demux_viv: fix unsafe code
[mplayer.git] / libao2 / ao_coreaudio.c
blobd1a93c85e25be8888fc8e962e917f921c63e57d4
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 int vol = ao->b_muted ? 0 : 100;
173 *control_vol = (ao_control_vol_t) {
174 .left = vol, .right = vol,
176 return CONTROL_TRUE;
178 err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol);
180 if(err==0) {
181 // printf("GET VOL=%f\n", vol);
182 control_vol->left=control_vol->right=vol*100.0/4.0;
183 return CONTROL_TRUE;
185 else {
186 ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err);
187 return CONTROL_FALSE;
190 case AOCONTROL_SET_VOLUME:
191 control_vol = (ao_control_vol_t*)arg;
193 if (ao->b_digital) {
194 // Digital output can not set volume. Here we have to return true
195 // to make mixer forget it. Else mixer will add a soft filter,
196 // that's not we expected and the filter not support ac3 stream
197 // will cause mplayer die.
199 // Although not support set volume, but at least we support mute.
200 // MPlayer set mute by set volume to zero, we handle it.
201 if (control_vol->left == 0 && control_vol->right == 0)
202 ao->b_muted = 1;
203 else
204 ao->b_muted = 0;
205 return CONTROL_TRUE;
208 vol=(control_vol->left+control_vol->right)*4.0/200.0;
209 err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0);
210 if(err==0) {
211 // printf("SET VOL=%f\n", vol);
212 return CONTROL_TRUE;
214 else {
215 ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err);
216 return CONTROL_FALSE;
218 /* Everything is currently unimplemented */
219 default:
220 return CONTROL_FALSE;
226 static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){
227 uint32_t flags=(uint32_t) f->mFormatFlags;
228 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",
229 str, f->mSampleRate, f->mBitsPerChannel,
230 (int)(f->mFormatID & 0xff000000) >> 24,
231 (int)(f->mFormatID & 0x00ff0000) >> 16,
232 (int)(f->mFormatID & 0x0000ff00) >> 8,
233 (int)(f->mFormatID & 0x000000ff) >> 0,
234 f->mFormatFlags, f->mBytesPerPacket,
235 f->mFramesPerPacket, f->mBytesPerFrame,
236 f->mChannelsPerFrame,
237 (flags&kAudioFormatFlagIsFloat) ? "float" : "int",
238 (flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
239 (flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U",
240 (flags&kAudioFormatFlagIsPacked) ? " packed" : "",
241 (flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
242 (flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" );
245 static OSStatus GetAudioProperty(AudioObjectID id,
246 AudioObjectPropertySelector selector,
247 UInt32 outSize, void *outData)
249 AudioObjectPropertyAddress property_address;
251 property_address.mSelector = selector;
252 property_address.mScope = kAudioObjectPropertyScopeGlobal;
253 property_address.mElement = kAudioObjectPropertyElementMaster;
255 return AudioObjectGetPropertyData(id, &property_address, 0, NULL, &outSize, outData);
258 static UInt32 GetAudioPropertyArray(AudioObjectID id,
259 AudioObjectPropertySelector selector,
260 AudioObjectPropertyScope scope,
261 void **outData)
263 OSStatus err;
264 AudioObjectPropertyAddress property_address;
265 UInt32 i_param_size;
267 property_address.mSelector = selector;
268 property_address.mScope = scope;
269 property_address.mElement = kAudioObjectPropertyElementMaster;
271 err = AudioObjectGetPropertyDataSize(id, &property_address, 0, NULL, &i_param_size);
273 if (err != noErr)
274 return 0;
276 *outData = malloc(i_param_size);
279 err = AudioObjectGetPropertyData(id, &property_address, 0, NULL, &i_param_size, *outData);
281 if (err != noErr) {
282 free(*outData);
283 return 0;
286 return i_param_size;
289 static UInt32 GetGlobalAudioPropertyArray(AudioObjectID id,
290 AudioObjectPropertySelector selector,
291 void **outData)
293 return GetAudioPropertyArray(id, selector, kAudioObjectPropertyScopeGlobal, outData);
296 static OSStatus GetAudioPropertyString(AudioObjectID id,
297 AudioObjectPropertySelector selector,
298 char **outData)
300 OSStatus err;
301 AudioObjectPropertyAddress property_address;
302 UInt32 i_param_size;
303 CFStringRef string;
304 CFIndex string_length;
306 property_address.mSelector = selector;
307 property_address.mScope = kAudioObjectPropertyScopeGlobal;
308 property_address.mElement = kAudioObjectPropertyElementMaster;
310 i_param_size = sizeof(CFStringRef);
311 err = AudioObjectGetPropertyData(id, &property_address, 0, NULL, &i_param_size, &string);
312 if (err != noErr)
313 return err;
315 string_length = CFStringGetMaximumSizeForEncoding(CFStringGetLength(string),
316 kCFStringEncodingASCII);
317 *outData = malloc(string_length + 1);
318 CFStringGetCString(string, *outData, string_length + 1, kCFStringEncodingASCII);
320 CFRelease(string);
322 return err;
325 static OSStatus SetAudioProperty(AudioObjectID id,
326 AudioObjectPropertySelector selector,
327 UInt32 inDataSize, void *inData)
329 AudioObjectPropertyAddress property_address;
331 property_address.mSelector = selector;
332 property_address.mScope = kAudioObjectPropertyScopeGlobal;
333 property_address.mElement = kAudioObjectPropertyElementMaster;
335 return AudioObjectSetPropertyData(id, &property_address, 0, NULL, inDataSize, inData);
338 static Boolean IsAudioPropertySettable(AudioObjectID id,
339 AudioObjectPropertySelector selector,
340 Boolean *outData)
342 AudioObjectPropertyAddress property_address;
344 property_address.mSelector = selector;
345 property_address.mScope = kAudioObjectPropertyScopeGlobal;
346 property_address.mElement = kAudioObjectPropertyElementMaster;
348 return AudioObjectIsPropertySettable(id, &property_address, outData);
351 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
352 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
353 static int OpenSPDIF(void);
354 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format );
355 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
356 const AudioTimeStamp * inNow,
357 const void * inInputData,
358 const AudioTimeStamp * inInputTime,
359 AudioBufferList * outOutputData,
360 const AudioTimeStamp * inOutputTime,
361 void * threadGlobals );
362 static OSStatus StreamListener( AudioObjectID inObjectID,
363 UInt32 inNumberAddresses,
364 const AudioObjectPropertyAddress inAddresses[],
365 void *inClientData );
366 static OSStatus DeviceListener( AudioObjectID inObjectID,
367 UInt32 inNumberAddresses,
368 const AudioObjectPropertyAddress inAddresses[],
369 void *inClientData );
371 static void print_help(void)
373 OSStatus err;
374 UInt32 i_param_size;
375 int num_devices;
376 AudioDeviceID *devids;
377 char *device_name;
379 mp_msg(MSGT_AO, MSGL_FATAL,
380 "\n-ao coreaudio commandline help:\n"
381 "Example: mplayer -ao coreaudio:device_id=266\n"
382 " open Core Audio with output device ID 266.\n"
383 "\nOptions:\n"
384 " device_id\n"
385 " ID of output device to use (0 = default device)\n"
386 " help\n"
387 " This help including list of available devices.\n"
388 "\n"
389 "Available output devices:\n");
391 i_param_size = GetGlobalAudioPropertyArray(kAudioObjectSystemObject, kAudioHardwarePropertyDevices, (void **)&devids);
393 if (!i_param_size) {
394 mp_msg(MSGT_AO, MSGL_FATAL, "Failed to get list of output devices.\n");
395 return;
398 num_devices = i_param_size / sizeof(AudioDeviceID);
400 for (int i = 0; i < num_devices; ++i) {
401 err = GetAudioPropertyString(devids[i], kAudioObjectPropertyName, &device_name);
403 if (err == noErr) {
404 mp_msg(MSGT_AO, MSGL_FATAL, "%s (id: %"PRIu32")\n", device_name, devids[i]);
405 free(device_name);
406 } else
407 mp_msg(MSGT_AO, MSGL_FATAL, "Unknown (id: %"PRIu32")\n", devids[i]);
410 mp_msg(MSGT_AO, MSGL_FATAL, "\n");
412 free(devids);
415 static int init(int rate,int channels,int format,int flags)
417 AudioStreamBasicDescription inDesc;
418 ComponentDescription desc;
419 Component comp;
420 AURenderCallbackStruct renderCallback;
421 OSStatus err;
422 UInt32 size, maxFrames, b_alive;
423 char *psz_name;
424 AudioDeviceID devid_def = 0;
425 int device_id, display_help = 0;
427 const opt_t subopts[] = {
428 {"device_id", OPT_ARG_INT, &device_id, NULL},
429 {"help", OPT_ARG_BOOL, &display_help, NULL},
430 {NULL}
433 // set defaults
434 device_id = 0;
436 if (subopt_parse(ao_subdevice, subopts) != 0 || display_help) {
437 print_help();
438 if (!display_help)
439 return 0;
442 ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
444 ao = calloc(1, sizeof(ao_coreaudio_t));
446 ao->i_selected_dev = 0;
447 ao->b_supports_digital = 0;
448 ao->b_digital = 0;
449 ao->b_muted = 0;
450 ao->b_stream_format_changed = 0;
451 ao->i_hog_pid = -1;
452 ao->i_stream_id = 0;
453 ao->i_stream_index = -1;
454 ao->b_revert = 0;
455 ao->b_changed_mixing = 0;
457 global_ao->no_persistent_volume = true;
459 if (device_id == 0) {
460 /* Find the ID of the default Device. */
461 err = GetAudioProperty(kAudioObjectSystemObject,
462 kAudioHardwarePropertyDefaultOutputDevice,
463 sizeof(UInt32), &devid_def);
464 if (err != noErr)
466 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
467 goto err_out;
469 } else {
470 devid_def = device_id;
473 /* Retrieve the name of the device. */
474 err = GetAudioPropertyString(devid_def,
475 kAudioObjectPropertyName,
476 &psz_name);
477 if (err != noErr)
479 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
480 goto err_out;
483 ao_msg(MSGT_AO,MSGL_V, "got audio output device ID: %"PRIu32" Name: %s\n", devid_def, psz_name );
485 /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
486 if (AF_FORMAT_IS_AC3(format)) {
487 if (AudioDeviceSupportsDigital(devid_def))
489 ao->b_supports_digital = 1;
491 ao_msg(MSGT_AO, MSGL_V,
492 "probe default audio output device about support for digital s/pdif output: %d\n",
493 ao->b_supports_digital );
496 free(psz_name);
498 // Save selected device id
499 ao->i_selected_dev = devid_def;
501 // Build Description for the input format
502 inDesc.mSampleRate=rate;
503 inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
504 inDesc.mChannelsPerFrame=channels;
505 inDesc.mBitsPerChannel=af_fmt2bits(format);
507 if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
508 // float
509 inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
511 else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) {
512 // signed int
513 inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
515 else {
516 // unsigned int
517 inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
519 if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
520 inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
522 inDesc.mFramesPerPacket = 1;
523 ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
524 print_format(MSGL_V, "source:",&inDesc);
526 if (ao->b_supports_digital)
528 b_alive = 1;
529 err = GetAudioProperty(ao->i_selected_dev,
530 kAudioDevicePropertyDeviceIsAlive,
531 sizeof(UInt32), &b_alive);
532 if (err != noErr)
533 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
534 if (!b_alive)
535 ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
537 /* S/PDIF output need device in HogMode. */
538 err = GetAudioProperty(ao->i_selected_dev,
539 kAudioDevicePropertyHogMode,
540 sizeof(pid_t), &ao->i_hog_pid);
541 if (err != noErr)
543 /* This is not a fatal error. Some drivers simply don't support this property. */
544 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
545 (char *)&err);
546 ao->i_hog_pid = -1;
549 if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
551 ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
552 goto err_out;
554 ao->stream_format = inDesc;
555 return OpenSPDIF();
558 /* original analog output code */
559 desc.componentType = kAudioUnitType_Output;
560 desc.componentSubType = (device_id == 0) ? kAudioUnitSubType_DefaultOutput : kAudioUnitSubType_HALOutput;
561 desc.componentManufacturer = kAudioUnitManufacturer_Apple;
562 desc.componentFlags = 0;
563 desc.componentFlagsMask = 0;
565 comp = FindNextComponent(NULL, &desc); //Finds an component that meets the desc spec's
566 if (comp == NULL) {
567 ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
568 goto err_out;
571 err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component
572 if (err) {
573 ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
574 goto err_out;
577 // Initialize AudioUnit
578 err = AudioUnitInitialize(ao->theOutputUnit);
579 if (err) {
580 ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
581 goto err_out1;
584 size = sizeof(AudioStreamBasicDescription);
585 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
587 if (err) {
588 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
589 goto err_out2;
592 size = sizeof(UInt32);
593 err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
595 if (err)
597 ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
598 goto err_out2;
601 //Set the Current Device to the Default Output Unit.
602 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &ao->i_selected_dev, sizeof(ao->i_selected_dev));
604 ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
606 ao_data.samplerate = inDesc.mSampleRate;
607 ao_data.channels = inDesc.mChannelsPerFrame;
608 ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
609 ao_data.outburst = ao->chunk_size;
610 ao_data.buffersize = ao_data.bps;
612 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
613 ao->buffer_len = ao->num_chunks * ao->chunk_size;
614 ao->buffer = av_fifo_alloc(ao->buffer_len);
616 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);
618 renderCallback.inputProc = theRenderProc;
619 renderCallback.inputProcRefCon = 0;
620 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
621 if (err) {
622 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
623 goto err_out2;
626 reset();
628 return CONTROL_OK;
630 err_out2:
631 AudioUnitUninitialize(ao->theOutputUnit);
632 err_out1:
633 CloseComponent(ao->theOutputUnit);
634 err_out:
635 av_fifo_free(ao->buffer);
636 free(ao);
637 ao = NULL;
638 return CONTROL_FALSE;
641 /*****************************************************************************
642 * Setup a encoded digital stream (SPDIF)
643 *****************************************************************************/
644 static int OpenSPDIF(void)
646 OSStatus err = noErr;
647 UInt32 i_param_size, b_mix = 0;
648 Boolean b_writeable = 0;
649 AudioStreamID *p_streams = NULL;
650 int i, i_streams = 0;
651 AudioObjectPropertyAddress property_address;
653 /* Start doing the SPDIF setup process. */
654 ao->b_digital = 1;
656 /* Hog the device. */
657 ao->i_hog_pid = getpid() ;
659 err = SetAudioProperty(ao->i_selected_dev,
660 kAudioDevicePropertyHogMode,
661 sizeof(ao->i_hog_pid), &ao->i_hog_pid);
662 if (err != noErr)
664 ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
665 ao->i_hog_pid = -1;
666 goto err_out;
669 property_address.mSelector = kAudioDevicePropertySupportsMixing;
670 property_address.mScope = kAudioObjectPropertyScopeGlobal;
671 property_address.mElement = kAudioObjectPropertyElementMaster;
673 /* Set mixable to false if we are allowed to. */
674 if (AudioObjectHasProperty(ao->i_selected_dev, &property_address)) {
675 /* Set mixable to false if we are allowed to. */
676 err = IsAudioPropertySettable(ao->i_selected_dev,
677 kAudioDevicePropertySupportsMixing,
678 &b_writeable);
679 err = GetAudioProperty(ao->i_selected_dev,
680 kAudioDevicePropertySupportsMixing,
681 sizeof(UInt32), &b_mix);
682 if (err == noErr && b_writeable)
684 b_mix = 0;
685 err = SetAudioProperty(ao->i_selected_dev,
686 kAudioDevicePropertySupportsMixing,
687 sizeof(UInt32), &b_mix);
688 ao->b_changed_mixing = 1;
690 if (err != noErr)
692 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
693 goto err_out;
697 /* Get a list of all the streams on this device. */
698 i_param_size = GetAudioPropertyArray(ao->i_selected_dev,
699 kAudioDevicePropertyStreams,
700 kAudioDevicePropertyScopeOutput,
701 (void **)&p_streams);
703 if (!i_param_size) {
704 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n");
705 goto err_out;
708 i_streams = i_param_size / sizeof(AudioStreamID);
710 ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams);
712 for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i)
714 /* Find a stream with a cac3 stream. */
715 AudioStreamRangedDescription *p_format_list = NULL;
716 int i_formats = 0, j = 0, b_digital = 0;
718 i_param_size = GetGlobalAudioPropertyArray(p_streams[i],
719 kAudioStreamPropertyAvailablePhysicalFormats,
720 (void **)&p_format_list);
722 if (!i_param_size) {
723 ao_msg(MSGT_AO, MSGL_WARN,
724 "Could not get number of stream formats.\n");
725 continue;
728 i_formats = i_param_size / sizeof(AudioStreamRangedDescription);
730 /* Check if one of the supported formats is a digital format. */
731 for (j = 0; j < i_formats; ++j)
733 if (p_format_list[j].mFormat.mFormatID == 'IAC3' ||
734 p_format_list[j].mFormat.mFormatID == 'iac3' ||
735 p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 ||
736 p_format_list[j].mFormat.mFormatID == kAudioFormatAC3)
738 b_digital = 1;
739 break;
743 if (b_digital)
745 /* If this stream supports a digital (cac3) format, then set it. */
746 int i_requested_rate_format = -1;
747 int i_current_rate_format = -1;
748 int i_backup_rate_format = -1;
750 ao->i_stream_id = p_streams[i];
751 ao->i_stream_index = i;
753 if (ao->b_revert == 0)
755 /* Retrieve the original format of this stream first if not done so already. */
756 err = GetAudioProperty(ao->i_stream_id,
757 kAudioStreamPropertyPhysicalFormat,
758 sizeof(ao->sfmt_revert), &ao->sfmt_revert);
759 if (err != noErr)
761 ao_msg(MSGT_AO, MSGL_WARN,
762 "Could not retrieve the original stream format: [%4.4s]\n",
763 (char *)&err);
764 free(p_format_list);
765 continue;
767 ao->b_revert = 1;
770 for (j = 0; j < i_formats; ++j)
771 if (p_format_list[j].mFormat.mFormatID == 'IAC3' ||
772 p_format_list[j].mFormat.mFormatID == 'iac3' ||
773 p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 ||
774 p_format_list[j].mFormat.mFormatID == kAudioFormatAC3)
776 if (p_format_list[j].mFormat.mSampleRate == ao->stream_format.mSampleRate)
778 i_requested_rate_format = j;
779 break;
781 if (p_format_list[j].mFormat.mSampleRate == ao->sfmt_revert.mSampleRate)
782 i_current_rate_format = j;
783 else if (i_backup_rate_format < 0 || p_format_list[j].mFormat.mSampleRate > p_format_list[i_backup_rate_format].mFormat.mSampleRate)
784 i_backup_rate_format = j;
787 if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */
788 ao->stream_format = p_format_list[i_requested_rate_format].mFormat;
789 else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */
790 ao->stream_format = p_format_list[i_current_rate_format].mFormat;
791 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). */
793 free(p_format_list);
795 free(p_streams);
797 if (ao->i_stream_index < 0)
799 ao_msg(MSGT_AO, MSGL_WARN,
800 "Cannot find any digital output stream format when OpenSPDIF().\n");
801 goto err_out;
804 print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);
806 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
807 goto err_out;
809 property_address.mSelector = kAudioDevicePropertyDeviceHasChanged;
810 property_address.mScope = kAudioObjectPropertyScopeGlobal;
811 property_address.mElement = kAudioObjectPropertyElementMaster;
813 err = AudioObjectAddPropertyListener(ao->i_selected_dev,
814 &property_address,
815 DeviceListener,
816 NULL);
817 if (err != noErr)
818 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err);
821 /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */
822 /* Although there's no such case reported. */
823 #if HAVE_BIGENDIAN
824 if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
825 #else
826 /* tell mplayer that we need a byteswap on AC3 streams, */
827 if (ao->stream_format.mFormatID & kAudioFormat60958AC3)
828 ao_data.format = AF_FORMAT_AC3_LE;
830 if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
831 #endif
832 ao_msg(MSGT_AO, MSGL_WARN,
833 "Output stream has non-native byte order, digital output may fail.\n");
835 /* For ac3/dts, just use packet size 6144 bytes as chunk size. */
836 ao->chunk_size = ao->stream_format.mBytesPerPacket;
838 ao_data.samplerate = ao->stream_format.mSampleRate;
839 ao_data.channels = ao->stream_format.mChannelsPerFrame;
840 ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
841 ao_data.outburst = ao->chunk_size;
842 ao_data.buffersize = ao_data.bps;
844 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
845 ao->buffer_len = ao->num_chunks * ao->chunk_size;
846 ao->buffer = av_fifo_alloc(ao->buffer_len);
848 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);
851 /* Create IOProc callback. */
852 err = AudioDeviceCreateIOProcID(ao->i_selected_dev,
853 (AudioDeviceIOProc)RenderCallbackSPDIF,
854 (void *)ao,
855 &ao->renderCallback);
857 if (err != noErr || ao->renderCallback == NULL)
859 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
860 goto err_out1;
863 reset();
865 return CONTROL_TRUE;
867 err_out1:
868 if (ao->b_revert)
869 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
870 err_out:
871 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
873 int b_mix = 1;
874 err = SetAudioProperty(ao->i_selected_dev,
875 kAudioDevicePropertySupportsMixing,
876 sizeof(int), &b_mix);
877 if (err != noErr)
878 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
879 (char *)&err);
881 if (ao->i_hog_pid == getpid())
883 ao->i_hog_pid = -1;
884 err = SetAudioProperty(ao->i_selected_dev,
885 kAudioDevicePropertyHogMode,
886 sizeof(ao->i_hog_pid), &ao->i_hog_pid);
887 if (err != noErr)
888 ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
889 (char *)&err);
891 av_fifo_free(ao->buffer);
892 free(ao);
893 ao = NULL;
894 return CONTROL_FALSE;
897 /*****************************************************************************
898 * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support.
899 *****************************************************************************/
900 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id )
902 UInt32 i_param_size = 0;
903 AudioStreamID *p_streams = NULL;
904 int i = 0, i_streams = 0;
905 int b_return = CONTROL_FALSE;
907 /* Retrieve all the output streams. */
908 i_param_size = GetAudioPropertyArray(i_dev_id,
909 kAudioDevicePropertyStreams,
910 kAudioDevicePropertyScopeOutput,
911 (void **)&p_streams);
913 if (!i_param_size) {
914 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n");
915 return CONTROL_FALSE;
918 i_streams = i_param_size / sizeof(AudioStreamID);
920 for (i = 0; i < i_streams; ++i)
922 if (AudioStreamSupportsDigital(p_streams[i]))
923 b_return = CONTROL_OK;
926 free(p_streams);
927 return b_return;
930 /*****************************************************************************
931 * AudioStreamSupportsDigital: Check i_stream_id for digital stream support.
932 *****************************************************************************/
933 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id )
935 UInt32 i_param_size;
936 AudioStreamRangedDescription *p_format_list = NULL;
937 int i, i_formats, b_return = CONTROL_FALSE;
939 /* Retrieve all the stream formats supported by each output stream. */
940 i_param_size = GetGlobalAudioPropertyArray(i_stream_id,
941 kAudioStreamPropertyAvailablePhysicalFormats,
942 (void **)&p_format_list);
944 if (!i_param_size) {
945 ao_msg(MSGT_AO, MSGL_WARN, "Could not get number of stream formats.\n");
946 return CONTROL_FALSE;
949 i_formats = i_param_size / sizeof(AudioStreamRangedDescription);
951 for (i = 0; i < i_formats; ++i)
953 print_format(MSGL_V, "supported format:", &(p_format_list[i].mFormat));
955 if (p_format_list[i].mFormat.mFormatID == 'IAC3' ||
956 p_format_list[i].mFormat.mFormatID == 'iac3' ||
957 p_format_list[i].mFormat.mFormatID == kAudioFormat60958AC3 ||
958 p_format_list[i].mFormat.mFormatID == kAudioFormatAC3)
959 b_return = CONTROL_OK;
962 free(p_format_list);
963 return b_return;
966 /*****************************************************************************
967 * AudioStreamChangeFormat: Change i_stream_id to change_format
968 *****************************************************************************/
969 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format )
971 OSStatus err = noErr;
972 int i;
973 AudioObjectPropertyAddress property_address;
975 static volatile int stream_format_changed;
976 stream_format_changed = 0;
978 print_format(MSGL_V, "setting stream format:", &change_format);
980 /* Install the callback. */
981 property_address.mSelector = kAudioStreamPropertyPhysicalFormat;
982 property_address.mScope = kAudioObjectPropertyScopeGlobal;
983 property_address.mElement = kAudioObjectPropertyElementMaster;
985 err = AudioObjectAddPropertyListener(i_stream_id,
986 &property_address,
987 StreamListener,
988 (void *)&stream_format_changed);
989 if (err != noErr)
991 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err);
992 return CONTROL_FALSE;
995 /* Change the format. */
996 err = SetAudioProperty(i_stream_id,
997 kAudioStreamPropertyPhysicalFormat,
998 sizeof(AudioStreamBasicDescription), &change_format);
999 if (err != noErr)
1001 ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err);
1002 return CONTROL_FALSE;
1005 /* The AudioStreamSetProperty is not only asynchronious,
1006 * it is also not Atomic, in its behaviour.
1007 * Therefore we check 5 times before we really give up.
1008 * FIXME: failing isn't actually implemented yet. */
1009 for (i = 0; i < 5; ++i)
1011 AudioStreamBasicDescription actual_format;
1012 int j;
1013 for (j = 0; !stream_format_changed && j < 50; ++j)
1014 usec_sleep(10000);
1015 if (stream_format_changed)
1016 stream_format_changed = 0;
1017 else
1018 ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" );
1020 err = GetAudioProperty(i_stream_id,
1021 kAudioStreamPropertyPhysicalFormat,
1022 sizeof(AudioStreamBasicDescription), &actual_format);
1024 print_format(MSGL_V, "actual format in use:", &actual_format);
1025 if (actual_format.mSampleRate == change_format.mSampleRate &&
1026 actual_format.mFormatID == change_format.mFormatID &&
1027 actual_format.mFramesPerPacket == change_format.mFramesPerPacket)
1029 /* The right format is now active. */
1030 break;
1032 /* We need to check again. */
1035 /* Removing the property listener. */
1036 err = AudioObjectRemovePropertyListener(i_stream_id,
1037 &property_address,
1038 StreamListener,
1039 (void *)&stream_format_changed);
1040 if (err != noErr)
1042 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err);
1043 return CONTROL_FALSE;
1046 return CONTROL_TRUE;
1049 /*****************************************************************************
1050 * RenderCallbackSPDIF: callback for SPDIF audio output
1051 *****************************************************************************/
1052 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
1053 const AudioTimeStamp * inNow,
1054 const void * inInputData,
1055 const AudioTimeStamp * inInputTime,
1056 AudioBufferList * outOutputData,
1057 const AudioTimeStamp * inOutputTime,
1058 void * threadGlobals )
1060 int amt = av_fifo_size(ao->buffer);
1061 int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;
1063 if (amt > req)
1064 amt = req;
1065 if (amt)
1066 read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt);
1068 return noErr;
1072 static int play(void* output_samples,int num_bytes,int flags)
1074 int wrote, b_digital;
1075 SInt32 exit_reason;
1077 // Check whether we need to reset the digital output stream.
1078 if (ao->b_digital && ao->b_stream_format_changed)
1080 ao->b_stream_format_changed = 0;
1081 b_digital = AudioStreamSupportsDigital(ao->i_stream_id);
1082 if (b_digital)
1084 /* Current stream supports digital format output, let's set it. */
1085 ao_msg(MSGT_AO, MSGL_V,
1086 "Detected current stream supports digital, try to restore digital output...\n");
1088 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
1090 ao_msg(MSGT_AO, MSGL_WARN, "Restoring digital output failed.\n");
1092 else
1094 ao_msg(MSGT_AO, MSGL_WARN, "Restoring digital output succeeded.\n");
1095 reset();
1098 else
1099 ao_msg(MSGT_AO, MSGL_V, "Detected current stream does not support digital.\n");
1102 wrote=write_buffer(output_samples, num_bytes);
1103 audio_resume();
1105 do {
1106 exit_reason = CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.01, true);
1107 } while (exit_reason == kCFRunLoopRunHandledSource);
1109 return wrote;
1112 /* set variables and buffer to initial state */
1113 static void reset(void)
1115 audio_pause();
1116 av_fifo_reset(ao->buffer);
1120 /* return available space */
1121 static int get_space(void)
1123 return ao->buffer_len - av_fifo_size(ao->buffer);
1127 /* return delay until audio is played */
1128 static float get_delay(void)
1130 // inaccurate, should also contain the data buffered e.g. by the OS
1131 return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps;
1135 /* unload plugin and deregister from coreaudio */
1136 static void uninit(int immed)
1138 OSStatus err = noErr;
1140 if (!immed) {
1141 long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps;
1142 ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft);
1143 usec_sleep((int)timeleft);
1146 if (!ao->b_digital) {
1147 AudioOutputUnitStop(ao->theOutputUnit);
1148 AudioUnitUninitialize(ao->theOutputUnit);
1149 CloseComponent(ao->theOutputUnit);
1151 else {
1152 /* Stop device. */
1153 err = AudioDeviceStop(ao->i_selected_dev, ao->renderCallback);
1154 if (err != noErr)
1155 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1157 /* Remove IOProc callback. */
1158 err = AudioDeviceDestroyIOProcID(ao->i_selected_dev, ao->renderCallback);
1159 if (err != noErr)
1160 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err);
1162 if (ao->b_revert)
1163 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
1165 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
1167 UInt32 b_mix;
1168 Boolean b_writeable = 0;
1169 /* Revert mixable to true if we are allowed to. */
1170 err = IsAudioPropertySettable(ao->i_selected_dev,
1171 kAudioDevicePropertySupportsMixing,
1172 &b_writeable);
1173 err = GetAudioProperty(ao->i_selected_dev,
1174 kAudioDevicePropertySupportsMixing,
1175 sizeof(UInt32), &b_mix);
1176 if (err == noErr && b_writeable)
1178 b_mix = 1;
1179 err = SetAudioProperty(ao->i_selected_dev,
1180 kAudioDevicePropertySupportsMixing,
1181 sizeof(UInt32), &b_mix);
1183 if (err != noErr)
1184 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
1186 if (ao->i_hog_pid == getpid())
1188 ao->i_hog_pid = -1;
1189 err = SetAudioProperty(ao->i_selected_dev,
1190 kAudioDevicePropertyHogMode,
1191 sizeof(ao->i_hog_pid), &ao->i_hog_pid);
1192 if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err);
1196 av_fifo_free(ao->buffer);
1197 free(ao);
1198 ao = NULL;
1202 /* stop playing, keep buffers (for pause) */
1203 static void audio_pause(void)
1205 OSErr err=noErr;
1207 /* Stop callback. */
1208 if (!ao->b_digital)
1210 err=AudioOutputUnitStop(ao->theOutputUnit);
1211 if (err != noErr)
1212 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err);
1214 else
1216 err = AudioDeviceStop(ao->i_selected_dev, ao->renderCallback);
1217 if (err != noErr)
1218 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
1220 ao->paused = 1;
1224 /* resume playing, after audio_pause() */
1225 static void audio_resume(void)
1227 OSErr err=noErr;
1229 if (!ao->paused)
1230 return;
1232 /* Start callback. */
1233 if (!ao->b_digital)
1235 err = AudioOutputUnitStart(ao->theOutputUnit);
1236 if (err != noErr)
1237 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err);
1239 else
1241 err = AudioDeviceStart(ao->i_selected_dev, ao->renderCallback);
1242 if (err != noErr)
1243 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err);
1245 ao->paused = 0;
1248 /*****************************************************************************
1249 * StreamListener
1250 *****************************************************************************/
1251 static OSStatus StreamListener( AudioObjectID inObjectID,
1252 UInt32 inNumberAddresses,
1253 const AudioObjectPropertyAddress inAddresses[],
1254 void *inClientData )
1256 for (int i=0; i < inNumberAddresses; ++i)
1258 if (inAddresses[i].mSelector == kAudioStreamPropertyPhysicalFormat) {
1259 ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioStreamPropertyPhysicalFormat changed.\n");
1260 if (inClientData)
1261 *(volatile int *)inClientData = 1;
1262 break;
1265 return noErr;
1268 static OSStatus DeviceListener( AudioObjectID inObjectID,
1269 UInt32 inNumberAddresses,
1270 const AudioObjectPropertyAddress inAddresses[],
1271 void *inClientData )
1273 for (int i=0; i < inNumberAddresses; ++i)
1275 if (inAddresses[i].mSelector == kAudioDevicePropertyDeviceHasChanged) {
1276 ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n");
1277 ao->b_stream_format_changed = 1;
1278 break;
1281 return noErr;