mirror of https://github.com/FFmpeg/FFmpeg.git
Long overdue for removal, af_aresample should be used instead. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>pull/131/merge
Rostislav Pehlivanov
8 years ago
committed by
Rostislav Pehlivanov
parent
d7896e9b42
commit
a8fe8d6b4a
8 changed files with 2 additions and 367 deletions
@ -1,323 +0,0 @@ |
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/*
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include <stdint.h> |
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#include "libavresample/avresample.h" |
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#include "libavutil/attributes.h" |
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#include "libavutil/audio_fifo.h" |
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#include "libavutil/common.h" |
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#include "libavutil/mathematics.h" |
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#include "libavutil/opt.h" |
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#include "libavutil/samplefmt.h" |
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#include "audio.h" |
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#include "avfilter.h" |
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#include "internal.h" |
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typedef struct ASyncContext { |
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const AVClass *class; |
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AVAudioResampleContext *avr; |
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int64_t pts; ///< timestamp in samples of the first sample in fifo
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int min_delta; ///< pad/trim min threshold in samples
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int first_frame; ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE
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int64_t first_pts; ///< user-specified first expected pts, in samples
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int comp; ///< current resample compensation
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/* options */ |
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int resample; |
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float min_delta_sec; |
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int max_comp; |
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/* set by filter_frame() to signal an output frame to request_frame() */ |
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int got_output; |
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} ASyncContext; |
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#define OFFSET(x) offsetof(ASyncContext, x) |
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#define A AV_OPT_FLAG_AUDIO_PARAM |
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#define F AV_OPT_FLAG_FILTERING_PARAM |
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static const AVOption asyncts_options[] = { |
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{ "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, A|F }, |
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{ "min_delta", "Minimum difference between timestamps and audio data " |
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"(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { .dbl = 0.1 }, 0, INT_MAX, A|F }, |
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{ "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A|F }, |
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{ "first_pts", "Assume the first pts should be this value.", OFFSET(first_pts), AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_MAX, A|F }, |
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{ NULL } |
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}; |
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AVFILTER_DEFINE_CLASS(asyncts); |
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static av_cold int init(AVFilterContext *ctx) |
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{ |
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ASyncContext *s = ctx->priv; |
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s->pts = AV_NOPTS_VALUE; |
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s->first_frame = 1; |
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return 0; |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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ASyncContext *s = ctx->priv; |
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if (s->avr) { |
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avresample_close(s->avr); |
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avresample_free(&s->avr); |
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} |
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} |
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static int config_props(AVFilterLink *link) |
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{ |
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ASyncContext *s = link->src->priv; |
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int ret; |
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s->min_delta = s->min_delta_sec * link->sample_rate; |
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link->time_base = (AVRational){1, link->sample_rate}; |
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s->avr = avresample_alloc_context(); |
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if (!s->avr) |
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return AVERROR(ENOMEM); |
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av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0); |
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av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0); |
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av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0); |
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av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0); |
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av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0); |
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av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0); |
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if (s->resample) |
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av_opt_set_int(s->avr, "force_resampling", 1, 0); |
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if ((ret = avresample_open(s->avr)) < 0) |
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return ret; |
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return 0; |
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} |
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/* get amount of data currently buffered, in samples */ |
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static int64_t get_delay(ASyncContext *s) |
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{ |
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return avresample_available(s->avr) + avresample_get_delay(s->avr); |
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} |
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static void handle_trimming(AVFilterContext *ctx) |
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{ |
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ASyncContext *s = ctx->priv; |
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if (s->pts < s->first_pts) { |
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int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr)); |
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av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n", |
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delta); |
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avresample_read(s->avr, NULL, delta); |
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s->pts += delta; |
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} else if (s->first_frame) |
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s->pts = s->first_pts; |
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} |
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static int request_frame(AVFilterLink *link) |
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{ |
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AVFilterContext *ctx = link->src; |
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ASyncContext *s = ctx->priv; |
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int ret = 0; |
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int nb_samples; |
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s->got_output = 0; |
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ret = ff_request_frame(ctx->inputs[0]); |
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/* flush the fifo */ |
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if (ret == AVERROR_EOF) { |
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if (s->first_pts != AV_NOPTS_VALUE) |
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handle_trimming(ctx); |
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if (nb_samples = get_delay(s)) { |
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AVFrame *buf = ff_get_audio_buffer(link, nb_samples); |
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if (!buf) |
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return AVERROR(ENOMEM); |
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ret = avresample_convert(s->avr, buf->extended_data, |
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buf->linesize[0], nb_samples, NULL, 0, 0); |
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if (ret <= 0) { |
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av_frame_free(&buf); |
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return (ret < 0) ? ret : AVERROR_EOF; |
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} |
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buf->pts = s->pts; |
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return ff_filter_frame(link, buf); |
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} |
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} |
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return ret; |
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} |
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static int write_to_fifo(ASyncContext *s, AVFrame *buf) |
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{ |
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int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, |
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buf->linesize[0], buf->nb_samples); |
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av_frame_free(&buf); |
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return ret; |
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} |
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static int filter_frame(AVFilterLink *inlink, AVFrame *buf) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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ASyncContext *s = ctx->priv; |
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AVFilterLink *outlink = ctx->outputs[0]; |
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int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout); |
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int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts : |
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av_rescale_q(buf->pts, inlink->time_base, outlink->time_base); |
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int out_size, ret; |
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int64_t delta; |
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int64_t new_pts; |
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/* buffer data until we get the next timestamp */ |
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if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) { |
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if (pts != AV_NOPTS_VALUE) { |
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s->pts = pts - get_delay(s); |
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} |
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return write_to_fifo(s, buf); |
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} |
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if (s->first_pts != AV_NOPTS_VALUE) { |
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handle_trimming(ctx); |
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if (!avresample_available(s->avr)) |
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return write_to_fifo(s, buf); |
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} |
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/* when we have two timestamps, compute how many samples would we have
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* to add/remove to get proper sync between data and timestamps */ |
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delta = pts - s->pts - get_delay(s); |
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out_size = avresample_available(s->avr); |
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if (llabs(delta) > s->min_delta || |
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(s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) { |
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av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta); |
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out_size = av_clipl_int32((int64_t)out_size + delta); |
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} else { |
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if (s->resample) { |
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// adjust the compensation if delta is non-zero
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int delay = get_delay(s); |
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int comp = s->comp + av_clip(delta * inlink->sample_rate / delay, |
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-s->max_comp, s->max_comp); |
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if (comp != s->comp) { |
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av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp); |
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if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) { |
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s->comp = comp; |
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} |
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} |
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} |
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// adjust PTS to avoid monotonicity errors with input PTS jitter
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pts -= delta; |
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delta = 0; |
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} |
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if (out_size > 0) { |
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AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size); |
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if (!buf_out) { |
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ret = AVERROR(ENOMEM); |
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goto fail; |
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} |
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if (s->first_frame && delta > 0) { |
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int planar = av_sample_fmt_is_planar(buf_out->format); |
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int planes = planar ? nb_channels : 1; |
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int block_size = av_get_bytes_per_sample(buf_out->format) * |
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(planar ? 1 : nb_channels); |
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int ch; |
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av_samples_set_silence(buf_out->extended_data, 0, delta, |
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nb_channels, buf->format); |
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for (ch = 0; ch < planes; ch++) |
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buf_out->extended_data[ch] += delta * block_size; |
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avresample_read(s->avr, buf_out->extended_data, out_size); |
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for (ch = 0; ch < planes; ch++) |
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buf_out->extended_data[ch] -= delta * block_size; |
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} else { |
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avresample_read(s->avr, buf_out->extended_data, out_size); |
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if (delta > 0) { |
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av_samples_set_silence(buf_out->extended_data, out_size - delta, |
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delta, nb_channels, buf->format); |
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} |
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} |
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buf_out->pts = s->pts; |
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ret = ff_filter_frame(outlink, buf_out); |
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if (ret < 0) |
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goto fail; |
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s->got_output = 1; |
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} else if (avresample_available(s->avr)) { |
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av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " |
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"whole buffer.\n"); |
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} |
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/* drain any remaining buffered data */ |
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avresample_read(s->avr, NULL, avresample_available(s->avr)); |
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new_pts = pts - avresample_get_delay(s->avr); |
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/* check for s->pts monotonicity */ |
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if (new_pts > s->pts) { |
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s->pts = new_pts; |
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ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, |
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buf->linesize[0], buf->nb_samples); |
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} else { |
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av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " |
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"whole buffer.\n"); |
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ret = 0; |
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} |
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s->first_frame = 0; |
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fail: |
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av_frame_free(&buf); |
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return ret; |
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} |
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static const AVFilterPad avfilter_af_asyncts_inputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.filter_frame = filter_frame |
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}, |
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{ NULL } |
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}; |
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static const AVFilterPad avfilter_af_asyncts_outputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.config_props = config_props, |
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.request_frame = request_frame |
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}, |
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{ NULL } |
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}; |
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AVFilter ff_af_asyncts = { |
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.name = "asyncts", |
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.description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps."), |
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.init = init, |
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.uninit = uninit, |
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.priv_size = sizeof(ASyncContext), |
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.priv_class = &asyncts_class, |
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.query_formats = ff_query_formats_all_layouts, |
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.inputs = avfilter_af_asyncts_inputs, |
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.outputs = avfilter_af_asyncts_outputs, |
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}; |
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