Mercurial > mplayer.hg
annotate libao2/ao_coreaudio.c @ 29697:1cd8a8e2c144
Simplify aac_get_sample_rate_index
author | reimar |
---|---|
date | Wed, 30 Sep 2009 07:52:37 +0000 |
parents | 02dec439f717 |
children | 9d09ff127c25 |
rev | line source |
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29209 | 1 /* |
2 * CoreAudio audio output driver for Mac OS X | |
3 * | |
4 * original copyright (C) Timothy J. Wood - Aug 2000 | |
5 * ported to MPlayer libao2 by Dan Christiansen | |
6 * | |
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> | |
10 * | |
11 * This file is part of MPlayer. | |
12 * | |
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. | |
17 * | |
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. | |
22 * | |
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 | |
26 */ | |
27 | |
28 /* | |
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). | |
33 * | |
34 * AC-3 and MPEG audio passthrough is possible, but has never been tested | |
35 * due to lack of a soundcard that supports it. | |
36 */ | |
37 | |
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> | |
47 | |
48 #include "config.h" | |
49 #include "mp_msg.h" | |
50 | |
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 | |
57 static const ao_info_t info = | |
58 { | |
59 "Darwin/Mac OS X native audio output", | |
60 "coreaudio", | |
61 "Timothy J. Wood & Dan Christiansen & Chris Roccati", | |
62 "" | |
63 }; | |
64 | |
65 LIBAO_EXTERN(coreaudio) | |
66 | |
67 /* Prefix for all mp_msg() calls */ | |
68 #define ao_msg(a, b, c...) mp_msg(a, b, "AO: [coreaudio] " c) | |
69 | |
70 typedef struct ao_coreaudio_s | |
71 { | |
72 AudioDeviceID i_selected_dev; /* Keeps DeviceID of the selected device. */ | |
73 int b_supports_digital; /* Does the currently selected device support digital mode? */ | |
74 int b_digital; /* Are we running in digital mode? */ | |
75 int b_muted; /* Are we muted in digital mode? */ | |
76 | |
77 /* AudioUnit */ | |
78 AudioUnit theOutputUnit; | |
79 | |
80 /* CoreAudio SPDIF mode specific */ | |
81 pid_t i_hog_pid; /* Keeps the pid of our hog status. */ | |
82 AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */ | |
83 int i_stream_index; /* The index of i_stream_id in an AudioBufferList */ | |
84 AudioStreamBasicDescription stream_format;/* The format we changed the stream to */ | |
85 AudioStreamBasicDescription sfmt_revert; /* The original format of the stream */ | |
86 int b_revert; /* Whether we need to revert the stream format */ | |
87 int b_changed_mixing; /* Whether we need to set the mixing mode back */ | |
88 int b_stream_format_changed; /* Flag for main thread to reset stream's format to digital and reset buffer */ | |
89 | |
90 /* Original common part */ | |
91 int packetSize; | |
92 int paused; | |
93 | |
94 /* Ring-buffer */ | |
95 AVFifoBuffer *buffer; | |
96 unsigned int buffer_len; ///< must always be num_chunks * chunk_size | |
97 unsigned int num_chunks; | |
98 unsigned int chunk_size; | |
99 } ao_coreaudio_t; | |
100 | |
101 static ao_coreaudio_t *ao = NULL; | |
102 | |
103 /** | |
104 * \brief add data to ringbuffer | |
105 */ | |
106 static int write_buffer(unsigned char* data, int len){ | |
107 int free = ao->buffer_len - av_fifo_size(ao->buffer); | |
108 if (len > free) len = free; | |
109 return av_fifo_generic_write(ao->buffer, data, len, NULL); | |
110 } | |
111 | |
112 /** | |
113 * \brief remove data from ringbuffer | |
114 */ | |
115 static int read_buffer(unsigned char* data,int len){ | |
116 int buffered = av_fifo_size(ao->buffer); | |
117 if (len > buffered) len = buffered; | |
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118 av_fifo_generic_read(ao->buffer, data, len, NULL); |
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119 return len; |
29209 | 120 } |
121 | |
122 OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData) | |
123 { | |
124 int amt=av_fifo_size(ao->buffer); | |
125 int req=(inNumFrames)*ao->packetSize; | |
126 | |
127 if(amt>req) | |
128 amt=req; | |
129 | |
130 if(amt) | |
131 read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt); | |
132 else audio_pause(); | |
133 ioData->mBuffers[0].mDataByteSize = amt; | |
134 | |
135 return noErr; | |
136 } | |
137 | |
138 static int control(int cmd,void *arg){ | |
139 ao_control_vol_t *control_vol; | |
140 OSStatus err; | |
141 Float32 vol; | |
142 switch (cmd) { | |
143 case AOCONTROL_GET_VOLUME: | |
144 control_vol = (ao_control_vol_t*)arg; | |
145 if (ao->b_digital) { | |
146 // Digital output has no volume adjust. | |
147 return CONTROL_FALSE; | |
148 } | |
149 err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol); | |
150 | |
151 if(err==0) { | |
152 // printf("GET VOL=%f\n", vol); | |
153 control_vol->left=control_vol->right=vol*100.0/4.0; | |
154 return CONTROL_TRUE; | |
155 } | |
156 else { | |
157 ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err); | |
158 return CONTROL_FALSE; | |
159 } | |
160 | |
161 case AOCONTROL_SET_VOLUME: | |
162 control_vol = (ao_control_vol_t*)arg; | |
163 | |
164 if (ao->b_digital) { | |
165 // Digital output can not set volume. Here we have to return true | |
166 // to make mixer forget it. Else mixer will add a soft filter, | |
167 // that's not we expected and the filter not support ac3 stream | |
168 // will cause mplayer die. | |
169 | |
170 // Although not support set volume, but at least we support mute. | |
171 // MPlayer set mute by set volume to zero, we handle it. | |
172 if (control_vol->left == 0 && control_vol->right == 0) | |
173 ao->b_muted = 1; | |
174 else | |
175 ao->b_muted = 0; | |
176 return CONTROL_TRUE; | |
177 } | |
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178 |
29209 | 179 vol=(control_vol->left+control_vol->right)*4.0/200.0; |
180 err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0); | |
181 if(err==0) { | |
182 // printf("SET VOL=%f\n", vol); | |
183 return CONTROL_TRUE; | |
184 } | |
185 else { | |
186 ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err); | |
187 return CONTROL_FALSE; | |
188 } | |
189 /* Everything is currently unimplemented */ | |
190 default: | |
191 return CONTROL_FALSE; | |
192 } | |
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193 |
29209 | 194 } |
195 | |
196 | |
197 static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){ | |
198 uint32_t flags=(uint32_t) f->mFormatFlags; | |
199 ao_msg(MSGT_AO,lev, "%s %7.1fHz %lubit [%c%c%c%c][%lu][%lu][%lu][%lu][%lu] %s %s %s%s%s%s\n", | |
200 str, f->mSampleRate, f->mBitsPerChannel, | |
201 (int)(f->mFormatID & 0xff000000) >> 24, | |
202 (int)(f->mFormatID & 0x00ff0000) >> 16, | |
203 (int)(f->mFormatID & 0x0000ff00) >> 8, | |
204 (int)(f->mFormatID & 0x000000ff) >> 0, | |
205 f->mFormatFlags, f->mBytesPerPacket, | |
206 f->mFramesPerPacket, f->mBytesPerFrame, | |
207 f->mChannelsPerFrame, | |
208 (flags&kAudioFormatFlagIsFloat) ? "float" : "int", | |
209 (flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE", | |
210 (flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U", | |
211 (flags&kAudioFormatFlagIsPacked) ? " packed" : "", | |
212 (flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "", | |
213 (flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" ); | |
214 } | |
215 | |
216 | |
217 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id ); | |
218 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id ); | |
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219 static int OpenSPDIF(void); |
29209 | 220 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format ); |
221 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice, | |
222 const AudioTimeStamp * inNow, | |
223 const void * inInputData, | |
224 const AudioTimeStamp * inInputTime, | |
225 AudioBufferList * outOutputData, | |
226 const AudioTimeStamp * inOutputTime, | |
227 void * threadGlobals ); | |
228 static OSStatus StreamListener( AudioStreamID inStream, | |
229 UInt32 inChannel, | |
230 AudioDevicePropertyID inPropertyID, | |
231 void * inClientData ); | |
232 static OSStatus DeviceListener( AudioDeviceID inDevice, | |
233 UInt32 inChannel, | |
234 Boolean isInput, | |
235 AudioDevicePropertyID inPropertyID, | |
236 void* inClientData ); | |
237 | |
238 static int init(int rate,int channels,int format,int flags) | |
239 { | |
240 AudioStreamBasicDescription inDesc; | |
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241 ComponentDescription desc; |
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242 Component comp; |
29209 | 243 AURenderCallbackStruct renderCallback; |
244 OSStatus err; | |
245 UInt32 size, maxFrames, i_param_size; | |
246 char *psz_name; | |
247 AudioDeviceID devid_def = 0; | |
248 int b_alive; | |
249 | |
250 ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags); | |
251 | |
252 ao = calloc(1, sizeof(ao_coreaudio_t)); | |
253 | |
254 ao->i_selected_dev = 0; | |
255 ao->b_supports_digital = 0; | |
256 ao->b_digital = 0; | |
257 ao->b_muted = 0; | |
258 ao->b_stream_format_changed = 0; | |
259 ao->i_hog_pid = -1; | |
260 ao->i_stream_id = 0; | |
261 ao->i_stream_index = -1; | |
262 ao->b_revert = 0; | |
263 ao->b_changed_mixing = 0; | |
264 | |
265 /* Probe whether device support S/PDIF stream output if input is AC3 stream. */ | |
266 if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3) | |
267 { | |
268 /* Find the ID of the default Device. */ | |
269 i_param_size = sizeof(AudioDeviceID); | |
270 err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, | |
271 &i_param_size, &devid_def); | |
272 if (err != noErr) | |
273 { | |
274 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err); | |
275 goto err_out; | |
276 } | |
277 | |
278 /* Retrieve the length of the device name. */ | |
279 i_param_size = 0; | |
280 err = AudioDeviceGetPropertyInfo(devid_def, 0, 0, | |
281 kAudioDevicePropertyDeviceName, | |
282 &i_param_size, NULL); | |
283 if (err != noErr) | |
284 { | |
285 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name length: [%4.4s]\n", (char *)&err); | |
286 goto err_out; | |
287 } | |
288 | |
289 /* Retrieve the name of the device. */ | |
290 psz_name = (char *)malloc(i_param_size); | |
291 err = AudioDeviceGetProperty(devid_def, 0, 0, | |
292 kAudioDevicePropertyDeviceName, | |
293 &i_param_size, psz_name); | |
294 if (err != noErr) | |
295 { | |
296 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err); | |
297 free( psz_name); | |
298 goto err_out; | |
299 } | |
300 | |
301 ao_msg(MSGT_AO,MSGL_V, "got default audio output device ID: %#lx Name: %s\n", devid_def, psz_name ); | |
302 | |
303 if (AudioDeviceSupportsDigital(devid_def)) | |
304 { | |
305 ao->b_supports_digital = 1; | |
306 ao->i_selected_dev = devid_def; | |
307 } | |
308 ao_msg(MSGT_AO,MSGL_V, "probe default audio output device whether support digital s/pdif output:%d\n", ao->b_supports_digital ); | |
309 | |
310 free( psz_name); | |
311 } | |
312 | |
313 // Build Description for the input format | |
314 inDesc.mSampleRate=rate; | |
315 inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM; | |
316 inDesc.mChannelsPerFrame=channels; | |
317 switch(format&AF_FORMAT_BITS_MASK){ | |
318 case AF_FORMAT_8BIT: | |
319 inDesc.mBitsPerChannel=8; | |
320 break; | |
321 case AF_FORMAT_16BIT: | |
322 inDesc.mBitsPerChannel=16; | |
323 break; | |
324 case AF_FORMAT_24BIT: | |
325 inDesc.mBitsPerChannel=24; | |
326 break; | |
327 case AF_FORMAT_32BIT: | |
328 inDesc.mBitsPerChannel=32; | |
329 break; | |
330 default: | |
331 ao_msg(MSGT_AO, MSGL_WARN, "Unsupported format (0x%08x)\n", format); | |
332 goto err_out; | |
333 } | |
334 | |
335 if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) { | |
336 // float | |
337 inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked; | |
338 } | |
339 else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) { | |
340 // signed int | |
341 inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked; | |
342 } | |
343 else { | |
344 // unsigned int | |
345 inDesc.mFormatFlags = kAudioFormatFlagIsPacked; | |
346 } | |
347 if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3) { | |
348 // Currently ac3 input (comes from hwac3) is always in native byte-order. | |
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349 #if HAVE_BIGENDIAN |
29209 | 350 inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian; |
351 #endif | |
352 } | |
353 else if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE) | |
354 inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian; | |
355 | |
356 inDesc.mFramesPerPacket = 1; | |
357 ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8); | |
358 print_format(MSGL_V, "source:",&inDesc); | |
359 | |
360 if (ao->b_supports_digital) | |
361 { | |
362 b_alive = 1; | |
363 i_param_size = sizeof(b_alive); | |
364 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
365 kAudioDevicePropertyDeviceIsAlive, | |
366 &i_param_size, &b_alive); | |
367 if (err != noErr) | |
368 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err); | |
369 if (!b_alive) | |
370 ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" ); | |
371 /* S/PDIF output need device in HogMode. */ | |
372 i_param_size = sizeof(ao->i_hog_pid); | |
373 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
374 kAudioDevicePropertyHogMode, | |
375 &i_param_size, &ao->i_hog_pid); | |
376 | |
377 if (err != noErr) | |
378 { | |
379 /* This is not a fatal error. Some drivers simply don't support this property. */ | |
380 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n", | |
381 (char *)&err); | |
382 ao->i_hog_pid = -1; | |
383 } | |
384 | |
385 if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid()) | |
386 { | |
387 ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" ); | |
388 goto err_out; | |
389 } | |
390 ao->stream_format = inDesc; | |
391 return OpenSPDIF(); | |
392 } | |
393 | |
394 /* original analog output code */ | |
395 desc.componentType = kAudioUnitType_Output; | |
396 desc.componentSubType = kAudioUnitSubType_DefaultOutput; | |
397 desc.componentManufacturer = kAudioUnitManufacturer_Apple; | |
398 desc.componentFlags = 0; | |
399 desc.componentFlagsMask = 0; | |
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400 |
29209 | 401 comp = FindNextComponent(NULL, &desc); //Finds an component that meets the desc spec's |
402 if (comp == NULL) { | |
403 ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n"); | |
404 goto err_out; | |
405 } | |
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406 |
29209 | 407 err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component |
408 if (err) { | |
409 ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err); | |
410 goto err_out; | |
411 } | |
412 | |
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413 // Initialize AudioUnit |
29209 | 414 err = AudioUnitInitialize(ao->theOutputUnit); |
415 if (err) { | |
416 ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err); | |
417 goto err_out1; | |
418 } | |
419 | |
420 size = sizeof(AudioStreamBasicDescription); | |
421 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size); | |
422 | |
423 if (err) { | |
424 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err); | |
425 goto err_out2; | |
426 } | |
427 | |
428 size = sizeof(UInt32); | |
429 err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size); | |
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430 |
29209 | 431 if (err) |
432 { | |
433 ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err); | |
434 goto err_out2; | |
435 } | |
436 | |
437 ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame; | |
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438 |
29209 | 439 ao_data.samplerate = inDesc.mSampleRate; |
440 ao_data.channels = inDesc.mChannelsPerFrame; | |
441 ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame; | |
442 ao_data.outburst = ao->chunk_size; | |
443 ao_data.buffersize = ao_data.bps; | |
444 | |
445 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size; | |
446 ao->buffer_len = ao->num_chunks * ao->chunk_size; | |
447 ao->buffer = av_fifo_alloc(ao->buffer_len); | |
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448 |
29209 | 449 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); |
450 | |
451 renderCallback.inputProc = theRenderProc; | |
452 renderCallback.inputProcRefCon = 0; | |
453 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct)); | |
454 if (err) { | |
455 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err); | |
456 goto err_out2; | |
457 } | |
458 | |
459 reset(); | |
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460 |
29209 | 461 return CONTROL_OK; |
462 | |
463 err_out2: | |
464 AudioUnitUninitialize(ao->theOutputUnit); | |
465 err_out1: | |
466 CloseComponent(ao->theOutputUnit); | |
467 err_out: | |
468 av_fifo_free(ao->buffer); | |
469 free(ao); | |
470 ao = NULL; | |
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471 return CONTROL_FALSE; |
29209 | 472 } |
473 | |
474 /***************************************************************************** | |
475 * Setup a encoded digital stream (SPDIF) | |
476 *****************************************************************************/ | |
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477 static int OpenSPDIF(void) |
29209 | 478 { |
479 OSStatus err = noErr; | |
480 UInt32 i_param_size, b_mix = 0; | |
481 Boolean b_writeable = 0; | |
482 AudioStreamID *p_streams = NULL; | |
483 int i, i_streams = 0; | |
484 | |
485 /* Start doing the SPDIF setup process. */ | |
486 ao->b_digital = 1; | |
487 | |
488 /* Hog the device. */ | |
489 i_param_size = sizeof(ao->i_hog_pid); | |
490 ao->i_hog_pid = getpid() ; | |
491 | |
492 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
493 kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid); | |
494 | |
495 if (err != noErr) | |
496 { | |
497 ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err); | |
498 ao->i_hog_pid = -1; | |
499 goto err_out; | |
500 } | |
501 | |
502 /* Set mixable to false if we are allowed to. */ | |
503 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, | |
504 kAudioDevicePropertySupportsMixing, | |
505 &i_param_size, &b_writeable); | |
506 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
507 kAudioDevicePropertySupportsMixing, | |
508 &i_param_size, &b_mix); | |
509 if (err != noErr && b_writeable) | |
510 { | |
511 b_mix = 0; | |
512 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
513 kAudioDevicePropertySupportsMixing, | |
514 i_param_size, &b_mix); | |
515 ao->b_changed_mixing = 1; | |
516 } | |
517 if (err != noErr) | |
518 { | |
519 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); | |
520 goto err_out; | |
521 } | |
522 | |
523 /* Get a list of all the streams on this device. */ | |
524 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, | |
525 kAudioDevicePropertyStreams, | |
526 &i_param_size, NULL); | |
527 if (err != noErr) | |
528 { | |
529 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
530 goto err_out; | |
531 } | |
532 | |
533 i_streams = i_param_size / sizeof(AudioStreamID); | |
534 p_streams = (AudioStreamID *)malloc(i_param_size); | |
535 if (p_streams == NULL) | |
536 { | |
537 ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" ); | |
538 goto err_out; | |
539 } | |
540 | |
541 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
542 kAudioDevicePropertyStreams, | |
543 &i_param_size, p_streams); | |
544 if (err != noErr) | |
545 { | |
546 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
547 if (p_streams) free(p_streams); | |
548 goto err_out; | |
549 } | |
550 | |
551 ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams); | |
552 | |
553 for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i) | |
554 { | |
555 /* Find a stream with a cac3 stream. */ | |
556 AudioStreamBasicDescription *p_format_list = NULL; | |
557 int i_formats = 0, j = 0, b_digital = 0; | |
558 | |
559 /* Retrieve all the stream formats supported by each output stream. */ | |
560 err = AudioStreamGetPropertyInfo(p_streams[i], 0, | |
561 kAudioStreamPropertyPhysicalFormats, | |
562 &i_param_size, NULL); | |
563 if (err != noErr) | |
564 { | |
565 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streamformats: [%4.4s]\n", (char *)&err); | |
566 continue; | |
567 } | |
568 | |
569 i_formats = i_param_size / sizeof(AudioStreamBasicDescription); | |
570 p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size); | |
571 if (p_format_list == NULL) | |
572 { | |
573 ao_msg(MSGT_AO, MSGL_WARN, "could not malloc the memory\n" ); | |
574 continue; | |
575 } | |
576 | |
577 err = AudioStreamGetProperty(p_streams[i], 0, | |
578 kAudioStreamPropertyPhysicalFormats, | |
579 &i_param_size, p_format_list); | |
580 if (err != noErr) | |
581 { | |
582 ao_msg(MSGT_AO, MSGL_WARN, "could not get the list of streamformats: [%4.4s]\n", (char *)&err); | |
583 if (p_format_list) free(p_format_list); | |
584 continue; | |
585 } | |
586 | |
587 /* Check if one of the supported formats is a digital format. */ | |
588 for (j = 0; j < i_formats; ++j) | |
589 { | |
590 if (p_format_list[j].mFormatID == 'IAC3' || | |
591 p_format_list[j].mFormatID == kAudioFormat60958AC3) | |
592 { | |
593 b_digital = 1; | |
594 break; | |
595 } | |
596 } | |
597 | |
598 if (b_digital) | |
599 { | |
600 /* If this stream supports a digital (cac3) format, then set it. */ | |
601 int i_requested_rate_format = -1; | |
602 int i_current_rate_format = -1; | |
603 int i_backup_rate_format = -1; | |
604 | |
605 ao->i_stream_id = p_streams[i]; | |
606 ao->i_stream_index = i; | |
607 | |
608 if (ao->b_revert == 0) | |
609 { | |
610 /* Retrieve the original format of this stream first if not done so already. */ | |
611 i_param_size = sizeof(ao->sfmt_revert); | |
612 err = AudioStreamGetProperty(ao->i_stream_id, 0, | |
613 kAudioStreamPropertyPhysicalFormat, | |
614 &i_param_size, | |
615 &ao->sfmt_revert); | |
616 if (err != noErr) | |
617 { | |
618 ao_msg(MSGT_AO, MSGL_WARN, "could not retrieve the original streamformat: [%4.4s]\n", (char *)&err); | |
619 if (p_format_list) free(p_format_list); | |
620 continue; | |
621 } | |
622 ao->b_revert = 1; | |
623 } | |
624 | |
625 for (j = 0; j < i_formats; ++j) | |
626 if (p_format_list[j].mFormatID == 'IAC3' || | |
627 p_format_list[j].mFormatID == kAudioFormat60958AC3) | |
628 { | |
629 if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate) | |
630 { | |
631 i_requested_rate_format = j; | |
632 break; | |
633 } | |
634 if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate) | |
635 i_current_rate_format = j; | |
636 else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate) | |
637 i_backup_rate_format = j; | |
638 } | |
639 | |
640 if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */ | |
641 ao->stream_format = p_format_list[i_requested_rate_format]; | |
642 else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */ | |
643 ao->stream_format = p_format_list[i_current_rate_format]; | |
644 else ao->stream_format = p_format_list[i_backup_rate_format]; /* And if we have to, any digital format will be just fine (highest rate possible). */ | |
645 } | |
646 if (p_format_list) free(p_format_list); | |
647 } | |
648 if (p_streams) free(p_streams); | |
649 | |
650 if (ao->i_stream_index < 0) | |
651 { | |
652 ao_msg(MSGT_AO, MSGL_WARN, "can not find any digital output stream format when OpenSPDIF().\n"); | |
653 goto err_out; | |
654 } | |
655 | |
656 print_format(MSGL_V, "original stream format:", &ao->sfmt_revert); | |
657 | |
658 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) | |
659 goto err_out; | |
660 | |
661 err = AudioDeviceAddPropertyListener(ao->i_selected_dev, | |
662 kAudioPropertyWildcardChannel, | |
663 0, | |
664 kAudioDevicePropertyDeviceHasChanged, | |
665 DeviceListener, | |
666 NULL); | |
667 if (err != noErr) | |
668 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err); | |
669 | |
670 | |
671 /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */ | |
672 /* Although there's no such case reported. */ | |
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|
673 #if HAVE_BIGENDIAN |
29209 | 674 if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)) |
675 #else | |
676 if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian) | |
677 #endif | |
678 ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\n"); | |
679 | |
680 /* For ac3/dts, just use packet size 6144 bytes as chunk size. */ | |
681 ao->chunk_size = ao->stream_format.mBytesPerPacket; | |
682 | |
683 ao_data.samplerate = ao->stream_format.mSampleRate; | |
684 ao_data.channels = ao->stream_format.mChannelsPerFrame; | |
685 ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket); | |
686 ao_data.outburst = ao->chunk_size; | |
687 ao_data.buffersize = ao_data.bps; | |
688 | |
689 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size; | |
690 ao->buffer_len = ao->num_chunks * ao->chunk_size; | |
691 ao->buffer = av_fifo_alloc(ao->buffer_len); | |
692 | |
693 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); | |
694 | |
695 | |
696 /* Add IOProc callback. */ | |
697 err = AudioDeviceAddIOProc(ao->i_selected_dev, | |
698 (AudioDeviceIOProc)RenderCallbackSPDIF, | |
699 (void *)ao); | |
700 if (err != noErr) | |
701 { | |
702 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err); | |
703 goto err_out1; | |
704 } | |
705 | |
706 reset(); | |
707 | |
708 return CONTROL_TRUE; | |
709 | |
710 err_out1: | |
711 if (ao->b_revert) | |
712 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert); | |
713 err_out: | |
714 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3) | |
715 { | |
716 int b_mix = 1; | |
717 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
718 kAudioDevicePropertySupportsMixing, | |
719 i_param_size, &b_mix); | |
720 if (err != noErr) | |
721 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", | |
722 (char *)&err); | |
723 } | |
724 if (ao->i_hog_pid == getpid()) | |
725 { | |
726 ao->i_hog_pid = -1; | |
727 i_param_size = sizeof(ao->i_hog_pid); | |
728 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
729 kAudioDevicePropertyHogMode, | |
730 i_param_size, &ao->i_hog_pid); | |
731 if (err != noErr) | |
732 ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", | |
733 (char *)&err); | |
734 } | |
735 av_fifo_free(ao->buffer); | |
736 free(ao); | |
737 ao = NULL; | |
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738 return CONTROL_FALSE; |
29209 | 739 } |
740 | |
741 /***************************************************************************** | |
742 * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support. | |
743 *****************************************************************************/ | |
744 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id ) | |
745 { | |
746 OSStatus err = noErr; | |
747 UInt32 i_param_size = 0; | |
748 AudioStreamID *p_streams = NULL; | |
749 int i = 0, i_streams = 0; | |
750 int b_return = CONTROL_FALSE; | |
751 | |
752 /* Retrieve all the output streams. */ | |
753 err = AudioDeviceGetPropertyInfo(i_dev_id, 0, FALSE, | |
754 kAudioDevicePropertyStreams, | |
755 &i_param_size, NULL); | |
756 if (err != noErr) | |
757 { | |
758 ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
759 return CONTROL_FALSE; | |
760 } | |
761 | |
762 i_streams = i_param_size / sizeof(AudioStreamID); | |
763 p_streams = (AudioStreamID *)malloc(i_param_size); | |
764 if (p_streams == NULL) | |
765 { | |
766 ao_msg(MSGT_AO,MSGL_V, "out of memory\n"); | |
767 return CONTROL_FALSE; | |
768 } | |
769 | |
770 err = AudioDeviceGetProperty(i_dev_id, 0, FALSE, | |
771 kAudioDevicePropertyStreams, | |
772 &i_param_size, p_streams); | |
773 | |
774 if (err != noErr) | |
775 { | |
776 ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
777 free(p_streams); | |
778 return CONTROL_FALSE; | |
779 } | |
780 | |
781 for (i = 0; i < i_streams; ++i) | |
782 { | |
783 if (AudioStreamSupportsDigital(p_streams[i])) | |
784 b_return = CONTROL_OK; | |
785 } | |
786 | |
787 free(p_streams); | |
788 return b_return; | |
789 } | |
790 | |
791 /***************************************************************************** | |
792 * AudioStreamSupportsDigital: Check i_stream_id for digital stream support. | |
793 *****************************************************************************/ | |
794 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id ) | |
795 { | |
796 OSStatus err = noErr; | |
797 UInt32 i_param_size; | |
798 AudioStreamBasicDescription *p_format_list = NULL; | |
799 int i, i_formats, b_return = CONTROL_FALSE; | |
800 | |
801 /* Retrieve all the stream formats supported by each output stream. */ | |
802 err = AudioStreamGetPropertyInfo(i_stream_id, 0, | |
803 kAudioStreamPropertyPhysicalFormats, | |
804 &i_param_size, NULL); | |
805 if (err != noErr) | |
806 { | |
807 ao_msg(MSGT_AO,MSGL_V, "could not get number of streamformats: [%4.4s]\n", (char *)&err); | |
808 return CONTROL_FALSE; | |
809 } | |
810 | |
811 i_formats = i_param_size / sizeof(AudioStreamBasicDescription); | |
812 p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size); | |
813 if (p_format_list == NULL) | |
814 { | |
815 ao_msg(MSGT_AO,MSGL_V, "could not malloc the memory\n" ); | |
816 return CONTROL_FALSE; | |
817 } | |
818 | |
819 err = AudioStreamGetProperty(i_stream_id, 0, | |
820 kAudioStreamPropertyPhysicalFormats, | |
821 &i_param_size, p_format_list); | |
822 if (err != noErr) | |
823 { | |
824 ao_msg(MSGT_AO,MSGL_V, "could not get the list of streamformats: [%4.4s]\n", (char *)&err); | |
825 free(p_format_list); | |
826 return CONTROL_FALSE; | |
827 } | |
828 | |
829 for (i = 0; i < i_formats; ++i) | |
830 { | |
831 print_format(MSGL_V, "supported format:", &p_format_list[i]); | |
832 | |
833 if (p_format_list[i].mFormatID == 'IAC3' || | |
834 p_format_list[i].mFormatID == kAudioFormat60958AC3) | |
835 b_return = CONTROL_OK; | |
836 } | |
837 | |
838 free(p_format_list); | |
839 return b_return; | |
840 } | |
841 | |
842 /***************************************************************************** | |
843 * AudioStreamChangeFormat: Change i_stream_id to change_format | |
844 *****************************************************************************/ | |
845 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format ) | |
846 { | |
847 OSStatus err = noErr; | |
848 UInt32 i_param_size = 0; | |
849 int i; | |
850 | |
851 static volatile int stream_format_changed; | |
852 stream_format_changed = 0; | |
853 | |
854 print_format(MSGL_V, "setting stream format:", &change_format); | |
855 | |
856 /* Install the callback. */ | |
857 err = AudioStreamAddPropertyListener(i_stream_id, 0, | |
858 kAudioStreamPropertyPhysicalFormat, | |
859 StreamListener, | |
860 (void *)&stream_format_changed); | |
861 if (err != noErr) | |
862 { | |
863 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err); | |
864 return CONTROL_FALSE; | |
865 } | |
866 | |
867 /* Change the format. */ | |
868 err = AudioStreamSetProperty(i_stream_id, 0, 0, | |
869 kAudioStreamPropertyPhysicalFormat, | |
870 sizeof(AudioStreamBasicDescription), | |
871 &change_format); | |
872 if (err != noErr) | |
873 { | |
874 ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err); | |
875 return CONTROL_FALSE; | |
876 } | |
877 | |
878 /* The AudioStreamSetProperty is not only asynchronious, | |
879 * it is also not Atomic, in its behaviour. | |
880 * Therefore we check 5 times before we really give up. | |
881 * FIXME: failing isn't actually implemented yet. */ | |
882 for (i = 0; i < 5; ++i) | |
883 { | |
884 AudioStreamBasicDescription actual_format; | |
885 int j; | |
886 for (j = 0; !stream_format_changed && j < 50; ++j) | |
887 usec_sleep(10000); | |
888 if (stream_format_changed) | |
889 stream_format_changed = 0; | |
890 else | |
891 ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" ); | |
892 | |
893 i_param_size = sizeof(AudioStreamBasicDescription); | |
894 err = AudioStreamGetProperty(i_stream_id, 0, | |
895 kAudioStreamPropertyPhysicalFormat, | |
896 &i_param_size, | |
897 &actual_format); | |
898 | |
899 print_format(MSGL_V, "actual format in use:", &actual_format); | |
900 if (actual_format.mSampleRate == change_format.mSampleRate && | |
901 actual_format.mFormatID == change_format.mFormatID && | |
902 actual_format.mFramesPerPacket == change_format.mFramesPerPacket) | |
903 { | |
904 /* The right format is now active. */ | |
905 break; | |
906 } | |
907 /* We need to check again. */ | |
908 } | |
909 | |
910 /* Removing the property listener. */ | |
911 err = AudioStreamRemovePropertyListener(i_stream_id, 0, | |
912 kAudioStreamPropertyPhysicalFormat, | |
913 StreamListener); | |
914 if (err != noErr) | |
915 { | |
916 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err); | |
917 return CONTROL_FALSE; | |
918 } | |
919 | |
920 return CONTROL_TRUE; | |
921 } | |
922 | |
923 /***************************************************************************** | |
924 * RenderCallbackSPDIF: callback for SPDIF audio output | |
925 *****************************************************************************/ | |
926 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice, | |
927 const AudioTimeStamp * inNow, | |
928 const void * inInputData, | |
929 const AudioTimeStamp * inInputTime, | |
930 AudioBufferList * outOutputData, | |
931 const AudioTimeStamp * inOutputTime, | |
932 void * threadGlobals ) | |
933 { | |
934 int amt = av_fifo_size(ao->buffer); | |
935 int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize; | |
936 | |
937 if (amt > req) | |
938 amt = req; | |
939 if (amt) | |
940 read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt); | |
941 | |
942 return noErr; | |
943 } | |
944 | |
945 | |
946 static int play(void* output_samples,int num_bytes,int flags) | |
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|
947 { |
29209 | 948 int wrote, b_digital; |
949 | |
950 // Check whether we need to reset the digital output stream. | |
951 if (ao->b_digital && ao->b_stream_format_changed) | |
952 { | |
953 ao->b_stream_format_changed = 0; | |
954 b_digital = AudioStreamSupportsDigital(ao->i_stream_id); | |
955 if (b_digital) | |
956 { | |
957 /* Current stream support digital format output, let's set it. */ | |
958 ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n"); | |
959 | |
960 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) | |
961 { | |
962 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n"); | |
963 } | |
964 else | |
965 { | |
966 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n"); | |
967 reset(); | |
968 } | |
969 } | |
970 else | |
971 ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n"); | |
972 } | |
973 | |
974 wrote=write_buffer(output_samples, num_bytes); | |
975 audio_resume(); | |
976 return wrote; | |
977 } | |
978 | |
979 /* set variables and buffer to initial state */ | |
980 static void reset(void) | |
981 { | |
982 audio_pause(); | |
983 av_fifo_reset(ao->buffer); | |
984 } | |
985 | |
986 | |
987 /* return available space */ | |
988 static int get_space(void) | |
989 { | |
990 return ao->buffer_len - av_fifo_size(ao->buffer); | |
991 } | |
992 | |
993 | |
994 /* return delay until audio is played */ | |
995 static float get_delay(void) | |
996 { | |
997 // inaccurate, should also contain the data buffered e.g. by the OS | |
998 return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps; | |
999 } | |
1000 | |
1001 | |
1002 /* unload plugin and deregister from coreaudio */ | |
1003 static void uninit(int immed) | |
1004 { | |
1005 OSStatus err = noErr; | |
1006 UInt32 i_param_size = 0; | |
1007 | |
1008 if (!immed) { | |
1009 long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps; | |
1010 ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft); | |
1011 usec_sleep((int)timeleft); | |
1012 } | |
1013 | |
1014 if (!ao->b_digital) { | |
1015 AudioOutputUnitStop(ao->theOutputUnit); | |
1016 AudioUnitUninitialize(ao->theOutputUnit); | |
1017 CloseComponent(ao->theOutputUnit); | |
1018 } | |
1019 else { | |
1020 /* Stop device. */ | |
1021 err = AudioDeviceStop(ao->i_selected_dev, | |
1022 (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1023 if (err != noErr) | |
1024 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); | |
1025 | |
1026 /* Remove IOProc callback. */ | |
1027 err = AudioDeviceRemoveIOProc(ao->i_selected_dev, | |
1028 (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1029 if (err != noErr) | |
1030 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err); | |
1031 | |
1032 if (ao->b_revert) | |
1033 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert); | |
1034 | |
1035 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3) | |
1036 { | |
1037 int b_mix; | |
1038 Boolean b_writeable; | |
1039 /* Revert mixable to true if we are allowed to. */ | |
1040 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing, | |
1041 &i_param_size, &b_writeable); | |
1042 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing, | |
1043 &i_param_size, &b_mix); | |
1044 if (err != noErr && b_writeable) | |
1045 { | |
1046 b_mix = 1; | |
1047 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
1048 kAudioDevicePropertySupportsMixing, i_param_size, &b_mix); | |
1049 } | |
1050 if (err != noErr) | |
1051 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); | |
1052 } | |
1053 if (ao->i_hog_pid == getpid()) | |
1054 { | |
1055 ao->i_hog_pid = -1; | |
1056 i_param_size = sizeof(ao->i_hog_pid); | |
1057 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
1058 kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid); | |
1059 if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err); | |
1060 } | |
1061 } | |
1062 | |
1063 av_fifo_free(ao->buffer); | |
1064 free(ao); | |
1065 ao = NULL; | |
1066 } | |
1067 | |
1068 | |
1069 /* stop playing, keep buffers (for pause) */ | |
1070 static void audio_pause(void) | |
1071 { | |
1072 OSErr err=noErr; | |
1073 | |
1074 /* Stop callback. */ | |
1075 if (!ao->b_digital) | |
1076 { | |
1077 err=AudioOutputUnitStop(ao->theOutputUnit); | |
1078 if (err != noErr) | |
1079 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err); | |
1080 } | |
1081 else | |
1082 { | |
1083 err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1084 if (err != noErr) | |
1085 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); | |
1086 } | |
1087 ao->paused = 1; | |
1088 } | |
1089 | |
1090 | |
1091 /* resume playing, after audio_pause() */ | |
1092 static void audio_resume(void) | |
1093 { | |
1094 OSErr err=noErr; | |
1095 | |
1096 if (!ao->paused) | |
1097 return; | |
1098 | |
1099 /* Start callback. */ | |
1100 if (!ao->b_digital) | |
1101 { | |
1102 err = AudioOutputUnitStart(ao->theOutputUnit); | |
1103 if (err != noErr) | |
1104 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err); | |
1105 } | |
1106 else | |
1107 { | |
1108 err = AudioDeviceStart(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1109 if (err != noErr) | |
1110 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err); | |
1111 } | |
1112 ao->paused = 0; | |
1113 } | |
1114 | |
1115 /***************************************************************************** | |
1116 * StreamListener | |
1117 *****************************************************************************/ | |
1118 static OSStatus StreamListener( AudioStreamID inStream, | |
1119 UInt32 inChannel, | |
1120 AudioDevicePropertyID inPropertyID, | |
1121 void * inClientData ) | |
1122 { | |
1123 switch (inPropertyID) | |
1124 { | |
1125 case kAudioStreamPropertyPhysicalFormat: | |
1126 ao_msg(MSGT_AO, MSGL_V, "got notify kAudioStreamPropertyPhysicalFormat changed.\n"); | |
1127 if (inClientData) | |
1128 *(volatile int *)inClientData = 1; | |
1129 default: | |
1130 break; | |
1131 } | |
1132 return noErr; | |
1133 } | |
1134 | |
1135 static OSStatus DeviceListener( AudioDeviceID inDevice, | |
1136 UInt32 inChannel, | |
1137 Boolean isInput, | |
1138 AudioDevicePropertyID inPropertyID, | |
1139 void* inClientData ) | |
1140 { | |
1141 switch (inPropertyID) | |
1142 { | |
1143 case kAudioDevicePropertyDeviceHasChanged: | |
1144 ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n"); | |
1145 ao->b_stream_format_changed = 1; | |
1146 default: | |
1147 break; | |
1148 } | |
1149 return noErr; | |
1150 } |