mirror of
https://github.com/mpv-player/mpv
synced 2024-12-12 01:46:16 +00:00
ac64ce71d6
Was masked by FFmpeg's terrible headers, but failed with Libav.
282 lines
8.3 KiB
C
282 lines
8.3 KiB
C
/*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv 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
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <math.h>
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#include <assert.h>
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#include <libavutil/mem.h>
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#include "demux/codec_tags.h"
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#include "common/codecs.h"
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#include "common/msg.h"
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#include "misc/bstr.h"
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#include "stream/stream.h"
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#include "demux/demux.h"
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#include "demux/stheader.h"
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#include "dec_audio.h"
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#include "ad.h"
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#include "audio/format.h"
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#include "audio/audio.h"
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#include "audio/audio_buffer.h"
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#include "audio/filter/af.h"
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extern const struct ad_functions ad_lavc;
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extern const struct ad_functions ad_spdif;
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static const struct ad_functions * const ad_drivers[] = {
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&ad_lavc,
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&ad_spdif,
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NULL
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};
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static void uninit_decoder(struct dec_audio *d_audio)
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{
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audio_reset_decoding(d_audio);
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if (d_audio->ad_driver) {
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MP_VERBOSE(d_audio, "Uninit audio decoder.\n");
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d_audio->ad_driver->uninit(d_audio);
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}
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d_audio->ad_driver = NULL;
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talloc_free(d_audio->priv);
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d_audio->priv = NULL;
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d_audio->afilter->initialized = -1;
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d_audio->decode_format = (struct mp_audio){0};
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}
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static int init_audio_codec(struct dec_audio *d_audio, const char *decoder)
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{
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if (!d_audio->ad_driver->init(d_audio, decoder)) {
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MP_VERBOSE(d_audio, "Audio decoder init failed.\n");
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d_audio->ad_driver = NULL;
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uninit_decoder(d_audio);
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return 0;
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}
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return 1;
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}
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struct mp_decoder_list *audio_decoder_list(void)
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{
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struct mp_decoder_list *list = talloc_zero(NULL, struct mp_decoder_list);
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for (int i = 0; ad_drivers[i] != NULL; i++)
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ad_drivers[i]->add_decoders(list);
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return list;
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}
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static struct mp_decoder_list *audio_select_decoders(struct dec_audio *d_audio)
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{
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struct MPOpts *opts = d_audio->opts;
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const char *codec = d_audio->header->codec;
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struct mp_decoder_list *list = audio_decoder_list();
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struct mp_decoder_list *new =
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mp_select_decoders(list, codec, opts->audio_decoders);
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if (d_audio->spdif_passthrough) {
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struct mp_decoder_list *spdif =
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mp_select_decoder_list(list, codec, "spdif", opts->audio_spdif);
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mp_append_decoders(spdif, new);
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talloc_free(new);
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new = spdif;
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}
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talloc_free(list);
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return new;
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}
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static const struct ad_functions *find_driver(const char *name)
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{
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for (int i = 0; ad_drivers[i] != NULL; i++) {
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if (strcmp(ad_drivers[i]->name, name) == 0)
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return ad_drivers[i];
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}
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return NULL;
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}
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int audio_init_best_codec(struct dec_audio *d_audio)
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{
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uninit_decoder(d_audio);
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assert(!d_audio->ad_driver);
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struct mp_decoder_entry *decoder = NULL;
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struct mp_decoder_list *list = audio_select_decoders(d_audio);
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mp_print_decoders(d_audio->log, MSGL_V, "Codec list:", list);
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for (int n = 0; n < list->num_entries; n++) {
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struct mp_decoder_entry *sel = &list->entries[n];
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const struct ad_functions *driver = find_driver(sel->family);
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if (!driver)
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continue;
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MP_VERBOSE(d_audio, "Opening audio decoder %s:%s\n",
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sel->family, sel->decoder);
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d_audio->ad_driver = driver;
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if (init_audio_codec(d_audio, sel->decoder)) {
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decoder = sel;
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break;
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}
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MP_WARN(d_audio, "Audio decoder init failed for "
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"%s:%s\n", sel->family, sel->decoder);
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}
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if (d_audio->ad_driver) {
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d_audio->decoder_desc =
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talloc_asprintf(d_audio, "%s [%s:%s]", decoder->desc, decoder->family,
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decoder->decoder);
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MP_VERBOSE(d_audio, "Selected audio codec: %s\n", d_audio->decoder_desc);
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} else {
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MP_ERR(d_audio, "Failed to initialize an audio decoder for codec '%s'.\n",
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d_audio->header->codec ? d_audio->header->codec : "<unknown>");
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}
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talloc_free(list);
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return !!d_audio->ad_driver;
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}
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void audio_uninit(struct dec_audio *d_audio)
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{
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if (!d_audio)
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return;
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MP_VERBOSE(d_audio, "Uninit audio filters...\n");
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uninit_decoder(d_audio);
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af_destroy(d_audio->afilter);
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talloc_free(d_audio->waiting);
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talloc_free(d_audio);
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}
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static int decode_new_frame(struct dec_audio *da)
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{
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while (!da->waiting) {
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int ret = da->ad_driver->decode_packet(da, &da->waiting);
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if (ret < 0)
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return ret;
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if (da->waiting) {
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if (da->waiting->pts != MP_NOPTS_VALUE) {
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if (da->pts != MP_NOPTS_VALUE) {
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da->pts += da->pts_offset / (double)da->waiting->rate;
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da->pts_offset = 0;
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}
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// Keep the interpolated timestamp if it doesn't deviate more
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// than 1 ms from the real one. (MKV rounded timestamps.)
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if (da->pts == MP_NOPTS_VALUE || da->pts_offset != 0 ||
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fabs(da->pts - da->waiting->pts) > 0.001)
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{
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da->pts = da->waiting->pts;
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da->pts_offset = 0;
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}
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}
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da->pts_offset += da->waiting->samples;
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da->decode_format = *da->waiting;
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mp_audio_set_null_data(&da->decode_format);
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}
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if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
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da->pts = 0;
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}
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return mp_audio_config_valid(da->waiting) ? AD_OK : AD_ERR;
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}
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/* Decode packets until we know the audio format. Then reinit the buffer.
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* Returns AD_OK on success, negative AD_* code otherwise.
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* Also returns AD_OK if already initialized (and does nothing).
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*/
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int initial_audio_decode(struct dec_audio *da)
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{
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return decode_new_frame(da);
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}
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static bool copy_output(struct af_stream *afs, struct mp_audio_buffer *outbuf,
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int minsamples, bool eof)
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{
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while (mp_audio_buffer_samples(outbuf) < minsamples) {
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if (af_output_frame(afs, eof) < 0)
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return true; // error, stop doing stuff
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struct mp_audio *mpa = af_read_output_frame(afs);
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if (!mpa)
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return false; // out of data
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mp_audio_buffer_append(outbuf, mpa);
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talloc_free(mpa);
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}
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return true;
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}
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/* Try to get at least minsamples decoded+filtered samples in outbuf
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* (total length including possible existing data).
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* Return 0 on success, or negative AD_* error code.
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* In the former case outbuf has at least minsamples buffered on return.
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* In case of EOF/error it might or might not be. */
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int audio_decode(struct dec_audio *da, struct mp_audio_buffer *outbuf,
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int minsamples)
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{
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struct af_stream *afs = da->afilter;
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if (afs->initialized < 1)
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return AD_ERR;
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MP_STATS(da, "start audio");
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int res;
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while (1) {
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res = 0;
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if (copy_output(afs, outbuf, minsamples, false))
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break;
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res = decode_new_frame(da);
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if (res < 0) {
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// drain filters first (especially for true EOF case)
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copy_output(afs, outbuf, minsamples, true);
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break;
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}
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// On format change, make sure to drain the filter chain.
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if (!mp_audio_config_equals(&afs->input, da->waiting)) {
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copy_output(afs, outbuf, minsamples, true);
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res = AD_NEW_FMT;
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break;
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}
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struct mp_audio *mpa = da->waiting;
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da->waiting = NULL;
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if (af_filter_frame(afs, mpa) < 0)
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return AD_ERR;
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}
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MP_STATS(da, "end audio");
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return res;
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}
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void audio_reset_decoding(struct dec_audio *d_audio)
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{
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if (d_audio->ad_driver)
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d_audio->ad_driver->control(d_audio, ADCTRL_RESET, NULL);
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af_seek_reset(d_audio->afilter);
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d_audio->pts = MP_NOPTS_VALUE;
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d_audio->pts_offset = 0;
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if (d_audio->waiting) {
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talloc_free(d_audio->waiting);
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d_audio->waiting = NULL;
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}
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}
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