mpv/audio/out/internal.h

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/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* mpv 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef MP_AO_INTERNAL_H_
#define MP_AO_INTERNAL_H_
#include <stdbool.h>
#include <pthread.h>
#include "osdep/atomic.h"
#include "audio/out/ao.h"
/* global data used by ao.c and ao drivers */
struct ao {
int samplerate;
struct mp_chmap channels;
int format; // one of AF_FORMAT_...
int bps; // bytes per second (per plane)
int sstride; // size of a sample on each plane
// (format_size*num_channels/num_planes)
int num_planes;
bool probing; // if true, don't fail loudly on init
bool untimed; // don't assume realtime playback
int device_buffer; // device buffer in samples (guessed by
// common init code if not set by driver)
const struct ao_driver *driver;
audio: merge pull/push ring buffer glue code This is preparation to further cleanups (and eventually actual improvements) of the audio output code. AOs are split into two classes: pull and push. Pull AOs let an audio callback of the native audio API read from a ring buffer. Push AOs expose a function that works similar to write(), and for which we start a "feeder" thread. It seems making this split was beneficial, because of the different data flow, and emulating the one or other in the AOs directly would have created code duplication (all the "pull" AOs had their own ring buffer implementation before it was cleaned up). Unfortunately, both types had completely separate implementations (in pull.c and push.c). The idea was that little can be shared anyway. But that's very annoying now, because I want to change the API between AO and player. This commit attempts to merge them. I've moved everything from push.c to pull.c, the trivial entrypoints from ao.c to pull.c, and attempted to reconcile the differences. It's a mess, but at least there's only one ring buffer within the AO code now. Everything should work mostly the same. Pull AOs now always copy the audio data under a lock; before this commit, all ring buffer access was lock-free (except for the decoder wakeup callback, which acquired a mutex). In theory, this is "bad", and people obsessed with lock-free stuff will hate me, but in practice probably won't matter. The planned change will probably remove this copying-under-lock again, but who knows when this will happen. One change for the push AOs now makes it drop audio, where before only a warning was logged. This is only in case of AOs or drivers which exhibit unexpected (and now unsupported) behavior. This is a risky change. Although it's completely trivial conceptually, there are too many special cases. In addition, I barely tested it, and I've messed with it in a half-motivated state over a longer time, barely making any progress, and finishing it under a rush when I already should have been asleep. Most things seem to work, and I made superficial tests with alsa, sdl, and encode mode. This should cover most things, but there are a lot of tricky things that received no coverage. All this text means you should be prepared to roll back to an older commit and report your problem.
2020-05-24 23:53:41 +00:00
bool driver_initialized;
void *priv;
struct mpv_global *global;
struct encode_lavc_context *encode_lavc_ctx;
void (*wakeup_cb)(void *ctx);
void *wakeup_ctx;
struct mp_log *log; // Using e.g. "[ao/coreaudio]" as prefix
int init_flags; // AO_INIT_* flags
bool stream_silence; // if audio inactive, just play silence
// The device as selected by the user, usually using ao_device_desc.name
// from an entry from the list returned by driver->list_devices. If the
// default device should be used, this is set to NULL.
char *device;
// Application name to report to the audio API.
char *client_name;
// Used during init: if init fails, redirect to this ao
char *redirect;
// Internal events (use ao_request_reload(), ao_hotplug_event())
atomic_uint events_;
// Float gain multiplicator
mp_atomic_float gain;
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 20:30:10 +00:00
int buffer;
double def_buffer;
audio: merge pull/push ring buffer glue code This is preparation to further cleanups (and eventually actual improvements) of the audio output code. AOs are split into two classes: pull and push. Pull AOs let an audio callback of the native audio API read from a ring buffer. Push AOs expose a function that works similar to write(), and for which we start a "feeder" thread. It seems making this split was beneficial, because of the different data flow, and emulating the one or other in the AOs directly would have created code duplication (all the "pull" AOs had their own ring buffer implementation before it was cleaned up). Unfortunately, both types had completely separate implementations (in pull.c and push.c). The idea was that little can be shared anyway. But that's very annoying now, because I want to change the API between AO and player. This commit attempts to merge them. I've moved everything from push.c to pull.c, the trivial entrypoints from ao.c to pull.c, and attempted to reconcile the differences. It's a mess, but at least there's only one ring buffer within the AO code now. Everything should work mostly the same. Pull AOs now always copy the audio data under a lock; before this commit, all ring buffer access was lock-free (except for the decoder wakeup callback, which acquired a mutex). In theory, this is "bad", and people obsessed with lock-free stuff will hate me, but in practice probably won't matter. The planned change will probably remove this copying-under-lock again, but who knows when this will happen. One change for the push AOs now makes it drop audio, where before only a warning was logged. This is only in case of AOs or drivers which exhibit unexpected (and now unsupported) behavior. This is a risky change. Although it's completely trivial conceptually, there are too many special cases. In addition, I barely tested it, and I've messed with it in a half-motivated state over a longer time, barely making any progress, and finishing it under a rush when I already should have been asleep. Most things seem to work, and I made superficial tests with alsa, sdl, and encode mode. This should cover most things, but there are a lot of tricky things that received no coverage. All this text means you should be prepared to roll back to an older commit and report your problem.
2020-05-24 23:53:41 +00:00
struct buffer_state *buffer_state;
};
audio: merge pull/push ring buffer glue code This is preparation to further cleanups (and eventually actual improvements) of the audio output code. AOs are split into two classes: pull and push. Pull AOs let an audio callback of the native audio API read from a ring buffer. Push AOs expose a function that works similar to write(), and for which we start a "feeder" thread. It seems making this split was beneficial, because of the different data flow, and emulating the one or other in the AOs directly would have created code duplication (all the "pull" AOs had their own ring buffer implementation before it was cleaned up). Unfortunately, both types had completely separate implementations (in pull.c and push.c). The idea was that little can be shared anyway. But that's very annoying now, because I want to change the API between AO and player. This commit attempts to merge them. I've moved everything from push.c to pull.c, the trivial entrypoints from ao.c to pull.c, and attempted to reconcile the differences. It's a mess, but at least there's only one ring buffer within the AO code now. Everything should work mostly the same. Pull AOs now always copy the audio data under a lock; before this commit, all ring buffer access was lock-free (except for the decoder wakeup callback, which acquired a mutex). In theory, this is "bad", and people obsessed with lock-free stuff will hate me, but in practice probably won't matter. The planned change will probably remove this copying-under-lock again, but who knows when this will happen. One change for the push AOs now makes it drop audio, where before only a warning was logged. This is only in case of AOs or drivers which exhibit unexpected (and now unsupported) behavior. This is a risky change. Although it's completely trivial conceptually, there are too many special cases. In addition, I barely tested it, and I've messed with it in a half-motivated state over a longer time, barely making any progress, and finishing it under a rush when I already should have been asleep. Most things seem to work, and I made superficial tests with alsa, sdl, and encode mode. This should cover most things, but there are a lot of tricky things that received no coverage. All this text means you should be prepared to roll back to an older commit and report your problem.
2020-05-24 23:53:41 +00:00
void init_buffer_pre(struct ao *ao);
bool init_buffer_post(struct ao *ao);
struct mp_pcm_state {
// Note: free_samples+queued_samples <= ao->device_buffer; the sum may be
// less if the audio API can report partial periods played, while
// free_samples should be period-size aligned. If free_samples is not
// period-size aligned, the AO thread might get into a situation where
// it writes a very small number of samples in each iteration, leading
// to extremely inefficient behavior.
// Keep in mind that write() may write less than free_samples (or your
// period size alignment) anyway.
int free_samples; // number of free space in ring buffer
int queued_samples; // number of samples to play in ring buffer
double delay; // total latency in seconds (includes queued_samples)
bool playing; // set if underlying API is actually playing audio;
// the AO must unset it on underrun (accidental
// underrun and EOF are indistinguishable; the upper
// layers decide what it was)
// real pausing may assume playing=true
};
/* Note:
*
* In general, there are two types of audio drivers:
* a) push based (the user queues data that should be played)
* b) pull callback based (the audio API calls a callback to get audio)
*
* The ao.c code can handle both. It basically implements two audio paths
* and provides a uniform API for them. If ao_driver->write is NULL, it assumes
* that the driver uses a callback based audio API, otherwise push based.
*
* Requirements:
* a+b) Mandatory for both types:
* init
* uninit
* start
* Optional for both types:
* control
* a) ->write is called to queue audio. push.c creates a thread to regularly
* refill audio device buffers with ->write, but all driver functions are
* always called under an exclusive lock.
* Mandatory:
* reset
* write
* get_state
audio/out/pull: remove race conditions There were subtle and minor race conditions in the pull.c code, and AOs using it (jack, portaudio, sdl, wasapi). Attempt to remove these. There was at least a race condition in the ao_reset() implementation: mp_ring_reset() was called concurrently to the audio callback. While the ringbuffer uses atomics to allow concurrent access, the reset function wasn't concurrency-safe (and can't easily be made to). Fix this by stopping the audio callback before doing a reset. After that, we can do anything without needing synchronization. The callback is resumed when resuming playback at a later point. Don't call driver->pause, and make driver->resume and driver->reset start/stop the audio callback. In the initial state, the audio callback must be disabled. JackAudio of course is different. Maybe there is no way to suspend the audio callback without "disconnecting" it (what jack_deactivate() would do), so I'm not trying my luck, and implemented a really bad hack doing active waiting until we get the audio callback into a state where it won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we can be sure that the callback doesn't access the ringbuffer or anything else anymore. Since both sched_yield() and pthread_yield() apparently are not always available, use mp_sleep_us(1) to avoid burning CPU during active waiting. The ao_jack.c change also removes a race condition: apparently we didn't initialize _all_ ao fields before starting the audio callback. In ao_wasapi.c, I'm not sure whether reset really waits for the audio callback to return. Kovensky says it's not guaranteed, so disable the reset callback - for now the behavior of ao_wasapi.c is like with ao_jack.c, and active waiting is used to deal with the audio callback.
2014-05-29 00:24:17 +00:00
* Optional:
* set_pause
* b) ->write must be NULL. ->start must be provided, and should make the
* audio API start calling the audio callback. Your audio callback should
* in turn call ao_read_data() to get audio data. Most functions are
* optional and will be emulated if missing (e.g. pausing is emulated as
* silence).
audio/out/pull: remove race conditions There were subtle and minor race conditions in the pull.c code, and AOs using it (jack, portaudio, sdl, wasapi). Attempt to remove these. There was at least a race condition in the ao_reset() implementation: mp_ring_reset() was called concurrently to the audio callback. While the ringbuffer uses atomics to allow concurrent access, the reset function wasn't concurrency-safe (and can't easily be made to). Fix this by stopping the audio callback before doing a reset. After that, we can do anything without needing synchronization. The callback is resumed when resuming playback at a later point. Don't call driver->pause, and make driver->resume and driver->reset start/stop the audio callback. In the initial state, the audio callback must be disabled. JackAudio of course is different. Maybe there is no way to suspend the audio callback without "disconnecting" it (what jack_deactivate() would do), so I'm not trying my luck, and implemented a really bad hack doing active waiting until we get the audio callback into a state where it won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we can be sure that the callback doesn't access the ringbuffer or anything else anymore. Since both sched_yield() and pthread_yield() apparently are not always available, use mp_sleep_us(1) to avoid burning CPU during active waiting. The ao_jack.c change also removes a race condition: apparently we didn't initialize _all_ ao fields before starting the audio callback. In ao_wasapi.c, I'm not sure whether reset really waits for the audio callback to return. Kovensky says it's not guaranteed, so disable the reset callback - for now the behavior of ao_wasapi.c is like with ao_jack.c, and active waiting is used to deal with the audio callback.
2014-05-29 00:24:17 +00:00
* Also, the following optional callbacks can be provided:
* reset (stops the audio callback, start() restarts it)
*/
struct ao_driver {
// If true, use with encoding only.
bool encode;
// Name used for --ao.
const char *name;
// Description shown with --ao=help.
const char *description;
// This requires waiting for a AO_EVENT_INITIAL_UNBLOCK event before the
// first write() call is done. Encode mode uses this, and push mode
// respects it automatically (don't use with pull mode).
bool initially_blocked;
// If true, write units of entire frames. The write() call is modified to
// use data==mp_aframe. Useful for encoding AO only.
bool write_frames;
// Init the device using ao->format/ao->channels/ao->samplerate. If the
// device doesn't accept these parameters, you can attempt to negotiate
// fallback parameters, and set the ao format fields accordingly.
int (*init)(struct ao *ao);
// Optional. See ao_control() etc. in ao.c
int (*control)(struct ao *ao, enum aocontrol cmd, void *arg);
void (*uninit)(struct ao *ao);
// Stop all audio playback, clear buffers, back to state after init().
// Optional for pull AOs.
void (*reset)(struct ao *ao);
// push based: set pause state. Only called after start() and before reset().
// returns success (this is intended for paused=true; if it
// returns false, playback continues, and the core emulates via
// reset(); unpausing always works)
// The pausing state is also cleared by reset().
bool (*set_pause)(struct ao *ao, bool paused);
// pull based: start the audio callback
// push based: start playing queued data
// AO should call ao_wakeup_playthread() if a period boundary
// is crossed, or playback stops due to external reasons
// (including underruns or device removal)
// must set mp_pcm_state.playing; unset on error/underrun/end
void (*start)(struct ao *ao);
// push based: queue new data. This won't try to write more data than the
// reported free space (samples <= mp_pcm_state.free_samples).
// This must NOT start playback. start() does that, and write() may be
// called multiple times before start() is called. It may also happen that
// reset() is called to discard the buffer. start() without write() will
// immediately reported an underrun.
// Return false on failure.
bool (*write)(struct ao *ao, void **data, int samples);
// push based: return mandatory stream information
void (*get_state)(struct ao *ao, struct mp_pcm_state *state);
// Return the list of devices currently available in the system. Use
// ao_device_list_add() to add entries. The selected device will be set as
// ao->device (using ao_device_desc.name).
// Warning: the ao struct passed is not initialized with ao_driver->init().
// Instead, hotplug_init/hotplug_uninit is called. If these
// callbacks are not set, no driver initialization call is done
// on the ao struct.
void (*list_devs)(struct ao *ao, struct ao_device_list *list);
// If set, these are called before/after ao_driver->list_devs is called.
// It is also assumed that the driver can do hotplugging - which means
// it is expected to call ao_hotplug_event(ao) whenever the system's
// audio device list changes. The player will then call list_devs() again.
int (*hotplug_init)(struct ao *ao);
void (*hotplug_uninit)(struct ao *ao);
// For option parsing (see vo.h)
int priv_size;
const void *priv_defaults;
const struct m_option *options;
const char *options_prefix;
const struct m_sub_options *global_opts;
};
// These functions can be called by AOs.
int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us);
bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map);
bool ao_chmap_sel_adjust2(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map, bool safe_multichannel);
bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map, int num);
// Add a deep copy of e to the list.
// Call from ao_driver->list_devs callback only.
void ao_device_list_add(struct ao_device_list *list, struct ao *ao,
struct ao_device_desc *e);
audio: add audio softvol processing to AO This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
2017-11-29 20:30:10 +00:00
void ao_post_process_data(struct ao *ao, void **data, int num_samples);
struct ao_convert_fmt {
int src_fmt; // source AF_FORMAT_*
int channels; // number of channels
int dst_bits; // total target data sample size
int pad_msb; // padding in the MSB (i.e. required shifting)
int pad_lsb; // padding in LSB (required 0 bits) (ignored)
};
bool ao_can_convert_inplace(struct ao_convert_fmt *fmt);
bool ao_need_conversion(struct ao_convert_fmt *fmt);
void ao_convert_inplace(struct ao_convert_fmt *fmt, void **data, int num_samples);
void ao_wakeup_playthread(struct ao *ao);
int ao_read_data_converted(struct ao *ao, struct ao_convert_fmt *fmt,
void **data, int samples, int64_t out_time_us);
#endif