From b9f804b566c4c528714e4ec5e63675ad7ba5fefd Mon Sep 17 00:00:00 2001 From: wm4 Date: Thu, 18 Jan 2018 14:44:20 +0100 Subject: audio: rewrite filtering glue code Use the new filtering code for audio too. --- filters/f_swresample.c | 717 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 717 insertions(+) create mode 100644 filters/f_swresample.c (limited to 'filters/f_swresample.c') diff --git a/filters/f_swresample.c b/filters/f_swresample.c new file mode 100644 index 0000000000..48bd08d847 --- /dev/null +++ b/filters/f_swresample.c @@ -0,0 +1,717 @@ +/* + * This file is part of mpv. + * + * mpv 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. + * + * mpv 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 mpv. If not, see . + */ + +#include +#include +#include +#include +#include + +#include "config.h" + +#include "audio/aframe.h" +#include "audio/fmt-conversion.h" +#include "audio/format.h" +#include "common/common.h" +#include "common/av_common.h" +#include "common/msg.h" +#include "options/m_config.h" +#include "options/m_option.h" + +#include "f_swresample.h" +#include "filter_internal.h" + +#define HAVE_LIBSWRESAMPLE (!HAVE_LIBAV) +#define HAVE_LIBAVRESAMPLE HAVE_LIBAV + +#if HAVE_LIBAVRESAMPLE +#include +#elif HAVE_LIBSWRESAMPLE +#include +#define AVAudioResampleContext SwrContext +#define avresample_alloc_context swr_alloc +#define avresample_open swr_init +#define avresample_close(x) do { } while(0) +#define avresample_free swr_free +#define avresample_available(x) 0 +#define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \ + swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples) +#define avresample_set_channel_mapping swr_set_channel_mapping +#define avresample_set_compensation swr_set_compensation +#else +#error "config.h broken or no resampler found" +#endif + +struct priv { + struct mp_log *log; + bool is_resampling; + struct AVAudioResampleContext *avrctx; + struct mp_aframe *avrctx_fmt; // output format of avrctx + struct mp_aframe *pool_fmt; // format used to allocate frames for avrctx output + struct mp_aframe *pre_out_fmt; // format before final conversion + struct AVAudioResampleContext *avrctx_out; // for output channel reordering + struct mp_resample_opts *opts; // opts requested by the user + // At least libswresample keeps a pointer around for this: + int reorder_in[MP_NUM_CHANNELS]; + int reorder_out[MP_NUM_CHANNELS]; + struct mp_aframe_pool *reorder_buffer; + struct mp_aframe_pool *out_pool; + + int in_rate_user; // user input sample rate + int in_rate; // actual rate (used by lavr), adjusted for playback speed + int in_format; + struct mp_chmap in_channels; + int out_rate; + int out_format; + struct mp_chmap out_channels; + + double current_pts; + + double cmd_speed; + double speed; + + struct mp_swresample public; +}; + +#define OPT_BASE_STRUCT struct mp_resample_opts +const struct m_sub_options resample_conf = { + .opts = (const m_option_t[]) { + OPT_INTRANGE("audio-resample-filter-size", filter_size, 0, 0, 32), + OPT_INTRANGE("audio-resample-phase-shift", phase_shift, 0, 0, 30), + OPT_FLAG("audio-resample-linear", linear, 0), + OPT_DOUBLE("audio-resample-cutoff", cutoff, M_OPT_RANGE, + .min = 0, .max = 1), + OPT_FLAG("audio-normalize-downmix", normalize, 0), + OPT_KEYVALUELIST("audio-swresample-o", avopts, 0), + {0} + }, + .size = sizeof(struct mp_resample_opts), + .defaults = &(const struct mp_resample_opts)MP_RESAMPLE_OPTS_DEF, + .change_flags = UPDATE_AUDIO, +}; + +#if HAVE_LIBAVRESAMPLE +static double get_delay(struct priv *p) +{ + return avresample_get_delay(p->avrctx) / (double)p->in_rate + + avresample_available(p->avrctx) / (double)p->out_rate; +} +static int get_out_samples(struct priv *p, int in_samples) +{ + return avresample_get_out_samples(p->avrctx, in_samples); +} +#else +static double get_delay(struct priv *p) +{ + int64_t base = p->in_rate * (int64_t)p->out_rate; + return swr_get_delay(p->avrctx, base) / (double)base; +} +static int get_out_samples(struct priv *p, int in_samples) +{ + return swr_get_out_samples(p->avrctx, in_samples); +} +#endif + +static void close_lavrr(struct priv *p) +{ + if (p->avrctx) + avresample_close(p->avrctx); + avresample_free(&p->avrctx); + if (p->avrctx_out) + avresample_close(p->avrctx_out); + avresample_free(&p->avrctx_out); + + TA_FREEP(&p->pre_out_fmt); + TA_FREEP(&p->avrctx_fmt); + TA_FREEP(&p->pool_fmt); +} + +static int rate_from_speed(int rate, double speed) +{ + return lrint(rate * speed); +} + +static struct mp_chmap fudge_pairs[][2] = { + {MP_CHMAP2(BL, BR), MP_CHMAP2(SL, SR)}, + {MP_CHMAP2(SL, SR), MP_CHMAP2(BL, BR)}, + {MP_CHMAP2(SDL, SDR), MP_CHMAP2(SL, SR)}, + {MP_CHMAP2(SL, SR), MP_CHMAP2(SDL, SDR)}, +}; + +// Modify out_layout and return the new value. The intention is reducing the +// loss libswresample's rematrixing will cause by exchanging similar, but +// strictly speaking incompatible channel pairs. For example, 7.1 should be +// changed to 7.1(wide) without dropping the SL/SR channels. (We still leave +// it to libswresample to create the remix matrix.) +static uint64_t fudge_layout_conversion(struct priv *p, + uint64_t in, uint64_t out) +{ + for (int n = 0; n < MP_ARRAY_SIZE(fudge_pairs); n++) { + uint64_t a = mp_chmap_to_lavc(&fudge_pairs[n][0]); + uint64_t b = mp_chmap_to_lavc(&fudge_pairs[n][1]); + if ((in & a) == a && (in & b) == 0 && + (out & a) == 0 && (out & b) == b) + { + out = (out & ~b) | a; + + MP_VERBOSE(p, "Fudge: %s -> %s\n", + mp_chmap_to_str(&fudge_pairs[n][0]), + mp_chmap_to_str(&fudge_pairs[n][1])); + } + } + return out; +} + +// mp_chmap_get_reorder() performs: +// to->speaker[n] = from->speaker[src[n]] +// but libavresample does: +// to->speaker[dst[n]] = from->speaker[n] +static void transpose_order(int *map, int num) +{ + int nmap[MP_NUM_CHANNELS] = {0}; + for (int n = 0; n < num; n++) { + for (int i = 0; i < num; i++) { + if (map[n] == i) + nmap[i] = n; + } + } + memcpy(map, nmap, sizeof(nmap)); +} + +static bool configure_lavrr(struct priv *p, bool verbose) +{ + close_lavrr(p); + + p->in_rate = rate_from_speed(p->in_rate_user, p->speed); + + MP_VERBOSE(p, "%dHz %s %s -> %dHz %s %s\n", + p->in_rate, mp_chmap_to_str(&p->in_channels), + af_fmt_to_str(p->in_format), + p->out_rate, mp_chmap_to_str(&p->out_channels), + af_fmt_to_str(p->out_format)); + + p->avrctx = avresample_alloc_context(); + p->avrctx_out = avresample_alloc_context(); + if (!p->avrctx || !p->avrctx_out) + goto error; + + enum AVSampleFormat in_samplefmt = af_to_avformat(p->in_format); + enum AVSampleFormat out_samplefmt = af_to_avformat(p->out_format); + enum AVSampleFormat out_samplefmtp = av_get_planar_sample_fmt(out_samplefmt); + + if (in_samplefmt == AV_SAMPLE_FMT_NONE || + out_samplefmt == AV_SAMPLE_FMT_NONE || + out_samplefmtp == AV_SAMPLE_FMT_NONE) + { + MP_ERR(p, "unsupported conversion: %s -> %s\n", + af_fmt_to_str(p->in_format), af_fmt_to_str(p->out_format)); + goto error; + } + + av_opt_set_int(p->avrctx, "filter_size", p->opts->filter_size, 0); + av_opt_set_int(p->avrctx, "phase_shift", p->opts->phase_shift, 0); + av_opt_set_int(p->avrctx, "linear_interp", p->opts->linear, 0); + + double cutoff = p->opts->cutoff; + if (cutoff <= 0.0) + cutoff = MPMAX(1.0 - 6.5 / (p->opts->filter_size + 8), 0.80); + av_opt_set_double(p->avrctx, "cutoff", cutoff, 0); + + int normalize = p->opts->normalize; +#if HAVE_LIBSWRESAMPLE + av_opt_set_double(p->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0); +#else + av_opt_set_int(p->avrctx, "normalize_mix_level", !!normalize, 0); +#endif + + if (mp_set_avopts(p->log, p->avrctx, p->opts->avopts) < 0) + goto error; + + struct mp_chmap map_in = p->in_channels; + struct mp_chmap map_out = p->out_channels; + + // Try not to do any remixing if at least one is "unknown". Some corner + // cases also benefit from disabling all channel handling logic if the + // src/dst layouts are the same (like fl-fr-na -> fl-fr-na). + if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out) || + mp_chmap_equals(&map_in, &map_out)) + { + mp_chmap_set_unknown(&map_in, map_in.num); + mp_chmap_set_unknown(&map_out, map_out.num); + } + + // unchecked: don't take any channel reordering into account + uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in); + uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out); + + struct mp_chmap in_lavc, out_lavc; + mp_chmap_from_lavc(&in_lavc, in_ch_layout); + mp_chmap_from_lavc(&out_lavc, out_ch_layout); + + if (verbose && !mp_chmap_equals(&in_lavc, &out_lavc)) { + MP_VERBOSE(p, "Remix: %s -> %s\n", mp_chmap_to_str(&in_lavc), + mp_chmap_to_str(&out_lavc)); + } + + if (in_lavc.num != map_in.num) { + // For handling NA channels, we would have to add a planarization step. + MP_FATAL(p, "Unsupported input channel layout %s.\n", + mp_chmap_to_str(&map_in)); + goto error; + } + + mp_chmap_get_reorder(p->reorder_in, &map_in, &in_lavc); + transpose_order(p->reorder_in, map_in.num); + + if (mp_chmap_equals(&out_lavc, &map_out)) { + // No intermediate step required - output new format directly. + out_samplefmtp = out_samplefmt; + } else { + // Verify that we really just reorder and/or insert NA channels. + struct mp_chmap withna = out_lavc; + mp_chmap_fill_na(&withna, map_out.num); + if (withna.num != map_out.num) + goto error; + } + mp_chmap_get_reorder(p->reorder_out, &out_lavc, &map_out); + + p->pre_out_fmt = mp_aframe_create(); + mp_aframe_set_rate(p->pre_out_fmt, p->out_rate); + mp_aframe_set_chmap(p->pre_out_fmt, &p->out_channels); + mp_aframe_set_format(p->pre_out_fmt, p->out_format); + + p->avrctx_fmt = mp_aframe_create(); + mp_aframe_config_copy(p->avrctx_fmt, p->pre_out_fmt); + mp_aframe_set_chmap(p->avrctx_fmt, &out_lavc); + mp_aframe_set_format(p->avrctx_fmt, af_from_avformat(out_samplefmtp)); + + // If there are NA channels, the final output will have more channels than + // the avrctx output. Also, avrctx will output planar (out_samplefmtp was + // not overwritten). Allocate the output frame with more channels, so the + // NA channels can be trivially added. + p->pool_fmt = mp_aframe_create(); + mp_aframe_config_copy(p->pool_fmt, p->avrctx_fmt); + if (map_out.num > out_lavc.num) + mp_aframe_set_chmap(p->pool_fmt, &map_out); + + out_ch_layout = fudge_layout_conversion(p, in_ch_layout, out_ch_layout); + + // Real conversion; output is input to avrctx_out. + av_opt_set_int(p->avrctx, "in_channel_layout", in_ch_layout, 0); + av_opt_set_int(p->avrctx, "out_channel_layout", out_ch_layout, 0); + av_opt_set_int(p->avrctx, "in_sample_rate", p->in_rate, 0); + av_opt_set_int(p->avrctx, "out_sample_rate", p->out_rate, 0); + av_opt_set_int(p->avrctx, "in_sample_fmt", in_samplefmt, 0); + av_opt_set_int(p->avrctx, "out_sample_fmt", out_samplefmtp, 0); + + // Just needs the correct number of channels for deplanarization. + struct mp_chmap fake_chmap; + mp_chmap_set_unknown(&fake_chmap, map_out.num); + uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap); + if (!fake_out_ch_layout) + goto error; + av_opt_set_int(p->avrctx_out, "in_channel_layout", fake_out_ch_layout, 0); + av_opt_set_int(p->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0); + + av_opt_set_int(p->avrctx_out, "in_sample_fmt", out_samplefmtp, 0); + av_opt_set_int(p->avrctx_out, "out_sample_fmt", out_samplefmt, 0); + av_opt_set_int(p->avrctx_out, "in_sample_rate", p->out_rate, 0); + av_opt_set_int(p->avrctx_out, "out_sample_rate", p->out_rate, 0); + + // API has weird requirements, quoting avresample.h: + // * This function can only be called when the allocated context is not open. + // * Also, the input channel layout must have already been set. + avresample_set_channel_mapping(p->avrctx, p->reorder_in); + + p->is_resampling = false; + + if (avresample_open(p->avrctx) < 0 || avresample_open(p->avrctx_out) < 0) { + MP_ERR(p, "Cannot open Libavresample context.\n"); + goto error; + } + return true; + +error: + close_lavrr(p); + mp_filter_internal_mark_failed(p->public.f); + MP_FATAL(p, "libswresample failed to initialize.\n"); + return false; +} + +static void reset(struct mp_filter *f) +{ + struct priv *p = f->priv; + + p->current_pts = MP_NOPTS_VALUE; + + if (!p->avrctx) + return; +#if HAVE_LIBSWRESAMPLE + swr_close(p->avrctx); + if (swr_init(p->avrctx) < 0) + close_lavrr(p); +#else + while (avresample_read(p->avrctx, NULL, 1000) > 0) {} +#endif +} + +static void extra_output_conversion(struct mp_aframe *mpa) +{ + int format = af_fmt_from_planar(mp_aframe_get_format(mpa)); + int num_planes = mp_aframe_get_planes(mpa); + uint8_t **planes = mp_aframe_get_data_rw(mpa); + if (!planes) + return; + for (int p = 0; p < num_planes; p++) { + void *ptr = planes[p]; + int total = mp_aframe_get_total_plane_samples(mpa); + if (format == AF_FORMAT_FLOAT) { + for (int s = 0; s < total; s++) + ((float *)ptr)[s] = av_clipf(((float *)ptr)[s], -1.0f, 1.0f); + } else if (format == AF_FORMAT_DOUBLE) { + for (int s = 0; s < total; s++) + ((double *)ptr)[s] = MPCLAMP(((double *)ptr)[s], -1.0, 1.0); + } + } +} + +// This relies on the tricky way mpa was allocated. +static bool reorder_planes(struct mp_aframe *mpa, int *reorder, + struct mp_chmap *newmap) +{ + if (!mp_aframe_set_chmap(mpa, newmap)) + return false; + + int num_planes = newmap->num; + uint8_t **planes = mp_aframe_get_data_rw(mpa); + uint8_t *old_planes[MP_NUM_CHANNELS]; + assert(num_planes <= MP_NUM_CHANNELS); + for (int n = 0; n < num_planes; n++) + old_planes[n] = planes[n]; + + int next_na = 0; + for (int n = 0; n < num_planes; n++) + next_na += newmap->speaker[n] != MP_SPEAKER_ID_NA; + + for (int n = 0; n < num_planes; n++) { + int src = reorder[n]; + assert(src >= -1 && src < num_planes); + if (src >= 0) { + planes[n] = old_planes[src]; + } else { + assert(next_na < num_planes); + planes[n] = old_planes[next_na++]; + // The NA planes were never written by avrctx, so clear them. + af_fill_silence(planes[n], + mp_aframe_get_sstride(mpa) * mp_aframe_get_size(mpa), + mp_aframe_get_format(mpa)); + } + } + + return true; +} + +static int resample_frame(struct AVAudioResampleContext *r, + struct mp_aframe *out, struct mp_aframe *in) +{ + // Be aware that the channel layout and count can be different for in and + // out frames. In some situations the caller will fix up the frames before + // or after conversion. The sample rates can also be different. + AVFrame *av_i = in ? mp_aframe_get_raw_avframe(in) : NULL; + AVFrame *av_o = out ? mp_aframe_get_raw_avframe(out) : NULL; + return avresample_convert(r, + av_o ? av_o->extended_data : NULL, + av_o ? av_o->linesize[0] : 0, + av_o ? av_o->nb_samples : 0, + av_i ? av_i->extended_data : NULL, + av_i ? av_i->linesize[0] : 0, + av_i ? av_i->nb_samples : 0); +} + +static struct mp_frame filter_resample_output(struct priv *p, + struct mp_aframe *in) +{ + struct mp_aframe *out = NULL; + + if (!p->avrctx) + goto error; + + int samples = get_out_samples(p, in ? mp_aframe_get_size(in) : 0); + out = mp_aframe_create(); + mp_aframe_config_copy(out, p->pool_fmt); + if (mp_aframe_pool_allocate(p->out_pool, out, samples) < 0) + goto error; + + int out_samples = 0; + if (samples) { + out_samples = resample_frame(p->avrctx, out, in); + if (out_samples < 0 || out_samples > samples) + goto error; + mp_aframe_set_size(out, out_samples); + } + + struct mp_chmap out_chmap; + if (!mp_aframe_get_chmap(p->pool_fmt, &out_chmap)) + goto error; + if (!reorder_planes(out, p->reorder_out, &out_chmap)) + goto error; + + if (!mp_aframe_config_equals(out, p->pre_out_fmt)) { + struct mp_aframe *new = mp_aframe_create(); + mp_aframe_config_copy(new, p->pre_out_fmt); + if (mp_aframe_pool_allocate(p->reorder_buffer, new, out_samples) < 0) { + talloc_free(new); + goto error; + } + int got = 0; + if (out_samples) + got = resample_frame(p->avrctx_out, new, out); + talloc_free(out); + out = new; + if (got != out_samples) + goto error; + } + + extra_output_conversion(out); + + if (in) { + mp_aframe_copy_attributes(out, in); + p->current_pts = mp_aframe_end_pts(in); + } + + if (out_samples) { + if (p->current_pts != MP_NOPTS_VALUE) { + double delay = get_delay(p) * mp_aframe_get_speed(out) + + mp_aframe_duration(out); + mp_aframe_set_pts(out, p->current_pts - delay); + mp_aframe_mul_speed(out, p->speed); + } + } else { + TA_FREEP(&out); + } + + return out ? MAKE_FRAME(MP_FRAME_AUDIO, out) : MP_NO_FRAME; +error: + talloc_free(out); + MP_ERR(p, "Error on resampling.\n"); + mp_filter_internal_mark_failed(p->public.f); + return MP_NO_FRAME; +} + +static void process(struct mp_filter *f) +{ + struct priv *p = f->priv; + + if (!mp_pin_can_transfer_data(f->ppins[1], f->ppins[0])) + return; + + p->speed = p->cmd_speed * p->public.speed; + + struct mp_frame frame = mp_pin_out_read(f->ppins[0]); + + struct mp_aframe *input = NULL; + if (frame.type == MP_FRAME_AUDIO) { + input = frame.data; + } else if (frame.type != MP_FRAME_EOF) { + MP_ERR(p, "Unsupported frame type.\n"); + mp_frame_unref(&frame); + mp_filter_internal_mark_failed(f); + return; + } + + if (!input && !p->avrctx) { + // Obviously no draining needed. + mp_pin_in_write(f->ppins[1], MP_EOF_FRAME); + return; + } + + if (input) { + struct mp_swresample *s = &p->public; + + int in_rate = mp_aframe_get_rate(input); + int in_format = mp_aframe_get_format(input); + struct mp_chmap in_channels = {0}; + mp_aframe_get_chmap(input, &in_channels); + + if (!in_rate || !in_format || !in_channels.num) { + MP_ERR(p, "Frame with invalid format unsupported\n"); + mp_frame_unref(&frame); + mp_filter_internal_mark_failed(f); + return; + } + + int out_rate = s->out_rate ? s->out_rate : in_rate; + int out_format = s->out_format ? s->out_format : in_format; + struct mp_chmap out_channels = + s->out_channels.num ? s->out_channels : in_channels; + + if (p->in_rate_user != in_rate || + p->in_format != in_format || + !mp_chmap_equals(&p->in_channels, &in_channels) || + p->out_rate != out_rate || + p->out_format != out_format || + !mp_chmap_equals(&p->out_channels, &out_channels) || + !p->avrctx) + { + if (p->avrctx) { + // drain remaining audio + struct mp_frame out = filter_resample_output(p, NULL); + if (out.type) { + mp_pin_in_write(f->ppins[1], out); + // continue filtering next time. + mp_pin_out_unread(f->ppins[0], frame); + input = NULL; + } + } + + MP_VERBOSE(p, "format change, reinitializing resampler\n"); + + p->in_rate_user = in_rate; + p->in_format = in_format; + p->in_channels = in_channels; + p->out_rate = out_rate; + p->out_format = out_format; + p->out_channels = out_channels; + + if (!configure_lavrr(p, true)) { + talloc_free(input); + return; + } + + if (!input) { + // continue filtering next time + mp_filter_internal_mark_progress(f); + return; + } + } + } + + int new_rate = rate_from_speed(p->in_rate_user, p->speed); + if (p->avrctx && !(!p->is_resampling && new_rate == p->in_rate)) { + AVRational r = + av_d2q(p->speed * p->in_rate_user / p->in_rate, INT_MAX / 2); + // Essentially, swr/avresample_set_compensation() does 2 things: + // - adjust output sample rate by sample_delta/compensation_distance + // - reset the adjustment after compensation_distance output samples + // Increase the compensation_distance to avoid undesired reset + // semantics - we want to keep the ratio for the whole frame we're + // feeding it, until the next filter() call. + int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1); + r = (AVRational){ r.num * mult, r.den * mult }; + if (avresample_set_compensation(p->avrctx, r.den - r.num, r.den) >= 0) { + new_rate = p->in_rate; + p->is_resampling = true; + } + } + + bool need_reinit = fabs(new_rate / (double)p->in_rate - 1) > 0.01; + if (need_reinit && new_rate != p->in_rate) { + // Before reconfiguring, drain the audio that is still buffered + // in the resampler. + struct mp_frame out = filter_resample_output(p, NULL); + bool need_drain = !!out.type; + if (need_drain) { + mp_pin_in_write(f->ppins[1], out); + // Drain; continue filtering next time. + mp_pin_out_unread(f->ppins[0], frame); + } + // Reinitialize resampler. + configure_lavrr(p, false); + if (need_drain) { + mp_filter_internal_mark_progress(f); + return; + } + } + + struct mp_frame out = filter_resample_output(p, input); + + if (input && out.type) { + mp_pin_in_write(f->ppins[1], out); + mp_pin_out_request_data(f->ppins[0]); + } else if (!input && out.type) { + mp_pin_in_write(f->ppins[1], out); + mp_pin_out_repeat_eof(f->ppins[0]); + } else if (!input) { + mp_pin_in_write(f->ppins[1], MP_EOF_FRAME); + } + + talloc_free(input); +} + +double mp_swresample_get_delay(struct mp_swresample *s) +{ + struct priv *p = s->f->priv; + + return get_delay(p); +} + +static bool command(struct mp_filter *f, struct mp_filter_command *cmd) +{ + struct priv *p = f->priv; + + if (cmd->type == MP_FILTER_COMMAND_SET_SPEED_RESAMPLE) { + p->cmd_speed = cmd->speed; + return true; + } + + return false; +} + +static void destroy(struct mp_filter *f) +{ + struct priv *p = f->priv; + + close_lavrr(p); +} + +static const struct mp_filter_info swresample_filter = { + .name = "swresample", + .priv_size = sizeof(struct priv), + .process = process, + .command = command, + .reset = reset, + .destroy = destroy, +}; + +struct mp_swresample *mp_swresample_create(struct mp_filter *parent, + struct mp_resample_opts *opts) +{ + struct mp_filter *f = mp_filter_create(parent, &swresample_filter); + if (!f) + return NULL; + + mp_filter_add_pin(f, MP_PIN_IN, "in"); + mp_filter_add_pin(f, MP_PIN_OUT, "out"); + + struct priv *p = f->priv; + p->public.f = f; + p->public.speed = 1.0; + p->cmd_speed = 1.0; + p->log = f->log; + + if (opts) { + p->opts = talloc_dup(p, opts); + p->opts->avopts = mp_dup_str_array(p, p->opts->avopts); + } else { + p->opts = mp_get_config_group(p, f->global, &resample_conf); + } + + p->reorder_buffer = mp_aframe_pool_create(p); + p->out_pool = mp_aframe_pool_create(p); + + return &p->public; +} -- cgit v1.2.3