mirror of
https://github.com/mpv-player/mpv
synced 2025-03-30 23:38:10 +00:00
8125252399
ao_lavc.c accesses ao->buffer, which I consider internal. The access was done in ao_lavc.c/uninit(), which tried to get the left-over audio in order to write the last (possibly partial) audio frame. The play() function didn't accept partial frames, because the AOPLAY_FINAL_CHUNK flag was not correctly set, and handling it otherwise would require an internal FIFO. Fix this by making sure that with gapless audio (used with encoding), the AOPLAY_FINAL_CHUNK is set only once, instead when each file ends. Basically, move the hack in ao_lavc's uninit to uninit_player. One thing can not be entirely correctly handled: if gapless audio is active, we don't know really whether the AO is closed because the file ended playing (i.e. we want to send the buffered remainder of the audio to the AO), or whether the user is quitting the player. (The stop_play flag is overwritten, fixing that is perhaps not worth it.) Handle this by adding additional code to drain the AO and the buffers when playback is quit (see play_current_file() change). Test case: mpv avdevice://lavfi:sine=441 avdevice://lavfi:sine=441 -length 0.2267 -gapless-audio
424 lines
14 KiB
C
424 lines
14 KiB
C
/*
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* This file is part of MPlayer.
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*
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* MPlayer 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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* MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <stddef.h>
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#include <stdbool.h>
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#include <inttypes.h>
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#include <math.h>
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#include <assert.h>
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#include "config.h"
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#include "talloc.h"
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#include "mpvcore/mp_msg.h"
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#include "mpvcore/options.h"
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#include "mpvcore/mp_common.h"
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#include "audio/mixer.h"
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#include "audio/decode/dec_audio.h"
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#include "audio/filter/af.h"
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#include "audio/out/ao.h"
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#include "demux/demux.h"
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#include "mp_core.h"
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static int build_afilter_chain(struct MPContext *mpctx)
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{
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struct sh_audio *sh_audio = mpctx->sh_audio;
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struct ao *ao = mpctx->ao;
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struct MPOpts *opts = mpctx->opts;
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int new_srate;
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if (af_control_any_rev(sh_audio->afilter,
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AF_CONTROL_PLAYBACK_SPEED | AF_CONTROL_SET,
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&opts->playback_speed))
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new_srate = sh_audio->samplerate;
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else {
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new_srate = sh_audio->samplerate * opts->playback_speed;
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if (new_srate != ao->samplerate) {
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// limits are taken from libaf/af_resample.c
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if (new_srate < 8000)
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new_srate = 8000;
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if (new_srate > 192000)
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new_srate = 192000;
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opts->playback_speed = (double)new_srate / sh_audio->samplerate;
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}
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}
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return init_audio_filters(sh_audio, new_srate,
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&ao->samplerate, &ao->channels, &ao->format);
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}
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static int recreate_audio_filters(struct MPContext *mpctx)
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{
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assert(mpctx->sh_audio);
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// init audio filters:
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if (!build_afilter_chain(mpctx)) {
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MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
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return -1;
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}
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mixer_reinit_audio(mpctx->mixer, mpctx->ao, mpctx->sh_audio->afilter);
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return 0;
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}
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int reinit_audio_filters(struct MPContext *mpctx)
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{
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struct sh_audio *sh_audio = mpctx->sh_audio;
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if (!sh_audio)
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return -2;
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af_uninit(mpctx->sh_audio->afilter);
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if (af_init(mpctx->sh_audio->afilter) < 0)
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return -1;
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if (recreate_audio_filters(mpctx) < 0)
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return -1;
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return 0;
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}
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void reinit_audio_chain(struct MPContext *mpctx)
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{
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struct MPOpts *opts = mpctx->opts;
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init_demux_stream(mpctx, STREAM_AUDIO);
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if (!mpctx->sh_audio) {
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uninit_player(mpctx, INITIALIZED_AO);
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goto no_audio;
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}
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if (!(mpctx->initialized_flags & INITIALIZED_ACODEC)) {
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if (!init_best_audio_codec(mpctx->sh_audio, opts->audio_decoders))
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goto init_error;
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mpctx->initialized_flags |= INITIALIZED_ACODEC;
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}
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int ao_srate = opts->force_srate;
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int ao_format = opts->audio_output_format;
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struct mp_chmap ao_channels = {0};
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if (mpctx->initialized_flags & INITIALIZED_AO) {
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ao_srate = mpctx->ao->samplerate;
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ao_format = mpctx->ao->format;
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ao_channels = mpctx->ao->channels;
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} else {
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// Automatic downmix
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if (mp_chmap_is_stereo(&opts->audio_output_channels) &&
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!mp_chmap_is_stereo(&mpctx->sh_audio->channels))
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{
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mp_chmap_from_channels(&ao_channels, 2);
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}
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}
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// Determine what the filter chain outputs. build_afilter_chain() also
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// needs this for testing whether playback speed is changed by resampling
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// or using a special filter.
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if (!init_audio_filters(mpctx->sh_audio, // preliminary init
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// input:
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mpctx->sh_audio->samplerate,
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// output:
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&ao_srate, &ao_channels, &ao_format)) {
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MP_ERR(mpctx, "Error at audio filter chain pre-init!\n");
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goto init_error;
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}
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if (!(mpctx->initialized_flags & INITIALIZED_AO)) {
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mpctx->initialized_flags |= INITIALIZED_AO;
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mp_chmap_remove_useless_channels(&ao_channels,
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&opts->audio_output_channels);
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mpctx->ao = ao_init_best(mpctx->global, mpctx->input,
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mpctx->encode_lavc_ctx, ao_srate, ao_format,
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ao_channels);
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struct ao *ao = mpctx->ao;
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if (!ao) {
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MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
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goto init_error;
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}
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ao->buffer.start = talloc_new(ao);
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char *s = mp_audio_fmt_to_str(ao->samplerate, &ao->channels, ao->format);
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MP_INFO(mpctx, "AO: [%s] %s\n", ao->driver->name, s);
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talloc_free(s);
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MP_VERBOSE(mpctx, "AO: Description: %s\n", ao->driver->description);
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}
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if (recreate_audio_filters(mpctx) < 0)
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goto init_error;
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mpctx->syncing_audio = true;
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return;
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init_error:
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uninit_player(mpctx, INITIALIZED_ACODEC | INITIALIZED_AO);
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cleanup_demux_stream(mpctx, STREAM_AUDIO);
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no_audio:
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mpctx->current_track[STREAM_AUDIO] = NULL;
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MP_INFO(mpctx, "Audio: no audio\n");
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}
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// Return pts value corresponding to the end point of audio written to the
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// ao so far.
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double written_audio_pts(struct MPContext *mpctx)
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{
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sh_audio_t *sh_audio = mpctx->sh_audio;
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if (!sh_audio)
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return MP_NOPTS_VALUE;
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double bps = sh_audio->channels.num * sh_audio->samplerate *
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sh_audio->samplesize;
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// first calculate the end pts of audio that has been output by decoder
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double a_pts = sh_audio->pts;
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if (a_pts == MP_NOPTS_VALUE)
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return MP_NOPTS_VALUE;
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// sh_audio->pts is the timestamp of the latest input packet with
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// known pts that the decoder has decoded. sh_audio->pts_bytes is
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// the amount of bytes the decoder has written after that timestamp.
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a_pts += sh_audio->pts_bytes / bps;
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// Now a_pts hopefully holds the pts for end of audio from decoder.
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// Subtract data in buffers between decoder and audio out.
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// Decoded but not filtered
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a_pts -= sh_audio->a_buffer_len / bps;
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// Data buffered in audio filters, measured in bytes of "missing" output
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double buffered_output = af_calc_delay(sh_audio->afilter);
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// Data that was ready for ao but was buffered because ao didn't fully
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// accept everything to internal buffers yet
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buffered_output += mpctx->ao->buffer.len;
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// Filters divide audio length by playback_speed, so multiply by it
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// to get the length in original units without speedup or slowdown
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a_pts -= buffered_output * mpctx->opts->playback_speed / mpctx->ao->bps;
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return a_pts + mpctx->video_offset;
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}
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// Return pts value corresponding to currently playing audio.
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double playing_audio_pts(struct MPContext *mpctx)
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{
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double pts = written_audio_pts(mpctx);
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if (pts == MP_NOPTS_VALUE)
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return pts;
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return pts - mpctx->opts->playback_speed * ao_get_delay(mpctx->ao);
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}
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static int write_to_ao(struct MPContext *mpctx, void *data, int len, int flags,
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double pts)
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{
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if (mpctx->paused)
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return 0;
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struct ao *ao = mpctx->ao;
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double bps = ao->bps / mpctx->opts->playback_speed;
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int unitsize = ao->channels.num * af_fmt2bits(ao->format) / 8;
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ao->pts = pts;
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int played = ao_play(mpctx->ao, data, len, flags);
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assert(played <= len);
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assert(played % unitsize == 0);
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if (played > 0) {
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mpctx->shown_aframes += played / unitsize;
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mpctx->delay += played / bps;
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// Keep correct pts for remaining data - could be used to flush
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// remaining buffer when closing ao.
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ao->pts += played / bps;
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return played;
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}
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return 0;
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}
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#define ASYNC_PLAY_DONE -3
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static int audio_start_sync(struct MPContext *mpctx, int playsize)
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{
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struct ao *ao = mpctx->ao;
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struct MPOpts *opts = mpctx->opts;
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sh_audio_t * const sh_audio = mpctx->sh_audio;
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int res;
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// Timing info may not be set without
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res = decode_audio(sh_audio, &ao->buffer, 1);
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if (res < 0)
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return res;
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int bytes;
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bool did_retry = false;
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double written_pts;
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double bps = ao->bps / opts->playback_speed;
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bool hrseek = mpctx->hrseek_active; // audio only hrseek
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mpctx->hrseek_active = false;
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while (1) {
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written_pts = written_audio_pts(mpctx);
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double ptsdiff;
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if (hrseek)
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ptsdiff = written_pts - mpctx->hrseek_pts;
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else
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ptsdiff = written_pts - mpctx->sh_video->pts - mpctx->delay
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- mpctx->audio_delay;
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bytes = ptsdiff * bps;
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bytes -= bytes % (ao->channels.num * af_fmt2bits(ao->format) / 8);
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// ogg demuxers give packets without timing
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if (written_pts <= 1 && sh_audio->pts == MP_NOPTS_VALUE) {
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if (!did_retry) {
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// Try to read more data to see packets that have pts
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res = decode_audio(sh_audio, &ao->buffer, ao->bps);
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if (res < 0)
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return res;
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did_retry = true;
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continue;
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}
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bytes = 0;
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}
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if (fabs(ptsdiff) > 300 || isnan(ptsdiff)) // pts reset or just broken?
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bytes = 0;
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if (bytes > 0)
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break;
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mpctx->syncing_audio = false;
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int a = MPMIN(-bytes, MPMAX(playsize, 20000));
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res = decode_audio(sh_audio, &ao->buffer, a);
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bytes += ao->buffer.len;
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if (bytes >= 0) {
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memmove(ao->buffer.start,
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ao->buffer.start + ao->buffer.len - bytes, bytes);
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ao->buffer.len = bytes;
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if (res < 0)
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return res;
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return decode_audio(sh_audio, &ao->buffer, playsize);
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}
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ao->buffer.len = 0;
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if (res < 0)
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return res;
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}
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if (hrseek)
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// Don't add silence in audio-only case even if position is too late
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return 0;
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int fillbyte = 0;
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if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_US)
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fillbyte = 0x80;
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if (bytes >= playsize) {
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/* This case could fall back to the one below with
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* bytes = playsize, but then silence would keep accumulating
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* in a_out_buffer if the AO accepts less data than it asks for
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* in playsize. */
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char *p = malloc(playsize);
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memset(p, fillbyte, playsize);
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write_to_ao(mpctx, p, playsize, 0, written_pts - bytes / bps);
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free(p);
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return ASYNC_PLAY_DONE;
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}
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mpctx->syncing_audio = false;
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decode_audio_prepend_bytes(&ao->buffer, bytes, fillbyte);
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return decode_audio(sh_audio, &ao->buffer, playsize);
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}
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int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
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{
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struct MPOpts *opts = mpctx->opts;
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struct ao *ao = mpctx->ao;
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int playsize;
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int playflags = 0;
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bool audio_eof = false;
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bool signal_eof = false;
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bool partial_fill = false;
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sh_audio_t * const sh_audio = mpctx->sh_audio;
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bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK);
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int unitsize = ao->channels.num * af_fmt2bits(ao->format) / 8;
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if (mpctx->paused)
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playsize = 1; // just initialize things (audio pts at least)
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else
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playsize = ao_get_space(ao);
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// Coming here with hrseek_active still set means audio-only
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if (!mpctx->sh_video || !mpctx->sync_audio_to_video)
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mpctx->syncing_audio = false;
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if (!opts->initial_audio_sync || !modifiable_audio_format) {
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mpctx->syncing_audio = false;
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mpctx->hrseek_active = false;
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}
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int res;
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if (mpctx->syncing_audio || mpctx->hrseek_active)
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res = audio_start_sync(mpctx, playsize);
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else
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res = decode_audio(sh_audio, &ao->buffer, playsize);
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if (res < 0) { // EOF, error or format change
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if (res == -2) {
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/* The format change isn't handled too gracefully. A more precise
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* implementation would require draining buffered old-format audio
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* while displaying video, then doing the output format switch.
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*/
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if (!mpctx->opts->gapless_audio)
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uninit_player(mpctx, INITIALIZED_AO);
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reinit_audio_chain(mpctx);
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return -1;
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} else if (res == ASYNC_PLAY_DONE)
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return 0;
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else if (demux_stream_eof(mpctx->sh_audio->gsh))
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audio_eof = true;
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}
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if (endpts != MP_NOPTS_VALUE && modifiable_audio_format) {
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double bytes = (endpts - written_audio_pts(mpctx) + mpctx->audio_delay)
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* ao->bps / opts->playback_speed;
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if (playsize > bytes) {
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playsize = MPMAX(bytes, 0);
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audio_eof = true;
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partial_fill = true;
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}
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}
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assert(ao->buffer.len % unitsize == 0);
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if (playsize > ao->buffer.len) {
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partial_fill = true;
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playsize = ao->buffer.len;
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}
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playsize -= playsize % unitsize;
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if (!playsize)
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return partial_fill && audio_eof ? -2 : -partial_fill;
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if (audio_eof && partial_fill) {
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if (opts->gapless_audio) {
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// With gapless audio, delay this to ao_uninit. There must be only
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// 1 final chunk, and that is handled when calling ao_uninit().
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signal_eof = true;
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} else {
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playflags |= AOPLAY_FINAL_CHUNK;
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}
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}
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assert(ao->buffer_playable_size <= ao->buffer.len);
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int played = write_to_ao(mpctx, ao->buffer.start, playsize, playflags,
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written_audio_pts(mpctx));
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ao->buffer_playable_size = playsize - played;
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if (played > 0) {
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ao->buffer.len -= played;
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memmove(ao->buffer.start, ao->buffer.start + played, ao->buffer.len);
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} else if (!mpctx->paused && audio_eof && ao_get_delay(ao) < .04) {
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// Sanity check to avoid hanging in case current ao doesn't output
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// partial chunks and doesn't check for AOPLAY_FINAL_CHUNK
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signal_eof = true;
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}
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return signal_eof ? -2 : -partial_fill;
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}
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