mpv/audio/out/push.c

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/*
* This file is part of mpv.
*
Relicense some non-MPlayer source files to LGPL 2.1 or later This covers source files which were added in mplayer2 and mpv times only, and where all code is covered by LGPL relicensing agreements. There are probably more files to which this applies, but I'm being conservative here. A file named ao_sdl.c exists in MPlayer too, but the mpv one is a complete rewrite, and was added some time after the original ao_sdl.c was removed. The same applies to vo_sdl.c, for which the SDL2 API is radically different in addition (MPlayer supports SDL 1.2 only). common.c contains only code written by me. But common.h is a strange case: although it originally was named mp_common.h and exists in MPlayer too, by now it contains only definitions written by uau and me. The exceptions are the CONTROL_ defines - thus not changing the license of common.h yet. codec_tags.c contained once large tables generated from MPlayer's codecs.conf, but all of these tables were removed. From demux_playlist.c I'm removing a code fragment from someone who was not asked; this probably could be done later (see commit 15dccc37). misc.c is a bit complicated to reason about (it was split off mplayer.c and thus contains random functions out of this file), but actually all functions have been added post-MPlayer. Except get_relative_time(), which was written by uau, but looks similar to 3 different versions of something similar in each of the Unix/win32/OSX timer source files. I'm not sure what that means in regards to copyright, so I've just moved it into another still-GPL source file for now. screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but they're all gone.
2016-01-19 17:36:06 +00:00
* 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
Relicense some non-MPlayer source files to LGPL 2.1 or later This covers source files which were added in mplayer2 and mpv times only, and where all code is covered by LGPL relicensing agreements. There are probably more files to which this applies, but I'm being conservative here. A file named ao_sdl.c exists in MPlayer too, but the mpv one is a complete rewrite, and was added some time after the original ao_sdl.c was removed. The same applies to vo_sdl.c, for which the SDL2 API is radically different in addition (MPlayer supports SDL 1.2 only). common.c contains only code written by me. But common.h is a strange case: although it originally was named mp_common.h and exists in MPlayer too, by now it contains only definitions written by uau and me. The exceptions are the CONTROL_ defines - thus not changing the license of common.h yet. codec_tags.c contained once large tables generated from MPlayer's codecs.conf, but all of these tables were removed. From demux_playlist.c I'm removing a code fragment from someone who was not asked; this probably could be done later (see commit 15dccc37). misc.c is a bit complicated to reason about (it was split off mplayer.c and thus contains random functions out of this file), but actually all functions have been added post-MPlayer. Except get_relative_time(), which was written by uau, but looks similar to 3 different versions of something similar in each of the Unix/win32/OSX timer source files. I'm not sure what that means in regards to copyright, so I've just moved it into another still-GPL source file for now. screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but they're all gone.
2016-01-19 17:36:06 +00:00
* GNU Lesser General Public License for more details.
*
Relicense some non-MPlayer source files to LGPL 2.1 or later This covers source files which were added in mplayer2 and mpv times only, and where all code is covered by LGPL relicensing agreements. There are probably more files to which this applies, but I'm being conservative here. A file named ao_sdl.c exists in MPlayer too, but the mpv one is a complete rewrite, and was added some time after the original ao_sdl.c was removed. The same applies to vo_sdl.c, for which the SDL2 API is radically different in addition (MPlayer supports SDL 1.2 only). common.c contains only code written by me. But common.h is a strange case: although it originally was named mp_common.h and exists in MPlayer too, by now it contains only definitions written by uau and me. The exceptions are the CONTROL_ defines - thus not changing the license of common.h yet. codec_tags.c contained once large tables generated from MPlayer's codecs.conf, but all of these tables were removed. From demux_playlist.c I'm removing a code fragment from someone who was not asked; this probably could be done later (see commit 15dccc37). misc.c is a bit complicated to reason about (it was split off mplayer.c and thus contains random functions out of this file), but actually all functions have been added post-MPlayer. Except get_relative_time(), which was written by uau, but looks similar to 3 different versions of something similar in each of the Unix/win32/OSX timer source files. I'm not sure what that means in regards to copyright, so I've just moved it into another still-GPL source file for now. screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but they're all gone.
2016-01-19 17:36:06 +00:00
* 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 <pthread.h>
#include <inttypes.h>
#include <unistd.h>
#include <errno.h>
#include <assert.h>
#include "osdep/io.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/threads.h"
#include "osdep/timer.h"
#include "osdep/atomic.h"
#include "audio/audio_buffer.h"
struct ao_push_state {
pthread_t thread;
pthread_mutex_t lock;
pthread_cond_t wakeup;
// --- protected by lock
struct mp_audio_buffer *buffer;
uint8_t *silence[MP_NUM_CHANNELS];
int silence_samples;
bool terminate;
bool wait_on_ao;
bool still_playing;
bool need_wakeup;
bool paused;
// Whether the current buffer contains the complete audio.
bool final_chunk;
double expected_end_time;
int wakeup_pipe[2];
};
// lock must be held
static void wakeup_playthread(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
if (ao->driver->wakeup)
ao->driver->wakeup(ao);
p->need_wakeup = true;
pthread_cond_signal(&p->wakeup);
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
int r = CONTROL_UNKNOWN;
if (ao->driver->control) {
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
r = ao->driver->control(ao, cmd, arg);
pthread_mutex_unlock(&p->lock);
}
return r;
}
static double unlocked_get_delay(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
double driver_delay = 0;
if (ao->driver->get_delay)
driver_delay = ao->driver->get_delay(ao);
return driver_delay + mp_audio_buffer_seconds(p->buffer);
}
static double get_delay(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
double delay = unlocked_get_delay(ao);
pthread_mutex_unlock(&p->lock);
return delay;
}
static void reset(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
if (ao->driver->reset)
ao->driver->reset(ao);
mp_audio_buffer_clear(p->buffer);
p->paused = false;
if (p->still_playing)
wakeup_playthread(ao);
p->still_playing = false;
pthread_mutex_unlock(&p->lock);
}
static void audio_pause(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
if (ao->driver->pause)
ao->driver->pause(ao);
p->paused = true;
wakeup_playthread(ao);
pthread_mutex_unlock(&p->lock);
}
static void resume(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
if (ao->driver->resume)
ao->driver->resume(ao);
p->paused = false;
p->expected_end_time = 0;
wakeup_playthread(ao);
pthread_mutex_unlock(&p->lock);
}
static void drain(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
double maxbuffer = ao->buffer / (double)ao->samplerate + 1;
MP_VERBOSE(ao, "draining...\n");
pthread_mutex_lock(&p->lock);
if (p->paused)
goto done;
p->final_chunk = true;
wakeup_playthread(ao);
// Wait until everything is done. Since the audio API (especially ALSA)
// can't be trusted to do this right, and we're hard-blocking here, apply
// an upper bound timeout.
struct timespec until = mp_rel_time_to_timespec(maxbuffer);
while (p->still_playing && mp_audio_buffer_samples(p->buffer) > 0) {
if (pthread_cond_timedwait(&p->wakeup, &p->lock, &until)) {
MP_WARN(ao, "Draining is taking too long, aborting.\n");
goto done;
}
}
if (ao->driver->drain) {
ao->driver->drain(ao);
} else {
double time = unlocked_get_delay(ao);
mp_sleep_us(MPMIN(time, maxbuffer) * 1e6);
}
done:
pthread_mutex_unlock(&p->lock);
reset(ao);
}
static int unlocked_get_space(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
int space = mp_audio_buffer_get_write_available(p->buffer);
if (ao->driver->get_space) {
int align = af_format_sample_alignment(ao->format);
// The following code attempts to keep the total buffered audio to
// ao->buffer in order to improve latency.
int device_space = ao->driver->get_space(ao);
int device_buffered = ao->device_buffer - device_space;
int soft_buffered = mp_audio_buffer_samples(p->buffer);
// The extra margin helps avoiding too many wakeups if the AO is fully
// byte based and doesn't do proper chunked processing.
int min_buffer = ao->buffer + 64;
int missing = min_buffer - device_buffered - soft_buffered;
missing = (missing + align - 1) / align * align;
// But always keep the device's buffer filled as much as we can.
int device_missing = device_space - soft_buffered;
missing = MPMAX(missing, device_missing);
space = MPMIN(space, missing);
space = MPMAX(0, space);
}
return space;
}
static int get_space(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
int space = unlocked_get_space(ao);
pthread_mutex_unlock(&p->lock);
return space;
}
static bool get_eof(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
bool eof = !p->still_playing;
pthread_mutex_unlock(&p->lock);
return eof;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
int write_samples = mp_audio_buffer_get_write_available(p->buffer);
write_samples = MPMIN(write_samples, samples);
MP_TRACE(ao, "samples=%d flags=%d r=%d\n", samples, flags, write_samples);
if (write_samples < samples)
flags = flags & ~AOPLAY_FINAL_CHUNK;
bool is_final = flags & AOPLAY_FINAL_CHUNK;
mp_audio_buffer_append(p->buffer, data, samples);
bool got_data = write_samples > 0 || p->paused || p->final_chunk != is_final;
p->final_chunk = is_final;
p->paused = false;
if (got_data) {
p->still_playing = true;
p->expected_end_time = 0;
// If we don't have new data, the decoder thread basically promises it
// will send new data as soon as it's available.
wakeup_playthread(ao);
}
pthread_mutex_unlock(&p->lock);
return write_samples;
}
static bool realloc_silence(struct ao *ao, int samples)
{
struct ao_push_state *p = ao->api_priv;
if (samples <= 0 || !af_fmt_is_pcm(ao->format))
return false;
if (samples > p->silence_samples) {
talloc_free(p->silence[0]);
int bytes = af_fmt_to_bytes(ao->format) * samples * ao->channels.num;
p->silence[0] = talloc_size(p, bytes);
for (int n = 1; n < MP_NUM_CHANNELS; n++)
p->silence[n] = p->silence[0];
p->silence_samples = samples;
af_fill_silence(p->silence[0], bytes, ao->format);
}
return true;
}
// called locked
static void ao_play_data(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
int space = ao->driver->get_space(ao);
bool play_silence = p->paused || (ao->stream_silence && !p->still_playing);
space = MPMAX(space, 0);
if (space % ao->period_size)
MP_ERR(ao, "Audio device reports unaligned available buffer size.\n");
uint8_t **planes;
int samples;
if (play_silence) {
planes = p->silence;
samples = realloc_silence(ao, space) ? space : 0;
} else {
mp_audio_buffer_peek(p->buffer, &planes, &samples);
}
int max = samples;
if (samples > space)
samples = space;
int flags = 0;
if (p->final_chunk && samples == max) {
flags |= AOPLAY_FINAL_CHUNK;
} else {
samples = samples / ao->period_size * ao->period_size;
}
MP_STATS(ao, "start ao fill");
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
ao_post_process_data(ao, (void **)planes, samples);
int r = 0;
if (samples)
r = ao->driver->play(ao, (void **)planes, samples, flags);
MP_STATS(ao, "end ao fill");
if (r > samples) {
2017-10-23 08:53:28 +00:00
MP_ERR(ao, "Audio device returned nonsense value.\n");
r = samples;
} else if (r < 0) {
MP_ERR(ao, "Error writing audio to device.\n");
} else if (r != samples) {
MP_ERR(ao, "Audio device returned broken buffer state (sent %d samples, "
"got %d samples, %d period%s)!\n", samples, r,
ao->period_size, flags & AOPLAY_FINAL_CHUNK ? " final" : "");
}
r = MPMAX(r, 0);
// Probably can't copy the rest of the buffer due to period alignment.
bool stuck_eof = r <= 0 && space >= max && samples > 0;
if ((flags & AOPLAY_FINAL_CHUNK) && stuck_eof) {
MP_ERR(ao, "Audio output driver seems to ignore AOPLAY_FINAL_CHUNK.\n");
r = max;
}
if (!play_silence)
mp_audio_buffer_skip(p->buffer, r);
if (r > 0)
p->expected_end_time = 0;
// Nothing written, but more input data than space - this must mean the
// AO's get_space() doesn't do period alignment correctly.
bool stuck = r == 0 && max >= space && space > 0;
if (stuck)
MP_ERR(ao, "Audio output is reporting incorrect buffer status.\n");
// Wait until space becomes available. Also wait if we actually wrote data,
// so the AO wakes us up properly if it needs more data.
p->wait_on_ao = space == 0 || r > 0 || stuck;
p->still_playing |= r > 0 && !play_silence;
// If we just filled the AO completely (r == space), don't refill for a
// while. Prevents wakeup feedback with byte-granular AOs.
int needed = unlocked_get_space(ao);
bool more = needed >= (r == space ? ao->device_buffer / 4 : 1) && !stuck &&
!(flags & AOPLAY_FINAL_CHUNK);
if (more)
ao->wakeup_cb(ao->wakeup_ctx); // request more data
MP_TRACE(ao, "in=%d flags=%d space=%d r=%d wa/pl=%d/%d needed=%d more=%d\n",
max, flags, space, r, p->wait_on_ao, p->still_playing, needed, more);
}
static void *playthread(void *arg)
{
struct ao *ao = arg;
struct ao_push_state *p = ao->api_priv;
mpthread_set_name("ao");
pthread_mutex_lock(&p->lock);
while (!p->terminate) {
bool playing = !p->paused || ao->stream_silence;
if (playing)
ao_play_data(ao);
if (!p->need_wakeup) {
MP_STATS(ao, "start audio wait");
if (!p->wait_on_ao || !playing) {
// Avoid busy waiting, because the audio API will still report
// that it needs new data, even if we're not ready yet, or if
// get_space() decides that the amount of audio buffered in the
// device is enough, and p->buffer can be empty.
// The most important part is that the decoder is woken up, so
// that the decoder will wake up us in turn.
MP_TRACE(ao, "buffer inactive.\n");
bool was_playing = p->still_playing;
double timeout = -1;
if (p->still_playing && !p->paused && p->final_chunk &&
!mp_audio_buffer_samples(p->buffer))
{
double now = mp_time_sec();
if (!p->expected_end_time)
p->expected_end_time = now + unlocked_get_delay(ao);
if (p->expected_end_time < now) {
p->still_playing = false;
} else {
timeout = p->expected_end_time - now;
}
}
if (was_playing && !p->still_playing)
ao->wakeup_cb(ao->wakeup_ctx);
pthread_cond_signal(&p->wakeup); // for draining
if (p->still_playing && timeout > 0) {
struct timespec ts = mp_rel_time_to_timespec(timeout);
pthread_cond_timedwait(&p->wakeup, &p->lock, &ts);
} else {
pthread_cond_wait(&p->wakeup, &p->lock);
}
} else {
// Wait until the device wants us to write more data to it.
if (!ao->driver->wait || ao->driver->wait(ao, &p->lock) < 0) {
// Fallback to guessing.
double timeout = 0;
if (ao->driver->get_delay)
timeout = ao->driver->get_delay(ao);
timeout *= 0.25; // wake up if 25% played
if (!p->need_wakeup) {
struct timespec ts = mp_rel_time_to_timespec(timeout);
pthread_cond_timedwait(&p->wakeup, &p->lock, &ts);
}
}
}
MP_STATS(ao, "end audio wait");
}
p->need_wakeup = false;
}
pthread_mutex_unlock(&p->lock);
return NULL;
}
static void destroy_no_thread(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
ao->driver->uninit(ao);
for (int n = 0; n < 2; n++) {
int h = p->wakeup_pipe[n];
if (h >= 0)
close(h);
}
pthread_cond_destroy(&p->wakeup);
pthread_mutex_destroy(&p->lock);
}
static void uninit(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
p->terminate = true;
wakeup_playthread(ao);
pthread_mutex_unlock(&p->lock);
pthread_join(p->thread, NULL);
destroy_no_thread(ao);
}
static int init(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_init(&p->lock, NULL);
pthread_cond_init(&p->wakeup, NULL);
mp_make_wakeup_pipe(p->wakeup_pipe);
if (ao->device_buffer <= 0) {
MP_FATAL(ao, "Couldn't probe device buffer size.\n");
goto err;
}
p->buffer = mp_audio_buffer_create(ao);
mp_audio_buffer_reinit_fmt(p->buffer, ao->format,
&ao->channels, ao->samplerate);
mp_audio_buffer_preallocate_min(p->buffer, ao->buffer);
if (pthread_create(&p->thread, NULL, playthread, ao))
goto err;
return 0;
err:
destroy_no_thread(ao);
return -1;
}
const struct ao_driver ao_api_push = {
.init = init,
.control = control,
.uninit = uninit,
.reset = reset,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = audio_pause,
.resume = resume,
.drain = drain,
.get_eof = get_eof,
.priv_size = sizeof(struct ao_push_state),
};
// Must be called locked.
int ao_play_silence(struct ao *ao, int samples)
{
assert(ao->api == &ao_api_push);
struct ao_push_state *p = ao->api_priv;
if (!realloc_silence(ao, samples) || !ao->driver->play)
return 0;
return ao->driver->play(ao, (void **)p->silence, samples, 0);
}
#ifndef __MINGW32__
#include <poll.h>
#define MAX_POLL_FDS 20
// Call poll() for the given fds. This will extend the given fds with the
// wakeup pipe, so ao_wakeup_poll() will basically interrupt this function.
// Unlocks the lock temporarily.
// Returns <0 on error, 0 on success, 1 if the caller should return immediately.
int ao_wait_poll(struct ao *ao, struct pollfd *fds, int num_fds,
pthread_mutex_t *lock)
{
struct ao_push_state *p = ao->api_priv;
assert(ao->api == &ao_api_push);
assert(&p->lock == lock);
if (num_fds >= MAX_POLL_FDS || p->wakeup_pipe[0] < 0)
return -1;
struct pollfd p_fds[MAX_POLL_FDS];
memcpy(p_fds, fds, num_fds * sizeof(p_fds[0]));
p_fds[num_fds] = (struct pollfd){
.fd = p->wakeup_pipe[0],
.events = POLLIN,
};
pthread_mutex_unlock(&p->lock);
int r = poll(p_fds, num_fds + 1, -1);
r = r < 0 ? -errno : 0;
pthread_mutex_lock(&p->lock);
memcpy(fds, p_fds, num_fds * sizeof(fds[0]));
bool wakeup = false;
if (p_fds[num_fds].revents & POLLIN) {
wakeup = true;
// might "drown" some wakeups, but that's ok for our use-case
mp_flush_wakeup_pipe(p->wakeup_pipe[0]);
}
return (r >= 0 || r == -EINTR) ? wakeup : -1;
}
void ao_wakeup_poll(struct ao *ao)
{
assert(ao->api == &ao_api_push);
struct ao_push_state *p = ao->api_priv;
(void)write(p->wakeup_pipe[1], &(char){0}, 1);
}
#endif