53 -0800 (PST) From: Rostislav Pehlivanov To: ffmpeg-devel@ffmpeg.org Date: Mon, 6 Mar 2017 02:46:48 +0000 Message-Id: <20170306024651.66336-1-atomnuker@gmail.com> X-Mailer: git-send-email 2.12.0.rc1.440.g5b76565f74 Subject: [FFmpeg-devel] [PATCH 1/4] lavfi: remove af_resample 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 Cc: Rostislav Pehlivanov MIME-Version: 1.0 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" af_aresample does the same thing better and doesn't depend on libavresample Signed-off-by: Rostislav Pehlivanov --- libavfilter/Makefile | 1 - libavfilter/af_resample.c | 357 ---------------------------------------------- 2 files changed, 358 deletions(-) delete mode 100644 libavfilter/af_resample.c diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 0ba1c74a26..6b9fba2d4c 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -98,7 +98,6 @@ OBJS-$(CONFIG_LOUDNORM_FILTER) += af_loudnorm.o ebur128.o OBJS-$(CONFIG_LOWPASS_FILTER) += af_biquads.o OBJS-$(CONFIG_PAN_FILTER) += af_pan.o OBJS-$(CONFIG_REPLAYGAIN_FILTER) += af_replaygain.o -OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o OBJS-$(CONFIG_RUBBERBAND_FILTER) += af_rubberband.o OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER) += af_sidechaincompress.o OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o diff --git a/libavfilter/af_resample.c b/libavfilter/af_resample.c deleted file mode 100644 index e3c6a20696..0000000000 --- a/libavfilter/af_resample.c +++ /dev/null @@ -1,357 +0,0 @@ -/* - * 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 - */ - -/** - * @file - * sample format and channel layout conversion audio filter - */ - -#include "libavutil/avassert.h" -#include "libavutil/avstring.h" -#include "libavutil/common.h" -#include "libavutil/dict.h" -#include "libavutil/mathematics.h" -#include "libavutil/opt.h" - -#include "libavresample/avresample.h" - -#include "audio.h" -#include "avfilter.h" -#include "formats.h" -#include "internal.h" - -typedef struct ResampleContext { - const AVClass *class; - AVAudioResampleContext *avr; - AVDictionary *options; - - int resampling; - int64_t next_pts; - int64_t next_in_pts; - - /* set by filter_frame() to signal an output frame to request_frame() */ - int got_output; -} ResampleContext; - -static av_cold int init(AVFilterContext *ctx, AVDictionary **opts) -{ - ResampleContext *s = ctx->priv; - const AVClass *avr_class = avresample_get_class(); - AVDictionaryEntry *e = NULL; - - while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) { - if (av_opt_find(&avr_class, e->key, NULL, 0, - AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN)) - av_dict_set(&s->options, e->key, e->value, 0); - } - - e = NULL; - while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX))) - av_dict_set(opts, e->key, NULL, 0); - - /* do not allow the user to override basic format options */ - av_dict_set(&s->options, "in_channel_layout", NULL, 0); - av_dict_set(&s->options, "out_channel_layout", NULL, 0); - av_dict_set(&s->options, "in_sample_fmt", NULL, 0); - av_dict_set(&s->options, "out_sample_fmt", NULL, 0); - av_dict_set(&s->options, "in_sample_rate", NULL, 0); - av_dict_set(&s->options, "out_sample_rate", NULL, 0); - - return 0; -} - -static av_cold void uninit(AVFilterContext *ctx) -{ - ResampleContext *s = ctx->priv; - - if (s->avr) { - avresample_close(s->avr); - avresample_free(&s->avr); - } - av_dict_free(&s->options); -} - -static int query_formats(AVFilterContext *ctx) -{ - AVFilterLink *inlink = ctx->inputs[0]; - AVFilterLink *outlink = ctx->outputs[0]; - AVFilterFormats *in_formats, *out_formats, *in_samplerates, *out_samplerates; - AVFilterChannelLayouts *in_layouts, *out_layouts; - int ret; - - if (!(in_formats = ff_all_formats (AVMEDIA_TYPE_AUDIO)) || - !(out_formats = ff_all_formats (AVMEDIA_TYPE_AUDIO)) || - !(in_samplerates = ff_all_samplerates ( )) || - !(out_samplerates = ff_all_samplerates ( )) || - !(in_layouts = ff_all_channel_layouts ( )) || - !(out_layouts = ff_all_channel_layouts ( ))) - return AVERROR(ENOMEM); - - if ((ret = ff_formats_ref (in_formats, &inlink->out_formats )) < 0 || - (ret = ff_formats_ref (out_formats, &outlink->in_formats )) < 0 || - (ret = ff_formats_ref (in_samplerates, &inlink->out_samplerates )) < 0 || - (ret = ff_formats_ref (out_samplerates, &outlink->in_samplerates )) < 0 || - (ret = ff_channel_layouts_ref (in_layouts, &inlink->out_channel_layouts)) < 0 || - (ret = ff_channel_layouts_ref (out_layouts, &outlink->in_channel_layouts)) < 0) - return ret; - - return 0; -} - -static int config_output(AVFilterLink *outlink) -{ - AVFilterContext *ctx = outlink->src; - AVFilterLink *inlink = ctx->inputs[0]; - ResampleContext *s = ctx->priv; - char buf1[64], buf2[64]; - int ret; - - int64_t resampling_forced; - - if (s->avr) { - avresample_close(s->avr); - avresample_free(&s->avr); - } - - if (inlink->channel_layout == outlink->channel_layout && - inlink->sample_rate == outlink->sample_rate && - (inlink->format == outlink->format || - (av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 && - av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 && - av_get_planar_sample_fmt(inlink->format) == - av_get_planar_sample_fmt(outlink->format)))) - return 0; - - if (!(s->avr = avresample_alloc_context())) - return AVERROR(ENOMEM); - - if (s->options) { - int ret; - AVDictionaryEntry *e = NULL; - while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX))) - av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value); - - ret = av_opt_set_dict(s->avr, &s->options); - if (ret < 0) - return ret; - } - - av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0); - av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0); - av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0); - av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0); - av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0); - av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0); - - if ((ret = avresample_open(s->avr)) < 0) - return ret; - - av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced); - s->resampling = resampling_forced || (inlink->sample_rate != outlink->sample_rate); - - if (s->resampling) { - outlink->time_base = (AVRational){ 1, outlink->sample_rate }; - s->next_pts = AV_NOPTS_VALUE; - s->next_in_pts = AV_NOPTS_VALUE; - } else - outlink->time_base = inlink->time_base; - - av_get_channel_layout_string(buf1, sizeof(buf1), - -1, inlink ->channel_layout); - av_get_channel_layout_string(buf2, sizeof(buf2), - -1, outlink->channel_layout); - av_log(ctx, AV_LOG_VERBOSE, - "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n", - av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1, - av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2); - - return 0; -} - -static int request_frame(AVFilterLink *outlink) -{ - AVFilterContext *ctx = outlink->src; - ResampleContext *s = ctx->priv; - int ret = 0; - - s->got_output = 0; - while (ret >= 0 && !s->got_output) - ret = ff_request_frame(ctx->inputs[0]); - - /* flush the lavr delay buffer */ - if (ret == AVERROR_EOF && s->avr) { - AVFrame *frame; - int nb_samples = avresample_get_out_samples(s->avr, 0); - - if (!nb_samples) - return ret; - - frame = ff_get_audio_buffer(outlink, nb_samples); - if (!frame) - return AVERROR(ENOMEM); - - ret = avresample_convert(s->avr, frame->extended_data, - frame->linesize[0], nb_samples, - NULL, 0, 0); - if (ret <= 0) { - av_frame_free(&frame); - return (ret == 0) ? AVERROR_EOF : ret; - } - - frame->nb_samples = ret; - frame->pts = s->next_pts; - return ff_filter_frame(outlink, frame); - } - return ret; -} - -static int filter_frame(AVFilterLink *inlink, AVFrame *in) -{ - AVFilterContext *ctx = inlink->dst; - ResampleContext *s = ctx->priv; - AVFilterLink *outlink = ctx->outputs[0]; - int ret; - - if (s->avr) { - AVFrame *out; - int delay, nb_samples; - - /* maximum possible samples lavr can output */ - delay = avresample_get_delay(s->avr); - nb_samples = avresample_get_out_samples(s->avr, in->nb_samples); - - out = ff_get_audio_buffer(outlink, nb_samples); - if (!out) { - ret = AVERROR(ENOMEM); - goto fail; - } - - ret = avresample_convert(s->avr, out->extended_data, out->linesize[0], - nb_samples, in->extended_data, in->linesize[0], - in->nb_samples); - if (ret <= 0) { - av_frame_free(&out); - if (ret < 0) - goto fail; - } - - av_assert0(!avresample_available(s->avr)); - - if (s->resampling && s->next_pts == AV_NOPTS_VALUE) { - if (in->pts == AV_NOPTS_VALUE) { - av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, " - "assuming 0.\n"); - s->next_pts = 0; - } else - s->next_pts = av_rescale_q(in->pts, inlink->time_base, - outlink->time_base); - } - - if (ret > 0) { - out->nb_samples = ret; - - ret = av_frame_copy_props(out, in); - if (ret < 0) { - av_frame_free(&out); - goto fail; - } - - if (s->resampling) { - out->sample_rate = outlink->sample_rate; - /* Only convert in->pts if there is a discontinuous jump. - This ensures that out->pts tracks the number of samples actually - output by the resampler in the absence of such a jump. - Otherwise, the rounding in av_rescale_q() and av_rescale() - causes off-by-1 errors. */ - if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) { - out->pts = av_rescale_q(in->pts, inlink->time_base, - outlink->time_base) - - av_rescale(delay, outlink->sample_rate, - inlink->sample_rate); - } else - out->pts = s->next_pts; - - s->next_pts = out->pts + out->nb_samples; - s->next_in_pts = in->pts + in->nb_samples; - } else - out->pts = in->pts; - - ret = ff_filter_frame(outlink, out); - s->got_output = 1; - } - -fail: - av_frame_free(&in); - } else { - in->format = outlink->format; - ret = ff_filter_frame(outlink, in); - s->got_output = 1; - } - - return ret; -} - -static const AVClass *resample_child_class_next(const AVClass *prev) -{ - return prev ? NULL : avresample_get_class(); -} - -static void *resample_child_next(void *obj, void *prev) -{ - ResampleContext *s = obj; - return prev ? NULL : s->avr; -} - -static const AVClass resample_class = { - .class_name = "resample", - .item_name = av_default_item_name, - .version = LIBAVUTIL_VERSION_INT, - .child_class_next = resample_child_class_next, - .child_next = resample_child_next, -}; - -static const AVFilterPad avfilter_af_resample_inputs[] = { - { - .name = "default", - .type = AVMEDIA_TYPE_AUDIO, - .filter_frame = filter_frame, - }, - { NULL } -}; - -static const AVFilterPad avfilter_af_resample_outputs[] = { - { - .name = "default", - .type = AVMEDIA_TYPE_AUDIO, - .config_props = config_output, - .request_frame = request_frame - }, - { NULL } -}; - -AVFilter ff_af_resample = { - .name = "resample", - .description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."), - .priv_size = sizeof(ResampleContext), - .priv_class = &resample_class, - .init_dict = init, - .uninit = uninit, - .query_formats = query_formats, - .inputs = avfilter_af_resample_inputs, - .outputs = avfilter_af_resample_outputs, -};