mirror of
https://github.com/mpv-player/mpv
synced 2025-02-18 05:37:04 +00:00
This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
288 lines
8.2 KiB
C
288 lines
8.2 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
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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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* 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 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 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 "common/recorder.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/filter/af.h"
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extern const struct ad_functions ad_lavc;
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// Not a real codec - specially treated.
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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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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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}
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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->codec->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(d_audio->log, list, codec, opts->audio_decoders);
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if (d_audio->try_spdif && codec) {
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struct mp_decoder_list *spdif =
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select_spdif_codec(codec, 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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if (strcmp(name, "spdif") == 0)
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return &ad_spdif;
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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\n", 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 %s\n", 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)", decoder->decoder, decoder->desc);
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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->codec->codec);
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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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uninit_decoder(d_audio);
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talloc_free(d_audio);
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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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d_audio->pts = MP_NOPTS_VALUE;
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talloc_free(d_audio->current_frame);
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d_audio->current_frame = NULL;
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talloc_free(d_audio->packet);
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d_audio->packet = NULL;
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talloc_free(d_audio->new_segment);
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d_audio->new_segment = NULL;
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d_audio->start = d_audio->end = MP_NOPTS_VALUE;
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}
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static void fix_audio_pts(struct dec_audio *da)
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{
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if (!da->current_frame)
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return;
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double frame_pts = mp_aframe_get_pts(da->current_frame);
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if (frame_pts != MP_NOPTS_VALUE) {
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if (da->pts != MP_NOPTS_VALUE)
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MP_STATS(da, "value %f audio-pts-err", da->pts - frame_pts);
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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 || fabs(da->pts - frame_pts) > 0.001)
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da->pts = frame_pts;
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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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mp_aframe_set_pts(da->current_frame, da->pts);
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if (da->pts != MP_NOPTS_VALUE)
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da->pts += mp_aframe_duration(da->current_frame);
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}
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void audio_work(struct dec_audio *da)
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{
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if (da->current_frame || !da->ad_driver)
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return;
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if (!da->packet && !da->new_segment &&
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demux_read_packet_async(da->header, &da->packet) == 0)
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{
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da->current_state = DATA_WAIT;
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return;
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}
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if (da->packet && da->packet->new_segment) {
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assert(!da->new_segment);
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da->new_segment = da->packet;
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da->packet = NULL;
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}
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if (da->ad_driver->send_packet(da, da->packet)) {
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if (da->recorder_sink)
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mp_recorder_feed_packet(da->recorder_sink, da->packet);
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talloc_free(da->packet);
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da->packet = NULL;
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}
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bool progress = da->ad_driver->receive_frame(da, &da->current_frame);
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da->current_state = da->current_frame ? DATA_OK : DATA_AGAIN;
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if (!progress)
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da->current_state = DATA_EOF;
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fix_audio_pts(da);
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bool segment_end = da->current_state == DATA_EOF;
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if (da->current_frame) {
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mp_aframe_clip_timestamps(da->current_frame, da->start, da->end);
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double frame_pts = mp_aframe_get_pts(da->current_frame);
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if (frame_pts != MP_NOPTS_VALUE && da->start != MP_NOPTS_VALUE)
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segment_end = frame_pts >= da->end;
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if (mp_aframe_get_size(da->current_frame) == 0) {
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talloc_free(da->current_frame);
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da->current_frame = NULL;
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}
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}
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// If there's a new segment, start it as soon as we're drained/finished.
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if (segment_end && da->new_segment) {
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struct demux_packet *new_segment = da->new_segment;
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da->new_segment = NULL;
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if (da->codec == new_segment->codec) {
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audio_reset_decoding(da);
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} else {
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da->codec = new_segment->codec;
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da->ad_driver->uninit(da);
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da->ad_driver = NULL;
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audio_init_best_codec(da);
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}
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da->start = new_segment->start;
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da->end = new_segment->end;
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new_segment->new_segment = false;
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da->packet = new_segment;
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da->current_state = DATA_AGAIN;
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}
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}
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// Fetch an audio frame decoded with audio_work(). Returns one of:
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// DATA_OK: *out_frame is set to a new image
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// DATA_WAIT: waiting for demuxer; will receive a wakeup signal
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// DATA_EOF: end of file, no more frames to be expected
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// DATA_AGAIN: dropped frame or something similar
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int audio_get_frame(struct dec_audio *da, struct mp_aframe **out_frame)
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{
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*out_frame = NULL;
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if (da->current_frame) {
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*out_frame = da->current_frame;
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da->current_frame = NULL;
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return DATA_OK;
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}
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if (da->current_state == DATA_OK)
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return DATA_AGAIN;
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return da->current_state;
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}
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