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https://github.com/mpv-player/mpv
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d842b017e4
Since the addition of the AO feed thread, 200ms of latency (MIN_BUFFER) was added to all push-based AOs. This is not so nice, because even AOs with relatively small buffering (e.g. ao_alsa on my system with ~170ms of buffer size), the additional latency becomes noticable when e.g. toggling mute with softvol. Fix this by trying to keep not only 200ms minimum buffer, but also 200ms maximum buffer. In other words, never buffer beyond 200ms in total. Do this by estimating the AO's buffer fill status using get_space and the initially known AO buffer size (the get_space return value on initialization, before any audio was played). We limit the maximum amount of data written to the soft buffer so that soft buffer size and audio buffer size equal to 200ms (MIN_BUFFER). To avoid weird problems with weird AOs, we buffer beyond MIN_BUFFER if the AO's get_space requests more data than that, and as long as the soft buffer is large enough. Note that this is just a hack to improve the latency. When the audio chain gains the ability to refilter data, this won't be needed anymore, and instead we can introduce some sort of buffer replacement function in order to update data in the soft buffer.
133 lines
4.8 KiB
C
133 lines
4.8 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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#ifndef MP_AO_INTERNAL_H_
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#define MP_AO_INTERNAL_H_
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#include <stdbool.h>
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#include "audio/chmap.h"
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#include "audio/chmap_sel.h"
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// Minimum buffer size in seconds.
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#define MIN_BUFFER 0.2
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/* global data used by ao.c and ao drivers */
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struct ao {
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int samplerate;
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struct mp_chmap channels;
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int format; // one of AF_FORMAT_...
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int bps; // bytes per second (per plane)
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int sstride; // size of a sample on each plane
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// (format_size*num_channels/num_planes)
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int num_planes;
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bool probing; // if true, don't fail loudly on init
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bool untimed; // don't assume realtime playback
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bool no_persistent_volume; // the AO does the equivalent of af_volume
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bool per_application_mixer; // like above, but volume persists (per app)
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int device_buffer; // device buffer in samples (guessed by
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// common init code if not set by driver)
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const struct ao_driver *api; // entrypoints to the wrapper (push.c/pull.c)
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const struct ao_driver *driver;
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void *priv;
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struct encode_lavc_context *encode_lavc_ctx;
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struct input_ctx *input_ctx;
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struct mp_log *log; // Using e.g. "[ao/coreaudio]" as prefix
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int buffer;
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void *api_priv;
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};
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extern const struct ao_driver ao_api_push;
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extern const struct ao_driver ao_api_pull;
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/* Note:
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*
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* In general, there are two types of audio drivers:
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* a) push based (the user queues data that should be played)
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* b) pull callback based (the audio API calls a callback to get audio)
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*
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* The ao.c code can handle both. It basically implements two audio paths
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* and provides a uniform API for them. If ao_driver->play is NULL, it assumes
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* that the driver uses a callback based audio API, otherwise push based.
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*
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* Requirements:
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* a) Most functions (except ->control) must be provided. ->play is called to
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* queue audio. ao.c creates a thread to regularly refill audio device
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* buffers with ->play, but all driver functions are always called under
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* an exclusive lock.
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* Mandatory:
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* init
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* uninit
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* reset
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* get_space
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* play
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* get_delay
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* pause
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* resume
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* b) ->play must be NULL. The driver can start the audio API in init(). The
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* audio API in turn will start a thread and call a callback provided by the
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* driver. That callback calls ao_read_data() to get audio data. Most
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* functions are optional and will be emulated if missing (e.g. pausing
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* is emulated as silence). ->get_delay and ->get_space are never called.
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* Mandatory:
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* init
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* uninit
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*/
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struct ao_driver {
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// If true, use with encoding only.
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bool encode;
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// Name used for --ao.
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const char *name;
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// Description shown with --ao=help.
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const char *description;
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// Init the device using ao->format/ao->channels/ao->samplerate. If the
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// device doesn't accept these parameters, you can attempt to negotiate
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// fallback parameters, and set the ao format fields accordingly.
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int (*init)(struct ao *ao);
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// Optional. See ao_control() etc. in ao.c
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int (*control)(struct ao *ao, enum aocontrol cmd, void *arg);
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void (*uninit)(struct ao *ao);
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void (*reset)(struct ao*ao);
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int (*get_space)(struct ao *ao);
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int (*play)(struct ao *ao, void **data, int samples, int flags);
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float (*get_delay)(struct ao *ao);
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void (*pause)(struct ao *ao);
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void (*resume)(struct ao *ao);
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void (*drain)(struct ao *ao);
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// For option parsing (see vo.h)
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int priv_size;
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const void *priv_defaults;
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const struct m_option *options;
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};
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// These functions can be called by AOs.
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int ao_play_silence(struct ao *ao, int samples);
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void ao_wait_drain(struct ao *ao);
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int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us);
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bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
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struct mp_chmap *map);
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bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
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struct mp_chmap *map, int num);
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#endif
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