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mpv/audio/out/ao.c
Stefano Pigozzi 899e0bd16b input: add gamepad support through SDL2 
The code is very basic:

- only handles gamepads, could be extended for generic joysticks in the
  future.
- only has button mappings for controllers natively supported by SDL2.
  I heard more can be added through env vars, there's also ways to load
  mappings from text files, but I'd rather not go there yet. Common ones
  like Dualshock are supported natively.
- analog buttons (TRIGGER and AXIS) are mapped to discrete buttons using an
  activation threshold.
- only supports one gamepad at a time. the feature is intented to use
  gamepads as evolved remote controls, not play multiplayer games in mpv :)
2019-10-23 09:40:30 +02:00

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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
&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,
&audio_out_lavc,
#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
static 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,
};
#define OPT_BASE_STRUCT struct ao_opts
const struct m_sub_options ao_conf = {
.opts = (const struct m_option[]) {
OPT_SETTINGSLIST("ao", audio_driver_list, 0, &ao_obj_list, ),
OPT_STRING("audio-device", audio_device, UPDATE_AUDIO),
OPT_STRING("audio-client-name", audio_client_name, UPDATE_AUDIO),
OPT_DOUBLE("audio-buffer", audio_buffer, M_OPT_MIN | M_OPT_MAX,
.min = 0, .max = 10),
{0}
},
.size = sizeof(OPT_BASE_STRUCT),
.defaults = &(const OPT_BASE_STRUCT){
.audio_buffer = 0.2,
.audio_device = "auto",
.audio_client_name = "mpv",
},
};
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 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_opts *opts = mp_get_config_group(NULL, global, &ao_conf);
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),
};
talloc_free(opts);
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)
{
void *tmp = talloc_new(NULL);
struct ao_opts *opts = mp_get_config_group(tmp, global, &ao_conf);
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);
atomic_fetch_and(&ao->events_, ~(unsigned int)AO_EVENT_UNDERRUN);
}
// 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;
}
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);
}
void ao_underrun_event(struct ao *ao)
{
// Racy check, but it's just for the message.
if (!(atomic_load(&ao->events_) & AO_EVENT_UNDERRUN))
MP_WARN(ao, "Device underrun detected.\n");
ao_add_events(ao, AO_EVENT_UNDERRUN);
}
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_get_reports_underruns(struct ao *ao)
{
return ao->driver->reports_underruns;
}
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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