Mercurial > mplayer.hg
annotate libao2/ao_coreaudio.c @ 30603:c5d8b7640b7d
Call mpcodecs_config_vo with the proper image format for dmo and dshow codecs.
author | reimar |
---|---|
date | Thu, 18 Feb 2010 22:28:55 +0000 |
parents | 03c1ad03f29d |
children | 13b7aa964af6 |
rev | line source |
---|---|
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. */ | |
30234 | 266 if (AF_FORMAT_IS_AC3(format)) |
29209 | 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; | |
30235 | 317 inDesc.mBitsPerChannel=af_fmt2bits(format); |
29209 | 318 |
319 if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) { | |
320 // float | |
321 inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked; | |
322 } | |
323 else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) { | |
324 // signed int | |
325 inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked; | |
326 } | |
327 else { | |
328 // unsigned int | |
329 inDesc.mFormatFlags = kAudioFormatFlagIsPacked; | |
330 } | |
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331 if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE) |
29209 | 332 inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian; |
333 | |
334 inDesc.mFramesPerPacket = 1; | |
335 ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8); | |
336 print_format(MSGL_V, "source:",&inDesc); | |
337 | |
338 if (ao->b_supports_digital) | |
339 { | |
340 b_alive = 1; | |
341 i_param_size = sizeof(b_alive); | |
342 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
343 kAudioDevicePropertyDeviceIsAlive, | |
344 &i_param_size, &b_alive); | |
345 if (err != noErr) | |
346 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err); | |
347 if (!b_alive) | |
348 ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" ); | |
349 /* S/PDIF output need device in HogMode. */ | |
350 i_param_size = sizeof(ao->i_hog_pid); | |
351 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
352 kAudioDevicePropertyHogMode, | |
353 &i_param_size, &ao->i_hog_pid); | |
354 | |
355 if (err != noErr) | |
356 { | |
357 /* This is not a fatal error. Some drivers simply don't support this property. */ | |
358 ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n", | |
359 (char *)&err); | |
360 ao->i_hog_pid = -1; | |
361 } | |
362 | |
363 if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid()) | |
364 { | |
365 ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" ); | |
366 goto err_out; | |
367 } | |
368 ao->stream_format = inDesc; | |
369 return OpenSPDIF(); | |
370 } | |
371 | |
372 /* original analog output code */ | |
373 desc.componentType = kAudioUnitType_Output; | |
374 desc.componentSubType = kAudioUnitSubType_DefaultOutput; | |
375 desc.componentManufacturer = kAudioUnitManufacturer_Apple; | |
376 desc.componentFlags = 0; | |
377 desc.componentFlagsMask = 0; | |
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378 |
29209 | 379 comp = FindNextComponent(NULL, &desc); //Finds an component that meets the desc spec's |
380 if (comp == NULL) { | |
381 ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n"); | |
382 goto err_out; | |
383 } | |
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384 |
29209 | 385 err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component |
386 if (err) { | |
387 ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err); | |
388 goto err_out; | |
389 } | |
390 | |
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391 // Initialize AudioUnit |
29209 | 392 err = AudioUnitInitialize(ao->theOutputUnit); |
393 if (err) { | |
394 ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err); | |
395 goto err_out1; | |
396 } | |
397 | |
398 size = sizeof(AudioStreamBasicDescription); | |
399 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size); | |
400 | |
401 if (err) { | |
402 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err); | |
403 goto err_out2; | |
404 } | |
405 | |
406 size = sizeof(UInt32); | |
407 err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size); | |
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408 |
29209 | 409 if (err) |
410 { | |
411 ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err); | |
412 goto err_out2; | |
413 } | |
414 | |
415 ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame; | |
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416 |
29209 | 417 ao_data.samplerate = inDesc.mSampleRate; |
418 ao_data.channels = inDesc.mChannelsPerFrame; | |
419 ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame; | |
420 ao_data.outburst = ao->chunk_size; | |
421 ao_data.buffersize = ao_data.bps; | |
422 | |
423 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size; | |
424 ao->buffer_len = ao->num_chunks * ao->chunk_size; | |
425 ao->buffer = av_fifo_alloc(ao->buffer_len); | |
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426 |
29209 | 427 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); |
428 | |
429 renderCallback.inputProc = theRenderProc; | |
430 renderCallback.inputProcRefCon = 0; | |
431 err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct)); | |
432 if (err) { | |
433 ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err); | |
434 goto err_out2; | |
435 } | |
436 | |
437 reset(); | |
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438 |
29209 | 439 return CONTROL_OK; |
440 | |
441 err_out2: | |
442 AudioUnitUninitialize(ao->theOutputUnit); | |
443 err_out1: | |
444 CloseComponent(ao->theOutputUnit); | |
445 err_out: | |
446 av_fifo_free(ao->buffer); | |
447 free(ao); | |
448 ao = NULL; | |
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449 return CONTROL_FALSE; |
29209 | 450 } |
451 | |
452 /***************************************************************************** | |
453 * Setup a encoded digital stream (SPDIF) | |
454 *****************************************************************************/ | |
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455 static int OpenSPDIF(void) |
29209 | 456 { |
457 OSStatus err = noErr; | |
458 UInt32 i_param_size, b_mix = 0; | |
459 Boolean b_writeable = 0; | |
460 AudioStreamID *p_streams = NULL; | |
461 int i, i_streams = 0; | |
462 | |
463 /* Start doing the SPDIF setup process. */ | |
464 ao->b_digital = 1; | |
465 | |
466 /* Hog the device. */ | |
467 i_param_size = sizeof(ao->i_hog_pid); | |
468 ao->i_hog_pid = getpid() ; | |
469 | |
470 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
471 kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid); | |
472 | |
473 if (err != noErr) | |
474 { | |
475 ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err); | |
476 ao->i_hog_pid = -1; | |
477 goto err_out; | |
478 } | |
479 | |
480 /* Set mixable to false if we are allowed to. */ | |
481 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, | |
482 kAudioDevicePropertySupportsMixing, | |
483 &i_param_size, &b_writeable); | |
484 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
485 kAudioDevicePropertySupportsMixing, | |
486 &i_param_size, &b_mix); | |
487 if (err != noErr && b_writeable) | |
488 { | |
489 b_mix = 0; | |
490 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
491 kAudioDevicePropertySupportsMixing, | |
492 i_param_size, &b_mix); | |
493 ao->b_changed_mixing = 1; | |
494 } | |
495 if (err != noErr) | |
496 { | |
497 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); | |
498 goto err_out; | |
499 } | |
500 | |
501 /* Get a list of all the streams on this device. */ | |
502 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, | |
503 kAudioDevicePropertyStreams, | |
504 &i_param_size, NULL); | |
505 if (err != noErr) | |
506 { | |
507 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
508 goto err_out; | |
509 } | |
510 | |
511 i_streams = i_param_size / sizeof(AudioStreamID); | |
512 p_streams = (AudioStreamID *)malloc(i_param_size); | |
513 if (p_streams == NULL) | |
514 { | |
515 ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" ); | |
516 goto err_out; | |
517 } | |
518 | |
519 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, | |
520 kAudioDevicePropertyStreams, | |
521 &i_param_size, p_streams); | |
522 if (err != noErr) | |
523 { | |
524 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
525 if (p_streams) free(p_streams); | |
526 goto err_out; | |
527 } | |
528 | |
529 ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams); | |
530 | |
531 for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i) | |
532 { | |
533 /* Find a stream with a cac3 stream. */ | |
534 AudioStreamBasicDescription *p_format_list = NULL; | |
535 int i_formats = 0, j = 0, b_digital = 0; | |
536 | |
537 /* Retrieve all the stream formats supported by each output stream. */ | |
538 err = AudioStreamGetPropertyInfo(p_streams[i], 0, | |
539 kAudioStreamPropertyPhysicalFormats, | |
540 &i_param_size, NULL); | |
541 if (err != noErr) | |
542 { | |
543 ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streamformats: [%4.4s]\n", (char *)&err); | |
544 continue; | |
545 } | |
546 | |
547 i_formats = i_param_size / sizeof(AudioStreamBasicDescription); | |
548 p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size); | |
549 if (p_format_list == NULL) | |
550 { | |
551 ao_msg(MSGT_AO, MSGL_WARN, "could not malloc the memory\n" ); | |
552 continue; | |
553 } | |
554 | |
555 err = AudioStreamGetProperty(p_streams[i], 0, | |
556 kAudioStreamPropertyPhysicalFormats, | |
557 &i_param_size, p_format_list); | |
558 if (err != noErr) | |
559 { | |
560 ao_msg(MSGT_AO, MSGL_WARN, "could not get the list of streamformats: [%4.4s]\n", (char *)&err); | |
561 if (p_format_list) free(p_format_list); | |
562 continue; | |
563 } | |
564 | |
565 /* Check if one of the supported formats is a digital format. */ | |
566 for (j = 0; j < i_formats; ++j) | |
567 { | |
568 if (p_format_list[j].mFormatID == 'IAC3' || | |
569 p_format_list[j].mFormatID == kAudioFormat60958AC3) | |
570 { | |
571 b_digital = 1; | |
572 break; | |
573 } | |
574 } | |
575 | |
576 if (b_digital) | |
577 { | |
578 /* If this stream supports a digital (cac3) format, then set it. */ | |
579 int i_requested_rate_format = -1; | |
580 int i_current_rate_format = -1; | |
581 int i_backup_rate_format = -1; | |
582 | |
583 ao->i_stream_id = p_streams[i]; | |
584 ao->i_stream_index = i; | |
585 | |
586 if (ao->b_revert == 0) | |
587 { | |
588 /* Retrieve the original format of this stream first if not done so already. */ | |
589 i_param_size = sizeof(ao->sfmt_revert); | |
590 err = AudioStreamGetProperty(ao->i_stream_id, 0, | |
591 kAudioStreamPropertyPhysicalFormat, | |
592 &i_param_size, | |
593 &ao->sfmt_revert); | |
594 if (err != noErr) | |
595 { | |
596 ao_msg(MSGT_AO, MSGL_WARN, "could not retrieve the original streamformat: [%4.4s]\n", (char *)&err); | |
597 if (p_format_list) free(p_format_list); | |
598 continue; | |
599 } | |
600 ao->b_revert = 1; | |
601 } | |
602 | |
603 for (j = 0; j < i_formats; ++j) | |
604 if (p_format_list[j].mFormatID == 'IAC3' || | |
605 p_format_list[j].mFormatID == kAudioFormat60958AC3) | |
606 { | |
607 if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate) | |
608 { | |
609 i_requested_rate_format = j; | |
610 break; | |
611 } | |
612 if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate) | |
613 i_current_rate_format = j; | |
614 else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate) | |
615 i_backup_rate_format = j; | |
616 } | |
617 | |
618 if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */ | |
619 ao->stream_format = p_format_list[i_requested_rate_format]; | |
620 else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */ | |
621 ao->stream_format = p_format_list[i_current_rate_format]; | |
622 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). */ | |
623 } | |
624 if (p_format_list) free(p_format_list); | |
625 } | |
626 if (p_streams) free(p_streams); | |
627 | |
628 if (ao->i_stream_index < 0) | |
629 { | |
630 ao_msg(MSGT_AO, MSGL_WARN, "can not find any digital output stream format when OpenSPDIF().\n"); | |
631 goto err_out; | |
632 } | |
633 | |
634 print_format(MSGL_V, "original stream format:", &ao->sfmt_revert); | |
635 | |
636 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) | |
637 goto err_out; | |
638 | |
639 err = AudioDeviceAddPropertyListener(ao->i_selected_dev, | |
640 kAudioPropertyWildcardChannel, | |
641 0, | |
642 kAudioDevicePropertyDeviceHasChanged, | |
643 DeviceListener, | |
644 NULL); | |
645 if (err != noErr) | |
646 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err); | |
647 | |
648 | |
649 /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */ | |
650 /* Although there's no such case reported. */ | |
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651 #if HAVE_BIGENDIAN |
29209 | 652 if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)) |
653 #else | |
654 if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian) | |
655 #endif | |
656 ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\n"); | |
657 | |
658 /* For ac3/dts, just use packet size 6144 bytes as chunk size. */ | |
659 ao->chunk_size = ao->stream_format.mBytesPerPacket; | |
660 | |
661 ao_data.samplerate = ao->stream_format.mSampleRate; | |
662 ao_data.channels = ao->stream_format.mChannelsPerFrame; | |
663 ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket); | |
664 ao_data.outburst = ao->chunk_size; | |
665 ao_data.buffersize = ao_data.bps; | |
666 | |
667 ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size; | |
668 ao->buffer_len = ao->num_chunks * ao->chunk_size; | |
669 ao->buffer = av_fifo_alloc(ao->buffer_len); | |
670 | |
671 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); | |
672 | |
673 | |
674 /* Add IOProc callback. */ | |
675 err = AudioDeviceAddIOProc(ao->i_selected_dev, | |
676 (AudioDeviceIOProc)RenderCallbackSPDIF, | |
677 (void *)ao); | |
678 if (err != noErr) | |
679 { | |
680 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err); | |
681 goto err_out1; | |
682 } | |
683 | |
684 reset(); | |
685 | |
686 return CONTROL_TRUE; | |
687 | |
688 err_out1: | |
689 if (ao->b_revert) | |
690 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert); | |
691 err_out: | |
692 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3) | |
693 { | |
694 int b_mix = 1; | |
695 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
696 kAudioDevicePropertySupportsMixing, | |
697 i_param_size, &b_mix); | |
698 if (err != noErr) | |
699 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", | |
700 (char *)&err); | |
701 } | |
702 if (ao->i_hog_pid == getpid()) | |
703 { | |
704 ao->i_hog_pid = -1; | |
705 i_param_size = sizeof(ao->i_hog_pid); | |
706 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
707 kAudioDevicePropertyHogMode, | |
708 i_param_size, &ao->i_hog_pid); | |
709 if (err != noErr) | |
710 ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", | |
711 (char *)&err); | |
712 } | |
713 av_fifo_free(ao->buffer); | |
714 free(ao); | |
715 ao = NULL; | |
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716 return CONTROL_FALSE; |
29209 | 717 } |
718 | |
719 /***************************************************************************** | |
720 * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support. | |
721 *****************************************************************************/ | |
722 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id ) | |
723 { | |
724 OSStatus err = noErr; | |
725 UInt32 i_param_size = 0; | |
726 AudioStreamID *p_streams = NULL; | |
727 int i = 0, i_streams = 0; | |
728 int b_return = CONTROL_FALSE; | |
729 | |
730 /* Retrieve all the output streams. */ | |
731 err = AudioDeviceGetPropertyInfo(i_dev_id, 0, FALSE, | |
732 kAudioDevicePropertyStreams, | |
733 &i_param_size, NULL); | |
734 if (err != noErr) | |
735 { | |
736 ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
737 return CONTROL_FALSE; | |
738 } | |
739 | |
740 i_streams = i_param_size / sizeof(AudioStreamID); | |
741 p_streams = (AudioStreamID *)malloc(i_param_size); | |
742 if (p_streams == NULL) | |
743 { | |
744 ao_msg(MSGT_AO,MSGL_V, "out of memory\n"); | |
745 return CONTROL_FALSE; | |
746 } | |
747 | |
748 err = AudioDeviceGetProperty(i_dev_id, 0, FALSE, | |
749 kAudioDevicePropertyStreams, | |
750 &i_param_size, p_streams); | |
751 | |
752 if (err != noErr) | |
753 { | |
754 ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err); | |
755 free(p_streams); | |
756 return CONTROL_FALSE; | |
757 } | |
758 | |
759 for (i = 0; i < i_streams; ++i) | |
760 { | |
761 if (AudioStreamSupportsDigital(p_streams[i])) | |
762 b_return = CONTROL_OK; | |
763 } | |
764 | |
765 free(p_streams); | |
766 return b_return; | |
767 } | |
768 | |
769 /***************************************************************************** | |
770 * AudioStreamSupportsDigital: Check i_stream_id for digital stream support. | |
771 *****************************************************************************/ | |
772 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id ) | |
773 { | |
774 OSStatus err = noErr; | |
775 UInt32 i_param_size; | |
776 AudioStreamBasicDescription *p_format_list = NULL; | |
777 int i, i_formats, b_return = CONTROL_FALSE; | |
778 | |
779 /* Retrieve all the stream formats supported by each output stream. */ | |
780 err = AudioStreamGetPropertyInfo(i_stream_id, 0, | |
781 kAudioStreamPropertyPhysicalFormats, | |
782 &i_param_size, NULL); | |
783 if (err != noErr) | |
784 { | |
785 ao_msg(MSGT_AO,MSGL_V, "could not get number of streamformats: [%4.4s]\n", (char *)&err); | |
786 return CONTROL_FALSE; | |
787 } | |
788 | |
789 i_formats = i_param_size / sizeof(AudioStreamBasicDescription); | |
790 p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size); | |
791 if (p_format_list == NULL) | |
792 { | |
793 ao_msg(MSGT_AO,MSGL_V, "could not malloc the memory\n" ); | |
794 return CONTROL_FALSE; | |
795 } | |
796 | |
797 err = AudioStreamGetProperty(i_stream_id, 0, | |
798 kAudioStreamPropertyPhysicalFormats, | |
799 &i_param_size, p_format_list); | |
800 if (err != noErr) | |
801 { | |
802 ao_msg(MSGT_AO,MSGL_V, "could not get the list of streamformats: [%4.4s]\n", (char *)&err); | |
803 free(p_format_list); | |
804 return CONTROL_FALSE; | |
805 } | |
806 | |
807 for (i = 0; i < i_formats; ++i) | |
808 { | |
809 print_format(MSGL_V, "supported format:", &p_format_list[i]); | |
810 | |
811 if (p_format_list[i].mFormatID == 'IAC3' || | |
812 p_format_list[i].mFormatID == kAudioFormat60958AC3) | |
813 b_return = CONTROL_OK; | |
814 } | |
815 | |
816 free(p_format_list); | |
817 return b_return; | |
818 } | |
819 | |
820 /***************************************************************************** | |
821 * AudioStreamChangeFormat: Change i_stream_id to change_format | |
822 *****************************************************************************/ | |
823 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format ) | |
824 { | |
825 OSStatus err = noErr; | |
826 UInt32 i_param_size = 0; | |
827 int i; | |
828 | |
829 static volatile int stream_format_changed; | |
830 stream_format_changed = 0; | |
831 | |
832 print_format(MSGL_V, "setting stream format:", &change_format); | |
833 | |
834 /* Install the callback. */ | |
835 err = AudioStreamAddPropertyListener(i_stream_id, 0, | |
836 kAudioStreamPropertyPhysicalFormat, | |
837 StreamListener, | |
838 (void *)&stream_format_changed); | |
839 if (err != noErr) | |
840 { | |
841 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err); | |
842 return CONTROL_FALSE; | |
843 } | |
844 | |
845 /* Change the format. */ | |
846 err = AudioStreamSetProperty(i_stream_id, 0, 0, | |
847 kAudioStreamPropertyPhysicalFormat, | |
848 sizeof(AudioStreamBasicDescription), | |
849 &change_format); | |
850 if (err != noErr) | |
851 { | |
852 ao_msg(MSGT_AO, MSGL_WARN, "could not set the stream format: [%4.4s]\n", (char *)&err); | |
853 return CONTROL_FALSE; | |
854 } | |
855 | |
856 /* The AudioStreamSetProperty is not only asynchronious, | |
857 * it is also not Atomic, in its behaviour. | |
858 * Therefore we check 5 times before we really give up. | |
859 * FIXME: failing isn't actually implemented yet. */ | |
860 for (i = 0; i < 5; ++i) | |
861 { | |
862 AudioStreamBasicDescription actual_format; | |
863 int j; | |
864 for (j = 0; !stream_format_changed && j < 50; ++j) | |
865 usec_sleep(10000); | |
866 if (stream_format_changed) | |
867 stream_format_changed = 0; | |
868 else | |
869 ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" ); | |
870 | |
871 i_param_size = sizeof(AudioStreamBasicDescription); | |
872 err = AudioStreamGetProperty(i_stream_id, 0, | |
873 kAudioStreamPropertyPhysicalFormat, | |
874 &i_param_size, | |
875 &actual_format); | |
876 | |
877 print_format(MSGL_V, "actual format in use:", &actual_format); | |
878 if (actual_format.mSampleRate == change_format.mSampleRate && | |
879 actual_format.mFormatID == change_format.mFormatID && | |
880 actual_format.mFramesPerPacket == change_format.mFramesPerPacket) | |
881 { | |
882 /* The right format is now active. */ | |
883 break; | |
884 } | |
885 /* We need to check again. */ | |
886 } | |
887 | |
888 /* Removing the property listener. */ | |
889 err = AudioStreamRemovePropertyListener(i_stream_id, 0, | |
890 kAudioStreamPropertyPhysicalFormat, | |
891 StreamListener); | |
892 if (err != noErr) | |
893 { | |
894 ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err); | |
895 return CONTROL_FALSE; | |
896 } | |
897 | |
898 return CONTROL_TRUE; | |
899 } | |
900 | |
901 /***************************************************************************** | |
902 * RenderCallbackSPDIF: callback for SPDIF audio output | |
903 *****************************************************************************/ | |
904 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice, | |
905 const AudioTimeStamp * inNow, | |
906 const void * inInputData, | |
907 const AudioTimeStamp * inInputTime, | |
908 AudioBufferList * outOutputData, | |
909 const AudioTimeStamp * inOutputTime, | |
910 void * threadGlobals ) | |
911 { | |
912 int amt = av_fifo_size(ao->buffer); | |
913 int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize; | |
914 | |
915 if (amt > req) | |
916 amt = req; | |
917 if (amt) | |
918 read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt); | |
919 | |
920 return noErr; | |
921 } | |
922 | |
923 | |
924 static int play(void* output_samples,int num_bytes,int flags) | |
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925 { |
29209 | 926 int wrote, b_digital; |
927 | |
928 // Check whether we need to reset the digital output stream. | |
929 if (ao->b_digital && ao->b_stream_format_changed) | |
930 { | |
931 ao->b_stream_format_changed = 0; | |
932 b_digital = AudioStreamSupportsDigital(ao->i_stream_id); | |
933 if (b_digital) | |
934 { | |
935 /* Current stream support digital format output, let's set it. */ | |
936 ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n"); | |
937 | |
938 if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) | |
939 { | |
940 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n"); | |
941 } | |
942 else | |
943 { | |
944 ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n"); | |
945 reset(); | |
946 } | |
947 } | |
948 else | |
949 ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n"); | |
950 } | |
951 | |
952 wrote=write_buffer(output_samples, num_bytes); | |
953 audio_resume(); | |
954 return wrote; | |
955 } | |
956 | |
957 /* set variables and buffer to initial state */ | |
958 static void reset(void) | |
959 { | |
960 audio_pause(); | |
961 av_fifo_reset(ao->buffer); | |
962 } | |
963 | |
964 | |
965 /* return available space */ | |
966 static int get_space(void) | |
967 { | |
968 return ao->buffer_len - av_fifo_size(ao->buffer); | |
969 } | |
970 | |
971 | |
972 /* return delay until audio is played */ | |
973 static float get_delay(void) | |
974 { | |
975 // inaccurate, should also contain the data buffered e.g. by the OS | |
976 return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps; | |
977 } | |
978 | |
979 | |
980 /* unload plugin and deregister from coreaudio */ | |
981 static void uninit(int immed) | |
982 { | |
983 OSStatus err = noErr; | |
984 UInt32 i_param_size = 0; | |
985 | |
986 if (!immed) { | |
987 long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps; | |
988 ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft); | |
989 usec_sleep((int)timeleft); | |
990 } | |
991 | |
992 if (!ao->b_digital) { | |
993 AudioOutputUnitStop(ao->theOutputUnit); | |
994 AudioUnitUninitialize(ao->theOutputUnit); | |
995 CloseComponent(ao->theOutputUnit); | |
996 } | |
997 else { | |
998 /* Stop device. */ | |
999 err = AudioDeviceStop(ao->i_selected_dev, | |
1000 (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1001 if (err != noErr) | |
1002 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); | |
1003 | |
1004 /* Remove IOProc callback. */ | |
1005 err = AudioDeviceRemoveIOProc(ao->i_selected_dev, | |
1006 (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1007 if (err != noErr) | |
1008 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err); | |
1009 | |
1010 if (ao->b_revert) | |
1011 AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert); | |
1012 | |
1013 if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3) | |
1014 { | |
1015 int b_mix; | |
1016 Boolean b_writeable; | |
1017 /* Revert mixable to true if we are allowed to. */ | |
1018 err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing, | |
1019 &i_param_size, &b_writeable); | |
1020 err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing, | |
1021 &i_param_size, &b_mix); | |
1022 if (err != noErr && b_writeable) | |
1023 { | |
1024 b_mix = 1; | |
1025 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
1026 kAudioDevicePropertySupportsMixing, i_param_size, &b_mix); | |
1027 } | |
1028 if (err != noErr) | |
1029 ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); | |
1030 } | |
1031 if (ao->i_hog_pid == getpid()) | |
1032 { | |
1033 ao->i_hog_pid = -1; | |
1034 i_param_size = sizeof(ao->i_hog_pid); | |
1035 err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE, | |
1036 kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid); | |
1037 if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err); | |
1038 } | |
1039 } | |
1040 | |
1041 av_fifo_free(ao->buffer); | |
1042 free(ao); | |
1043 ao = NULL; | |
1044 } | |
1045 | |
1046 | |
1047 /* stop playing, keep buffers (for pause) */ | |
1048 static void audio_pause(void) | |
1049 { | |
1050 OSErr err=noErr; | |
1051 | |
1052 /* Stop callback. */ | |
1053 if (!ao->b_digital) | |
1054 { | |
1055 err=AudioOutputUnitStop(ao->theOutputUnit); | |
1056 if (err != noErr) | |
1057 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err); | |
1058 } | |
1059 else | |
1060 { | |
1061 err = AudioDeviceStop(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1062 if (err != noErr) | |
1063 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err); | |
1064 } | |
1065 ao->paused = 1; | |
1066 } | |
1067 | |
1068 | |
1069 /* resume playing, after audio_pause() */ | |
1070 static void audio_resume(void) | |
1071 { | |
1072 OSErr err=noErr; | |
1073 | |
1074 if (!ao->paused) | |
1075 return; | |
1076 | |
1077 /* Start callback. */ | |
1078 if (!ao->b_digital) | |
1079 { | |
1080 err = AudioOutputUnitStart(ao->theOutputUnit); | |
1081 if (err != noErr) | |
1082 ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err); | |
1083 } | |
1084 else | |
1085 { | |
1086 err = AudioDeviceStart(ao->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF); | |
1087 if (err != noErr) | |
1088 ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err); | |
1089 } | |
1090 ao->paused = 0; | |
1091 } | |
1092 | |
1093 /***************************************************************************** | |
1094 * StreamListener | |
1095 *****************************************************************************/ | |
1096 static OSStatus StreamListener( AudioStreamID inStream, | |
1097 UInt32 inChannel, | |
1098 AudioDevicePropertyID inPropertyID, | |
1099 void * inClientData ) | |
1100 { | |
1101 switch (inPropertyID) | |
1102 { | |
1103 case kAudioStreamPropertyPhysicalFormat: | |
1104 ao_msg(MSGT_AO, MSGL_V, "got notify kAudioStreamPropertyPhysicalFormat changed.\n"); | |
1105 if (inClientData) | |
1106 *(volatile int *)inClientData = 1; | |
1107 default: | |
1108 break; | |
1109 } | |
1110 return noErr; | |
1111 } | |
1112 | |
1113 static OSStatus DeviceListener( AudioDeviceID inDevice, | |
1114 UInt32 inChannel, | |
1115 Boolean isInput, | |
1116 AudioDevicePropertyID inPropertyID, | |
1117 void* inClientData ) | |
1118 { | |
1119 switch (inPropertyID) | |
1120 { | |
1121 case kAudioDevicePropertyDeviceHasChanged: | |
1122 ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n"); | |
1123 ao->b_stream_format_changed = 1; | |
1124 default: | |
1125 break; | |
1126 } | |
1127 return noErr; | |
1128 } |