mpv/audio/out/pull.c

338 lines
11 KiB
C

/*
* 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 <stddef.h>
#include <inttypes.h>
#include <assert.h>
#include "ao.h"
#include "internal.h"
#include "audio/format.h"
#include "common/msg.h"
#include "common/common.h"
#include "input/input.h"
#include "osdep/timer.h"
#include "osdep/threads.h"
#include "osdep/atomic.h"
#include "misc/ring.h"
/*
* Note: there is some stupid stuff in this file in order to avoid mutexes.
* This requirement is dictated by several audio APIs, at least jackaudio.
*/
enum {
AO_STATE_NONE, // idle (e.g. before playback started, or after playback
// finished, but device is open)
AO_STATE_WAIT, // wait for callback to go into AO_STATE_NONE state
AO_STATE_PLAY, // play the buffer
AO_STATE_BUSY, // like AO_STATE_PLAY, but ao_read_data() is being called
};
#define IS_PLAYING(st) ((st) == AO_STATE_PLAY || (st) == AO_STATE_BUSY)
struct ao_pull_state {
// Be very careful with the order when accessing planes.
struct mp_ring *buffers[MP_NUM_CHANNELS];
// AO_STATE_*
atomic_int state;
// Set when the buffer is intentionally not fed anymore in PLAY state.
atomic_bool draining;
// Set by the audio thread when an underflow was detected.
// It adds the number of samples.
atomic_int underflow;
// Device delay of the last written sample, in realtime.
atomic_llong end_time_us;
char *convert_buffer;
};
static void set_state(struct ao *ao, int new_state)
{
struct ao_pull_state *p = ao->api_priv;
while (1) {
int old = atomic_load(&p->state);
if (old == AO_STATE_BUSY) {
// A spinlock, because some audio APIs don't want us to use mutexes.
mp_sleep_us(1);
continue;
}
if (atomic_compare_exchange_strong(&p->state, &old, new_state))
break;
}
}
static int get_space(struct ao *ao)
{
struct ao_pull_state *p = ao->api_priv;
// Since the reader will read the last plane last, its free space is the
// minimum free space across all planes.
return mp_ring_available(p->buffers[ao->num_planes - 1]) / ao->sstride;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct ao_pull_state *p = ao->api_priv;
int write_samples = get_space(ao);
write_samples = MPMIN(write_samples, samples);
// Write starting from the last plane - this way, the first plane will
// always contain the minimum amount of data readable across all planes
// (assumes the reader starts with the first plane).
int write_bytes = write_samples * ao->sstride;
for (int n = ao->num_planes - 1; n >= 0; n--) {
int r = mp_ring_write(p->buffers[n], data[n], write_bytes);
assert(r == write_bytes);
}
int state = atomic_load(&p->state);
if (!IS_PLAYING(state)) {
atomic_store(&p->draining, false);
atomic_store(&p->underflow, 0);
set_state(ao, AO_STATE_PLAY);
if (!ao->stream_silence)
ao->driver->resume(ao);
}
bool draining = write_samples == samples && (flags & AOPLAY_FINAL_CHUNK);
atomic_store(&p->draining, draining);
int underflow = atomic_fetch_and(&p->underflow, 0);
if (underflow)
MP_WARN(ao, "Audio underflow by %d samples.\n", underflow);
return write_samples;
}
// Read the given amount of samples in the user-provided data buffer. Returns
// the number of samples copied. If there is not enough data (buffer underrun
// or EOF), return the number of samples that could be copied, and fill the
// rest of the user-provided buffer with silence.
// This basically assumes that the audio device doesn't care about underruns.
// If this is called in paused mode, it will always return 0.
// The caller should set out_time_us to the expected delay until the last sample
// reaches the speakers, in microseconds, using mp_time_us() as reference.
int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us)
{
assert(ao->api == &ao_api_pull);
struct ao_pull_state *p = ao->api_priv;
int full_bytes = samples * ao->sstride;
bool need_wakeup = false;
int bytes = 0;
// Play silence in states other than AO_STATE_PLAY.
if (!atomic_compare_exchange_strong(&p->state, &(int){AO_STATE_PLAY},
AO_STATE_BUSY))
goto end;
// Since the writer will write the first plane last, its buffered amount
// of data is the minimum amount across all planes.
int buffered_bytes = mp_ring_buffered(p->buffers[0]);
bytes = MPMIN(buffered_bytes, full_bytes);
if (buffered_bytes < bytes && !atomic_load(&p->draining))
atomic_fetch_add(&p->underflow, (bytes - buffered_bytes) / ao->sstride);
if (bytes > 0)
atomic_store(&p->end_time_us, out_time_us);
for (int n = 0; n < ao->num_planes; n++) {
int r = mp_ring_read(p->buffers[n], data[n], bytes);
bytes = MPMIN(bytes, r);
}
// Half of the buffer played -> request more.
need_wakeup = buffered_bytes - bytes <= mp_ring_size(p->buffers[0]) / 2;
// Should never fail.
atomic_compare_exchange_strong(&p->state, &(int){AO_STATE_BUSY}, AO_STATE_PLAY);
end:
if (need_wakeup)
ao->wakeup_cb(ao->wakeup_ctx);
// pad with silence (underflow/paused/eof)
for (int n = 0; n < ao->num_planes; n++)
af_fill_silence((char *)data[n] + bytes, full_bytes - bytes, ao->format);
ao_post_process_data(ao, data, samples);
return bytes / ao->sstride;
}
// Same as ao_read_data(), but convert data according to *fmt.
// fmt->src_fmt and fmt->channels must be the same as the AO parameters.
int ao_read_data_converted(struct ao *ao, struct ao_convert_fmt *fmt,
void **data, int samples, int64_t out_time_us)
{
assert(ao->api == &ao_api_pull);
struct ao_pull_state *p = ao->api_priv;
void *ndata[MP_NUM_CHANNELS] = {0};
if (!ao_need_conversion(fmt))
return ao_read_data(ao, data, samples, out_time_us);
assert(ao->format == fmt->src_fmt);
assert(ao->channels.num == fmt->channels);
bool planar = af_fmt_is_planar(fmt->src_fmt);
int planes = planar ? fmt->channels : 1;
int plane_samples = samples * (planar ? 1: fmt->channels);
int src_plane_size = plane_samples * af_fmt_to_bytes(fmt->src_fmt);
int dst_plane_size = plane_samples * fmt->dst_bits / 8;
int needed = src_plane_size * planes;
if (needed > talloc_get_size(p->convert_buffer) || !p->convert_buffer) {
talloc_free(p->convert_buffer);
p->convert_buffer = talloc_size(NULL, needed);
}
for (int n = 0; n < planes; n++)
ndata[n] = p->convert_buffer + n * src_plane_size;
int res = ao_read_data(ao, ndata, samples, out_time_us);
ao_convert_inplace(fmt, ndata, samples);
for (int n = 0; n < planes; n++)
memcpy(data[n], ndata[n], dst_plane_size);
return res;
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
if (ao->driver->control)
return ao->driver->control(ao, cmd, arg);
return 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.
static double get_delay(struct ao *ao)
{
struct ao_pull_state *p = ao->api_priv;
int64_t end = atomic_load(&p->end_time_us);
int64_t now = mp_time_us();
double driver_delay = MPMAX(0, (end - now) / (1000.0 * 1000.0));
return mp_ring_buffered(p->buffers[0]) / (double)ao->bps + driver_delay;
}
static void reset(struct ao *ao)
{
struct ao_pull_state *p = ao->api_priv;
if (!ao->stream_silence && ao->driver->reset)
ao->driver->reset(ao); // assumes the audio callback thread is stopped
set_state(ao, AO_STATE_NONE);
for (int n = 0; n < ao->num_planes; n++)
mp_ring_reset(p->buffers[n]);
atomic_store(&p->end_time_us, 0);
}
static void pause(struct ao *ao)
{
if (!ao->stream_silence && ao->driver->reset)
ao->driver->reset(ao);
set_state(ao, AO_STATE_NONE);
}
static void resume(struct ao *ao)
{
set_state(ao, AO_STATE_PLAY);
if (!ao->stream_silence)
ao->driver->resume(ao);
}
static bool get_eof(struct ao *ao)
{
struct ao_pull_state *p = ao->api_priv;
// For simplicity, ignore the latency. Otherwise, we would have to run an
// extra thread to time it.
return mp_ring_buffered(p->buffers[0]) == 0;
}
static void drain(struct ao *ao)
{
struct ao_pull_state *p = ao->api_priv;
int state = atomic_load(&p->state);
if (IS_PLAYING(state)) {
atomic_store(&p->draining, true);
// Wait for lower bound.
mp_sleep_us(mp_ring_buffered(p->buffers[0]) / (double)ao->bps * 1e6);
// And then poll for actual end. (Unfortunately, this code considers
// audio APIs which do not want you to use mutexes in the audio
// callback, and an extra semaphore would require slightly more effort.)
// Limit to arbitrary ~250ms max. waiting for robustness.
int64_t max = mp_time_us() + 250000;
while (mp_time_us() < max && !get_eof(ao))
mp_sleep_us(1);
}
reset(ao);
}
static void uninit(struct ao *ao)
{
struct ao_pull_state *p = ao->api_priv;
ao->driver->uninit(ao);
talloc_free(p->convert_buffer);
}
static int init(struct ao *ao)
{
struct ao_pull_state *p = ao->api_priv;
for (int n = 0; n < ao->num_planes; n++)
p->buffers[n] = mp_ring_new(ao, ao->buffer * ao->sstride);
atomic_store(&p->state, AO_STATE_NONE);
assert(ao->driver->resume);
if (ao->stream_silence)
ao->driver->resume(ao);
return 0;
}
const struct ao_driver ao_api_pull = {
.init = init,
.control = control,
.uninit = uninit,
.drain = drain,
.reset = reset,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.get_eof = get_eof,
.pause = pause,
.resume = resume,
.priv_size = sizeof(struct ao_pull_state),
};