Mercurial > mplayer.hg
annotate libao2/ao_coreaudio.c @ 31109:729ce4709ce1
Create output directory when copying stylesheet.
This eliminates one rule and simplifies dependencies.
author | diego |
---|---|
date | Tue, 11 May 2010 21:31:36 +0000 |
parents | 9fc9d1e788aa |
children | b0da003fadf2 |
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 | |
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122 static OSStatus theRenderProc(void *inRefCon, |
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123 AudioUnitRenderActionFlags *inActionFlags, |
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124 const AudioTimeStamp *inTimeStamp, |
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125 UInt32 inBusNumber, UInt32 inNumFrames, |
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126 AudioBufferList *ioData) |
29209 | 127 { |
128 int amt=av_fifo_size(ao->buffer); | |
129 int req=(inNumFrames)*ao->packetSize; | |
130 | |
131 if(amt>req) | |
132 amt=req; | |
133 | |
134 if(amt) | |
135 read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt); | |
136 else audio_pause(); | |
137 ioData->mBuffers[0].mDataByteSize = amt; | |
138 | |
139 return noErr; | |
140 } | |
141 | |
142 static int control(int cmd,void *arg){ | |
143 ao_control_vol_t *control_vol; | |
144 OSStatus err; | |
145 Float32 vol; | |
146 switch (cmd) { | |
147 case AOCONTROL_GET_VOLUME: | |
148 control_vol = (ao_control_vol_t*)arg; | |
149 if (ao->b_digital) { | |
150 // Digital output has no volume adjust. | |
151 return CONTROL_FALSE; | |
152 } | |
153 err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol); | |
154 | |
155 if(err==0) { | |
156 // printf("GET VOL=%f\n", vol); | |
157 control_vol->left=control_vol->right=vol*100.0/4.0; | |
158 return CONTROL_TRUE; | |
159 } | |
160 else { | |
161 ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err); | |
162 return CONTROL_FALSE; | |
163 } | |
164 | |
165 case AOCONTROL_SET_VOLUME: | |
166 control_vol = (ao_control_vol_t*)arg; | |
167 | |
168 if (ao->b_digital) { | |
169 // Digital output can not set volume. Here we have to return true | |
170 // to make mixer forget it. Else mixer will add a soft filter, | |
171 // that's not we expected and the filter not support ac3 stream | |
172 // will cause mplayer die. | |
173 | |
174 // Although not support set volume, but at least we support mute. | |
175 // MPlayer set mute by set volume to zero, we handle it. | |
176 if (control_vol->left == 0 && control_vol->right == 0) | |
177 ao->b_muted = 1; | |
178 else | |
179 ao->b_muted = 0; | |
180 return CONTROL_TRUE; | |
181 } | |
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182 |
29209 | 183 vol=(control_vol->left+control_vol->right)*4.0/200.0; |
184 err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0); | |
185 if(err==0) { | |
186 // printf("SET VOL=%f\n", vol); | |
187 return CONTROL_TRUE; | |
188 } | |
189 else { | |
190 ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err); | |
191 return CONTROL_FALSE; | |
192 } | |
193 /* Everything is currently unimplemented */ | |
194 default: | |
195 return CONTROL_FALSE; | |
196 } | |
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197 |
29209 | 198 } |
199 | |
200 | |
201 static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){ | |
202 uint32_t flags=(uint32_t) f->mFormatFlags; | |
203 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", | |
204 str, f->mSampleRate, f->mBitsPerChannel, | |
205 (int)(f->mFormatID & 0xff000000) >> 24, | |
206 (int)(f->mFormatID & 0x00ff0000) >> 16, | |
207 (int)(f->mFormatID & 0x0000ff00) >> 8, | |
208 (int)(f->mFormatID & 0x000000ff) >> 0, | |
209 f->mFormatFlags, f->mBytesPerPacket, | |
210 f->mFramesPerPacket, f->mBytesPerFrame, | |
211 f->mChannelsPerFrame, | |
212 (flags&kAudioFormatFlagIsFloat) ? "float" : "int", | |
213 (flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE", | |
214 (flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U", | |
215 (flags&kAudioFormatFlagIsPacked) ? " packed" : "", | |
216 (flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "", | |
217 (flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" ); | |
218 } | |
219 | |
220 | |
221 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id ); | |
222 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id ); | |
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223 static int OpenSPDIF(void); |
29209 | 224 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format ); |
225 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice, | |
226 const AudioTimeStamp * inNow, | |
227 const void * inInputData, | |
228 const AudioTimeStamp * inInputTime, | |
229 AudioBufferList * outOutputData, | |
230 const AudioTimeStamp * inOutputTime, | |
231 void * threadGlobals ); | |
232 static OSStatus StreamListener( AudioStreamID inStream, | |
233 UInt32 inChannel, | |
234 AudioDevicePropertyID inPropertyID, | |
235 void * inClientData ); | |
236 static OSStatus DeviceListener( AudioDeviceID inDevice, | |
237 UInt32 inChannel, | |
238 Boolean isInput, | |
239 AudioDevicePropertyID inPropertyID, | |
240 void* inClientData ); | |
241 | |
242 static int init(int rate,int channels,int format,int flags) | |
243 { | |
244 AudioStreamBasicDescription inDesc; | |
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245 ComponentDescription desc; |
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246 Component comp; |
29209 | 247 AURenderCallbackStruct renderCallback; |
248 OSStatus err; | |
249 UInt32 size, maxFrames, i_param_size; | |
250 char *psz_name; | |
251 AudioDeviceID devid_def = 0; | |
252 int b_alive; | |
253 | |
254 ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags); | |
255 | |
256 ao = calloc(1, sizeof(ao_coreaudio_t)); | |
257 | |
258 ao->i_selected_dev = 0; | |
259 ao->b_supports_digital = 0; | |
260 ao->b_digital = 0; | |
261 ao->b_muted = 0; | |
262 ao->b_stream_format_changed = 0; | |
263 ao->i_hog_pid = -1; | |
264 ao->i_stream_id = 0; | |
265 ao->i_stream_index = -1; | |
266 ao->b_revert = 0; | |
267 ao->b_changed_mixing = 0; | |
268 | |
269 /* Probe whether device support S/PDIF stream output if input is AC3 stream. */ | |
30234 | 270 if (AF_FORMAT_IS_AC3(format)) |
29209 | 271 { |
272 /* Find the ID of the default Device. */ | |
273 i_param_size = sizeof(AudioDeviceID); | |
274 err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, | |
275 &i_param_size, &devid_def); | |
276 if (err != noErr) | |
277 { | |
278 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err); | |
279 goto err_out; | |
280 } | |
281 | |
282 /* Retrieve the length of the device name. */ | |
283 i_param_size = 0; | |
284 err = AudioDeviceGetPropertyInfo(devid_def, 0, 0, | |
285 kAudioDevicePropertyDeviceName, | |
286 &i_param_size, NULL); | |
287 if (err != noErr) | |
288 { | |
289 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name length: [%4.4s]\n", (char *)&err); | |
290 goto err_out; | |
291 } | |
292 | |
293 /* Retrieve the name of the device. */ | |
30702 | 294 psz_name = malloc(i_param_size); |
29209 | 295 err = AudioDeviceGetProperty(devid_def, 0, 0, |
296 kAudioDevicePropertyDeviceName, | |
297 &i_param_size, psz_name); | |
298 if (err != noErr) | |
299 { | |
300 ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err); | |
301 free( psz_name); | |
302 goto err_out; | |
303 } | |
304 | |
305 ao_msg(MSGT_AO,MSGL_V, "got default audio output device ID: %#lx Name: %s\n", devid_def, psz_name ); | |
306 | |
307 if (AudioDeviceSupportsDigital(devid_def)) | |
308 { | |
309 ao->b_supports_digital = 1; | |
310 ao->i_selected_dev = devid_def; | |
311 } | |
312 ao_msg(MSGT_AO,MSGL_V, "probe default audio output device whether support digital s/pdif output:%d\n", ao->b_supports_digital ); | |
313 | |
314 free( psz_name); | |
315 } | |
316 | |
317 // Build Description for the input format | |
318 inDesc.mSampleRate=rate; | |
319 inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM; | |
320 inDesc.mChannelsPerFrame=channels; | |
30235 | 321 inDesc.mBitsPerChannel=af_fmt2bits(format); |
29209 | 322 |
323 if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) { | |
324 // float | |
325 inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked; | |
326 } | |
327 else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) { | |
328 // signed int | |
329 inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked; | |
330 } | |
331 else { | |
332 // unsigned int | |
333 inDesc.mFormatFlags = kAudioFormatFlagIsPacked; | |
334 } | |
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335 if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE) |
29209 | 336 inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian; |
337 | |
338 inDesc.mFramesPerPacket = 1; | |
339 ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8); | |
340 print_format(MSGL_V, "source:",&inDesc); | |
341 | |
342 if (ao->b_supports_digital) | |
343 { | |
344 b_alive = 1; | |
345 i_param_size = sizeof(b_alive); | |
346 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
347 kAudioDevicePropertyDeviceIsAlive, | |
348 &i_param_size, &b_alive); | |
349 if (err != noErr) | |
350 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err); | |
351 if (!b_alive) | |
352 ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" ); | |
353 /* S/PDIF output need device in HogMode. */ | |
354 i_param_size = sizeof(ao->i_hog_pid); | |
355 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
356 kAudioDevicePropertyHogMode, | |
357 &i_param_size, &ao->i_hog_pid); | |
358 | |
359 if (err != noErr) | |
360 { | |
361 /* This is not a fatal error. Some drivers simply don't support this property. */ | |
362 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n", | |
363 (char *)&err); | |
364 ao->i_hog_pid = -1; | |
365 } | |
366 | |
367 if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid()) | |
368 { | |
369 ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" ); | |
370 goto err_out; | |
371 } | |
372 ao->stream_format = inDesc; | |
373 return OpenSPDIF(); | |
374 } | |
375 | |
376 /* original analog output code */ | |
377 desc.componentType = kAudioUnitType_Output; | |
378 desc.componentSubType = kAudioUnitSubType_DefaultOutput; | |
379 desc.componentManufacturer = kAudioUnitManufacturer_Apple; | |
380 desc.componentFlags = 0; | |
381 desc.componentFlagsMask = 0; | |
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382 |
29209 | 383 comp = FindNextComponent(NULL, &desc); //Finds an component that meets the desc spec's |
384 if (comp == NULL) { | |
385 ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n"); | |
386 goto err_out; | |
387 } | |
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388 |
29209 | 389 err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component |
390 if (err) { | |
391 ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err); | |
392 goto err_out; | |
393 } | |
394 | |
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395 // Initialize AudioUnit |
29209 | 396 err = AudioUnitInitialize(ao->theOutputUnit); |
397 if (err) { | |
398 ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err); | |
399 goto err_out1; | |
400 } | |
401 | |
402 size = sizeof(AudioStreamBasicDescription); | |
403 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size); | |
404 | |
405 if (err) { | |
406 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err); | |
407 goto err_out2; | |
408 } | |
409 | |
410 size = sizeof(UInt32); | |
411 err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size); | |
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412 |
29209 | 413 if (err) |
414 { | |
415 ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err); | |
416 goto err_out2; | |
417 } | |
418 | |
419 ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame; | |
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420 |
29209 | 421 ao_data.samplerate = inDesc.mSampleRate; |
422 ao_data.channels = inDesc.mChannelsPerFrame; | |
423 ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame; | |
424 ao_data.outburst = ao->chunk_size; | |
425 ao_data.buffersize = ao_data.bps; | |
426 | |
427 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size; | |
428 ao->buffer_len = ao->num_chunks * ao->chunk_size; | |
429 ao->buffer = av_fifo_alloc(ao->buffer_len); | |
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430 |
29209 | 431 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); |
432 | |
433 renderCallback.inputProc = theRenderProc; | |
434 renderCallback.inputProcRefCon = 0; | |
435 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct)); | |
436 if (err) { | |
437 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err); | |
438 goto err_out2; | |
439 } | |
440 | |
441 reset(); | |
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442 |
29209 | 443 return CONTROL_OK; |
444 | |
445 err_out2: | |
446 AudioUnitUninitialize(ao->theOutputUnit); | |
447 err_out1: | |
448 CloseComponent(ao->theOutputUnit); | |
449 err_out: | |
450 av_fifo_free(ao->buffer); | |
451 free(ao); | |
452 ao = NULL; | |
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453 return CONTROL_FALSE; |
29209 | 454 } |
455 | |
456 /***************************************************************************** | |
457 * Setup a encoded digital stream (SPDIF) | |
458 *****************************************************************************/ | |
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459 static int OpenSPDIF(void) |
29209 | 460 { |
461 OSStatus err = noErr; | |
462 UInt32 i_param_size, b_mix = 0; | |
463 Boolean b_writeable = 0; | |
464 AudioStreamID *p_streams = NULL; | |
465 int i, i_streams = 0; | |
466 | |
467 /* Start doing the SPDIF setup process. */ | |
468 ao->b_digital = 1; | |
469 | |
470 /* Hog the device. */ | |
471 i_param_size = sizeof(ao->i_hog_pid); | |
472 ao->i_hog_pid = getpid() ; | |
473 | |
474 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
475 kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid); | |
476 | |
477 if (err != noErr) | |
478 { | |
479 ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err); | |
480 ao->i_hog_pid = -1; | |
481 goto err_out; | |
482 } | |
483 | |
484 /* Set mixable to false if we are allowed to. */ | |
485 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, | |
486 kAudioDevicePropertySupportsMixing, | |
487 &i_param_size, &b_writeable); | |
488 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
489 kAudioDevicePropertySupportsMixing, | |
490 &i_param_size, &b_mix); | |
491 if (err != noErr && b_writeable) | |
492 { | |
493 b_mix = 0; | |
494 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
495 kAudioDevicePropertySupportsMixing, | |
496 i_param_size, &b_mix); | |
497 ao->b_changed_mixing = 1; | |
498 } | |
499 if (err != noErr) | |
500 { | |
501 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); | |
502 goto err_out; | |
503 } | |
504 | |
505 /* Get a list of all the streams on this device. */ | |
506 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, | |
507 kAudioDevicePropertyStreams, | |
508 &i_param_size, NULL); | |
509 if (err != noErr) | |
510 { | |
511 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
512 goto err_out; | |
513 } | |
514 | |
515 i_streams = i_param_size / sizeof(AudioStreamID); | |
30702 | 516 p_streams = malloc(i_param_size); |
29209 | 517 if (p_streams == NULL) |
518 { | |
519 ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" ); | |
520 goto err_out; | |
521 } | |
522 | |
523 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
524 kAudioDevicePropertyStreams, | |
525 &i_param_size, p_streams); | |
526 if (err != noErr) | |
527 { | |
528 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
529 if (p_streams) free(p_streams); | |
530 goto err_out; | |
531 } | |
532 | |
533 ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams); | |
534 | |
535 for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i) | |
536 { | |
537 /* Find a stream with a cac3 stream. */ | |
538 AudioStreamBasicDescription *p_format_list = NULL; | |
539 int i_formats = 0, j = 0, b_digital = 0; | |
540 | |
541 /* Retrieve all the stream formats supported by each output stream. */ | |
542 err = AudioStreamGetPropertyInfo(p_streams[i], 0, | |
543 kAudioStreamPropertyPhysicalFormats, | |
544 &i_param_size, NULL); | |
545 if (err != noErr) | |
546 { | |
547 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streamformats: [%4.4s]\n", (char *)&err); | |
548 continue; | |
549 } | |
550 | |
551 i_formats = i_param_size / sizeof(AudioStreamBasicDescription); | |
30702 | 552 p_format_list = malloc(i_param_size); |
29209 | 553 if (p_format_list == NULL) |
554 { | |
555 ao_msg(MSGT_AO, MSGL_WARN, "could not malloc the memory\n" ); | |
556 continue; | |
557 } | |
558 | |
559 err = AudioStreamGetProperty(p_streams[i], 0, | |
560 kAudioStreamPropertyPhysicalFormats, | |
561 &i_param_size, p_format_list); | |
562 if (err != noErr) | |
563 { | |
564 ao_msg(MSGT_AO, MSGL_WARN, "could not get the list of streamformats: [%4.4s]\n", (char *)&err); | |
565 if (p_format_list) free(p_format_list); | |
566 continue; | |
567 } | |
568 | |
569 /* Check if one of the supported formats is a digital format. */ | |
570 for (j = 0; j < i_formats; ++j) | |
571 { | |
572 if (p_format_list[j].mFormatID == 'IAC3' || | |
573 p_format_list[j].mFormatID == kAudioFormat60958AC3) | |
574 { | |
575 b_digital = 1; | |
576 break; | |
577 } | |
578 } | |
579 | |
580 if (b_digital) | |
581 { | |
582 /* If this stream supports a digital (cac3) format, then set it. */ | |
583 int i_requested_rate_format = -1; | |
584 int i_current_rate_format = -1; | |
585 int i_backup_rate_format = -1; | |
586 | |
587 ao->i_stream_id = p_streams[i]; | |
588 ao->i_stream_index = i; | |
589 | |
590 if (ao->b_revert == 0) | |
591 { | |
592 /* Retrieve the original format of this stream first if not done so already. */ | |
593 i_param_size = sizeof(ao->sfmt_revert); | |
594 err = AudioStreamGetProperty(ao->i_stream_id, 0, | |
595 kAudioStreamPropertyPhysicalFormat, | |
596 &i_param_size, | |
597 &ao->sfmt_revert); | |
598 if (err != noErr) | |
599 { | |
600 ao_msg(MSGT_AO, MSGL_WARN, "could not retrieve the original streamformat: [%4.4s]\n", (char *)&err); | |
601 if (p_format_list) free(p_format_list); | |
602 continue; | |
603 } | |
604 ao->b_revert = 1; | |
605 } | |
606 | |
607 for (j = 0; j < i_formats; ++j) | |
608 if (p_format_list[j].mFormatID == 'IAC3' || | |
609 p_format_list[j].mFormatID == kAudioFormat60958AC3) | |
610 { | |
611 if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate) | |
612 { | |
613 i_requested_rate_format = j; | |
614 break; | |
615 } | |
616 if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate) | |
617 i_current_rate_format = j; | |
618 else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate) | |
619 i_backup_rate_format = j; | |
620 } | |
621 | |
622 if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */ | |
623 ao->stream_format = p_format_list[i_requested_rate_format]; | |
624 else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */ | |
625 ao->stream_format = p_format_list[i_current_rate_format]; | |
626 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). */ | |
627 } | |
628 if (p_format_list) free(p_format_list); | |
629 } | |
630 if (p_streams) free(p_streams); | |
631 | |
632 if (ao->i_stream_index < 0) | |
633 { | |
634 ao_msg(MSGT_AO, MSGL_WARN, "can not find any digital output stream format when OpenSPDIF().\n"); | |
635 goto err_out; | |
636 } | |
637 | |
638 print_format(MSGL_V, "original stream format:", &ao->sfmt_revert); | |
639 | |
640 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) | |
641 goto err_out; | |
642 | |
643 err = AudioDeviceAddPropertyListener(ao->i_selected_dev, | |
644 kAudioPropertyWildcardChannel, | |
645 0, | |
646 kAudioDevicePropertyDeviceHasChanged, | |
647 DeviceListener, | |
648 NULL); | |
649 if (err != noErr) | |
650 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err); | |
651 | |
652 | |
653 /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */ | |
654 /* Although there's no such case reported. */ | |
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|
655 #if HAVE_BIGENDIAN |
29209 | 656 if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)) |
657 #else | |
658 if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian) | |
659 #endif | |
660 ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\n"); | |
661 | |
662 /* For ac3/dts, just use packet size 6144 bytes as chunk size. */ | |
663 ao->chunk_size = ao->stream_format.mBytesPerPacket; | |
664 | |
665 ao_data.samplerate = ao->stream_format.mSampleRate; | |
666 ao_data.channels = ao->stream_format.mChannelsPerFrame; | |
667 ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket); | |
668 ao_data.outburst = ao->chunk_size; | |
669 ao_data.buffersize = ao_data.bps; | |
670 | |
671 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size; | |
672 ao->buffer_len = ao->num_chunks * ao->chunk_size; | |
673 ao->buffer = av_fifo_alloc(ao->buffer_len); | |
674 | |
675 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); | |
676 | |
677 | |
678 /* Add IOProc callback. */ | |
679 err = AudioDeviceAddIOProc(ao->i_selected_dev, | |
680 (AudioDeviceIOProc)RenderCallbackSPDIF, | |
681 (void *)ao); | |
682 if (err != noErr) | |
683 { | |
684 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err); | |
685 goto err_out1; | |
686 } | |
687 | |
688 reset(); | |
689 | |
690 return CONTROL_TRUE; | |
691 | |
692 err_out1: | |
693 if (ao->b_revert) | |
694 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert); | |
695 err_out: | |
696 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3) | |
697 { | |
698 int b_mix = 1; | |
699 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
700 kAudioDevicePropertySupportsMixing, | |
701 i_param_size, &b_mix); | |
702 if (err != noErr) | |
703 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", | |
704 (char *)&err); | |
705 } | |
706 if (ao->i_hog_pid == getpid()) | |
707 { | |
708 ao->i_hog_pid = -1; | |
709 i_param_size = sizeof(ao->i_hog_pid); | |
710 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
711 kAudioDevicePropertyHogMode, | |
712 i_param_size, &ao->i_hog_pid); | |
713 if (err != noErr) | |
714 ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", | |
715 (char *)&err); | |
716 } | |
717 av_fifo_free(ao->buffer); | |
718 free(ao); | |
719 ao = NULL; | |
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|
720 return CONTROL_FALSE; |
29209 | 721 } |
722 | |
723 /***************************************************************************** | |
724 * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support. | |
725 *****************************************************************************/ | |
726 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id ) | |
727 { | |
728 OSStatus err = noErr; | |
729 UInt32 i_param_size = 0; | |
730 AudioStreamID *p_streams = NULL; | |
731 int i = 0, i_streams = 0; | |
732 int b_return = CONTROL_FALSE; | |
733 | |
734 /* Retrieve all the output streams. */ | |
735 err = AudioDeviceGetPropertyInfo(i_dev_id, 0, FALSE, | |
736 kAudioDevicePropertyStreams, | |
737 &i_param_size, NULL); | |
738 if (err != noErr) | |
739 { | |
740 ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
741 return CONTROL_FALSE; | |
742 } | |
743 | |
744 i_streams = i_param_size / sizeof(AudioStreamID); | |
30702 | 745 p_streams = malloc(i_param_size); |
29209 | 746 if (p_streams == NULL) |
747 { | |
748 ao_msg(MSGT_AO,MSGL_V, "out of memory\n"); | |
749 return CONTROL_FALSE; | |
750 } | |
751 | |
752 err = AudioDeviceGetProperty(i_dev_id, 0, FALSE, | |
753 kAudioDevicePropertyStreams, | |
754 &i_param_size, p_streams); | |
755 | |
756 if (err != noErr) | |
757 { | |
758 ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
759 free(p_streams); | |
760 return CONTROL_FALSE; | |
761 } | |
762 | |
763 for (i = 0; i < i_streams; ++i) | |
764 { | |
765 if (AudioStreamSupportsDigital(p_streams[i])) | |
766 b_return = CONTROL_OK; | |
767 } | |
768 | |
769 free(p_streams); | |
770 return b_return; | |
771 } | |
772 | |
773 /***************************************************************************** | |
774 * AudioStreamSupportsDigital: Check i_stream_id for digital stream support. | |
775 *****************************************************************************/ | |
776 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id ) | |
777 { | |
778 OSStatus err = noErr; | |
779 UInt32 i_param_size; | |
780 AudioStreamBasicDescription *p_format_list = NULL; | |
781 int i, i_formats, b_return = CONTROL_FALSE; | |
782 | |
783 /* Retrieve all the stream formats supported by each output stream. */ | |
784 err = AudioStreamGetPropertyInfo(i_stream_id, 0, | |
785 kAudioStreamPropertyPhysicalFormats, | |
786 &i_param_size, NULL); | |
787 if (err != noErr) | |
788 { | |
789 ao_msg(MSGT_AO,MSGL_V, "could not get number of streamformats: [%4.4s]\n", (char *)&err); | |
790 return CONTROL_FALSE; | |
791 } | |
792 | |
793 i_formats = i_param_size / sizeof(AudioStreamBasicDescription); | |
30702 | 794 p_format_list = malloc(i_param_size); |
29209 | 795 if (p_format_list == NULL) |
796 { | |
797 ao_msg(MSGT_AO,MSGL_V, "could not malloc the memory\n" ); | |
798 return CONTROL_FALSE; | |
799 } | |
800 | |
801 err = AudioStreamGetProperty(i_stream_id, 0, | |
802 kAudioStreamPropertyPhysicalFormats, | |
803 &i_param_size, p_format_list); | |
804 if (err != noErr) | |
805 { | |
806 ao_msg(MSGT_AO,MSGL_V, "could not get the list of streamformats: [%4.4s]\n", (char *)&err); | |
807 free(p_format_list); | |
808 return CONTROL_FALSE; | |
809 } | |
810 | |
811 for (i = 0; i < i_formats; ++i) | |
812 { | |
813 print_format(MSGL_V, "supported format:", &p_format_list[i]); | |
814 | |
815 if (p_format_list[i].mFormatID == 'IAC3' || | |
816 p_format_list[i].mFormatID == kAudioFormat60958AC3) | |
817 b_return = CONTROL_OK; | |
818 } | |
819 | |
820 free(p_format_list); | |
821 return b_return; | |
822 } | |
823 | |
824 /***************************************************************************** | |
825 * AudioStreamChangeFormat: Change i_stream_id to change_format | |
826 *****************************************************************************/ | |
827 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format ) | |
828 { | |
829 OSStatus err = noErr; | |
830 UInt32 i_param_size = 0; | |
831 int i; | |
832 | |
833 static volatile int stream_format_changed; | |
834 stream_format_changed = 0; | |
835 | |
836 print_format(MSGL_V, "setting stream format:", &change_format); | |
837 | |
838 /* Install the callback. */ | |
839 err = AudioStreamAddPropertyListener(i_stream_id, 0, | |
840 kAudioStreamPropertyPhysicalFormat, | |
841 StreamListener, | |
842 (void *)&stream_format_changed); | |
843 if (err != noErr) | |
844 { | |
845 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err); | |
846 return CONTROL_FALSE; | |
847 } | |
848 | |
849 /* Change the format. */ | |
850 err = AudioStreamSetProperty(i_stream_id, 0, 0, | |
851 kAudioStreamPropertyPhysicalFormat, | |
852 sizeof(AudioStreamBasicDescription), | |
853 &change_format); | |
854 if (err != noErr) | |
855 { | |
856 ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err); | |
857 return CONTROL_FALSE; | |
858 } | |
859 | |
860 /* The AudioStreamSetProperty is not only asynchronious, | |
861 * it is also not Atomic, in its behaviour. | |
862 * Therefore we check 5 times before we really give up. | |
863 * FIXME: failing isn't actually implemented yet. */ | |
864 for (i = 0; i < 5; ++i) | |
865 { | |
866 AudioStreamBasicDescription actual_format; | |
867 int j; | |
868 for (j = 0; !stream_format_changed && j < 50; ++j) | |
869 usec_sleep(10000); | |
870 if (stream_format_changed) | |
871 stream_format_changed = 0; | |
872 else | |
873 ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" ); | |
874 | |
875 i_param_size = sizeof(AudioStreamBasicDescription); | |
876 err = AudioStreamGetProperty(i_stream_id, 0, | |
877 kAudioStreamPropertyPhysicalFormat, | |
878 &i_param_size, | |
879 &actual_format); | |
880 | |
881 print_format(MSGL_V, "actual format in use:", &actual_format); | |
882 if (actual_format.mSampleRate == change_format.mSampleRate && | |
883 actual_format.mFormatID == change_format.mFormatID && | |
884 actual_format.mFramesPerPacket == change_format.mFramesPerPacket) | |
885 { | |
886 /* The right format is now active. */ | |
887 break; | |
888 } | |
889 /* We need to check again. */ | |
890 } | |
891 | |
892 /* Removing the property listener. */ | |
893 err = AudioStreamRemovePropertyListener(i_stream_id, 0, | |
894 kAudioStreamPropertyPhysicalFormat, | |
895 StreamListener); | |
896 if (err != noErr) | |
897 { | |
898 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err); | |
899 return CONTROL_FALSE; | |
900 } | |
901 | |
902 return CONTROL_TRUE; | |
903 } | |
904 | |
905 /***************************************************************************** | |
906 * RenderCallbackSPDIF: callback for SPDIF audio output | |
907 *****************************************************************************/ | |
908 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice, | |
909 const AudioTimeStamp * inNow, | |
910 const void * inInputData, | |
911 const AudioTimeStamp * inInputTime, | |
912 AudioBufferList * outOutputData, | |
913 const AudioTimeStamp * inOutputTime, | |
914 void * threadGlobals ) | |
915 { | |
916 int amt = av_fifo_size(ao->buffer); | |
917 int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize; | |
918 | |
919 if (amt > req) | |
920 amt = req; | |
921 if (amt) | |
922 read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt); | |
923 | |
924 return noErr; | |
925 } | |
926 | |
927 | |
928 static int play(void* output_samples,int num_bytes,int flags) | |
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|
929 { |
29209 | 930 int wrote, b_digital; |
931 | |
932 // Check whether we need to reset the digital output stream. | |
933 if (ao->b_digital && ao->b_stream_format_changed) | |
934 { | |
935 ao->b_stream_format_changed = 0; | |
936 b_digital = AudioStreamSupportsDigital(ao->i_stream_id); | |
937 if (b_digital) | |
938 { | |
939 /* Current stream support digital format output, let's set it. */ | |
940 ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n"); | |
941 | |
942 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) | |
943 { | |
944 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n"); | |
945 } | |
946 else | |
947 { | |
948 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n"); | |
949 reset(); | |
950 } | |
951 } | |
952 else | |
953 ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n"); | |
954 } | |
955 | |
956 wrote=write_buffer(output_samples, num_bytes); | |
957 audio_resume(); | |
958 return wrote; | |
959 } | |
960 | |
961 /* set variables and buffer to initial state */ | |
962 static void reset(void) | |
963 { | |
964 audio_pause(); | |
965 av_fifo_reset(ao->buffer); | |
966 } | |
967 | |
968 | |
969 /* return available space */ | |
970 static int get_space(void) | |
971 { | |
972 return ao->buffer_len - av_fifo_size(ao->buffer); | |
973 } | |
974 | |
975 | |
976 /* return delay until audio is played */ | |
977 static float get_delay(void) | |
978 { | |
979 // inaccurate, should also contain the data buffered e.g. by the OS | |
980 return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps; | |
981 } | |
982 | |
983 | |
984 /* unload plugin and deregister from coreaudio */ | |
985 static void uninit(int immed) | |
986 { | |
987 OSStatus err = noErr; | |
988 UInt32 i_param_size = 0; | |
989 | |
990 if (!immed) { | |
991 long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps; | |
992 ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft); | |
993 usec_sleep((int)timeleft); | |
994 } | |
995 | |
996 if (!ao->b_digital) { | |
997 AudioOutputUnitStop(ao->theOutputUnit); | |
998 AudioUnitUninitialize(ao->theOutputUnit); | |
999 CloseComponent(ao->theOutputUnit); | |
1000 } | |
1001 else { | |
1002 /* Stop device. */ | |
1003 err = AudioDeviceStop(ao->i_selected_dev, | |
1004 (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1005 if (err != noErr) | |
1006 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); | |
1007 | |
1008 /* Remove IOProc callback. */ | |
1009 err = AudioDeviceRemoveIOProc(ao->i_selected_dev, | |
1010 (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1011 if (err != noErr) | |
1012 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err); | |
1013 | |
1014 if (ao->b_revert) | |
1015 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert); | |
1016 | |
1017 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3) | |
1018 { | |
1019 int b_mix; | |
1020 Boolean b_writeable; | |
1021 /* Revert mixable to true if we are allowed to. */ | |
1022 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing, | |
1023 &i_param_size, &b_writeable); | |
1024 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing, | |
1025 &i_param_size, &b_mix); | |
1026 if (err != noErr && b_writeable) | |
1027 { | |
1028 b_mix = 1; | |
1029 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
1030 kAudioDevicePropertySupportsMixing, i_param_size, &b_mix); | |
1031 } | |
1032 if (err != noErr) | |
1033 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); | |
1034 } | |
1035 if (ao->i_hog_pid == getpid()) | |
1036 { | |
1037 ao->i_hog_pid = -1; | |
1038 i_param_size = sizeof(ao->i_hog_pid); | |
1039 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
1040 kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid); | |
1041 if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err); | |
1042 } | |
1043 } | |
1044 | |
1045 av_fifo_free(ao->buffer); | |
1046 free(ao); | |
1047 ao = NULL; | |
1048 } | |
1049 | |
1050 | |
1051 /* stop playing, keep buffers (for pause) */ | |
1052 static void audio_pause(void) | |
1053 { | |
1054 OSErr err=noErr; | |
1055 | |
1056 /* Stop callback. */ | |
1057 if (!ao->b_digital) | |
1058 { | |
1059 err=AudioOutputUnitStop(ao->theOutputUnit); | |
1060 if (err != noErr) | |
1061 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err); | |
1062 } | |
1063 else | |
1064 { | |
1065 err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1066 if (err != noErr) | |
1067 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); | |
1068 } | |
1069 ao->paused = 1; | |
1070 } | |
1071 | |
1072 | |
1073 /* resume playing, after audio_pause() */ | |
1074 static void audio_resume(void) | |
1075 { | |
1076 OSErr err=noErr; | |
1077 | |
1078 if (!ao->paused) | |
1079 return; | |
1080 | |
1081 /* Start callback. */ | |
1082 if (!ao->b_digital) | |
1083 { | |
1084 err = AudioOutputUnitStart(ao->theOutputUnit); | |
1085 if (err != noErr) | |
1086 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err); | |
1087 } | |
1088 else | |
1089 { | |
1090 err = AudioDeviceStart(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1091 if (err != noErr) | |
1092 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err); | |
1093 } | |
1094 ao->paused = 0; | |
1095 } | |
1096 | |
1097 /***************************************************************************** | |
1098 * StreamListener | |
1099 *****************************************************************************/ | |
1100 static OSStatus StreamListener( AudioStreamID inStream, | |
1101 UInt32 inChannel, | |
1102 AudioDevicePropertyID inPropertyID, | |
1103 void * inClientData ) | |
1104 { | |
1105 switch (inPropertyID) | |
1106 { | |
1107 case kAudioStreamPropertyPhysicalFormat: | |
1108 ao_msg(MSGT_AO, MSGL_V, "got notify kAudioStreamPropertyPhysicalFormat changed.\n"); | |
1109 if (inClientData) | |
1110 *(volatile int *)inClientData = 1; | |
1111 default: | |
1112 break; | |
1113 } | |
1114 return noErr; | |
1115 } | |
1116 | |
1117 static OSStatus DeviceListener( AudioDeviceID inDevice, | |
1118 UInt32 inChannel, | |
1119 Boolean isInput, | |
1120 AudioDevicePropertyID inPropertyID, | |
1121 void* inClientData ) | |
1122 { | |
1123 switch (inPropertyID) | |
1124 { | |
1125 case kAudioDevicePropertyDeviceHasChanged: | |
1126 ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n"); | |
1127 ao->b_stream_format_changed = 1; | |
1128 default: | |
1129 break; | |
1130 } | |
1131 return noErr; | |
1132 } |