mirror of
https://github.com/mpv-player/mpv
synced 2024-12-24 07:42:17 +00:00
685 lines
20 KiB
C
685 lines
20 KiB
C
/*
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* CoreAudio audio output driver for Mac OS X
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*
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* original copyright (C) Timothy J. Wood - Aug 2000
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* ported to MPlayer libao2 by Dan Christiansen
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*
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* Chris Roccati
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* Stefano Pigozzi
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*
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* The S/PDIF part of the code is based on the auhal audio output
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* module from VideoLAN:
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* Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
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*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* along with MPlayer; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/*
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* The MacOS X CoreAudio framework doesn't mesh as simply as some
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* simpler frameworks do. This is due to the fact that CoreAudio pulls
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* audio samples rather than having them pushed at it (which is nice
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* when you are wanting to do good buffering of audio).
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*/
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#include "config.h"
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#include "ao.h"
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#include "internal.h"
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#include "audio/format.h"
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#include "osdep/timer.h"
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#include "options/m_option.h"
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#include "misc/ring.h"
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#include "common/msg.h"
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#include "audio/out/ao_coreaudio_properties.h"
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#include "audio/out/ao_coreaudio_utils.h"
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static void audio_pause(struct ao *ao);
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static void audio_resume(struct ao *ao);
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static void reset(struct ao *ao);
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static bool ca_format_is_digital(AudioStreamBasicDescription asbd)
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{
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switch (asbd.mFormatID)
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case 'IAC3':
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case 'iac3':
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case kAudioFormat60958AC3:
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case kAudioFormatAC3:
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return true;
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return false;
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}
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static bool ca_stream_supports_digital(struct ao *ao, AudioStreamID stream)
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{
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AudioStreamRangedDescription *formats = NULL;
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size_t n_formats;
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OSStatus err =
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CA_GET_ARY(stream, kAudioStreamPropertyAvailablePhysicalFormats,
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&formats, &n_formats);
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CHECK_CA_ERROR("Could not get number of stream formats.");
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for (int i = 0; i < n_formats; i++) {
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AudioStreamBasicDescription asbd = formats[i].mFormat;
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ca_print_asbd(ao, "supported format:", &(asbd));
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if (ca_format_is_digital(asbd)) {
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talloc_free(formats);
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return true;
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}
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}
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talloc_free(formats);
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coreaudio_error:
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return false;
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}
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static bool ca_device_supports_digital(struct ao *ao, AudioDeviceID device)
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{
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AudioStreamID *streams = NULL;
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size_t n_streams;
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/* Retrieve all the output streams. */
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OSStatus err =
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CA_GET_ARY_O(device, kAudioDevicePropertyStreams, &streams, &n_streams);
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CHECK_CA_ERROR("could not get number of streams.");
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for (int i = 0; i < n_streams; i++) {
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if (ca_stream_supports_digital(ao, streams[i])) {
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talloc_free(streams);
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return true;
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}
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}
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talloc_free(streams);
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coreaudio_error:
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return false;
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}
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static OSStatus ca_property_listener(
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AudioObjectPropertySelector selector,
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AudioObjectID object, uint32_t n_addresses,
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const AudioObjectPropertyAddress addresses[],
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void *data)
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{
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void *talloc_ctx = talloc_new(NULL);
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for (int i = 0; i < n_addresses; i++) {
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if (addresses[i].mSelector == selector) {
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if (data) *(volatile int *)data = 1;
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break;
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}
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}
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talloc_free(talloc_ctx);
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return noErr;
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}
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static OSStatus ca_stream_listener(
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AudioObjectID object, uint32_t n_addresses,
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const AudioObjectPropertyAddress addresses[],
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void *data)
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{
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return ca_property_listener(kAudioStreamPropertyPhysicalFormat,
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object, n_addresses, addresses, data);
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}
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static OSStatus ca_device_listener(
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AudioObjectID object, uint32_t n_addresses,
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const AudioObjectPropertyAddress addresses[],
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void *data)
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{
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return ca_property_listener(kAudioDevicePropertyDeviceHasChanged,
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object, n_addresses, addresses, data);
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}
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static OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid) {
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*pid = getpid();
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OSStatus err = CA_SET(device, kAudioDevicePropertyHogMode, pid);
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if (err != noErr)
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*pid = -1;
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return err;
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}
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static OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid) {
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if (*pid == getpid()) {
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*pid = -1;
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return CA_SET(device, kAudioDevicePropertyHogMode, &pid);
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}
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return noErr;
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}
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static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
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uint32_t val, bool *changed) {
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*changed = false;
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AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
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.mSelector = kAudioDevicePropertySupportsMixing,
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.mScope = kAudioObjectPropertyScopeGlobal,
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.mElement = kAudioObjectPropertyElementMaster,
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};
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if (AudioObjectHasProperty(device, &p_addr)) {
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OSStatus err;
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Boolean writeable = 0;
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err = CA_SETTABLE(device, kAudioDevicePropertySupportsMixing,
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&writeable);
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if (!CHECK_CA_WARN("can't tell if mixing property is settable")) {
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return err;
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}
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if (!writeable)
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return noErr;
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err = CA_SET(device, kAudioDevicePropertySupportsMixing, &val);
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if (err != noErr)
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return err;
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if (!CHECK_CA_WARN("can't set mix mode")) {
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return err;
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}
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*changed = true;
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}
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return noErr;
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}
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static OSStatus ca_disable_mixing(struct ao *ao,
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AudioDeviceID device, bool *changed) {
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return ca_change_mixing(ao, device, 0, changed);
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}
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static OSStatus ca_enable_mixing(struct ao *ao,
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AudioDeviceID device, bool changed) {
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if (changed) {
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bool dont_care = false;
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return ca_change_mixing(ao, device, 1, &dont_care);
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}
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return noErr;
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}
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static OSStatus ca_change_device_listening(AudioDeviceID device,
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void *flag, bool enabled)
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{
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AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
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.mSelector = kAudioDevicePropertyDeviceHasChanged,
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.mScope = kAudioObjectPropertyScopeGlobal,
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.mElement = kAudioObjectPropertyElementMaster,
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};
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if (enabled) {
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return AudioObjectAddPropertyListener(
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device, &p_addr, ca_device_listener, flag);
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} else {
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return AudioObjectRemovePropertyListener(
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device, &p_addr, ca_device_listener, flag);
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}
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}
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static OSStatus ca_enable_device_listener(AudioDeviceID device, void *flag) {
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return ca_change_device_listening(device, flag, true);
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}
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static OSStatus ca_disable_device_listener(AudioDeviceID device, void *flag) {
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return ca_change_device_listening(device, flag, false);
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}
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static bool ca_change_format(struct ao *ao, AudioStreamID stream,
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AudioStreamBasicDescription change_format)
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{
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OSStatus err = noErr;
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AudioObjectPropertyAddress p_addr;
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volatile int stream_format_changed = 0;
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ca_print_asbd(ao, "setting stream format:", &change_format);
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/* Install the callback. */
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p_addr = (AudioObjectPropertyAddress) {
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.mSelector = kAudioStreamPropertyPhysicalFormat,
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.mScope = kAudioObjectPropertyScopeGlobal,
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.mElement = kAudioObjectPropertyElementMaster,
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};
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err = AudioObjectAddPropertyListener(stream, &p_addr, ca_stream_listener,
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(void *)&stream_format_changed);
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if (!CHECK_CA_WARN("can't add property listener during format change")) {
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return false;
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}
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/* Change the format. */
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err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &change_format);
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if (!CHECK_CA_WARN("error changing physical format")) {
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return false;
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}
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/* The AudioStreamSetProperty is not only asynchronious,
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* it is also not Atomic, in its behaviour.
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* Therefore we check 5 times before we really give up. */
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bool format_set = false;
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for (int i = 0; !format_set && i < 5; i++) {
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for (int j = 0; !stream_format_changed && j < 50; j++)
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mp_sleep_us(10000);
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if (stream_format_changed) {
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stream_format_changed = 0;
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} else {
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MP_VERBOSE(ao, "reached timeout\n");
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}
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AudioStreamBasicDescription actual_format;
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err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);
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ca_print_asbd(ao, "actual format in use:", &actual_format);
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if (actual_format.mSampleRate == change_format.mSampleRate &&
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actual_format.mFormatID == change_format.mFormatID &&
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actual_format.mFramesPerPacket == change_format.mFramesPerPacket) {
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format_set = true;
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}
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}
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err = AudioObjectRemovePropertyListener(stream, &p_addr, ca_stream_listener,
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(void *)&stream_format_changed);
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if (!CHECK_CA_WARN("can't remove property listener")) {
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return false;
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}
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return format_set;
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}
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struct priv {
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AudioDeviceID device; // selected device
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bool paused;
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struct mp_ring *buffer;
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// digital render callback
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AudioDeviceIOProcID render_cb;
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// pid set for hog mode, (-1) means that hog mode on the device was
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// released. hog mode is exclusive access to a device
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pid_t hog_pid;
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// stream selected for digital playback by the detection in init
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AudioStreamID stream;
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// stream index in an AudioBufferList
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int stream_idx;
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// format we changed the stream to: for the digital case each application
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// sets the stream format for a device to what it needs
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AudioStreamBasicDescription stream_asbd;
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AudioStreamBasicDescription original_asbd;
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bool changed_mixing;
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int stream_asbd_changed;
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bool muted;
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// options
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int opt_device_id;
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int opt_list;
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};
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static int get_ring_size(struct ao *ao)
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{
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return af_fmt_seconds_to_bytes(
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ao->format, 0.5, ao->channels.num, ao->samplerate);
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}
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static OSStatus render_cb_digital(
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AudioDeviceID device, const AudioTimeStamp *ts,
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const void *in_data, const AudioTimeStamp *in_ts,
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AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx)
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{
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struct ao *ao = ctx;
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struct priv *p = ao->priv;
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AudioBuffer buf = out_data->mBuffers[p->stream_idx];
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int requested = buf.mDataByteSize;
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if (p->muted)
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mp_ring_drain(p->buffer, requested);
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else
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mp_ring_read(p->buffer, buf.mData, requested);
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return noErr;
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}
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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struct priv *p = ao->priv;
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ao_control_vol_t *control_vol;
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switch (cmd) {
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case AOCONTROL_GET_VOLUME:
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control_vol = (ao_control_vol_t *)arg;
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// Digital output has no volume adjust.
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int digitalvol = p->muted ? 0 : 100;
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*control_vol = (ao_control_vol_t) {
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.left = digitalvol, .right = digitalvol,
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};
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return CONTROL_TRUE;
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case AOCONTROL_SET_VOLUME:
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control_vol = (ao_control_vol_t *)arg;
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// Digital output can not set volume. Here we have to return true
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// to make mixer forget it. Else mixer will add a soft filter,
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// that's not we expected and the filter not support ac3 stream
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// will cause mplayer die.
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// Although not support set volume, but at least we support mute.
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// MPlayer set mute by set volume to zero, we handle it.
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if (control_vol->left == 0 && control_vol->right == 0)
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p->muted = true;
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else
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p->muted = false;
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return CONTROL_TRUE;
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} // end switch
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return CONTROL_UNKNOWN;
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}
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static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd);
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static int init(struct ao *ao)
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{
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struct priv *p = ao->priv;
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if (p->opt_list) ca_print_device_list(ao);
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*p = (struct priv) {
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.muted = false,
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.stream_asbd_changed = 0,
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.hog_pid = -1,
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.stream = 0,
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.stream_idx = -1,
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.changed_mixing = false,
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};
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OSStatus err = ca_select_device(ao, p->opt_device_id, &p->device);
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CHECK_CA_ERROR("failed to select device");
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ao->format = af_fmt_from_planar(ao->format);
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bool supports_digital = false;
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/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
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if (AF_FORMAT_IS_AC3(ao->format)) {
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if (ca_device_supports_digital(ao, p->device))
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supports_digital = true;
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}
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if (!supports_digital) {
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MP_ERR(ao, "selected device doesn't support digital formats\n");
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goto coreaudio_error;
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} // closes if (!supports_digital)
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// Build ASBD for the input format
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AudioStreamBasicDescription asbd;
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ca_fill_asbd(ao, &asbd);
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return init_digital(ao, asbd);
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coreaudio_error:
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return CONTROL_ERROR;
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}
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static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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{
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struct priv *p = ao->priv;
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OSStatus err = noErr;
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uint32_t is_alive = 1;
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err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
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CHECK_CA_WARN("could not check whether device is alive");
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if (!is_alive)
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MP_WARN(ao , "device is not alive\n");
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err = ca_lock_device(p->device, &p->hog_pid);
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CHECK_CA_WARN("failed to set hogmode");
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err = ca_disable_mixing(ao, p->device, &p->changed_mixing);
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CHECK_CA_WARN("failed to disable mixing");
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AudioStreamID *streams;
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size_t n_streams;
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/* Get a list of all the streams on this device. */
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err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
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&streams, &n_streams);
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CHECK_CA_ERROR("could not get number of streams");
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for (int i = 0; i < n_streams && p->stream_idx < 0; i++) {
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bool digital = ca_stream_supports_digital(ao, streams[i]);
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if (digital) {
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err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
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&p->original_asbd);
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if (!CHECK_CA_WARN("could not get stream's physical format to "
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"revert to, getting the next one"))
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continue;
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AudioStreamRangedDescription *formats;
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size_t n_formats;
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err = CA_GET_ARY(streams[i],
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kAudioStreamPropertyAvailablePhysicalFormats,
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&formats, &n_formats);
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if (!CHECK_CA_WARN("could not get number of stream formats"))
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continue; // try next one
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int req_rate_format = -1;
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int max_rate_format = -1;
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p->stream = streams[i];
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p->stream_idx = i;
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for (int j = 0; j < n_formats; j++)
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if (ca_format_is_digital(formats[j].mFormat)) {
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// select the digital format that has exactly the same
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// samplerate. If an exact match cannot be found, select
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// the format with highest samplerate as backup.
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if (formats[j].mFormat.mSampleRate == asbd.mSampleRate) {
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req_rate_format = j;
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break;
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} else if (max_rate_format < 0 ||
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formats[j].mFormat.mSampleRate >
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formats[max_rate_format].mFormat.mSampleRate)
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max_rate_format = j;
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}
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if (req_rate_format >= 0)
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p->stream_asbd = formats[req_rate_format].mFormat;
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else
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p->stream_asbd = formats[max_rate_format].mFormat;
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talloc_free(formats);
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}
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}
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talloc_free(streams);
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if (p->stream_idx < 0) {
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MP_WARN(ao , "can't find any digital output stream format\n");
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goto coreaudio_error;
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}
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if (!ca_change_format(ao, p->stream, p->stream_asbd))
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goto coreaudio_error;
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void *changed = (void *) &(p->stream_asbd_changed);
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err = ca_enable_device_listener(p->device, changed);
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CHECK_CA_ERROR("cannot install format change listener during init");
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#if BYTE_ORDER == BIG_ENDIAN
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if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian))
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#else
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/* 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),
|
|
.options = (const struct m_option[]) {
|
|
OPT_INT("device_id", opt_device_id, 0, OPTDEF_INT(-1)),
|
|
OPT_FLAG("list", opt_list, 0),
|
|
{0}
|
|
},
|
|
};
|