mpv/player/audio.c

472 lines
16 KiB
C

/*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stddef.h>
#include <stdbool.h>
#include <inttypes.h>
#include <math.h>
#include <assert.h>
#include "config.h"
#include "talloc.h"
#include "common/msg.h"
#include "options/options.h"
#include "common/common.h"
#include "audio/mixer.h"
#include "audio/audio.h"
#include "audio/audio_buffer.h"
#include "audio/decode/dec_audio.h"
#include "audio/filter/af.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "video/decode/dec_video.h"
#include "core.h"
static int build_afilter_chain(struct MPContext *mpctx)
{
struct dec_audio *d_audio = mpctx->d_audio;
struct ao *ao = mpctx->ao;
struct MPOpts *opts = mpctx->opts;
if (!d_audio)
return 0;
struct mp_audio in_format;
mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
int new_srate;
if (af_control_any_rev(d_audio->afilter, AF_CONTROL_SET_PLAYBACK_SPEED,
&opts->playback_speed))
new_srate = in_format.rate;
else {
new_srate = in_format.rate * opts->playback_speed;
if (new_srate != ao->samplerate) {
// limits are taken from libaf/af_resample.c
if (new_srate < 8000)
new_srate = 8000;
if (new_srate > 192000)
new_srate = 192000;
opts->playback_speed = new_srate / (double)in_format.rate;
}
}
return audio_init_filters(d_audio, new_srate,
&ao->samplerate, &ao->channels, &ao->format);
}
static int recreate_audio_filters(struct MPContext *mpctx)
{
assert(mpctx->d_audio);
// init audio filters:
if (!build_afilter_chain(mpctx)) {
MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
return -1;
}
mixer_reinit_audio(mpctx->mixer, mpctx->ao, mpctx->d_audio->afilter);
return 0;
}
int reinit_audio_filters(struct MPContext *mpctx)
{
struct dec_audio *d_audio = mpctx->d_audio;
if (!d_audio)
return -2;
af_uninit(mpctx->d_audio->afilter);
if (af_init(mpctx->d_audio->afilter) < 0)
return -1;
if (recreate_audio_filters(mpctx) < 0)
return -1;
return 0;
}
void reinit_audio_chain(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct sh_stream *sh = init_demux_stream(mpctx, STREAM_AUDIO);
if (!sh) {
uninit_player(mpctx, INITIALIZED_AO);
goto no_audio;
}
if (!(mpctx->initialized_flags & INITIALIZED_ACODEC)) {
mpctx->initialized_flags |= INITIALIZED_ACODEC;
assert(!mpctx->d_audio);
mpctx->d_audio = talloc_zero(NULL, struct dec_audio);
mpctx->d_audio->opts = opts;
mpctx->d_audio->header = sh;
if (!audio_init_best_codec(mpctx->d_audio, opts->audio_decoders))
goto init_error;
}
assert(mpctx->d_audio);
struct mp_audio in_format;
mp_audio_buffer_get_format(mpctx->d_audio->decode_buffer, &in_format);
int ao_srate = opts->force_srate;
int ao_format = opts->audio_output_format;
struct mp_chmap ao_channels = {0};
if (mpctx->initialized_flags & INITIALIZED_AO) {
ao_srate = mpctx->ao->samplerate;
ao_format = mpctx->ao->format;
ao_channels = mpctx->ao->channels;
} else {
// Automatic downmix
if (mp_chmap_is_stereo(&opts->audio_output_channels) &&
!mp_chmap_is_stereo(&in_format.channels))
{
mp_chmap_from_channels(&ao_channels, 2);
}
}
// Determine what the filter chain outputs. build_afilter_chain() also
// needs this for testing whether playback speed is changed by resampling
// or using a special filter.
if (!audio_init_filters(mpctx->d_audio, // preliminary init
// input:
in_format.rate,
// output:
&ao_srate, &ao_channels, &ao_format)) {
MP_ERR(mpctx, "Error at audio filter chain pre-init!\n");
goto init_error;
}
if (!(mpctx->initialized_flags & INITIALIZED_AO)) {
mpctx->initialized_flags |= INITIALIZED_AO;
mp_chmap_remove_useless_channels(&ao_channels,
&opts->audio_output_channels);
mpctx->ao = ao_init_best(mpctx->global, mpctx->input,
mpctx->encode_lavc_ctx, ao_srate, ao_format,
ao_channels);
struct ao *ao = mpctx->ao;
if (!ao) {
MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
goto init_error;
}
ao->buffer = mp_audio_buffer_create(ao);
mp_audio_buffer_reinit_fmt(ao->buffer, ao->format, &ao->channels,
ao->samplerate);
char *s = mp_audio_fmt_to_str(ao->samplerate, &ao->channels, ao->format);
MP_INFO(mpctx, "AO: [%s] %s\n", ao->driver->name, s);
talloc_free(s);
MP_VERBOSE(mpctx, "AO: Description: %s\n", ao->driver->description);
update_window_title(mpctx, true);
}
if (recreate_audio_filters(mpctx) < 0)
goto init_error;
mpctx->syncing_audio = true;
return;
init_error:
uninit_player(mpctx, INITIALIZED_ACODEC | INITIALIZED_AO);
cleanup_demux_stream(mpctx, STREAM_AUDIO);
no_audio:
mpctx->current_track[STREAM_AUDIO] = NULL;
MP_INFO(mpctx, "Audio: no audio\n");
}
// Return pts value corresponding to the end point of audio written to the
// ao so far.
double written_audio_pts(struct MPContext *mpctx)
{
struct dec_audio *d_audio = mpctx->d_audio;
if (!d_audio)
return MP_NOPTS_VALUE;
struct mp_audio in_format;
mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
// first calculate the end pts of audio that has been output by decoder
double a_pts = d_audio->pts;
if (a_pts == MP_NOPTS_VALUE)
return MP_NOPTS_VALUE;
// d_audio->pts is the timestamp of the latest input packet with
// known pts that the decoder has decoded. d_audio->pts_bytes is
// the amount of bytes the decoder has written after that timestamp.
a_pts += d_audio->pts_offset / (double)in_format.rate;
// 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
a_pts -= mp_audio_buffer_seconds(d_audio->decode_buffer);
// Data buffered in audio filters, measured in seconds of "missing" output
double buffered_output = af_calc_delay(d_audio->afilter);
// Data that was ready for ao but was buffered because ao didn't fully
// accept everything to internal buffers yet
buffered_output += mp_audio_buffer_seconds(mpctx->ao->buffer);
// Filters divide audio length by playback_speed, so multiply by it
// to get the length in original units without speedup or slowdown
a_pts -= buffered_output * mpctx->opts->playback_speed;
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);
}
static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags,
double pts)
{
if (mpctx->paused)
return 0;
struct ao *ao = mpctx->ao;
ao->pts = pts;
double real_samplerate = ao->samplerate / mpctx->opts->playback_speed;
int played = ao_play(mpctx->ao, data->planes, data->samples, flags);
assert(played <= data->samples);
if (played > 0) {
mpctx->shown_aframes += played;
mpctx->delay += played / real_samplerate;
// Keep correct pts for remaining data - could be used to flush
// remaining buffer when closing ao.
ao->pts += played / real_samplerate;
return played;
}
return 0;
}
static int write_silence_to_ao(struct MPContext *mpctx, int samples, int flags,
double pts)
{
struct mp_audio tmp = {0};
mp_audio_buffer_get_format(mpctx->ao->buffer, &tmp);
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;
}
#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;
struct dec_audio *d_audio = mpctx->d_audio;
int res;
assert(d_audio);
// Timing info may not be set without
res = audio_decode(d_audio, ao->buffer, 1);
if (res < 0)
return res;
int samples;
bool did_retry = false;
double written_pts;
double real_samplerate = ao->samplerate / opts->playback_speed;
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
ptsdiff = written_pts - mpctx->video_next_pts - mpctx->delay
- mpctx->audio_delay;
samples = ptsdiff * real_samplerate;
// ogg demuxers give packets without timing
if (written_pts <= 1 && d_audio->pts == MP_NOPTS_VALUE) {
if (!did_retry) {
// Try to read more data to see packets that have pts
res = audio_decode(d_audio, ao->buffer, ao->samplerate);
if (res < 0)
return res;
did_retry = true;
continue;
}
samples = 0;
}
if (fabs(ptsdiff) > 300 || isnan(ptsdiff)) // pts reset or just broken?
samples = 0;
if (samples > 0)
break;
mpctx->syncing_audio = false;
int skip_samples = -samples;
int a = MPMIN(skip_samples, MPMAX(playsize, 2500));
res = audio_decode(d_audio, ao->buffer, a);
if (skip_samples <= mp_audio_buffer_samples(ao->buffer)) {
mp_audio_buffer_skip(ao->buffer, skip_samples);
if (res < 0)
return res;
return audio_decode(d_audio, ao->buffer, playsize);
}
mp_audio_buffer_clear(ao->buffer);
if (res < 0)
return res;
}
if (hrseek)
// Don't add silence in audio-only case even if position is too late
return 0;
if (samples >= playsize) {
/* This case could fall back to the one below with
* samples = playsize, but then silence would keep accumulating
* in ao->buffer if the AO accepts less data than it asks for
* in playsize. */
write_silence_to_ao(mpctx, playsize, 0,
written_pts - samples / real_samplerate);
return ASYNC_PLAY_DONE;
}
mpctx->syncing_audio = false;
mp_audio_buffer_prepend_silence(ao->buffer, samples);
return audio_decode(d_audio, ao->buffer, playsize);
}
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;
bool signal_eof = false;
bool partial_fill = false;
struct dec_audio *d_audio = mpctx->d_audio;
// Can't adjust the start of audio with spdif pass-through.
bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK);
assert(d_audio);
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
if (!mpctx->d_video || !mpctx->sync_audio_to_video)
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
res = audio_decode(d_audio, ao->buffer, playsize);
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;
else if (demux_stream_eof(d_audio->header))
audio_eof = true;
}
if (endpts != MP_NOPTS_VALUE) {
double samples = (endpts - written_audio_pts(mpctx) + mpctx->audio_delay)
* ao->samplerate / opts->playback_speed;
if (playsize > samples) {
playsize = MPMAX(samples, 0);
audio_eof = true;
partial_fill = true;
}
}
if (playsize > mp_audio_buffer_samples(ao->buffer)) {
playsize = mp_audio_buffer_samples(ao->buffer);
partial_fill = true;
}
if (!playsize)
return partial_fill && audio_eof ? -2 : -partial_fill;
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;
}
}
assert(ao->buffer_playable_samples <= mp_audio_buffer_samples(ao->buffer));
struct mp_audio data;
mp_audio_buffer_peek(ao->buffer, &data);
data.samples = MPMIN(data.samples, playsize);
int played = write_to_ao(mpctx, &data, playflags, written_audio_pts(mpctx));
ao->buffer_playable_samples = playsize - played;
if (played > 0) {
mp_audio_buffer_skip(ao->buffer, played);
} 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
signal_eof = true;
}
return signal_eof ? -2 : -partial_fill;
}
// 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);
mp_audio_buffer_clear(mpctx->ao->buffer);
mpctx->ao->buffer_playable_samples = 0;
}
}
// Drop decoded data queued for filtering.
void clear_audio_decode_buffers(struct MPContext *mpctx)
{
if (mpctx->d_audio)
mp_audio_buffer_clear(mpctx->d_audio->decode_buffer);
}