cleanup: Silence compilation warnings on MinGW-w64
[mplayer.git] / libao2 / ao_coreaudio.c
blob34374f4c9cc31fbaf5a6caa20f22e554a1ba0965
1 /*
2 * CoreAudio audio output driver for Mac OS X
4 * original copyright (C) Timothy J. Wood - Aug 2000
5 * ported to MPlayer libao2 by Dan Christiansen
7 * The S/PDIF part of the code is based on the auhal audio output
8 * module from VideoLAN:
9 * Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
11 * This file is part of MPlayer.
13 * MPlayer is free software; you can redistribute it and/or modify
14 * it under the terms of the GNU General Public License as published by
15 * the Free Software Foundation; either version 2 of the License, or
16 * (at your option) any later version.
18 * MPlayer is distributed in the hope that it will be useful,
19 * but WITHOUT ANY WARRANTY; without even the implied warranty of
20 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
21 * GNU General Public License for more details.
23 * You should have received a copy of the GNU General Public License along
24 * along with MPlayer; if not, write to the Free Software
25 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
29 * The MacOS X CoreAudio framework doesn't mesh as simply as some
30 * simpler frameworks do. This is due to the fact that CoreAudio pulls
31 * audio samples rather than having them pushed at it (which is nice
32 * when you are wanting to do good buffering of audio).
34 * AC-3 and MPEG audio passthrough is possible, but has never been tested
35 * due to lack of a soundcard that supports it.
38 #include <CoreServices/CoreServices.h>
39 #include <AudioUnit/AudioUnit.h>
40 #include <AudioToolbox/AudioToolbox.h>
41 #include <stdio.h>
42 #include <string.h>
43 #include <stdlib.h>
44 #include <inttypes.h>
45 #include <sys/types.h>
46 #include <unistd.h>
48 #include "config.h"
49 #include "mp_msg.h"
51 #include "audio_out.h"
52 #include "audio_out_internal.h"
53 #include "libaf/af_format.h"
54 #include "osdep/timer.h"
55 #include "libavutil/fifo.h"
56 #include "subopt-helper.h"
58 static const ao_info_t info =
60 "Darwin/Mac OS X native audio output",
61 "coreaudio",
62 "Timothy J. Wood & Dan Christiansen & Chris Roccati",
66 LIBAO_EXTERN(coreaudio)
68 /* Prefix for all mp_msg() calls */
69 #define ao_msg(a, b, c...) mp_msg(a, b, "AO: [coreaudio] " c)
71 #if MAC_OS_X_VERSION_MAX_ALLOWED <= 1040
72 /* AudioDeviceIOProcID does not exist in Mac OS X 10.4. We can emulate
73 * this by using AudioDeviceAddIOProc() and AudioDeviceRemoveIOProc(). */
74 #define AudioDeviceIOProcID AudioDeviceIOProc
75 #define AudioDeviceDestroyIOProcID AudioDeviceRemoveIOProc
76 static OSStatus AudioDeviceCreateIOProcID(AudioDeviceID dev,
77 AudioDeviceIOProc proc,
78 void *data,
79 AudioDeviceIOProcID *procid)
81 *procid = proc;
82 return AudioDeviceAddIOProc(dev, proc, data);
84 #endif
86 typedef struct ao_coreaudio_s
88 AudioDeviceID i_selected_dev; /* Keeps DeviceID of the selected device. */
89 int b_supports_digital; /* Does the currently selected device support digital mode? */
90 int b_digital; /* Are we running in digital mode? */
91 int b_muted; /* Are we muted in digital mode? */
93 AudioDeviceIOProcID renderCallback; /* Render callback used for SPDIF */
95 /* AudioUnit */
96 AudioUnit theOutputUnit;
98 /* CoreAudio SPDIF mode specific */
99 pid_t i_hog_pid; /* Keeps the pid of our hog status. */
100 AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */
101 int i_stream_index; /* The index of i_stream_id in an AudioBufferList */
102 AudioStreamBasicDescription stream_format;/* The format we changed the stream to */
103 AudioStreamBasicDescription sfmt_revert; /* The original format of the stream */
104 int b_revert; /* Whether we need to revert the stream format */
105 int b_changed_mixing; /* Whether we need to set the mixing mode back */
106 int b_stream_format_changed; /* Flag for main thread to reset stream's format to digital and reset buffer */
108 /* Original common part */
109 int packetSize;
110 int paused;
112 /* Ring-buffer */
113 AVFifoBuffer *buffer;
114 unsigned int buffer_len; ///< must always be num_chunks * chunk_size
115 unsigned int num_chunks;
116 unsigned int chunk_size;
117 } ao_coreaudio_t;
119 static ao_coreaudio_t *ao = NULL;
122 * \brief add data to ringbuffer
124 static int write_buffer(unsigned char* data, int len){
125 int free = ao->buffer_len - av_fifo_size(ao->buffer);
126 if (len > free) len = free;
127 return av_fifo_generic_write(ao->buffer, data, len, NULL);
131 * \brief remove data from ringbuffer
133 static int read_buffer(unsigned char* data,int len){
134 int buffered = av_fifo_size(ao->buffer);
135 if (len > buffered) len = buffered;
136 if (data)
137 av_fifo_generic_read(ao->buffer, data, len, NULL);
138 else
139 av_fifo_drain(ao->buffer, len);
140 return len;
143 static OSStatus theRenderProc(void *inRefCon,
144 AudioUnitRenderActionFlags *inActionFlags,
145 const AudioTimeStamp *inTimeStamp,
146 UInt32 inBusNumber, UInt32 inNumFrames,
147 AudioBufferList *ioData)
149 int amt=av_fifo_size(ao->buffer);
150 int req=(inNumFrames)*ao->packetSize;
152 if(amt>req)
153 amt=req;
155 if(amt)
156 read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt);
157 else audio_pause();
158 ioData->mBuffers[0].mDataByteSize = amt;
160 return noErr;
163 static int control(int cmd,void *arg){
164 ao_control_vol_t *control_vol;
165 OSStatus err;
166 Float32 vol;
167 switch (cmd) {
168 case AOCONTROL_GET_VOLUME:
169 control_vol = (ao_control_vol_t*)arg;
170 if (ao->b_digital) {
171 // Digital output has no volume adjust.
172 return CONTROL_FALSE;
174 err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol);
176 if(err==0) {
177 // printf("GET VOL=%f\n", vol);
178 control_vol->left=control_vol->right=vol*100.0/4.0;
179 return CONTROL_TRUE;
181 else {
182 ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err);
183 return CONTROL_FALSE;
186 case AOCONTROL_SET_VOLUME:
187 control_vol = (ao_control_vol_t*)arg;
189 if (ao->b_digital) {
190 // Digital output can not set volume. Here we have to return true
191 // to make mixer forget it. Else mixer will add a soft filter,
192 // that's not we expected and the filter not support ac3 stream
193 // will cause mplayer die.
195 // Although not support set volume, but at least we support mute.
196 // MPlayer set mute by set volume to zero, we handle it.
197 if (control_vol->left == 0 && control_vol->right == 0)
198 ao->b_muted = 1;
199 else
200 ao->b_muted = 0;
201 return CONTROL_TRUE;
204 vol=(control_vol->left+control_vol->right)*4.0/200.0;
205 err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0);
206 if(err==0) {
207 // printf("SET VOL=%f\n", vol);
208 return CONTROL_TRUE;
210 else {
211 ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err);
212 return CONTROL_FALSE;
214 /* Everything is currently unimplemented */
215 default:
216 return CONTROL_FALSE;
222 static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){
223 uint32_t flags=(uint32_t) f->mFormatFlags;
224 ao_msg(MSGT_AO,lev, "%s %7.1fHz %"PRIu32"bit [%c%c%c%c][%"PRIu32"][%"PRIu32"][%"PRIu32"][%"PRIu32"][%"PRIu32"] %s %s %s%s%s%s\n",
225 str, f->mSampleRate, f->mBitsPerChannel,
226 (int)(f->mFormatID & 0xff000000) >> 24,
227 (int)(f->mFormatID & 0x00ff0000) >> 16,
228 (int)(f->mFormatID & 0x0000ff00) >> 8,
229 (int)(f->mFormatID & 0x000000ff) >> 0,
230 f->mFormatFlags, f->mBytesPerPacket,
231 f->mFramesPerPacket, f->mBytesPerFrame,
232 f->mChannelsPerFrame,
233 (flags&kAudioFormatFlagIsFloat) ? "float" : "int",
234 (flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
235 (flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U",
236 (flags&kAudioFormatFlagIsPacked) ? " packed" : "",
237 (flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
238 (flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" );
241 static OSStatus GetAudioProperty(AudioObjectID id,
242 AudioObjectPropertySelector selector,
243 UInt32 outSize, void *outData)
245 AudioObjectPropertyAddress property_address;
247 property_address.mSelector = selector;
248 property_address.mScope = kAudioObjectPropertyScopeGlobal;
249 property_address.mElement = kAudioObjectPropertyElementMaster;
251 return AudioObjectGetPropertyData(id, &property_address, 0, NULL, &outSize, outData);
254 static UInt32 GetAudioPropertyArray(AudioObjectID id,
255 AudioObjectPropertySelector selector,
256 AudioObjectPropertyScope scope,
257 void **outData)
259 OSStatus err;
260 AudioObjectPropertyAddress property_address;
261 UInt32 i_param_size;
263 property_address.mSelector = selector;
264 property_address.mScope = scope;
265 property_address.mElement = kAudioObjectPropertyElementMaster;
267 err = AudioObjectGetPropertyDataSize(id, &property_address, 0, NULL, &i_param_size);
269 if (err != noErr)
270 return 0;
272 *outData = malloc(i_param_size);
275 err = AudioObjectGetPropertyData(id, &property_address, 0, NULL, &i_param_size, *outData);
277 if (err != noErr) {
278 free(*outData);
279 return 0;
282 return i_param_size;
285 static UInt32 GetGlobalAudioPropertyArray(AudioObjectID id,
286 AudioObjectPropertySelector selector,
287 void **outData)
289 return GetAudioPropertyArray(id, selector, kAudioObjectPropertyScopeGlobal, outData);
292 static OSStatus GetAudioPropertyString(AudioObjectID id,
293 AudioObjectPropertySelector selector,
294 char **outData)
296 OSStatus err;
297 AudioObjectPropertyAddress property_address;
298 UInt32 i_param_size;
299 CFStringRef string;
300 CFIndex string_length;
302 property_address.mSelector = selector;
303 property_address.mScope = kAudioObjectPropertyScopeGlobal;
304 property_address.mElement = kAudioObjectPropertyElementMaster;
306 i_param_size = sizeof(CFStringRef);
307 err = AudioObjectGetPropertyData(id, &property_address, 0, NULL, &i_param_size, &string);
308 if (err != noErr)
309 return err;
311 string_length = CFStringGetMaximumSizeForEncoding(CFStringGetLength(string),
312 kCFStringEncodingASCII);
313 *outData = malloc(string_length + 1);
314 CFStringGetCString(string, *outData, string_length + 1, kCFStringEncodingASCII);
316 CFRelease(string);
318 return err;
321 static OSStatus SetAudioProperty(AudioObjectID id,
322 AudioObjectPropertySelector selector,
323 UInt32 inDataSize, void *inData)
325 AudioObjectPropertyAddress property_address;
327 property_address.mSelector = selector;
328 property_address.mScope = kAudioObjectPropertyScopeGlobal;
329 property_address.mElement = kAudioObjectPropertyElementMaster;
331 return AudioObjectSetPropertyData(id, &property_address, 0, NULL, inDataSize, inData);
334 static Boolean IsAudioPropertySettable(AudioObjectID id,
335 AudioObjectPropertySelector selector,
336 Boolean *outData)
338 AudioObjectPropertyAddress property_address;
340 property_address.mSelector = selector;
341 property_address.mScope = kAudioObjectPropertyScopeGlobal;
342 property_address.mElement = kAudioObjectPropertyElementMaster;
344 return AudioObjectIsPropertySettable(id, &property_address, outData);
347 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
348 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
349 static int OpenSPDIF(void);
350 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format );
351 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
352 const AudioTimeStamp * inNow,
353 const void * inInputData,
354 const AudioTimeStamp * inInputTime,
355 AudioBufferList * outOutputData,
356 const AudioTimeStamp * inOutputTime,
357 void * threadGlobals );
358 static OSStatus StreamListener( AudioObjectID inObjectID,
359 UInt32 inNumberAddresses,
360 const AudioObjectPropertyAddress inAddresses[],
361 void *inClientData );
362 static OSStatus DeviceListener( AudioObjectID inObjectID,
363 UInt32 inNumberAddresses,
364 const AudioObjectPropertyAddress inAddresses[],
365 void *inClientData );
367 static void print_help(void)
369 OSStatus err;
370 UInt32 i_param_size;
371 int num_devices;
372 AudioDeviceID *devids;
373 char *device_name;
375 mp_msg(MSGT_AO, MSGL_FATAL,
376 "\n-ao coreaudio commandline help:\n"
377 "Example: mplayer -ao coreaudio:device_id=266\n"
378 " open Core Audio with output device ID 266.\n"
379 "\nOptions:\n"
380 " device_id\n"
381 " ID of output device to use (0 = default device)\n"
382 " help\n"
383 " This help including list of available devices.\n"
384 "\n"
385 "Available output devices:\n");
387 i_param_size = GetGlobalAudioPropertyArray(kAudioObjectSystemObject, kAudioHardwarePropertyDevices, (void **)&devids);
389 if (!i_param_size) {
390 mp_msg(MSGT_AO, MSGL_FATAL, "Failed to get list of output devices.\n");
391 return;
394 num_devices = i_param_size / sizeof(AudioDeviceID);
396 for (int i = 0; i < num_devices; ++i) {
397 err = GetAudioPropertyString(devids[i], kAudioObjectPropertyName, &device_name);
399 if (err == noErr) {
400 mp_msg(MSGT_AO, MSGL_FATAL, "%s (id: %"PRIu32")\n", device_name, devids[i]);
401 free(device_name);
402 } else
403 mp_msg(MSGT_AO, MSGL_FATAL, "Unknown (id: %"PRIu32")\n", devids[i]);
406 mp_msg(MSGT_AO, MSGL_FATAL, "\n");
408 free(devids);
411 static int init(int rate,int channels,int format,int flags)
413 AudioStreamBasicDescription inDesc;
414 ComponentDescription desc;
415 Component comp;
416 AURenderCallbackStruct renderCallback;
417 OSStatus err;
418 UInt32 size, maxFrames, b_alive;
419 char *psz_name;
420 AudioDeviceID devid_def = 0;
421 int device_id, display_help = 0;
423 const opt_t subopts[] = {
424 {"device_id", OPT_ARG_INT, &device_id, NULL},
425 {"help", OPT_ARG_BOOL, &display_help, NULL},
426 {NULL}
429 // set defaults
430 device_id = 0;
432 if (subopt_parse(ao_subdevice, subopts) != 0 || display_help) {
433 print_help();
434 if (!display_help)
435 return 0;
438 ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
440 ao = calloc(1, sizeof(ao_coreaudio_t));
442 ao->i_selected_dev = 0;
443 ao->b_supports_digital = 0;
444 ao->b_digital = 0;
445 ao->b_muted = 0;
446 ao->b_stream_format_changed = 0;
447 ao->i_hog_pid = -1;
448 ao->i_stream_id = 0;
449 ao->i_stream_index = -1;
450 ao->b_revert = 0;
451 ao->b_changed_mixing = 0;
453 if (device_id == 0) {
454 /* Find the ID of the default Device. */
455 err = GetAudioProperty(kAudioObjectSystemObject,
456 kAudioHardwarePropertyDefaultOutputDevice,
457 sizeof(UInt32), &devid_def);
458 if (err != noErr)
460 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
461 goto err_out;
463 } else {
464 devid_def = device_id;
467 /* Retrieve the name of the device. */
468 err = GetAudioPropertyString(devid_def,
469 kAudioObjectPropertyName,
470 &psz_name);
471 if (err != noErr)
473 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
474 goto err_out;
477 ao_msg(MSGT_AO,MSGL_V, "got audio output device ID: %"PRIu32" Name: %s\n", devid_def, psz_name );
479 /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
480 if (AF_FORMAT_IS_AC3(format)) {
481 if (AudioDeviceSupportsDigital(devid_def))
483 ao->b_supports_digital = 1;
485 ao_msg(MSGT_AO, MSGL_V,
486 "probe default audio output device about support for digital s/pdif output: %d\n",
487 ao->b_supports_digital );
490 free(psz_name);
492 // Save selected device id
493 ao->i_selected_dev = devid_def;
495 // Build Description for the input format
496 inDesc.mSampleRate=rate;
497 inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
498 inDesc.mChannelsPerFrame=channels;
499 inDesc.mBitsPerChannel=af_fmt2bits(format);
501 if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
502 // float
503 inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
505 else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) {
506 // signed int
507 inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
509 else {
510 // unsigned int
511 inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
513 if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
514 inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
516 inDesc.mFramesPerPacket = 1;
517 ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
518 print_format(MSGL_V, "source:",&inDesc);
520 if (ao->b_supports_digital)
522 b_alive = 1;
523 err = GetAudioProperty(ao->i_selected_dev,
524 kAudioDevicePropertyDeviceIsAlive,
525 sizeof(UInt32), &b_alive);
526 if (err != noErr)
527 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
528 if (!b_alive)
529 ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
531 /* S/PDIF output need device in HogMode. */
532 err = GetAudioProperty(ao->i_selected_dev,
533 kAudioDevicePropertyHogMode,
534 sizeof(pid_t), &ao->i_hog_pid);
535 if (err != noErr)
537 /* This is not a fatal error. Some drivers simply don't support this property. */
538 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
539 (char *)&err);
540 ao->i_hog_pid = -1;
543 if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
545 ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
546 goto err_out;
548 ao->stream_format = inDesc;
549 return OpenSPDIF();
552 /* original analog output code */
553 desc.componentType = kAudioUnitType_Output;
554 desc.componentSubType = (device_id == 0) ? kAudioUnitSubType_DefaultOutput : kAudioUnitSubType_HALOutput;
555 desc.componentManufacturer = kAudioUnitManufacturer_Apple;
556 desc.componentFlags = 0;
557 desc.componentFlagsMask = 0;
559 comp = FindNextComponent(NULL, &desc); //Finds an component that meets the desc spec's
560 if (comp == NULL) {
561 ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
562 goto err_out;
565 err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component
566 if (err) {
567 ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
568 goto err_out;
571 // Initialize AudioUnit
572 err = AudioUnitInitialize(ao->theOutputUnit);
573 if (err) {
574 ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
575 goto err_out1;
578 size = sizeof(AudioStreamBasicDescription);
579 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
581 if (err) {
582 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
583 goto err_out2;
586 size = sizeof(UInt32);
587 err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
589 if (err)
591 ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
592 goto err_out2;
595 //Set the Current Device to the Default Output Unit.
596 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &ao->i_selected_dev, sizeof(ao->i_selected_dev));
598 ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
600 ao_data.samplerate = inDesc.mSampleRate;
601 ao_data.channels = inDesc.mChannelsPerFrame;
602 ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
603 ao_data.outburst = ao->chunk_size;
604 ao_data.buffersize = ao_data.bps;
606 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
607 ao->buffer_len = ao->num_chunks * ao->chunk_size;
608 ao->buffer = av_fifo_alloc(ao->buffer_len);
610 ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
612 renderCallback.inputProc = theRenderProc;
613 renderCallback.inputProcRefCon = 0;
614 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
615 if (err) {
616 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
617 goto err_out2;
620 reset();
622 return CONTROL_OK;
624 err_out2:
625 AudioUnitUninitialize(ao->theOutputUnit);
626 err_out1:
627 CloseComponent(ao->theOutputUnit);
628 err_out:
629 av_fifo_free(ao->buffer);
630 free(ao);
631 ao = NULL;
632 return CONTROL_FALSE;
635 /*****************************************************************************
636 * Setup a encoded digital stream (SPDIF)
637 *****************************************************************************/
638 static int OpenSPDIF(void)
640 OSStatus err = noErr;
641 UInt32 i_param_size, b_mix = 0;
642 Boolean b_writeable = 0;
643 AudioStreamID *p_streams = NULL;
644 int i, i_streams = 0;
645 AudioObjectPropertyAddress property_address;
647 /* Start doing the SPDIF setup process. */
648 ao->b_digital = 1;
650 /* Hog the device. */
651 ao->i_hog_pid = getpid() ;
653 err = SetAudioProperty(ao->i_selected_dev,
654 kAudioDevicePropertyHogMode,
655 sizeof(ao->i_hog_pid), &ao->i_hog_pid);
656 if (err != noErr)
658 ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
659 ao->i_hog_pid = -1;
660 goto err_out;
663 property_address.mSelector = kAudioDevicePropertySupportsMixing;
664 property_address.mScope = kAudioObjectPropertyScopeGlobal;
665 property_address.mElement = kAudioObjectPropertyElementMaster;
667 /* Set mixable to false if we are allowed to. */
668 if (AudioObjectHasProperty(ao->i_selected_dev, &property_address)) {
669 /* Set mixable to false if we are allowed to. */
670 err = IsAudioPropertySettable(ao->i_selected_dev,
671 kAudioDevicePropertySupportsMixing,
672 &b_writeable);
673 err = GetAudioProperty(ao->i_selected_dev,
674 kAudioDevicePropertySupportsMixing,
675 sizeof(UInt32), &b_mix);
676 if (err == noErr && b_writeable)
678 b_mix = 0;
679 err = SetAudioProperty(ao->i_selected_dev,
680 kAudioDevicePropertySupportsMixing,
681 sizeof(UInt32), &b_mix);
682 ao->b_changed_mixing = 1;
684 if (err != noErr)
686 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
687 goto err_out;
691 /* Get a list of all the streams on this device. */
692 i_param_size = GetAudioPropertyArray(ao->i_selected_dev,
693 kAudioDevicePropertyStreams,
694 kAudioDevicePropertyScopeOutput,
695 (void **)&p_streams);
697 if (!i_param_size) {
698 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n");
699 goto err_out;
702 i_streams = i_param_size / sizeof(AudioStreamID);
704 ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams);
706 for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i)
708 /* Find a stream with a cac3 stream. */
709 AudioStreamRangedDescription *p_format_list = NULL;
710 int i_formats = 0, j = 0, b_digital = 0;
712 i_param_size = GetGlobalAudioPropertyArray(p_streams[i],
713 kAudioStreamPropertyAvailablePhysicalFormats,
714 (void **)&p_format_list);
716 if (!i_param_size) {
717 ao_msg(MSGT_AO, MSGL_WARN,
718 "Could not get number of stream formats.\n");
719 continue;
722 i_formats = i_param_size / sizeof(AudioStreamRangedDescription);
724 /* Check if one of the supported formats is a digital format. */
725 for (j = 0; j < i_formats; ++j)
727 if (p_format_list[j].mFormat.mFormatID == 'IAC3' ||
728 p_format_list[j].mFormat.mFormatID == 'iac3' ||
729 p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 ||
730 p_format_list[j].mFormat.mFormatID == kAudioFormatAC3)
732 b_digital = 1;
733 break;
737 if (b_digital)
739 /* If this stream supports a digital (cac3) format, then set it. */
740 int i_requested_rate_format = -1;
741 int i_current_rate_format = -1;
742 int i_backup_rate_format = -1;
744 ao->i_stream_id = p_streams[i];
745 ao->i_stream_index = i;
747 if (ao->b_revert == 0)
749 /* Retrieve the original format of this stream first if not done so already. */
750 err = GetAudioProperty(ao->i_stream_id,
751 kAudioStreamPropertyPhysicalFormat,
752 sizeof(ao->sfmt_revert), &ao->sfmt_revert);
753 if (err != noErr)
755 ao_msg(MSGT_AO, MSGL_WARN,
756 "Could not retrieve the original stream format: [%4.4s]\n",
757 (char *)&err);
758 free(p_format_list);
759 continue;
761 ao->b_revert = 1;
764 for (j = 0; j < i_formats; ++j)
765 if (p_format_list[j].mFormat.mFormatID == 'IAC3' ||
766 p_format_list[j].mFormat.mFormatID == 'iac3' ||
767 p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 ||
768 p_format_list[j].mFormat.mFormatID == kAudioFormatAC3)
770 if (p_format_list[j].mFormat.mSampleRate == ao->stream_format.mSampleRate)
772 i_requested_rate_format = j;
773 break;
775 if (p_format_list[j].mFormat.mSampleRate == ao->sfmt_revert.mSampleRate)
776 i_current_rate_format = j;
777 else if (i_backup_rate_format < 0 || p_format_list[j].mFormat.mSampleRate > p_format_list[i_backup_rate_format].mFormat.mSampleRate)
778 i_backup_rate_format = j;
781 if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */
782 ao->stream_format = p_format_list[i_requested_rate_format].mFormat;
783 else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */
784 ao->stream_format = p_format_list[i_current_rate_format].mFormat;
785 else ao->stream_format = p_format_list[i_backup_rate_format].mFormat; /* And if we have to, any digital format will be just fine (highest rate possible). */
787 free(p_format_list);
789 free(p_streams);
791 if (ao->i_stream_index < 0)
793 ao_msg(MSGT_AO, MSGL_WARN,
794 "Cannot find any digital output stream format when OpenSPDIF().\n");
795 goto err_out;
798 print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);
800 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
801 goto err_out;
803 property_address.mSelector = kAudioDevicePropertyDeviceHasChanged;
804 property_address.mScope = kAudioObjectPropertyScopeGlobal;
805 property_address.mElement = kAudioObjectPropertyElementMaster;
807 err = AudioObjectAddPropertyListener(ao->i_selected_dev,
808 &property_address,
809 DeviceListener,
810 NULL);
811 if (err != noErr)
812 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err);
815 /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */
816 /* Although there's no such case reported. */
817 #if HAVE_BIGENDIAN
818 if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
819 #else
820 /* tell mplayer that we need a byteswap on AC3 streams, */
821 if (ao->stream_format.mFormatID & kAudioFormat60958AC3)
822 ao_data.format = AF_FORMAT_AC3_LE;
824 if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
825 #endif
826 ao_msg(MSGT_AO, MSGL_WARN,
827 "Output stream has non-native byte order, digital output may fail.\n");
829 /* For ac3/dts, just use packet size 6144 bytes as chunk size. */
830 ao->chunk_size = ao->stream_format.mBytesPerPacket;
832 ao_data.samplerate = ao->stream_format.mSampleRate;
833 ao_data.channels = ao->stream_format.mChannelsPerFrame;
834 ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
835 ao_data.outburst = ao->chunk_size;
836 ao_data.buffersize = ao_data.bps;
838 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
839 ao->buffer_len = ao->num_chunks * ao->chunk_size;
840 ao->buffer = av_fifo_alloc(ao->buffer_len);
842 ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
845 /* Create IOProc callback. */
846 err = AudioDeviceCreateIOProcID(ao->i_selected_dev,
847 (AudioDeviceIOProc)RenderCallbackSPDIF,
848 (void *)ao,
849 &ao->renderCallback);
851 if (err != noErr || ao->renderCallback == NULL)
853 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
854 goto err_out1;
857 reset();
859 return CONTROL_TRUE;
861 err_out1:
862 if (ao->b_revert)
863 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
864 err_out:
865 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
867 int b_mix = 1;
868 err = SetAudioProperty(ao->i_selected_dev,
869 kAudioDevicePropertySupportsMixing,
870 sizeof(int), &b_mix);
871 if (err != noErr)
872 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
873 (char *)&err);
875 if (ao->i_hog_pid == getpid())
877 ao->i_hog_pid = -1;
878 err = SetAudioProperty(ao->i_selected_dev,
879 kAudioDevicePropertyHogMode,
880 sizeof(ao->i_hog_pid), &ao->i_hog_pid);
881 if (err != noErr)
882 ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
883 (char *)&err);
885 av_fifo_free(ao->buffer);
886 free(ao);
887 ao = NULL;
888 return CONTROL_FALSE;
891 /*****************************************************************************
892 * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support.
893 *****************************************************************************/
894 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id )
896 UInt32 i_param_size = 0;
897 AudioStreamID *p_streams = NULL;
898 int i = 0, i_streams = 0;
899 int b_return = CONTROL_FALSE;
901 /* Retrieve all the output streams. */
902 i_param_size = GetAudioPropertyArray(i_dev_id,
903 kAudioDevicePropertyStreams,
904 kAudioDevicePropertyScopeOutput,
905 (void **)&p_streams);
907 if (!i_param_size) {
908 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n");
909 return CONTROL_FALSE;
912 i_streams = i_param_size / sizeof(AudioStreamID);
914 for (i = 0; i < i_streams; ++i)
916 if (AudioStreamSupportsDigital(p_streams[i]))
917 b_return = CONTROL_OK;
920 free(p_streams);
921 return b_return;
924 /*****************************************************************************
925 * AudioStreamSupportsDigital: Check i_stream_id for digital stream support.
926 *****************************************************************************/
927 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id )
929 UInt32 i_param_size;
930 AudioStreamRangedDescription *p_format_list = NULL;
931 int i, i_formats, b_return = CONTROL_FALSE;
933 /* Retrieve all the stream formats supported by each output stream. */
934 i_param_size = GetGlobalAudioPropertyArray(i_stream_id,
935 kAudioStreamPropertyAvailablePhysicalFormats,
936 (void **)&p_format_list);
938 if (!i_param_size) {
939 ao_msg(MSGT_AO, MSGL_WARN, "Could not get number of stream formats.\n");
940 return CONTROL_FALSE;
943 i_formats = i_param_size / sizeof(AudioStreamRangedDescription);
945 for (i = 0; i < i_formats; ++i)
947 print_format(MSGL_V, "supported format:", &(p_format_list[i].mFormat));
949 if (p_format_list[i].mFormat.mFormatID == 'IAC3' ||
950 p_format_list[i].mFormat.mFormatID == 'iac3' ||
951 p_format_list[i].mFormat.mFormatID == kAudioFormat60958AC3 ||
952 p_format_list[i].mFormat.mFormatID == kAudioFormatAC3)
953 b_return = CONTROL_OK;
956 free(p_format_list);
957 return b_return;
960 /*****************************************************************************
961 * AudioStreamChangeFormat: Change i_stream_id to change_format
962 *****************************************************************************/
963 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format )
965 OSStatus err = noErr;
966 int i;
967 AudioObjectPropertyAddress property_address;
969 static volatile int stream_format_changed;
970 stream_format_changed = 0;
972 print_format(MSGL_V, "setting stream format:", &change_format);
974 /* Install the callback. */
975 property_address.mSelector = kAudioStreamPropertyPhysicalFormat;
976 property_address.mScope = kAudioObjectPropertyScopeGlobal;
977 property_address.mElement = kAudioObjectPropertyElementMaster;
979 err = AudioObjectAddPropertyListener(i_stream_id,
980 &property_address,
981 StreamListener,
982 (void *)&stream_format_changed);
983 if (err != noErr)
985 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err);
986 return CONTROL_FALSE;
989 /* Change the format. */
990 err = SetAudioProperty(i_stream_id,
991 kAudioStreamPropertyPhysicalFormat,
992 sizeof(AudioStreamBasicDescription), &change_format);
993 if (err != noErr)
995 ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err);
996 return CONTROL_FALSE;
999 /* The AudioStreamSetProperty is not only asynchronious,
1000 * it is also not Atomic, in its behaviour.
1001 * Therefore we check 5 times before we really give up.
1002 * FIXME: failing isn't actually implemented yet. */
1003 for (i = 0; i < 5; ++i)
1005 AudioStreamBasicDescription actual_format;
1006 int j;
1007 for (j = 0; !stream_format_changed && j < 50; ++j)
1008 usec_sleep(10000);
1009 if (stream_format_changed)
1010 stream_format_changed = 0;
1011 else
1012 ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" );
1014 err = GetAudioProperty(i_stream_id,
1015 kAudioStreamPropertyPhysicalFormat,
1016 sizeof(AudioStreamBasicDescription), &actual_format);
1018 print_format(MSGL_V, "actual format in use:", &actual_format);
1019 if (actual_format.mSampleRate == change_format.mSampleRate &&
1020 actual_format.mFormatID == change_format.mFormatID &&
1021 actual_format.mFramesPerPacket == change_format.mFramesPerPacket)
1023 /* The right format is now active. */
1024 break;
1026 /* We need to check again. */
1029 /* Removing the property listener. */
1030 err = AudioObjectRemovePropertyListener(i_stream_id,
1031 &property_address,
1032 StreamListener,
1033 (void *)&stream_format_changed);
1034 if (err != noErr)
1036 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err);
1037 return CONTROL_FALSE;
1040 return CONTROL_TRUE;
1043 /*****************************************************************************
1044 * RenderCallbackSPDIF: callback for SPDIF audio output
1045 *****************************************************************************/
1046 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
1047 const AudioTimeStamp * inNow,
1048 const void * inInputData,
1049 const AudioTimeStamp * inInputTime,
1050 AudioBufferList * outOutputData,
1051 const AudioTimeStamp * inOutputTime,
1052 void * threadGlobals )
1054 int amt = av_fifo_size(ao->buffer);
1055 int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;
1057 if (amt > req)
1058 amt = req;
1059 if (amt)
1060 read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt);
1062 return noErr;
1066 static int play(void* output_samples,int num_bytes,int flags)
1068 int wrote, b_digital;
1069 SInt32 exit_reason;
1071 // Check whether we need to reset the digital output stream.
1072 if (ao->b_digital && ao->b_stream_format_changed)
1074 ao->b_stream_format_changed = 0;
1075 b_digital = AudioStreamSupportsDigital(ao->i_stream_id);
1076 if (b_digital)
1078 /* Current stream supports digital format output, let's set it. */
1079 ao_msg(MSGT_AO, MSGL_V,
1080 "Detected current stream supports digital, try to restore digital output...\n");
1082 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
1084 ao_msg(MSGT_AO, MSGL_WARN, "Restoring digital output failed.\n");
1086 else
1088 ao_msg(MSGT_AO, MSGL_WARN, "Restoring digital output succeeded.\n");
1089 reset();
1092 else
1093 ao_msg(MSGT_AO, MSGL_V, "Detected current stream does not support digital.\n");
1096 wrote=write_buffer(output_samples, num_bytes);
1097 audio_resume();
1099 do {
1100 exit_reason = CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.01, true);
1101 } while (exit_reason == kCFRunLoopRunHandledSource);
1103 return wrote;
1106 /* set variables and buffer to initial state */
1107 static void reset(void)
1109 audio_pause();
1110 av_fifo_reset(ao->buffer);
1114 /* return available space */
1115 static int get_space(void)
1117 return ao->buffer_len - av_fifo_size(ao->buffer);
1121 /* return delay until audio is played */
1122 static float get_delay(void)
1124 // inaccurate, should also contain the data buffered e.g. by the OS
1125 return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps;
1129 /* unload plugin and deregister from coreaudio */
1130 static void uninit(int immed)
1132 OSStatus err = noErr;
1134 if (!immed) {
1135 long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps;
1136 ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft);
1137 usec_sleep((int)timeleft);
1140 if (!ao->b_digital) {
1141 AudioOutputUnitStop(ao->theOutputUnit);
1142 AudioUnitUninitialize(ao->theOutputUnit);
1143 CloseComponent(ao->theOutputUnit);
1145 else {
1146 /* Stop device. */
1147 err = AudioDeviceStop(ao->i_selected_dev, ao->renderCallback);
1148 if (err != noErr)
1149 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1151 /* Remove IOProc callback. */
1152 err = AudioDeviceDestroyIOProcID(ao->i_selected_dev, ao->renderCallback);
1153 if (err != noErr)
1154 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err);
1156 if (ao->b_revert)
1157 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
1159 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
1161 UInt32 b_mix;
1162 Boolean b_writeable = 0;
1163 /* Revert mixable to true if we are allowed to. */
1164 err = IsAudioPropertySettable(ao->i_selected_dev,
1165 kAudioDevicePropertySupportsMixing,
1166 &b_writeable);
1167 err = GetAudioProperty(ao->i_selected_dev,
1168 kAudioDevicePropertySupportsMixing,
1169 sizeof(UInt32), &b_mix);
1170 if (err == noErr && b_writeable)
1172 b_mix = 1;
1173 err = SetAudioProperty(ao->i_selected_dev,
1174 kAudioDevicePropertySupportsMixing,
1175 sizeof(UInt32), &b_mix);
1177 if (err != noErr)
1178 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
1180 if (ao->i_hog_pid == getpid())
1182 ao->i_hog_pid = -1;
1183 err = SetAudioProperty(ao->i_selected_dev,
1184 kAudioDevicePropertyHogMode,
1185 sizeof(ao->i_hog_pid), &ao->i_hog_pid);
1186 if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err);
1190 av_fifo_free(ao->buffer);
1191 free(ao);
1192 ao = NULL;
1196 /* stop playing, keep buffers (for pause) */
1197 static void audio_pause(void)
1199 OSErr err=noErr;
1201 /* Stop callback. */
1202 if (!ao->b_digital)
1204 err=AudioOutputUnitStop(ao->theOutputUnit);
1205 if (err != noErr)
1206 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err);
1208 else
1210 err = AudioDeviceStop(ao->i_selected_dev, ao->renderCallback);
1211 if (err != noErr)
1212 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1214 ao->paused = 1;
1218 /* resume playing, after audio_pause() */
1219 static void audio_resume(void)
1221 OSErr err=noErr;
1223 if (!ao->paused)
1224 return;
1226 /* Start callback. */
1227 if (!ao->b_digital)
1229 err = AudioOutputUnitStart(ao->theOutputUnit);
1230 if (err != noErr)
1231 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err);
1233 else
1235 err = AudioDeviceStart(ao->i_selected_dev, ao->renderCallback);
1236 if (err != noErr)
1237 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err);
1239 ao->paused = 0;
1242 /*****************************************************************************
1243 * StreamListener
1244 *****************************************************************************/
1245 static OSStatus StreamListener( AudioObjectID inObjectID,
1246 UInt32 inNumberAddresses,
1247 const AudioObjectPropertyAddress inAddresses[],
1248 void *inClientData )
1250 for (int i=0; i < inNumberAddresses; ++i)
1252 if (inAddresses[i].mSelector == kAudioStreamPropertyPhysicalFormat) {
1253 ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioStreamPropertyPhysicalFormat changed.\n");
1254 if (inClientData)
1255 *(volatile int *)inClientData = 1;
1256 break;
1259 return noErr;
1262 static OSStatus DeviceListener( AudioObjectID inObjectID,
1263 UInt32 inNumberAddresses,
1264 const AudioObjectPropertyAddress inAddresses[],
1265 void *inClientData )
1267 for (int i=0; i < inNumberAddresses; ++i)
1269 if (inAddresses[i].mSelector == kAudioDevicePropertyDeviceHasChanged) {
1270 ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n");
1271 ao->b_stream_format_changed = 1;
1272 break;
1275 return noErr;