audio/out: feed AOs from a separate thread

This has 2 goals:
- Ensure that AOs have always enough data, even if the device buffers
  are very small.
- Reduce complexity in some AOs, which do their own buffering.

One disadvantage is that performance is slightly reduced due to more
copying.

Implementation-wise, we don't change ao.c much, and instead "redirect"
the driver's callback to an API wrapper in push.c.

Additionally, we add code for dealing with AOs that have a pull API.
These AOs usually do their own buffering (jack, coreaudio, portaudio),
and adding a thread is basically a waste. The code in pull.c manages
a ringbuffer, and allows callback-based AOs to read data directly.

1 files changed, 219 insertions, 0 deletions

diff --git a/audio/out/pull.c b/audio/out/pull.c
new file mode 100644
index 0000000000..a41ac714ba
--- /dev/null
+++ b/audio/out/pull.c
@@ -0,0 +1,219 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU 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 "osdep/timer.h"
+#include "osdep/threads.h"
+#include "compat/atomics.h"
+#include "misc/ring.h"
+
+enum {
+    AO_STATE_NONE,  // idle (e.g. before playback started, or after playback
+                    // finished, but device is open)
+    AO_STATE_PLAY,  // play the buffer
+    AO_STATE_PAUSE, // pause playback
+};
+
+struct ao_pull_state {
+    // Be very careful with the order when accessing planes.
+    struct mp_ring *buffers[MP_NUM_CHANNELS];
+
+    // AO_STATE_*
+    int state;
+
+    // Whether buffers[] can be accessed.
+    int ready;
+
+    // Device delay of the last written sample, in realtime.
+    int64_t end_time_us;
+};
+
+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);
+    }
+    if (p->state != AO_STATE_PLAY) {
+        p->end_time_us = 0;
+        p->state = AO_STATE_PLAY;
+        mp_memory_barrier();
+        if (ao->driver->resume)
+            ao->driver->resume(ao);
+    }
+
+    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 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;
+
+    mp_memory_barrier();
+    if (!p->ready) {
+        for (int n = 0; n < ao->num_planes; n++)
+            af_fill_silence(data[n], full_bytes, ao->format);
+        return 0;
+    }
+
+    // Since the writer will write the first plane last, its buffered amount
+    // of data is the minimum amount across all planes.
+    int bytes = mp_ring_buffered(p->buffers[0]);
+    bytes = MPMIN(bytes, full_bytes);
+
+    if (bytes > 0)
+        p->end_time_us = out_time_us;
+
+    mp_memory_barrier();
+    if (p->state == AO_STATE_PAUSE)
+        bytes = 0;
+
+    for (int n = 0; n < ao->num_planes; n++) {
+        int r = mp_ring_read(p->buffers[n], data[n], bytes);
+        assert(r == bytes);
+        // pad with silence (underflow/paused/eof)
+        int silence = full_bytes - bytes;
+        if (silence)
+            af_fill_silence((char *)data[n] + bytes, silence, ao->format);
+    }
+    return bytes / ao->sstride;
+}
+
+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 float get_delay(struct ao *ao)
+{
+    struct ao_pull_state *p = ao->api_priv;
+
+    mp_memory_barrier();
+    int64_t end = 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->driver->reset)
+        ao->driver->reset(ao);
+    // Not like this is race-condition free. It will work if ->reset
+    // stops the audio callback, though.
+    p->ready = 0;
+    p->state = AO_STATE_NONE;
+    mp_memory_barrier();
+    for (int n = 0; n < ao->num_planes; n++)
+        mp_ring_reset(p->buffers[n]);
+    p->end_time_us = 0;
+    mp_memory_barrier();
+    p->ready = 1;
+    mp_memory_barrier();
+}
+
+static void pause(struct ao *ao)
+{
+    struct ao_pull_state *p = ao->api_priv;
+    if (ao->driver->pause)
+        ao->driver->pause(ao);
+    p->state = AO_STATE_PAUSE;
+    mp_memory_barrier();
+}
+
+static void resume(struct ao *ao)
+{
+    struct ao_pull_state *p = ao->api_priv;
+    p->state = AO_STATE_PLAY;
+    mp_memory_barrier();
+    if (ao->driver->resume)
+        ao->driver->resume(ao);
+}
+
+static void uninit(struct ao *ao, bool cut_audio)
+{
+    ao->driver->uninit(ao, cut_audio);
+}
+
+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);
+    p->ready = 1;
+    mp_memory_barrier();
+    return 0;
+}
+
+const struct ao_driver ao_api_pull = {
+    .init = init,
+    .control = control,
+    .uninit = uninit,
+    .reset = reset,
+    .get_space = get_space,
+    .play = play,
+    .get_delay = get_delay,
+    .pause = pause,
+    .resume = resume,
+    .priv_size = sizeof(struct ao_pull_state),
+};