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mpv/audio/out/ao_coreaudio_device.c

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/*
* CoreAudio audio output driver for Mac OS X
*
* original copyright (C) Timothy J. Wood - Aug 2000
* ported to MPlayer libao2 by Dan Christiansen
*
* Chris Roccati
* Stefano Pigozzi
*
* The S/PDIF part of the code is based on the auhal audio output
* module from VideoLAN:
* Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
*
* This file is part of MPlayer.
*
* MPlayer 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.
*
* MPlayer 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
* along with MPlayer; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* The MacOS X CoreAudio framework doesn't mesh as simply as some
* simpler frameworks do. This is due to the fact that CoreAudio pulls
* audio samples rather than having them pushed at it (which is nice
* when you are wanting to do good buffering of audio).
*/
#include "config.h"
#include "ao.h"
#include "internal.h"
#include "audio/format.h"
#include "osdep/timer.h"
#include "options/m_option.h"
#include "misc/ring.h"
#include "common/msg.h"
#include "audio/out/ao_coreaudio_properties.h"
#include "audio/out/ao_coreaudio_utils.h"
static void audio_pause(struct ao *ao);
static void audio_resume(struct ao *ao);
static void reset(struct ao *ao);
static bool ca_format_is_digital(AudioStreamBasicDescription asbd)
{
switch (asbd.mFormatID)
case 'IAC3':
case 'iac3':
case kAudioFormat60958AC3:
case kAudioFormatAC3:
return true;
return false;
}
static bool ca_stream_supports_digital(struct ao *ao, AudioStreamID stream)
{
AudioStreamRangedDescription *formats = NULL;
size_t n_formats;
OSStatus err =
CA_GET_ARY(stream, kAudioStreamPropertyAvailablePhysicalFormats,
&formats, &n_formats);
CHECK_CA_ERROR("Could not get number of stream formats.");
for (int i = 0; i < n_formats; i++) {
AudioStreamBasicDescription asbd = formats[i].mFormat;
ca_print_asbd(ao, "supported format:", &(asbd));
if (ca_format_is_digital(asbd)) {
talloc_free(formats);
return true;
}
}
talloc_free(formats);
coreaudio_error:
return false;
}
static bool ca_device_supports_digital(struct ao *ao, AudioDeviceID device)
{
AudioStreamID *streams = NULL;
size_t n_streams;
/* Retrieve all the output streams. */
OSStatus err =
CA_GET_ARY_O(device, kAudioDevicePropertyStreams, &streams, &n_streams);
CHECK_CA_ERROR("could not get number of streams.");
for (int i = 0; i < n_streams; i++) {
if (ca_stream_supports_digital(ao, streams[i])) {
talloc_free(streams);
return true;
}
}
talloc_free(streams);
coreaudio_error:
return false;
}
static OSStatus ca_property_listener(
AudioObjectPropertySelector selector,
AudioObjectID object, uint32_t n_addresses,
const AudioObjectPropertyAddress addresses[],
void *data)
{
void *talloc_ctx = talloc_new(NULL);
for (int i = 0; i < n_addresses; i++) {
if (addresses[i].mSelector == selector) {
if (data) *(volatile int *)data = 1;
break;
}
}
talloc_free(talloc_ctx);
return noErr;
}
static OSStatus ca_stream_listener(
AudioObjectID object, uint32_t n_addresses,
const AudioObjectPropertyAddress addresses[],
void *data)
{
return ca_property_listener(kAudioStreamPropertyPhysicalFormat,
object, n_addresses, addresses, data);
}
static OSStatus ca_device_listener(
AudioObjectID object, uint32_t n_addresses,
const AudioObjectPropertyAddress addresses[],
void *data)
{
return ca_property_listener(kAudioDevicePropertyDeviceHasChanged,
object, n_addresses, addresses, data);
}
static OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid) {
*pid = getpid();
OSStatus err = CA_SET(device, kAudioDevicePropertyHogMode, pid);
if (err != noErr)
*pid = -1;
return err;
}
static OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid) {
if (*pid == getpid()) {
*pid = -1;
return CA_SET(device, kAudioDevicePropertyHogMode, &pid);
}
return noErr;
}
static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
uint32_t val, bool *changed) {
*changed = false;
AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
.mSelector = kAudioDevicePropertySupportsMixing,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster,
};
if (AudioObjectHasProperty(device, &p_addr)) {
OSStatus err;
Boolean writeable = 0;
err = CA_SETTABLE(device, kAudioDevicePropertySupportsMixing,
&writeable);
if (!CHECK_CA_WARN("can't tell if mixing property is settable")) {
return err;
}
if (!writeable)
return noErr;
err = CA_SET(device, kAudioDevicePropertySupportsMixing, &val);
if (err != noErr)
return err;
if (!CHECK_CA_WARN("can't set mix mode")) {
return err;
}
*changed = true;
}
return noErr;
}
static OSStatus ca_disable_mixing(struct ao *ao,
AudioDeviceID device, bool *changed) {
return ca_change_mixing(ao, device, 0, changed);
}
static OSStatus ca_enable_mixing(struct ao *ao,
AudioDeviceID device, bool changed) {
if (changed) {
bool dont_care = false;
return ca_change_mixing(ao, device, 1, &dont_care);
}
return noErr;
}
static OSStatus ca_change_device_listening(AudioDeviceID device,
void *flag, bool enabled)
{
AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
.mSelector = kAudioDevicePropertyDeviceHasChanged,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster,
};
if (enabled) {
return AudioObjectAddPropertyListener(
device, &p_addr, ca_device_listener, flag);
} else {
return AudioObjectRemovePropertyListener(
device, &p_addr, ca_device_listener, flag);
}
}
static OSStatus ca_enable_device_listener(AudioDeviceID device, void *flag) {
return ca_change_device_listening(device, flag, true);
}
static OSStatus ca_disable_device_listener(AudioDeviceID device, void *flag) {
return ca_change_device_listening(device, flag, false);
}
static bool ca_change_format(struct ao *ao, AudioStreamID stream,
AudioStreamBasicDescription change_format)
{
OSStatus err = noErr;
AudioObjectPropertyAddress p_addr;
volatile int stream_format_changed = 0;
ca_print_asbd(ao, "setting stream format:", &change_format);
/* Install the callback. */
p_addr = (AudioObjectPropertyAddress) {
.mSelector = kAudioStreamPropertyPhysicalFormat,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster,
};
err = AudioObjectAddPropertyListener(stream, &p_addr, ca_stream_listener,
(void *)&stream_format_changed);
if (!CHECK_CA_WARN("can't add property listener during format change")) {
return false;
}
/* Change the format. */
err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &change_format);
if (!CHECK_CA_WARN("error changing physical format")) {
return false;
}
/* The AudioStreamSetProperty is not only asynchronious,
* it is also not Atomic, in its behaviour.
* Therefore we check 5 times before we really give up. */
bool format_set = false;
for (int i = 0; !format_set && i < 5; i++) {
for (int j = 0; !stream_format_changed && j < 50; j++)
mp_sleep_us(10000);
if (stream_format_changed) {
stream_format_changed = 0;
} else {
MP_VERBOSE(ao, "reached timeout\n");
}
AudioStreamBasicDescription actual_format;
err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);
ca_print_asbd(ao, "actual format in use:", &actual_format);
if (actual_format.mSampleRate == change_format.mSampleRate &&
actual_format.mFormatID == change_format.mFormatID &&
actual_format.mFramesPerPacket == change_format.mFramesPerPacket) {
format_set = true;
}
}
err = AudioObjectRemovePropertyListener(stream, &p_addr, ca_stream_listener,
(void *)&stream_format_changed);
if (!CHECK_CA_WARN("can't remove property listener")) {
return false;
}
return format_set;
}
struct priv {
AudioDeviceID device; // selected device
bool paused;
struct mp_ring *buffer;
// digital render callback
AudioDeviceIOProcID render_cb;
// pid set for hog mode, (-1) means that hog mode on the device was
// released. hog mode is exclusive access to a device
pid_t hog_pid;
// stream selected for digital playback by the detection in init
AudioStreamID stream;
// stream index in an AudioBufferList
int stream_idx;
// format we changed the stream to: for the digital case each application
// sets the stream format for a device to what it needs
AudioStreamBasicDescription stream_asbd;
AudioStreamBasicDescription original_asbd;
bool changed_mixing;
int stream_asbd_changed;
bool muted;
// options
int opt_device_id;
int opt_list;
};
static int get_ring_size(struct ao *ao)
{
return af_fmt_seconds_to_bytes(
ao->format, 0.5, ao->channels.num, ao->samplerate);
}
static OSStatus render_cb_digital(
AudioDeviceID device, const AudioTimeStamp *ts,
const void *in_data, const AudioTimeStamp *in_ts,
AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx)
{
struct ao *ao = ctx;
struct priv *p = ao->priv;
AudioBuffer buf = out_data->mBuffers[p->stream_idx];
int requested = buf.mDataByteSize;
if (p->muted)
mp_ring_drain(p->buffer, requested);
else
mp_ring_read(p->buffer, buf.mData, requested);
return noErr;
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct priv *p = ao->priv;
ao_control_vol_t *control_vol;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
control_vol = (ao_control_vol_t *)arg;
// Digital output has no volume adjust.
int digitalvol = p->muted ? 0 : 100;
*control_vol = (ao_control_vol_t) {
.left = digitalvol, .right = digitalvol,
};
return CONTROL_TRUE;
case AOCONTROL_SET_VOLUME:
control_vol = (ao_control_vol_t *)arg;
// Digital output can not set volume. Here we have to return true
// to make mixer forget it. Else mixer will add a soft filter,
// that's not we expected and the filter not support ac3 stream
// will cause mplayer die.
// Although not support set volume, but at least we support mute.
// MPlayer set mute by set volume to zero, we handle it.
if (control_vol->left == 0 && control_vol->right == 0)
p->muted = true;
else
p->muted = false;
return CONTROL_TRUE;
} // end switch
return CONTROL_UNKNOWN;
}
static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd);
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
if (p->opt_list) ca_print_device_list(ao);
OSStatus err = ca_select_device(ao, p->opt_device_id, &p->device);
CHECK_CA_ERROR("failed to select device");
ao->format = af_fmt_from_planar(ao->format);
bool supports_digital = false;
/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
if (AF_FORMAT_IS_AC3(ao->format)) {
if (ca_device_supports_digital(ao, p->device))
supports_digital = true;
}
if (!supports_digital) {
MP_ERR(ao, "selected device doesn't support digital formats\n");
goto coreaudio_error;
} // closes if (!supports_digital)
// Build ASBD for the input format
AudioStreamBasicDescription asbd;
ca_fill_asbd(ao, &asbd);
return init_digital(ao, asbd);
coreaudio_error:
return CONTROL_ERROR;
}
static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
{
struct priv *p = ao->priv;
OSStatus err = noErr;
uint32_t is_alive = 1;
err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
CHECK_CA_WARN("could not check whether device is alive");
if (!is_alive)
MP_WARN(ao , "device is not alive\n");
err = ca_lock_device(p->device, &p->hog_pid);
CHECK_CA_WARN("failed to set hogmode");
err = ca_disable_mixing(ao, p->device, &p->changed_mixing);
CHECK_CA_WARN("failed to disable mixing");
AudioStreamID *streams;
size_t n_streams;
/* Get a list of all the streams on this device. */
err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
&streams, &n_streams);
CHECK_CA_ERROR("could not get number of streams");
for (int i = 0; i < n_streams && p->stream_idx < 0; i++) {
bool digital = ca_stream_supports_digital(ao, streams[i]);
if (digital) {
err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
&p->original_asbd);
if (!CHECK_CA_WARN("could not get stream's physical format to "
"revert to, getting the next one"))
continue;
AudioStreamRangedDescription *formats;
size_t n_formats;
err = CA_GET_ARY(streams[i],
kAudioStreamPropertyAvailablePhysicalFormats,
&formats, &n_formats);
if (!CHECK_CA_WARN("could not get number of stream formats"))
continue; // try next one
int req_rate_format = -1;
int max_rate_format = -1;
p->stream = streams[i];
p->stream_idx = i;
for (int j = 0; j < n_formats; j++)
if (ca_format_is_digital(formats[j].mFormat)) {
// select the digital format that has exactly the same
// samplerate. If an exact match cannot be found, select
// the format with highest samplerate as backup.
if (formats[j].mFormat.mSampleRate == asbd.mSampleRate) {
req_rate_format = j;
break;
} else if (max_rate_format < 0 ||
formats[j].mFormat.mSampleRate >
formats[max_rate_format].mFormat.mSampleRate)
max_rate_format = j;
}
if (req_rate_format >= 0)
p->stream_asbd = formats[req_rate_format].mFormat;
else
p->stream_asbd = formats[max_rate_format].mFormat;
talloc_free(formats);
}
}
talloc_free(streams);
if (p->stream_idx < 0) {
MP_WARN(ao , "can't find any digital output stream format\n");
goto coreaudio_error;
}
if (!ca_change_format(ao, p->stream, p->stream_asbd))
goto coreaudio_error;
void *changed = (void *) &(p->stream_asbd_changed);
err = ca_enable_device_listener(p->device, changed);
CHECK_CA_ERROR("cannot install format change listener during init");
#if BYTE_ORDER == BIG_ENDIAN
if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
/* tell mplayer that we need a byteswap on AC3 streams, */
if (p->stream_asbd.mFormatID & kAudioFormat60958AC3)
ao->format = AF_FORMAT_AC3_LE;
else if (p->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
MP_WARN(ao, "stream has non-native byte order, output may fail\n");
ao->samplerate = p->stream_asbd.mSampleRate;
ao->bps = ao->samplerate *
(p->stream_asbd.mBytesPerPacket /
p->stream_asbd.mFramesPerPacket);
p->buffer = mp_ring_new(p, get_ring_size(ao));
err = AudioDeviceCreateIOProcID(p->device,
(AudioDeviceIOProc)render_cb_digital,
(void *)ao,
&p->render_cb);
CHECK_CA_ERROR("failed to register digital render callback");
reset(ao);
return CONTROL_TRUE;
coreaudio_error:
err = ca_unlock_device(p->device, &p->hog_pid);
CHECK_CA_WARN("can't release hog mode");
return CONTROL_ERROR;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *p = ao->priv;
void *output_samples = data[0];
int num_bytes = samples * ao->sstride;
// Check whether we need to reset the digital output stream.
if (p->stream_asbd_changed) {
p->stream_asbd_changed = 0;
if (ca_stream_supports_digital(ao, p->stream)) {
if (!ca_change_format(ao, p->stream, p->stream_asbd)) {
MP_WARN(ao , "can't restore digital output\n");
} else {
MP_WARN(ao, "restoring digital output succeeded.\n");
reset(ao);
}
}
}
int wrote = mp_ring_write(p->buffer, output_samples, num_bytes);
audio_resume(ao);
return wrote / ao->sstride;
}
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
audio_pause(ao);
mp_ring_reset(p->buffer);
}
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
return mp_ring_available(p->buffer) / ao->sstride;
}
static float get_delay(struct ao *ao)
{
// FIXME: should also report the delay of coreaudio itself (hardware +
// internal buffers)
struct priv *p = ao->priv;
return mp_ring_buffered(p->buffer) / (float)ao->bps;
}
static void uninit(struct ao *ao)
{
struct priv *p = ao->priv;
OSStatus err = noErr;
void *changed = (void *) &(p->stream_asbd_changed);
err = ca_disable_device_listener(p->device, changed);
CHECK_CA_WARN("can't remove device listener, this may cause a crash");
err = AudioDeviceStop(p->device, p->render_cb);
CHECK_CA_WARN("failed to stop audio device");
err = AudioDeviceDestroyIOProcID(p->device, p->render_cb);
CHECK_CA_WARN("failed to remove device render callback");
if (!ca_change_format(ao, p->stream, p->original_asbd))
MP_WARN(ao, "can't revert to original device format");
err = ca_enable_mixing(ao, p->device, p->changed_mixing);
CHECK_CA_WARN("can't re-enable mixing");
err = ca_unlock_device(p->device, &p->hog_pid);
CHECK_CA_WARN("can't release hog mode");
}
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
if (p->paused)
return;
OSStatus err = AudioDeviceStop(p->device, p->render_cb);
CHECK_CA_WARN("can't stop digital device");
p->paused = true;
}
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
if (!p->paused)
return;
OSStatus err = AudioDeviceStart(p->device, p->render_cb);
CHECK_CA_WARN("can't start digital device");
p->paused = false;
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_coreaudio_exclusive = {
.description = "CoreAudio Exclusive Mode",
.name = "coreaudio_exclusive",
.uninit = uninit,
.init = init,
.play = play,
.control = control,
.get_space = get_space,
.get_delay = get_delay,
.reset = reset,
.pause = audio_pause,
.resume = audio_resume,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv){
.muted = false,
.stream_asbd_changed = 0,
.hog_pid = -1,
.stream = 0,
.stream_idx = -1,
.changed_mixing = false,
},
.options = (const struct m_option[]) {
OPT_INT("device_id", opt_device_id, 0, OPTDEF_INT(-1)),
OPT_FLAG("list", opt_list, 0),
{0}
},
};
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