2013-10-29 21:38:29 +00:00
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/*
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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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2013-12-17 01:39:45 +00:00
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#include "common/msg.h"
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2014-03-07 14:23:03 +00:00
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#include "common/encode.h"
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2013-12-17 01:02:25 +00:00
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#include "options/options.h"
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2013-12-17 01:39:45 +00:00
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#include "common/common.h"
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2013-10-29 21:38:29 +00:00
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#include "audio/mixer.h"
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2013-11-10 22:38:18 +00:00
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#include "audio/audio.h"
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#include "audio/audio_buffer.h"
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2013-10-29 21:38:29 +00:00
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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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2013-11-23 20:36:20 +00:00
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#include "video/decode/dec_video.h"
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2013-10-29 21:38:29 +00:00
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2013-12-17 00:08:53 +00:00
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#include "core.h"
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2014-02-17 01:52:26 +00:00
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#include "command.h"
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2013-10-29 21:38:29 +00:00
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static int build_afilter_chain(struct MPContext *mpctx)
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{
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2013-11-23 20:22:17 +00:00
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struct dec_audio *d_audio = mpctx->d_audio;
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2013-10-29 21:38:29 +00:00
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struct MPOpts *opts = mpctx->opts;
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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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2013-11-23 20:22:17 +00:00
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if (!d_audio)
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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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return 0;
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struct mp_audio in_format;
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2013-11-23 20:22:17 +00:00
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mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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2014-03-07 14:24:32 +00:00
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struct mp_audio out_format;
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ao_get_format(mpctx->ao, &out_format);
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2013-10-29 21:38:29 +00:00
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int new_srate;
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2013-11-23 20:22:17 +00:00
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if (af_control_any_rev(d_audio->afilter, AF_CONTROL_SET_PLAYBACK_SPEED,
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2013-10-29 21:38:29 +00:00
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&opts->playback_speed))
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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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new_srate = in_format.rate;
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2013-10-29 21:38:29 +00:00
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else {
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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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new_srate = in_format.rate * opts->playback_speed;
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2014-03-07 14:24:32 +00:00
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if (new_srate != out_format.rate) {
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2013-10-29 21:38:29 +00:00
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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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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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opts->playback_speed = new_srate / (double)in_format.rate;
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2013-10-29 21:38:29 +00:00
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}
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}
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2013-11-23 20:22:17 +00:00
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return audio_init_filters(d_audio, new_srate,
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2014-03-07 14:24:32 +00:00
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&out_format.rate, &out_format.channels, &out_format.format);
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2013-10-29 21:38:29 +00:00
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}
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static int recreate_audio_filters(struct MPContext *mpctx)
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{
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2013-11-23 20:22:17 +00:00
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assert(mpctx->d_audio);
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2013-10-29 21:38:29 +00:00
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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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2013-11-23 20:22:17 +00:00
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mixer_reinit_audio(mpctx->mixer, mpctx->ao, mpctx->d_audio->afilter);
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2013-10-29 21:38:29 +00:00
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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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2013-11-23 20:22:17 +00:00
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struct dec_audio *d_audio = mpctx->d_audio;
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if (!d_audio)
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2013-10-29 21:38:29 +00:00
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return -2;
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2013-11-23 20:22:17 +00:00
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af_uninit(mpctx->d_audio->afilter);
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if (af_init(mpctx->d_audio->afilter) < 0)
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2013-10-29 21:38:29 +00:00
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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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2013-12-24 16:46:08 +00:00
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struct track *track = mpctx->current_track[0][STREAM_AUDIO];
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2013-12-23 19:14:54 +00:00
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struct sh_stream *sh = init_demux_stream(mpctx, track);
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2013-11-23 20:22:17 +00:00
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if (!sh) {
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2013-10-29 21:38:29 +00:00
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uninit_player(mpctx, INITIALIZED_AO);
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goto no_audio;
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}
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2014-02-17 01:52:26 +00:00
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mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
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2013-10-29 21:38:29 +00:00
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if (!(mpctx->initialized_flags & INITIALIZED_ACODEC)) {
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mpctx->initialized_flags |= INITIALIZED_ACODEC;
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2013-11-23 20:22:17 +00:00
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assert(!mpctx->d_audio);
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mpctx->d_audio = talloc_zero(NULL, struct dec_audio);
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2013-12-21 17:23:59 +00:00
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mpctx->d_audio->log = mp_log_new(mpctx->d_audio, mpctx->log, "!ad");
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mpctx->d_audio->global = mpctx->global;
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2013-11-23 20:22:17 +00:00
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mpctx->d_audio->opts = opts;
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mpctx->d_audio->header = sh;
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if (!audio_init_best_codec(mpctx->d_audio, opts->audio_decoders))
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goto init_error;
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2013-10-29 21:38:29 +00:00
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}
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2013-11-23 20:22:17 +00:00
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assert(mpctx->d_audio);
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2013-10-29 21:38:29 +00:00
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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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struct mp_audio in_format;
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2013-11-23 20:22:17 +00:00
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mp_audio_buffer_get_format(mpctx->d_audio->decode_buffer, &in_format);
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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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2013-10-29 21:38:29 +00:00
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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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2014-03-07 14:24:32 +00:00
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struct mp_audio out_format;
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ao_get_format(mpctx->ao, &out_format);
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ao_srate = out_format.rate;
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ao_format = out_format.format;
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ao_channels = out_format.channels;
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2013-10-29 21:38:29 +00:00
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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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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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!mp_chmap_is_stereo(&in_format.channels))
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2013-10-29 21:38:29 +00:00
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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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2013-11-23 20:22:17 +00:00
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if (!audio_init_filters(mpctx->d_audio, // preliminary init
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2013-10-29 21:38:29 +00:00
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// input:
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audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
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in_format.rate,
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2013-10-29 21:38:29 +00:00
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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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}
|
2013-11-10 22:38:18 +00:00
|
|
|
|
2014-03-07 14:24:32 +00:00
|
|
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struct mp_audio fmt;
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ao_get_format(ao, &fmt);
|
2013-11-10 22:38:18 +00:00
|
|
|
|
2014-03-07 14:24:32 +00:00
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mpctx->ao_buffer = mp_audio_buffer_create(ao);
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mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt);
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char *s = mp_audio_config_to_str(&fmt);
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MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(ao), s);
|
2013-10-29 21:38:29 +00:00
|
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talloc_free(s);
|
2014-03-07 14:24:32 +00:00
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MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(ao));
|
2013-11-09 23:49:13 +00:00
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update_window_title(mpctx, true);
|
2013-10-29 21:38:29 +00:00
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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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|
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|
mpctx->syncing_audio = true;
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|
return;
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|
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|
init_error:
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|
uninit_player(mpctx, INITIALIZED_ACODEC | INITIALIZED_AO);
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|
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no_audio:
|
2013-12-23 19:14:54 +00:00
|
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|
mp_deselect_track(mpctx, track);
|
2013-10-29 21:38:29 +00:00
|
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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
|
|
|
|
// ao so far.
|
|
|
|
double written_audio_pts(struct MPContext *mpctx)
|
|
|
|
{
|
2013-11-23 20:22:17 +00:00
|
|
|
struct dec_audio *d_audio = mpctx->d_audio;
|
|
|
|
if (!d_audio)
|
2013-10-29 21:38:29 +00:00
|
|
|
return MP_NOPTS_VALUE;
|
|
|
|
|
audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
|
|
|
struct mp_audio in_format;
|
2013-11-23 20:22:17 +00:00
|
|
|
mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
|
audio: use the decoder buffer's format, not sh_audio
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
2013-11-18 12:57:17 +00:00
|
|
|
|
2013-10-29 21:38:29 +00:00
|
|
|
// first calculate the end pts of audio that has been output by decoder
|
2013-11-23 20:22:17 +00:00
|
|
|
double a_pts = d_audio->pts;
|
2013-10-29 21:38:29 +00:00
|
|
|
if (a_pts == MP_NOPTS_VALUE)
|
|
|
|
return MP_NOPTS_VALUE;
|
|
|
|
|
2013-11-23 20:22:17 +00:00
|
|
|
// d_audio->pts is the timestamp of the latest input packet with
|
|
|
|
// known pts that the decoder has decoded. d_audio->pts_bytes is
|
2013-10-29 21:38:29 +00:00
|
|
|
// the amount of bytes the decoder has written after that timestamp.
|
2013-11-23 20:22:17 +00:00
|
|
|
a_pts += d_audio->pts_offset / (double)in_format.rate;
|
2013-10-29 21:38:29 +00:00
|
|
|
|
|
|
|
// Now a_pts hopefully holds the pts for end of audio from decoder.
|
|
|
|
// Subtract data in buffers between decoder and audio out.
|
|
|
|
|
|
|
|
// Decoded but not filtered
|
2013-11-23 20:22:17 +00:00
|
|
|
a_pts -= mp_audio_buffer_seconds(d_audio->decode_buffer);
|
2013-10-29 21:38:29 +00:00
|
|
|
|
2013-11-10 22:38:18 +00:00
|
|
|
// Data buffered in audio filters, measured in seconds of "missing" output
|
2013-11-23 20:22:17 +00:00
|
|
|
double buffered_output = af_calc_delay(d_audio->afilter);
|
2013-10-29 21:38:29 +00:00
|
|
|
|
|
|
|
// Data that was ready for ao but was buffered because ao didn't fully
|
|
|
|
// accept everything to internal buffers yet
|
2014-03-07 14:24:32 +00:00
|
|
|
buffered_output += mp_audio_buffer_seconds(mpctx->ao_buffer);
|
2013-10-29 21:38:29 +00:00
|
|
|
|
|
|
|
// Filters divide audio length by playback_speed, so multiply by it
|
|
|
|
// to get the length in original units without speedup or slowdown
|
2013-11-10 22:38:18 +00:00
|
|
|
a_pts -= buffered_output * mpctx->opts->playback_speed;
|
2013-10-29 21:38:29 +00:00
|
|
|
|
|
|
|
return a_pts + mpctx->video_offset;
|
|
|
|
}
|
|
|
|
|
|
|
|
// Return pts value corresponding to currently playing audio.
|
|
|
|
double playing_audio_pts(struct MPContext *mpctx)
|
|
|
|
{
|
|
|
|
double pts = written_audio_pts(mpctx);
|
|
|
|
if (pts == MP_NOPTS_VALUE)
|
|
|
|
return pts;
|
|
|
|
return pts - mpctx->opts->playback_speed * ao_get_delay(mpctx->ao);
|
|
|
|
}
|
|
|
|
|
2013-11-10 22:38:18 +00:00
|
|
|
static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags,
|
2013-10-29 21:38:29 +00:00
|
|
|
double pts)
|
|
|
|
{
|
|
|
|
if (mpctx->paused)
|
|
|
|
return 0;
|
|
|
|
struct ao *ao = mpctx->ao;
|
2014-03-07 14:24:32 +00:00
|
|
|
struct mp_audio out_format;
|
|
|
|
ao_get_format(ao, &out_format);
|
|
|
|
mpctx->ao_pts = pts;
|
2014-03-07 14:23:03 +00:00
|
|
|
#if HAVE_ENCODING
|
2014-03-07 14:24:32 +00:00
|
|
|
encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, mpctx->ao_pts);
|
2014-03-07 14:23:03 +00:00
|
|
|
#endif
|
2014-03-07 14:24:32 +00:00
|
|
|
double real_samplerate = out_format.rate / mpctx->opts->playback_speed;
|
2013-11-10 22:38:18 +00:00
|
|
|
int played = ao_play(mpctx->ao, data->planes, data->samples, flags);
|
|
|
|
assert(played <= data->samples);
|
2013-10-29 21:38:29 +00:00
|
|
|
if (played > 0) {
|
2013-11-10 22:38:18 +00:00
|
|
|
mpctx->shown_aframes += played;
|
|
|
|
mpctx->delay += played / real_samplerate;
|
2013-10-29 21:38:29 +00:00
|
|
|
// Keep correct pts for remaining data - could be used to flush
|
|
|
|
// remaining buffer when closing ao.
|
2014-03-07 14:24:32 +00:00
|
|
|
mpctx->ao_pts += played / real_samplerate;
|
2013-10-29 21:38:29 +00:00
|
|
|
return played;
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2013-11-10 22:38:18 +00:00
|
|
|
static int write_silence_to_ao(struct MPContext *mpctx, int samples, int flags,
|
|
|
|
double pts)
|
|
|
|
{
|
|
|
|
struct mp_audio tmp = {0};
|
2014-03-07 14:24:32 +00:00
|
|
|
mp_audio_buffer_get_format(mpctx->ao_buffer, &tmp);
|
2013-11-10 22:38:18 +00:00
|
|
|
tmp.samples = samples;
|
|
|
|
char *p = talloc_size(NULL, tmp.samples * tmp.sstride);
|
|
|
|
for (int n = 0; n < tmp.num_planes; n++)
|
|
|
|
tmp.planes[n] = p;
|
|
|
|
mp_audio_fill_silence(&tmp, 0, tmp.samples);
|
|
|
|
int r = write_to_ao(mpctx, &tmp, 0, pts);
|
|
|
|
talloc_free(p);
|
|
|
|
return r;
|
|
|
|
}
|
|
|
|
|
2013-10-29 21:38:29 +00:00
|
|
|
#define ASYNC_PLAY_DONE -3
|
|
|
|
static int audio_start_sync(struct MPContext *mpctx, int playsize)
|
|
|
|
{
|
|
|
|
struct ao *ao = mpctx->ao;
|
|
|
|
struct MPOpts *opts = mpctx->opts;
|
2013-11-23 20:22:17 +00:00
|
|
|
struct dec_audio *d_audio = mpctx->d_audio;
|
2013-10-29 21:38:29 +00:00
|
|
|
int res;
|
|
|
|
|
2013-11-23 20:22:17 +00:00
|
|
|
assert(d_audio);
|
|
|
|
|
2014-03-07 14:24:32 +00:00
|
|
|
struct mp_audio out_format;
|
|
|
|
ao_get_format(ao, &out_format);
|
|
|
|
|
2013-10-29 21:38:29 +00:00
|
|
|
// Timing info may not be set without
|
2014-03-07 14:24:32 +00:00
|
|
|
res = audio_decode(d_audio, mpctx->ao_buffer, 1);
|
2013-10-29 21:38:29 +00:00
|
|
|
if (res < 0)
|
|
|
|
return res;
|
|
|
|
|
2013-11-10 22:38:18 +00:00
|
|
|
int samples;
|
2013-10-29 21:38:29 +00:00
|
|
|
bool did_retry = false;
|
|
|
|
double written_pts;
|
2014-03-07 14:24:32 +00:00
|
|
|
double real_samplerate = out_format.rate / opts->playback_speed;
|
2013-10-29 21:38:29 +00:00
|
|
|
bool hrseek = mpctx->hrseek_active; // audio only hrseek
|
|
|
|
mpctx->hrseek_active = false;
|
|
|
|
while (1) {
|
|
|
|
written_pts = written_audio_pts(mpctx);
|
|
|
|
double ptsdiff;
|
|
|
|
if (hrseek)
|
|
|
|
ptsdiff = written_pts - mpctx->hrseek_pts;
|
|
|
|
else
|
2013-11-27 19:57:08 +00:00
|
|
|
ptsdiff = written_pts - mpctx->video_next_pts - mpctx->delay
|
2014-01-06 17:39:49 +00:00
|
|
|
+ mpctx->audio_delay;
|
2013-11-10 22:38:18 +00:00
|
|
|
samples = ptsdiff * real_samplerate;
|
2013-10-29 21:38:29 +00:00
|
|
|
|
|
|
|
// ogg demuxers give packets without timing
|
2013-11-23 20:22:17 +00:00
|
|
|
if (written_pts <= 1 && d_audio->pts == MP_NOPTS_VALUE) {
|
2013-10-29 21:38:29 +00:00
|
|
|
if (!did_retry) {
|
|
|
|
// Try to read more data to see packets that have pts
|
2014-03-07 14:24:32 +00:00
|
|
|
res = audio_decode(d_audio, mpctx->ao_buffer, out_format.rate);
|
2013-10-29 21:38:29 +00:00
|
|
|
if (res < 0)
|
|
|
|
return res;
|
|
|
|
did_retry = true;
|
|
|
|
continue;
|
|
|
|
}
|
2013-11-10 22:38:18 +00:00
|
|
|
samples = 0;
|
2013-10-29 21:38:29 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
if (fabs(ptsdiff) > 300 || isnan(ptsdiff)) // pts reset or just broken?
|
2013-11-10 22:38:18 +00:00
|
|
|
samples = 0;
|
2013-10-29 21:38:29 +00:00
|
|
|
|
2013-11-10 22:38:18 +00:00
|
|
|
if (samples > 0)
|
2013-10-29 21:38:29 +00:00
|
|
|
break;
|
|
|
|
|
|
|
|
mpctx->syncing_audio = false;
|
2013-11-10 22:38:18 +00:00
|
|
|
int skip_samples = -samples;
|
|
|
|
int a = MPMIN(skip_samples, MPMAX(playsize, 2500));
|
2014-03-07 14:24:32 +00:00
|
|
|
res = audio_decode(d_audio, mpctx->ao_buffer, a);
|
|
|
|
if (skip_samples <= mp_audio_buffer_samples(mpctx->ao_buffer)) {
|
|
|
|
mp_audio_buffer_skip(mpctx->ao_buffer, skip_samples);
|
2013-10-29 21:38:29 +00:00
|
|
|
if (res < 0)
|
|
|
|
return res;
|
2014-03-07 14:24:32 +00:00
|
|
|
return audio_decode(d_audio, mpctx->ao_buffer, playsize);
|
2013-10-29 21:38:29 +00:00
|
|
|
}
|
2014-03-07 14:24:32 +00:00
|
|
|
mp_audio_buffer_clear(mpctx->ao_buffer);
|
2013-10-29 21:38:29 +00:00
|
|
|
if (res < 0)
|
|
|
|
return res;
|
|
|
|
}
|
|
|
|
if (hrseek)
|
|
|
|
// Don't add silence in audio-only case even if position is too late
|
|
|
|
return 0;
|
2013-11-10 22:38:18 +00:00
|
|
|
if (samples >= playsize) {
|
2013-10-29 21:38:29 +00:00
|
|
|
/* This case could fall back to the one below with
|
2013-11-10 22:38:18 +00:00
|
|
|
* samples = playsize, but then silence would keep accumulating
|
2014-03-07 14:24:32 +00:00
|
|
|
* in ao_buffer if the AO accepts less data than it asks for
|
2013-10-29 21:38:29 +00:00
|
|
|
* in playsize. */
|
2013-11-10 22:38:18 +00:00
|
|
|
write_silence_to_ao(mpctx, playsize, 0,
|
|
|
|
written_pts - samples / real_samplerate);
|
2013-10-29 21:38:29 +00:00
|
|
|
return ASYNC_PLAY_DONE;
|
|
|
|
}
|
|
|
|
mpctx->syncing_audio = false;
|
2014-03-07 14:24:32 +00:00
|
|
|
mp_audio_buffer_prepend_silence(mpctx->ao_buffer, samples);
|
|
|
|
return audio_decode(d_audio, mpctx->ao_buffer, playsize);
|
2013-10-29 21:38:29 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
|
|
|
|
{
|
|
|
|
struct MPOpts *opts = mpctx->opts;
|
|
|
|
struct ao *ao = mpctx->ao;
|
|
|
|
int playsize;
|
|
|
|
int playflags = 0;
|
|
|
|
bool audio_eof = false;
|
audio: don't let ao_lavc access frontend internals, change gapless audio
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
2013-11-08 19:00:58 +00:00
|
|
|
bool signal_eof = false;
|
2013-10-29 21:38:29 +00:00
|
|
|
bool partial_fill = false;
|
2013-11-23 20:22:17 +00:00
|
|
|
struct dec_audio *d_audio = mpctx->d_audio;
|
2014-03-07 14:24:32 +00:00
|
|
|
struct mp_audio out_format;
|
|
|
|
ao_get_format(ao, &out_format);
|
audio: respect --end/--length with spdif passthrough
In theory, we can't really do this, because we don't know when a spdif
frame ends. Spdif transports compressed audio through audio setups that
were originally designed for PCM only (which includes the audio filter
chain, the AO API, most audio output APIs, etc.), and to reach this
goal, spdif pretends to be PCM. Compressed data frames are padded with
zeros, until a certain data rate is reached, which corresponds to a
pseudo-PCM format with 2 bytes per sample and 2 channels at 48000 Hz.
Of course an actual spdif frame is significantly larger than a frame
of the PCM format it pretends to be, so cutting audio data on frame
boundaries (as according to the pseudo-PCM format) merely yields an
incomplete and broken frame, not audio that plays for the desired
duration.
However, sending an incomplete frame might still be much better than the
current behavior, which simply ignores --end/--length (but still lets
the video end at the exact end time).
Should this result in trouble with real spdif receivers, this commit
probably has to be reverted.
2013-11-23 20:42:31 +00:00
|
|
|
// Can't adjust the start of audio with spdif pass-through.
|
2014-03-07 14:24:32 +00:00
|
|
|
bool modifiable_audio_format = !(out_format.format & AF_FORMAT_SPECIAL_MASK);
|
2013-10-29 21:38:29 +00:00
|
|
|
|
2013-11-23 20:22:17 +00:00
|
|
|
assert(d_audio);
|
|
|
|
|
2013-10-29 21:38:29 +00:00
|
|
|
if (mpctx->paused)
|
|
|
|
playsize = 1; // just initialize things (audio pts at least)
|
|
|
|
else
|
|
|
|
playsize = ao_get_space(ao);
|
|
|
|
|
|
|
|
// Coming here with hrseek_active still set means audio-only
|
2013-11-23 20:36:20 +00:00
|
|
|
if (!mpctx->d_video || !mpctx->sync_audio_to_video)
|
2013-10-29 21:38:29 +00:00
|
|
|
mpctx->syncing_audio = false;
|
|
|
|
if (!opts->initial_audio_sync || !modifiable_audio_format) {
|
|
|
|
mpctx->syncing_audio = false;
|
|
|
|
mpctx->hrseek_active = false;
|
|
|
|
}
|
|
|
|
|
|
|
|
int res;
|
|
|
|
if (mpctx->syncing_audio || mpctx->hrseek_active)
|
|
|
|
res = audio_start_sync(mpctx, playsize);
|
|
|
|
else
|
2014-03-07 14:24:32 +00:00
|
|
|
res = audio_decode(d_audio, mpctx->ao_buffer, playsize);
|
2013-10-29 21:38:29 +00:00
|
|
|
|
|
|
|
if (res < 0) { // EOF, error or format change
|
|
|
|
if (res == -2) {
|
|
|
|
/* The format change isn't handled too gracefully. A more precise
|
|
|
|
* implementation would require draining buffered old-format audio
|
|
|
|
* while displaying video, then doing the output format switch.
|
|
|
|
*/
|
|
|
|
if (!mpctx->opts->gapless_audio)
|
|
|
|
uninit_player(mpctx, INITIALIZED_AO);
|
|
|
|
reinit_audio_chain(mpctx);
|
|
|
|
return -1;
|
|
|
|
} else if (res == ASYNC_PLAY_DONE)
|
|
|
|
return 0;
|
2013-11-23 20:22:17 +00:00
|
|
|
else if (demux_stream_eof(d_audio->header))
|
2013-10-29 21:38:29 +00:00
|
|
|
audio_eof = true;
|
|
|
|
}
|
|
|
|
|
audio: respect --end/--length with spdif passthrough
In theory, we can't really do this, because we don't know when a spdif
frame ends. Spdif transports compressed audio through audio setups that
were originally designed for PCM only (which includes the audio filter
chain, the AO API, most audio output APIs, etc.), and to reach this
goal, spdif pretends to be PCM. Compressed data frames are padded with
zeros, until a certain data rate is reached, which corresponds to a
pseudo-PCM format with 2 bytes per sample and 2 channels at 48000 Hz.
Of course an actual spdif frame is significantly larger than a frame
of the PCM format it pretends to be, so cutting audio data on frame
boundaries (as according to the pseudo-PCM format) merely yields an
incomplete and broken frame, not audio that plays for the desired
duration.
However, sending an incomplete frame might still be much better than the
current behavior, which simply ignores --end/--length (but still lets
the video end at the exact end time).
Should this result in trouble with real spdif receivers, this commit
probably has to be reverted.
2013-11-23 20:42:31 +00:00
|
|
|
if (endpts != MP_NOPTS_VALUE) {
|
2014-01-06 17:39:49 +00:00
|
|
|
double samples = (endpts - written_audio_pts(mpctx) - mpctx->audio_delay)
|
2014-03-07 14:24:32 +00:00
|
|
|
* out_format.rate / opts->playback_speed;
|
2013-11-10 22:38:18 +00:00
|
|
|
if (playsize > samples) {
|
|
|
|
playsize = MPMAX(samples, 0);
|
2013-10-29 21:38:29 +00:00
|
|
|
audio_eof = true;
|
|
|
|
partial_fill = true;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2014-03-07 14:24:32 +00:00
|
|
|
if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) {
|
|
|
|
playsize = mp_audio_buffer_samples(mpctx->ao_buffer);
|
2013-10-29 21:38:29 +00:00
|
|
|
partial_fill = true;
|
|
|
|
}
|
|
|
|
if (!playsize)
|
|
|
|
return partial_fill && audio_eof ? -2 : -partial_fill;
|
|
|
|
|
audio: don't let ao_lavc access frontend internals, change gapless audio
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
2013-11-08 19:00:58 +00:00
|
|
|
if (audio_eof && partial_fill) {
|
|
|
|
if (opts->gapless_audio) {
|
|
|
|
// With gapless audio, delay this to ao_uninit. There must be only
|
|
|
|
// 1 final chunk, and that is handled when calling ao_uninit().
|
|
|
|
signal_eof = true;
|
|
|
|
} else {
|
|
|
|
playflags |= AOPLAY_FINAL_CHUNK;
|
|
|
|
}
|
|
|
|
}
|
2013-10-29 21:38:29 +00:00
|
|
|
|
2013-11-10 22:38:18 +00:00
|
|
|
struct mp_audio data;
|
2014-03-07 14:24:32 +00:00
|
|
|
mp_audio_buffer_peek(mpctx->ao_buffer, &data);
|
2013-11-10 22:38:18 +00:00
|
|
|
data.samples = MPMIN(data.samples, playsize);
|
|
|
|
int played = write_to_ao(mpctx, &data, playflags, written_audio_pts(mpctx));
|
2014-02-09 15:21:17 +00:00
|
|
|
assert(played >= 0 && played <= data.samples);
|
2013-10-29 21:38:29 +00:00
|
|
|
|
|
|
|
if (played > 0) {
|
2014-03-07 14:24:32 +00:00
|
|
|
mp_audio_buffer_skip(mpctx->ao_buffer, played);
|
2013-10-29 21:38:29 +00:00
|
|
|
} else if (!mpctx->paused && audio_eof && ao_get_delay(ao) < .04) {
|
|
|
|
// Sanity check to avoid hanging in case current ao doesn't output
|
|
|
|
// partial chunks and doesn't check for AOPLAY_FINAL_CHUNK
|
audio: don't let ao_lavc access frontend internals, change gapless audio
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
2013-11-08 19:00:58 +00:00
|
|
|
signal_eof = true;
|
2013-10-29 21:38:29 +00:00
|
|
|
}
|
|
|
|
|
audio: don't let ao_lavc access frontend internals, change gapless audio
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
2013-11-08 19:00:58 +00:00
|
|
|
return signal_eof ? -2 : -partial_fill;
|
2013-10-29 21:38:29 +00:00
|
|
|
}
|
2013-11-08 19:02:09 +00:00
|
|
|
|
|
|
|
// Drop data queued for output, or which the AO is currently outputting.
|
|
|
|
void clear_audio_output_buffers(struct MPContext *mpctx)
|
|
|
|
{
|
|
|
|
if (mpctx->ao) {
|
|
|
|
ao_reset(mpctx->ao);
|
2014-03-07 14:24:32 +00:00
|
|
|
mp_audio_buffer_clear(mpctx->ao_buffer);
|
2013-11-08 19:02:09 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
// Drop decoded data queued for filtering.
|
|
|
|
void clear_audio_decode_buffers(struct MPContext *mpctx)
|
|
|
|
{
|
2013-11-23 20:22:17 +00:00
|
|
|
if (mpctx->d_audio)
|
|
|
|
mp_audio_buffer_clear(mpctx->d_audio->decode_buffer);
|
2013-11-08 19:02:09 +00:00
|
|
|
}
|