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mpv/audio/out/ao.c

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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/>.
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <assert.h>
#include "mpv_talloc.h"
#include "config.h"
#include "ao.h"
#include "internal.h"
#include "audio/format.h"
#include "options/options.h"
#include "options/m_config.h"
#include "osdep/endian.h"
#include "common/msg.h"
#include "common/common.h"
#include "common/global.h"
extern const struct ao_driver audio_out_oss;
extern const struct ao_driver audio_out_audiounit;
extern const struct ao_driver audio_out_coreaudio;
extern const struct ao_driver audio_out_coreaudio_exclusive;
extern const struct ao_driver audio_out_rsound;
extern const struct ao_driver audio_out_sndio;
extern const struct ao_driver audio_out_pulse;
extern const struct ao_driver audio_out_jack;
extern const struct ao_driver audio_out_openal;
extern const struct ao_driver audio_out_opensles;
extern const struct ao_driver audio_out_null;
extern const struct ao_driver audio_out_alsa;
extern const struct ao_driver audio_out_wasapi;
extern const struct ao_driver audio_out_pcm;
extern const struct ao_driver audio_out_lavc;
extern const struct ao_driver audio_out_sdl;
static const struct ao_driver * const audio_out_drivers[] = {
// native:
#if HAVE_AUDIOUNIT
&audio_out_audiounit,
#endif
#if HAVE_COREAUDIO
&audio_out_coreaudio,
#endif
#if HAVE_PULSE
&audio_out_pulse,
#endif
#if HAVE_ALSA
&audio_out_alsa,
#endif
#if HAVE_WASAPI
&audio_out_wasapi,
#endif
#if HAVE_OSS_AUDIO
&audio_out_oss,
#endif
// wrappers:
#if HAVE_JACK
&audio_out_jack,
#endif
#if HAVE_OPENAL
&audio_out_openal,
#endif
#if HAVE_OPENSLES
&audio_out_opensles,
#endif
#if HAVE_SDL2
&audio_out_sdl,
#endif
#if HAVE_SNDIO
&audio_out_sndio,
#endif
&audio_out_null,
#if HAVE_COREAUDIO
&audio_out_coreaudio_exclusive,
#endif
&audio_out_pcm,
#if HAVE_ENCODING
&audio_out_lavc,
#endif
#if HAVE_RSOUND
&audio_out_rsound,
#endif
NULL
};
static bool get_desc(struct m_obj_desc *dst, int index)
{
if (index >= MP_ARRAY_SIZE(audio_out_drivers) - 1)
return false;
const struct ao_driver *ao = audio_out_drivers[index];
*dst = (struct m_obj_desc) {
.name = ao->name,
.description = ao->description,
.priv_size = ao->priv_size,
.priv_defaults = ao->priv_defaults,
.options = ao->options,
.options_prefix = ao->options_prefix,
.global_opts = ao->global_opts,
.hidden = ao->encode,
.p = ao,
};
return true;
}
// For the ao option
const struct m_obj_list ao_obj_list = {
.get_desc = get_desc,
.description = "audio outputs",
.allow_unknown_entries = true,
.allow_trailer = true,
.disallow_positional_parameters = true,
.use_global_options = true,
};
static struct ao *ao_alloc(bool probing, struct mpv_global *global,
void (*wakeup_cb)(void *ctx), void *wakeup_ctx,
char *name)
{
assert(wakeup_cb);
struct MPOpts *opts = global->opts;
struct mp_log *log = mp_log_new(NULL, global->log, "ao");
struct m_obj_desc desc;
if (!m_obj_list_find(&desc, &ao_obj_list, bstr0(name))) {
mp_msg(log, MSGL_ERR, "Audio output %s not found!\n", name);
talloc_free(log);
return NULL;
};
struct ao *ao = talloc_ptrtype(NULL, ao);
talloc_steal(ao, log);
*ao = (struct ao) {
.driver = desc.p,
.probing = probing,
.global = global,
.wakeup_cb = wakeup_cb,
.wakeup_ctx = wakeup_ctx,
.log = mp_log_new(ao, log, name),
.def_buffer = opts->audio_buffer,
.client_name = talloc_strdup(ao, opts->audio_client_name),
};
ao->priv = m_config_group_from_desc(ao, ao->log, global, &desc, name);
if (!ao->priv)
goto error;
ao_set_gain(ao, 1.0f);
return ao;
error:
talloc_free(ao);
return NULL;
}
static struct ao *ao_init(bool probing, struct mpv_global *global,
void (*wakeup_cb)(void *ctx), void *wakeup_ctx,
struct encode_lavc_context *encode_lavc_ctx, int flags,
int samplerate, int format, struct mp_chmap channels,
char *dev, char *name)
{
struct ao *ao = ao_alloc(probing, global, wakeup_cb, wakeup_ctx, name);
if (!ao)
return NULL;
ao->samplerate = samplerate;
ao->channels = channels;
ao->format = format;
ao->encode_lavc_ctx = encode_lavc_ctx;
ao->init_flags = flags;
if (ao->driver->encode != !!ao->encode_lavc_ctx)
goto fail;
MP_VERBOSE(ao, "requested format: %d Hz, %s channels, %s\n",
ao->samplerate, mp_chmap_to_str(&ao->channels),
af_fmt_to_str(ao->format));
ao->device = talloc_strdup(ao, dev);
ao->api = ao->driver->play ? &ao_api_push : &ao_api_pull;
ao->api_priv = talloc_zero_size(ao, ao->api->priv_size);
assert(!ao->api->priv_defaults && !ao->api->options);
ao->stream_silence = flags & AO_INIT_STREAM_SILENCE;
ao->period_size = 1;
int r = ao->driver->init(ao);
if (r < 0) {
// Silly exception for coreaudio spdif redirection
if (ao->redirect) {
char redirect[80], rdevice[80];
snprintf(redirect, sizeof(redirect), "%s", ao->redirect);
snprintf(rdevice, sizeof(rdevice), "%s", ao->device ? ao->device : "");
talloc_free(ao);
return ao_init(probing, global, wakeup_cb, wakeup_ctx,
encode_lavc_ctx, flags, samplerate, format, channels,
rdevice, redirect);
}
goto fail;
}
if (ao->period_size < 1) {
MP_ERR(ao, "Invalid period size set.\n");
goto fail;
}
ao->sstride = af_fmt_to_bytes(ao->format);
ao->num_planes = 1;
if (af_fmt_is_planar(ao->format)) {
ao->num_planes = ao->channels.num;
} else {
ao->sstride *= ao->channels.num;
}
ao->bps = ao->samplerate * ao->sstride;
if (!ao->device_buffer && ao->driver->get_space)
ao->device_buffer = ao->driver->get_space(ao);
if (ao->device_buffer)
MP_VERBOSE(ao, "device buffer: %d samples.\n", ao->device_buffer);
ao->buffer = MPMAX(ao->device_buffer, ao->def_buffer * ao->samplerate);
ao->buffer = MPMAX(ao->buffer, 1);
int align = af_format_sample_alignment(ao->format);
ao->buffer = (ao->buffer + align - 1) / align * align;
MP_VERBOSE(ao, "using soft-buffer of %d samples.\n", ao->buffer);
if (ao->api->init(ao) < 0)
goto fail;
return ao;
fail:
talloc_free(ao);
return NULL;
}
static void split_ao_device(void *tmp, char *opt, char **out_ao, char **out_dev)
{
*out_ao = NULL;
*out_dev = NULL;
if (!opt)
return;
if (!opt[0] || strcmp(opt, "auto") == 0)
return;
// Split on "/". If "/" is the final character, or absent, out_dev is NULL.
bstr b_dev, b_ao;
bstr_split_tok(bstr0(opt), "/", &b_ao, &b_dev);
if (b_dev.len > 0)
*out_dev = bstrto0(tmp, b_dev);
*out_ao = bstrto0(tmp, b_ao);
}
struct ao *ao_init_best(struct mpv_global *global,
int init_flags,
void (*wakeup_cb)(void *ctx), void *wakeup_ctx,
struct encode_lavc_context *encode_lavc_ctx,
int samplerate, int format, struct mp_chmap channels)
{
struct MPOpts *opts = global->opts;
void *tmp = talloc_new(NULL);
struct mp_log *log = mp_log_new(tmp, global->log, "ao");
struct ao *ao = NULL;
struct m_obj_settings *ao_list = NULL;
int ao_num = 0;
for (int n = 0; opts->audio_driver_list && opts->audio_driver_list[n].name; n++)
MP_TARRAY_APPEND(tmp, ao_list, ao_num, opts->audio_driver_list[n]);
bool forced_dev = false;
char *pref_ao, *pref_dev;
split_ao_device(tmp, opts->audio_device, &pref_ao, &pref_dev);
if (!ao_num && pref_ao) {
// Reuse the autoselection code
MP_TARRAY_APPEND(tmp, ao_list, ao_num,
(struct m_obj_settings){.name = pref_ao});
forced_dev = true;
}
bool autoprobe = ao_num == 0;
// Something like "--ao=a,b," means do autoprobing after a and b fail.
if (ao_num && strlen(ao_list[ao_num - 1].name) == 0) {
ao_num -= 1;
autoprobe = true;
}
if (autoprobe) {
for (int n = 0; audio_out_drivers[n]; n++) {
const struct ao_driver *driver = audio_out_drivers[n];
if (driver == &audio_out_null)
break;
MP_TARRAY_APPEND(tmp, ao_list, ao_num,
(struct m_obj_settings){.name = (char *)driver->name});
}
}
if (init_flags & AO_INIT_NULL_FALLBACK) {
MP_TARRAY_APPEND(tmp, ao_list, ao_num,
(struct m_obj_settings){.name = "null"});
}
for (int n = 0; n < ao_num; n++) {
struct m_obj_settings *entry = &ao_list[n];
bool probing = n + 1 != ao_num;
mp_verbose(log, "Trying audio driver '%s'\n", entry->name);
char *dev = NULL;
if (pref_ao && pref_dev && strcmp(entry->name, pref_ao) == 0) {
dev = pref_dev;
mp_verbose(log, "Using preferred device '%s'\n", dev);
}
ao = ao_init(probing, global, wakeup_cb, wakeup_ctx, encode_lavc_ctx,
init_flags, samplerate, format, channels, dev, entry->name);
if (ao)
break;
if (!probing)
mp_err(log, "Failed to initialize audio driver '%s'\n", entry->name);
if (dev && forced_dev) {
mp_err(log, "This audio driver/device was forced with the "
"--audio-device option.\nTry unsetting it.\n");
}
}
talloc_free(tmp);
return ao;
}
// Uninitialize and destroy the AO. Remaining audio must be dropped.
void ao_uninit(struct ao *ao)
{
if (ao)
ao->api->uninit(ao);
talloc_free(ao);
}
// Queue the given audio data. Start playback if it hasn't started yet. Return
// the number of samples that was accepted (the core will try to queue the rest
// again later). Should never block.
// data: start pointer for each plane. If the audio data is packed, only
// data[0] is valid, otherwise there is a plane for each channel.
// samples: size of the audio data (see ao->sstride)
// flags: currently AOPLAY_FINAL_CHUNK can be set
int ao_play(struct ao *ao, void **data, int samples, int flags)
{
return ao->api->play(ao, data, samples, flags);
}
int ao_control(struct ao *ao, enum aocontrol cmd, void *arg)
{
return ao->api->control ? ao->api->control(ao, cmd, arg) : CONTROL_UNKNOWN;
}
// Return size of the buffered data in seconds. Can include the device latency.
// Basically, this returns how much data there is still to play, and how long
// it takes until the last sample in the buffer reaches the speakers. This is
// used for audio/video synchronization, so it's very important to implement
// this correctly.
double ao_get_delay(struct ao *ao)
{
return ao->api->get_delay(ao);
}
// Return free size of the internal audio buffer. This controls how much audio
// the core should decode and try to queue with ao_play().
int ao_get_space(struct ao *ao)
{
return ao->api->get_space(ao);
}
// Stop playback and empty buffers. Essentially go back to the state after
// ao->init().
void ao_reset(struct ao *ao)
{
if (ao->api->reset)
ao->api->reset(ao);
}
// Pause playback. Keep the current buffer. ao_get_delay() must return the
// same value as before pausing.
void ao_pause(struct ao *ao)
{
if (ao->api->pause)
ao->api->pause(ao);
}
// Resume playback. Play the remaining buffer. If the driver doesn't support
// pausing, it has to work around this and e.g. use ao_play_silence() to fill
// the lost audio.
void ao_resume(struct ao *ao)
{
if (ao->api->resume)
ao->api->resume(ao);
}
// Block until the current audio buffer has played completely.
void ao_drain(struct ao *ao)
{
if (ao->api->drain)
ao->api->drain(ao);
}
bool ao_eof_reached(struct ao *ao)
{
return ao->api->get_eof ? ao->api->get_eof(ao) : true;
}
// Query the AO_EVENT_*s as requested by the events parameter, and return them.
int ao_query_and_reset_events(struct ao *ao, int events)
{
return atomic_fetch_and(&ao->events_, ~(unsigned)events) & events;
}
static void ao_add_events(struct ao *ao, int events)
{
atomic_fetch_or(&ao->events_, events);
ao->wakeup_cb(ao->wakeup_ctx);
}
// Request that the player core destroys and recreates the AO. Fully thread-safe.
void ao_request_reload(struct ao *ao)
{
ao_add_events(ao, AO_EVENT_RELOAD);
}
// Notify the player that the device list changed. Fully thread-safe.
void ao_hotplug_event(struct ao *ao)
{
ao_add_events(ao, AO_EVENT_HOTPLUG);
}
bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map)
{
MP_VERBOSE(ao, "Channel layouts:\n");
mp_chmal_sel_log(s, ao->log, MSGL_V);
bool r = mp_chmap_sel_adjust(s, map);
if (r)
MP_VERBOSE(ao, "result: %s\n", mp_chmap_to_str(map));
return r;
}
// safe_multichannel=true behaves like ao_chmap_sel_adjust.
// safe_multichannel=false is a helper for callers which do not support safe
// handling of arbitrary channel layouts. If the multichannel layouts are not
// considered "always safe" (e.g. HDMI), then allow only stereo or mono, if
// they are part of the list in *s.
bool ao_chmap_sel_adjust2(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map, bool safe_multichannel)
{
if (!safe_multichannel && (ao->init_flags & AO_INIT_SAFE_MULTICHANNEL_ONLY)) {
struct mp_chmap res = *map;
if (mp_chmap_sel_adjust(s, &res)) {
if (!mp_chmap_equals(&res, &(struct mp_chmap)MP_CHMAP_INIT_MONO) &&
!mp_chmap_equals(&res, &(struct mp_chmap)MP_CHMAP_INIT_STEREO))
{
MP_VERBOSE(ao, "Disabling multichannel output.\n");
*map = (struct mp_chmap)MP_CHMAP_INIT_STEREO;
}
}
}
return ao_chmap_sel_adjust(ao, s, map);
}
bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map, int num)
{
return mp_chmap_sel_get_def(s, map, num);
}
// --- The following functions just return immutable information.
void ao_get_format(struct ao *ao,
int *samplerate, int *format, struct mp_chmap *channels)
{
*samplerate = ao->samplerate;
*format = ao->format;
*channels = ao->channels;
}
const char *ao_get_name(struct ao *ao)
{
return ao->driver->name;
}
const char *ao_get_description(struct ao *ao)
{
return ao->driver->description;
}
bool ao_untimed(struct ao *ao)
{
return ao->untimed;
}
// ---
struct ao_hotplug {
struct mpv_global *global;
void (*wakeup_cb)(void *ctx);
void *wakeup_ctx;
// A single AO instance is used to listen to hotplug events. It wouldn't
// make much sense to allow multiple AO drivers; all sane platforms have
// a single such audio API.
// This is _not_ the same AO instance as used for playing audio.
struct ao *ao;
// cached
struct ao_device_list *list;
bool needs_update;
};
struct ao_hotplug *ao_hotplug_create(struct mpv_global *global,
void (*wakeup_cb)(void *ctx),
void *wakeup_ctx)
{
struct ao_hotplug *hp = talloc_ptrtype(NULL, hp);
*hp = (struct ao_hotplug){
.global = global,
.wakeup_cb = wakeup_cb,
.wakeup_ctx = wakeup_ctx,
.needs_update = true,
};
return hp;
}
static void get_devices(struct ao *ao, struct ao_device_list *list)
{
if (ao->driver->list_devs) {
ao->driver->list_devs(ao, list);
} else {
ao_device_list_add(list, ao, &(struct ao_device_desc){"", ""});
}
}
bool ao_hotplug_check_update(struct ao_hotplug *hp)
{
if (hp->ao && ao_query_and_reset_events(hp->ao, AO_EVENT_HOTPLUG)) {
hp->needs_update = true;
return true;
}
return false;
}
// The return value is valid until the next call to this API.
struct ao_device_list *ao_hotplug_get_device_list(struct ao_hotplug *hp)
{
if (hp->list && !hp->needs_update)
return hp->list;
talloc_free(hp->list);
struct ao_device_list *list = talloc_zero(hp, struct ao_device_list);
hp->list = list;
MP_TARRAY_APPEND(list, list->devices, list->num_devices,
(struct ao_device_desc){"auto", "Autoselect device"});
for (int n = 0; audio_out_drivers[n]; n++) {
const struct ao_driver *d = audio_out_drivers[n];
if (d == &audio_out_null)
break; // don't add unsafe/special entries
struct ao *ao = ao_alloc(true, hp->global, hp->wakeup_cb, hp->wakeup_ctx,
(char *)d->name);
if (!ao)
continue;
if (ao->driver->hotplug_init) {
if (!hp->ao && ao->driver->hotplug_init(ao) >= 0)
hp->ao = ao; // keep this one
if (hp->ao && hp->ao->driver == d)
get_devices(hp->ao, list);
} else {
get_devices(ao, list);
}
if (ao != hp->ao)
talloc_free(ao);
}
hp->needs_update = false;
return list;
}
void ao_device_list_add(struct ao_device_list *list, struct ao *ao,
struct ao_device_desc *e)
{
struct ao_device_desc c = *e;
const char *dname = ao->driver->name;
char buf[80];
if (!c.desc || !c.desc[0]) {
if (c.name && c.name[0]) {
c.desc = c.name;
} else if (list->num_devices) {
// Assume this is the default device.
snprintf(buf, sizeof(buf), "Default (%s)", dname);
c.desc = buf;
} else {
// First default device (and maybe the only one).
c.desc = "Default";
}
}
c.name = (c.name && c.name[0]) ? talloc_asprintf(list, "%s/%s", dname, c.name)
: talloc_strdup(list, dname);
c.desc = talloc_strdup(list, c.desc);
MP_TARRAY_APPEND(list, list->devices, list->num_devices, c);
}
void ao_hotplug_destroy(struct ao_hotplug *hp)
{
if (!hp)
return;
if (hp->ao && hp->ao->driver->hotplug_uninit)
hp->ao->driver->hotplug_uninit(hp->ao);
talloc_free(hp->ao);
talloc_free(hp);
}
static void dummy_wakeup(void *ctx)
{
}
void ao_print_devices(struct mpv_global *global, struct mp_log *log)
{
struct ao_hotplug *hp = ao_hotplug_create(global, dummy_wakeup, NULL);
struct ao_device_list *list = ao_hotplug_get_device_list(hp);
mp_info(log, "List of detected audio devices:\n");
for (int n = 0; n < list->num_devices; n++) {
struct ao_device_desc *desc = &list->devices[n];
mp_info(log, " '%s' (%s)\n", desc->name, desc->desc);
}
ao_hotplug_destroy(hp);
}
void ao_set_gain(struct ao *ao, float gain)
{
atomic_store(&ao->gain, gain);
}
#define MUL_GAIN_i(d, num_samples, gain, low, center, high) \
for (int n = 0; n < (num_samples); n++) \
(d)[n] = MPCLAMP( \
((((int64_t)((d)[n]) - (center)) * (gain) + 128) >> 8) + (center), \
(low), (high))
#define MUL_GAIN_f(d, num_samples, gain) \
for (int n = 0; n < (num_samples); n++) \
(d)[n] = MPCLAMP(((d)[n]) * (gain), -1.0, 1.0)
static void process_plane(struct ao *ao, void *data, int num_samples)
{
float gain = atomic_load_explicit(&ao->gain, memory_order_relaxed);
int gi = lrint(256.0 * gain);
if (gi == 256)
return;
switch (af_fmt_from_planar(ao->format)) {
case AF_FORMAT_U8:
MUL_GAIN_i((uint8_t *)data, num_samples, gi, 0, 128, 255);
break;
case AF_FORMAT_S16:
MUL_GAIN_i((int16_t *)data, num_samples, gi, INT16_MIN, 0, INT16_MAX);
break;
case AF_FORMAT_S32:
MUL_GAIN_i((int32_t *)data, num_samples, gi, INT32_MIN, 0, INT32_MAX);
break;
case AF_FORMAT_FLOAT:
MUL_GAIN_f((float *)data, num_samples, gain);
break;
case AF_FORMAT_DOUBLE:
MUL_GAIN_f((double *)data, num_samples, gain);
break;
default:;
// all other sample formats are simply not supported
}
}
void ao_post_process_data(struct ao *ao, void **data, int num_samples)
{
bool planar = af_fmt_is_planar(ao->format);
int planes = planar ? ao->channels.num : 1;
int plane_samples = num_samples * (planar ? 1: ao->channels.num);
for (int n = 0; n < planes; n++)
process_plane(ao, data[n], plane_samples);
}
static int get_conv_type(struct ao_convert_fmt *fmt)
{
if (af_fmt_to_bytes(fmt->src_fmt) * 8 == fmt->dst_bits && !fmt->pad_msb)
return 0; // passthrough
if (fmt->src_fmt == AF_FORMAT_S32 && fmt->dst_bits == 24 && !fmt->pad_msb)
return 1; // simple 32->24 bit conversion
if (fmt->src_fmt == AF_FORMAT_S32 && fmt->dst_bits == 32 && fmt->pad_msb == 8)
return 2; // simple 32->24 bit conversion, with MSB padding
return -1; // unsupported
}
// Check whether ao_convert_inplace() can be called. As an exception, the
// planar-ness of the sample format and the number of channels is ignored.
// All other parameters must be as passed to ao_convert_inplace().
bool ao_can_convert_inplace(struct ao_convert_fmt *fmt)
{
return get_conv_type(fmt) >= 0;
}
bool ao_need_conversion(struct ao_convert_fmt *fmt)
{
return get_conv_type(fmt) != 0;
}
// The LSB is always ignored.
#if BYTE_ORDER == BIG_ENDIAN
#define SHIFT24(x) ((3-(x))*8)
#else
#define SHIFT24(x) (((x)+1)*8)
#endif
static void convert_plane(int type, void *data, int num_samples)
{
switch (type) {
case 0:
break;
case 1: /* fall through */
case 2: {
int bytes = type == 1 ? 3 : 4;
for (int s = 0; s < num_samples; s++) {
uint32_t val = *((uint32_t *)data + s);
uint8_t *ptr = (uint8_t *)data + s * bytes;
ptr[0] = val >> SHIFT24(0);
ptr[1] = val >> SHIFT24(1);
ptr[2] = val >> SHIFT24(2);
if (type == 2)
ptr[3] = 0;
}
break;
}
default:
abort();
}
}
// data[n] contains the pointer to the first sample of the n-th plane, in the
// format implied by fmt->src_fmt. src_fmt also controls whether the data is
// all in one plane, or if there is a plane per channel.
void ao_convert_inplace(struct ao_convert_fmt *fmt, void **data, int num_samples)
{
int type = get_conv_type(fmt);
bool planar = af_fmt_is_planar(fmt->src_fmt);
int planes = planar ? fmt->channels : 1;
int plane_samples = num_samples * (planar ? 1: fmt->channels);
for (int n = 0; n < planes; n++)
convert_plane(type, data[n], plane_samples);
}
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