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-rw-r--r--audio/out/ao_alsa.c29
-rw-r--r--audio/out/ao_audiounit.m2
-rw-r--r--audio/out/ao_jack.c3
-rw-r--r--audio/out/ao_openal.c14
-rw-r--r--audio/out/ao_opensles.c98
-rw-r--r--audio/out/ao_pulse.c6
6 files changed, 80 insertions, 72 deletions
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index d8a1ec5cae..df78a67178 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -939,17 +939,32 @@ static void drain(struct ao *ao)
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
- snd_pcm_status_t *status;
- int err;
- snd_pcm_status_alloca(&status);
+ // in case of pausing or the device still being configured,
+ // just return our buffer size.
+ if (p->paused || snd_pcm_state(p->alsa) == SND_PCM_STATE_SETUP)
+ return p->buffersize;
+
+ snd_pcm_sframes_t space = snd_pcm_avail(p->alsa);
+ if (space < 0) {
+ if (space == -EPIPE) {
+ MP_WARN(ao, "ALSA XRUN hit, attempting to recover...\n");
+ int err = snd_pcm_prepare(p->alsa);
+ CHECK_ALSA_ERROR("Unable to recover from under/overrun!");
+ return p->buffersize;
+ }
+
+ MP_ERR(ao, "Error received from snd_pcm_avail "
+ "(%ld, %s with ALSA state %s)!\n",
+ space, snd_strerror(space),
+ snd_pcm_state_name(snd_pcm_state(p->alsa)));
- err = snd_pcm_status(p->alsa, status);
- if (!check_device_present(ao, err))
+ // request a reload of the AO if device is not present,
+ // then error out.
+ check_device_present(ao, space);
goto alsa_error;
- CHECK_ALSA_ERROR("cannot get pcm status");
+ }
- unsigned space = snd_pcm_status_get_avail(status);
if (space > p->buffersize) // Buffer underrun?
space = p->buffersize;
return space / p->outburst * p->outburst;
diff --git a/audio/out/ao_audiounit.m b/audio/out/ao_audiounit.m
index fb18f1eb14..70a80f29f6 100644
--- a/audio/out/ao_audiounit.m
+++ b/audio/out/ao_audiounit.m
@@ -192,7 +192,7 @@ const struct ao_driver audio_out_audiounit = {
.name = "audiounit",
.uninit = uninit,
.init = init,
- .pause = stop,
+ .reset = stop,
.resume = start,
.priv_size = sizeof(struct priv),
};
diff --git a/audio/out/ao_jack.c b/audio/out/ao_jack.c
index b5413f77cf..0d5a2da207 100644
--- a/audio/out/ao_jack.c
+++ b/audio/out/ao_jack.c
@@ -143,7 +143,8 @@ connect_to_outports(struct ao *ao)
if (!port_name)
port_flags |= JackPortIsPhysical;
- matching_ports = jack_get_ports(p->client, port_name, NULL, port_flags);
+ const char *port_type = JACK_DEFAULT_AUDIO_TYPE; // exclude MIDI ports
+ matching_ports = jack_get_ports(p->client, port_name, port_type, port_flags);
if (!matching_ports || !matching_ports[0]) {
MP_FATAL(ao, "no ports to connect to\n");
diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c
index c1f405bf12..53571b6d6e 100644
--- a/audio/out/ao_openal.c
+++ b/audio/out/ao_openal.c
@@ -24,19 +24,6 @@
#include <stdlib.h>
#include <stdio.h>
#include <inttypes.h>
-
-#ifdef __APPLE__
-#ifndef AL_FORMAT_MONO_FLOAT32
-#define AL_FORMAT_MONO_FLOAT32 0x10010
-#endif
-#ifndef AL_FORMAT_STEREO_FLOAT32
-#define AL_FORMAT_STEREO_FLOAT32 0x10011
-#endif
-#ifndef AL_FORMAT_MONO_DOUBLE_EXT
-#define AL_FORMAT_MONO_DOUBLE_EXT 0x10012
-#endif
-#include <OpenAL/MacOSX_OALExtensions.h>
-#else
#ifdef OPENAL_AL_H
#include <OpenAL/alc.h>
#include <OpenAL/al.h>
@@ -46,7 +33,6 @@
#include <AL/al.h>
#include <AL/alext.h>
#endif
-#endif // __APPLE__
#include "common/msg.h"
diff --git a/audio/out/ao_opensles.c b/audio/out/ao_opensles.c
index ea48de892e..8c24320406 100644
--- a/audio/out/ao_opensles.c
+++ b/audio/out/ao_opensles.c
@@ -35,18 +35,13 @@ struct priv {
SLBufferQueueItf buffer_queue;
SLEngineItf engine;
SLPlayItf play;
- char *buf;
- size_t buffer_size;
+ void *buf;
+ int bytes_per_enqueue;
pthread_mutex_t buffer_lock;
double audio_latency;
- int cfg_frames_per_buffer;
-};
-
-static const int fmtmap[][2] = {
- { AF_FORMAT_U8, SL_PCMSAMPLEFORMAT_FIXED_8 },
- { AF_FORMAT_S16, SL_PCMSAMPLEFORMAT_FIXED_16 },
- { 0 }
+ int frames_per_enqueue;
+ int buffer_size_in_ms;
};
#define DESTROY(thing) \
@@ -71,7 +66,6 @@ static void uninit(struct ao *ao)
free(p->buf);
p->buf = NULL;
- p->buffer_size = 0;
}
#undef DESTROY
@@ -81,26 +75,22 @@ static void buffer_callback(SLBufferQueueItf buffer_queue, void *context)
struct ao *ao = context;
struct priv *p = ao->priv;
SLresult res;
- void *data[1];
double delay;
pthread_mutex_lock(&p->buffer_lock);
- data[0] = p->buf;
- delay = 2 * p->buffer_size / (double)ao->bps;
+ delay = 2 * p->frames_per_enqueue / (double)ao->samplerate;
delay += p->audio_latency;
- ao_read_data(ao, data, p->buffer_size / ao->sstride,
+ ao_read_data(ao, &p->buf, p->frames_per_enqueue,
mp_time_us() + 1000000LL * delay);
- res = (*buffer_queue)->Enqueue(buffer_queue, p->buf, p->buffer_size);
+ res = (*buffer_queue)->Enqueue(buffer_queue, p->buf, p->bytes_per_enqueue);
if (res != SL_RESULT_SUCCESS)
MP_ERR(ao, "Failed to Enqueue: %d\n", res);
pthread_mutex_unlock(&p->buffer_lock);
}
-#define DEFAULT_BUFFER_SIZE_MS 250
-
#define CHK(stmt) \
{ \
SLresult res = stmt; \
@@ -115,7 +105,7 @@ static int init(struct ao *ao)
struct priv *p = ao->priv;
SLDataLocator_BufferQueue locator_buffer_queue;
SLDataLocator_OutputMix locator_output_mix;
- SLDataFormat_PCM pcm;
+ SLAndroidDataFormat_PCM_EX pcm;
SLDataSource audio_source;
SLDataSink audio_sink;
@@ -129,43 +119,55 @@ static int init(struct ao *ao)
CHK((*p->output_mix)->Realize(p->output_mix, SL_BOOLEAN_FALSE));
locator_buffer_queue.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
- locator_buffer_queue.numBuffers = 1;
-
- pcm.formatType = SL_DATAFORMAT_PCM;
- pcm.numChannels = 2;
-
- int compatible_formats[AF_FORMAT_COUNT + 1];
- af_get_best_sample_formats(ao->format, compatible_formats);
- pcm.bitsPerSample = 0;
- for (int i = 0; compatible_formats[i] && !pcm.bitsPerSample; ++i)
- for (int j = 0; fmtmap[j][0]; ++j)
- if (compatible_formats[i] == fmtmap[j][0]) {
- ao->format = fmtmap[j][0];
- pcm.bitsPerSample = fmtmap[j][1];
- break;
- }
- if (!pcm.bitsPerSample) {
- MP_ERR(ao, "Cannot find compatible audio format\n");
- goto error;
+ locator_buffer_queue.numBuffers = 8;
+
+ if (af_fmt_is_int(ao->format)) {
+ // Be future-proof
+ if (af_fmt_to_bytes(ao->format) > 2)
+ ao->format = AF_FORMAT_S32;
+ else
+ ao->format = af_fmt_from_planar(ao->format);
+ pcm.formatType = SL_DATAFORMAT_PCM;
+ } else {
+ ao->format = AF_FORMAT_FLOAT;
+ pcm.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
+ pcm.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
}
- pcm.containerSize = 8 * af_fmt_to_bytes(ao->format);
+ pcm.numChannels = ao->channels.num;
+ pcm.containerSize = pcm.bitsPerSample = 8 * af_fmt_to_bytes(ao->format);
pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+ pcm.sampleRate = ao->samplerate * 1000;
+
+ if (p->buffer_size_in_ms) {
+ ao->device_buffer = ao->samplerate * p->buffer_size_in_ms / 1000;
+ // As the purpose of buffer_size_in_ms is to request a specific
+ // soft buffer size:
+ ao->def_buffer = 0;
+ }
- // samplesPerSec is misnamed, actually it's samples per ms
- pcm.samplesPerSec = ao->samplerate * 1000;
+ // But it does not make sense if it is smaller than the enqueue size:
+ if (p->frames_per_enqueue) {
+ ao->device_buffer = MPMAX(ao->device_buffer, p->frames_per_enqueue);
+ } else {
+ if (ao->device_buffer) {
+ p->frames_per_enqueue = ao->device_buffer;
+ } else if (ao->def_buffer) {
+ p->frames_per_enqueue = ao->def_buffer * ao->samplerate;
+ } else {
+ MP_ERR(ao, "Enqueue size is not set and can neither be derived\n");
+ goto error;
+ }
+ }
- if (p->cfg_frames_per_buffer)
- ao->device_buffer = p->cfg_frames_per_buffer;
- else
- ao->device_buffer = ao->samplerate * DEFAULT_BUFFER_SIZE_MS / 1000;
- p->buffer_size = ao->device_buffer * ao->channels.num *
+ p->bytes_per_enqueue = p->frames_per_enqueue * ao->channels.num *
af_fmt_to_bytes(ao->format);
- p->buf = calloc(1, p->buffer_size);
+ p->buf = calloc(1, p->bytes_per_enqueue);
if (!p->buf) {
MP_ERR(ao, "Failed to allocate device buffer\n");
goto error;
}
+
int r = pthread_mutex_init(&p->buffer_lock, NULL);
if (r) {
MP_ERR(ao, "Failed to initialize the mutex: %d\n", r);
@@ -248,8 +250,12 @@ const struct ao_driver audio_out_opensles = {
.resume = resume,
.priv_size = sizeof(struct priv),
+ .priv_defaults = &(const struct priv) {
+ .buffer_size_in_ms = 250,
+ },
.options = (const struct m_option[]) {
- OPT_INTRANGE("frames-per-buffer", cfg_frames_per_buffer, 0, 1, 96000),
+ OPT_INTRANGE("frames-per-enqueue", frames_per_enqueue, 0, 1, 96000),
+ OPT_INTRANGE("buffer-size-in-ms", buffer_size_in_ms, 0, 0, 500),
{0}
},
.options_prefix = "opensles",
diff --git a/audio/out/ao_pulse.c b/audio/out/ao_pulse.c
index ed45ba6549..e288841788 100644
--- a/audio/out/ao_pulse.c
+++ b/audio/out/ao_pulse.c
@@ -454,11 +454,11 @@ static int init(struct ao *ao)
pa_stream_set_write_callback(priv->stream, stream_request_cb, ao);
pa_stream_set_latency_update_callback(priv->stream,
stream_latency_update_cb, ao);
- int buf_size = af_fmt_seconds_to_bytes(ao->format, priv->cfg_buffer / 1000.0,
- ao->channels.num, ao->samplerate);
+ uint32_t buf_size = ao->samplerate * (priv->cfg_buffer / 1000.0) *
+ af_fmt_to_bytes(ao->format) * ao->channels.num;
pa_buffer_attr bufattr = {
.maxlength = -1,
- .tlength = buf_size > 0 ? buf_size : (uint32_t)-1,
+ .tlength = buf_size > 0 ? buf_size : -1,
.prebuf = -1,
.minreq = -1,
.fragsize = -1,
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-06 03:43:24 +0000