blob: ff36310acea80abf9ba784a9fc464ea7624f9722
1 | /* |
2 | * Copyright (C) 2011 The Android Open Source Project |
3 | * |
4 | * Licensed under the Apache License, Version 2.0 (the "License"); |
5 | * you may not use this file except in compliance with the License. |
6 | * You may obtain a copy of the License at |
7 | * |
8 | * http://www.apache.org/licenses/LICENSE-2.0 |
9 | * |
10 | * Unless required by applicable law or agreed to in writing, software |
11 | * distributed under the License is distributed on an "AS IS" BASIS, |
12 | * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
13 | * See the License for the specific language governing permissions and |
14 | * limitations under the License. |
15 | */ |
16 | |
17 | #define LOG_TAG "audio_hw_primary" |
18 | //#define LOG_NALOGV 0 |
19 | //#define LOG_NALOGV_FUNCTION |
20 | #ifdef LOG_NALOGV_FUNCTION |
21 | #define LOGFUNC(...) ((void)0) |
22 | #else |
23 | #define LOGFUNC(...) (ALOGD(__VA_ARGS__)) |
24 | #endif |
25 | |
26 | #include <errno.h> |
27 | #include <pthread.h> |
28 | #include <stdint.h> |
29 | #include <sys/time.h> |
30 | #include <stdlib.h> |
31 | #include <sys/stat.h> |
32 | #include <fcntl.h> |
33 | #include <time.h> |
34 | #include <utils/Timers.h> |
35 | #include <cutils/log.h> |
36 | #include <cutils/str_parms.h> |
37 | #include <cutils/properties.h> |
38 | #include <linux/ioctl.h> |
39 | #include <hardware/hardware.h> |
40 | #include <system/audio.h> |
41 | #include <hardware/audio.h> |
42 | #include <sound/asound.h> |
43 | #include <tinyalsa/asoundlib.h> |
44 | #include <audio_utils/echo_reference.h> |
45 | #include <hardware/audio_effect.h> |
46 | #include <audio_effects/effect_aec.h> |
47 | #include <audio_route/audio_route.h> |
48 | |
49 | #include "libTVaudio/audio/audio_effect_control.h" |
50 | #include "audio_hw.h" |
51 | #include "audio_hw_utils.h" |
52 | #include "spdifenc_wrap.h" |
53 | //extern int spdifenc_write(const void *buffer, size_t numBytes); |
54 | //extern uint64_t spdifenc_get_total(); |
55 | extern void aml_audio_hwsync_clear_status(struct aml_stream_out *out); |
56 | extern int aml_audio_hwsync_find_frame(struct aml_stream_out *out, |
57 | const void *in_buffer, size_t in_bytes, uint64_t *cur_pts, int *outsize); |
58 | |
59 | /* ALSA cards for AML */ |
60 | #define CARD_AMLOGIC_BOARD 0 |
61 | /* ALSA ports for AML */ |
62 | #define PORT_I2S 0 |
63 | #define PORT_SPDIF 1 |
64 | #define PORT_PCM 2 |
65 | /* number of frames per period */ |
66 | #define DEFAULT_PERIOD_SIZE 1024 |
67 | #define DEFAULT_CAPTURE_PERIOD_SIZE 1024 |
68 | //static unsigned PERIOD_SIZE = DEFAULT_PERIOD_SIZE; |
69 | static unsigned CAPTURE_PERIOD_SIZE = DEFAULT_CAPTURE_PERIOD_SIZE; |
70 | /* number of periods for low power playback */ |
71 | #define PLAYBACK_PERIOD_COUNT 4 |
72 | /* number of periods for capture */ |
73 | #define CAPTURE_PERIOD_COUNT 4 |
74 | |
75 | /* minimum sleep time in out_write() when write threshold is not reached */ |
76 | #define MIN_WRITE_SLEEP_US 5000 |
77 | |
78 | #define RESAMPLER_BUFFER_FRAMES (PERIOD_SIZE * 6) |
79 | #define RESAMPLER_BUFFER_SIZE (4 * RESAMPLER_BUFFER_FRAMES) |
80 | |
81 | #define NSEC_PER_SECOND 1000000000ULL |
82 | |
83 | //static unsigned int DEFAULT_OUT_SAMPLING_RATE = 48000; |
84 | |
85 | /* sampling rate when using MM low power port */ |
86 | #define MM_LOW_POWER_SAMPLING_RATE 44100 |
87 | /* sampling rate when using MM full power port */ |
88 | #define MM_FULL_POWER_SAMPLING_RATE 48000 |
89 | /* sampling rate when using VX port for narrow band */ |
90 | #define VX_NB_SAMPLING_RATE 8000 |
91 | #define MIXER_XML_PATH "/system/etc/mixer_paths.xml" |
92 | |
93 | #define TSYNC_FIRSTAPTS "/sys/class/tsync/firstapts" |
94 | #define TSYNC_PCRSCR "/sys/class/tsync/pts_pcrscr" |
95 | #define TSYNC_EVENT "/sys/class/tsync/event" |
96 | #define TSYNC_APTS "/sys/class/tsync/pts_audio" |
97 | #define SYSTIME_CORRECTION_THRESHOLD (90000*10/100) |
98 | #define APTS_DISCONTINUE_THRESHOLD_MIN (90000/1000*100) |
99 | #define APTS_DISCONTINUE_THRESHOLD_MAX (5*90000) |
100 | |
101 | static const struct pcm_config pcm_config_out = { |
102 | .channels = 2, |
103 | .rate = MM_FULL_POWER_SAMPLING_RATE, |
104 | .period_size = DEFAULT_PERIOD_SIZE, |
105 | .period_count = PLAYBACK_PERIOD_COUNT, |
106 | .format = PCM_FORMAT_S16_LE, |
107 | }; |
108 | |
109 | static const struct pcm_config pcm_config_out_direct = { |
110 | .channels = 2, |
111 | .rate = MM_FULL_POWER_SAMPLING_RATE, |
112 | .period_size = DEFAULT_PERIOD_SIZE, |
113 | .period_count = PLAYBACK_PERIOD_COUNT, |
114 | .format = PCM_FORMAT_S16_LE, |
115 | }; |
116 | |
117 | static const struct pcm_config pcm_config_in = { |
118 | .channels = 2, |
119 | .rate = MM_FULL_POWER_SAMPLING_RATE, |
120 | .period_size = DEFAULT_CAPTURE_PERIOD_SIZE, |
121 | .period_count = CAPTURE_PERIOD_COUNT, |
122 | .format = PCM_FORMAT_S16_LE, |
123 | }; |
124 | |
125 | static const struct pcm_config pcm_config_bt = { |
126 | .channels = 1, |
127 | .rate = VX_NB_SAMPLING_RATE, |
128 | .period_size = DEFAULT_PERIOD_SIZE, |
129 | .period_count = PLAYBACK_PERIOD_COUNT, |
130 | .format = PCM_FORMAT_S16_LE, |
131 | }; |
132 | |
133 | #define HW_SYNC_STATE_HEADER 0 |
134 | #define HW_SYNC_STATE_BODY 1 |
135 | #define HW_SYNC_STATE_RESYNC 2 |
136 | |
137 | static void select_output_device(struct aml_audio_device *adev); |
138 | static void select_input_device(struct aml_audio_device *adev); |
139 | static void select_devices(struct aml_audio_device *adev); |
140 | static int adev_set_voice_volume(struct audio_hw_device *dev, float volume); |
141 | static int do_input_standby(struct aml_stream_in *in); |
142 | static int do_output_standby(struct aml_stream_out *out); |
143 | static uint32_t out_get_sample_rate(const struct audio_stream *stream); |
144 | static int out_pause(struct audio_stream_out *stream); |
145 | static inline short CLIP(int r) |
146 | { |
147 | return (r > 0x7fff) ? 0x7fff : |
148 | (r < -0x8000) ? 0x8000 : |
149 | r; |
150 | } |
151 | //code here for audio hal mixer when hwsync with af mixer output stream output |
152 | //at the same,need do a software mixer in audio hal. |
153 | static int aml_hal_mixer_init(struct aml_hal_mixer *mixer) |
154 | { |
155 | pthread_mutex_lock(&mixer->lock); |
156 | mixer->wp = 0; |
157 | mixer->rp = 0; |
158 | mixer->buf_size = AML_HAL_MIXER_BUF_SIZE; |
159 | mixer->need_cache_flag = 1; |
160 | pthread_mutex_unlock(&mixer->lock); |
161 | return 0; |
162 | } |
163 | static uint aml_hal_mixer_get_space(struct aml_hal_mixer *mixer) |
164 | { |
165 | unsigned space; |
166 | if (mixer->wp >= mixer->rp) { |
167 | space = mixer->buf_size - (mixer->wp - mixer->rp); |
168 | } else { |
169 | space = mixer->rp - mixer->wp; |
170 | } |
171 | return space > 64 ? (space - 64) : 0; |
172 | } |
173 | static int aml_hal_mixer_get_content(struct aml_hal_mixer *mixer) |
174 | { |
175 | unsigned content = 0; |
176 | pthread_mutex_lock(&mixer->lock); |
177 | if (mixer->wp >= mixer->rp) { |
178 | content = mixer->wp - mixer->rp; |
179 | } else { |
180 | content = mixer->wp - mixer->rp + mixer->buf_size; |
181 | } |
182 | //ALOGI("wp %d,rp %d\n",mixer->wp,mixer->rp); |
183 | pthread_mutex_unlock(&mixer->lock); |
184 | return content; |
185 | } |
186 | //we assue the cached size is always smaller then buffer size |
187 | //need called by device mutux locked |
188 | static int aml_hal_mixer_write(struct aml_hal_mixer *mixer, const void *w_buf, uint size) |
189 | { |
190 | unsigned space; |
191 | unsigned write_size = size; |
192 | unsigned tail = 0; |
193 | pthread_mutex_lock(&mixer->lock); |
194 | space = aml_hal_mixer_get_space(mixer); |
195 | if (space < size) { |
196 | ALOGI("write data no space,space %d,size %d,rp %d,wp %d,reset all ptr\n", space, size, mixer->rp, mixer->wp); |
197 | mixer->wp = 0; |
198 | mixer->rp = 0; |
199 | } |
200 | //TODO |
201 | if (write_size > space) { |
202 | write_size = space; |
203 | } |
204 | if (write_size + mixer->wp > mixer->buf_size) { |
205 | tail = mixer->buf_size - mixer->wp; |
206 | memcpy(mixer->start_buf + mixer->wp, w_buf, tail); |
207 | write_size -= tail; |
208 | memcpy(mixer->start_buf, (unsigned char*)w_buf + tail, write_size); |
209 | mixer->wp = write_size; |
210 | } else { |
211 | memcpy(mixer->start_buf + mixer->wp, w_buf, write_size); |
212 | mixer->wp += write_size; |
213 | mixer->wp %= AML_HAL_MIXER_BUF_SIZE; |
214 | } |
215 | pthread_mutex_unlock(&mixer->lock); |
216 | return size; |
217 | } |
218 | //need called by device mutux locked |
219 | static int aml_hal_mixer_read(struct aml_hal_mixer *mixer, void *r_buf, uint size) |
220 | { |
221 | unsigned cached_size; |
222 | unsigned read_size = size; |
223 | unsigned tail = 0; |
224 | cached_size = aml_hal_mixer_get_content(mixer); |
225 | pthread_mutex_lock(&mixer->lock); |
226 | // we always assue we have enough data to read when hwsync enabled. |
227 | // if we do not have,insert zero data. |
228 | if (cached_size < size) { |
229 | ALOGI("read data has not enough data to mixer,read %d, have %d,rp %d,wp %d\n", size, cached_size, mixer->rp, mixer->wp); |
230 | memset((unsigned char*)r_buf + cached_size, 0, size - cached_size); |
231 | read_size = cached_size; |
232 | } |
233 | if (read_size + mixer->rp > mixer->buf_size) { |
234 | tail = mixer->buf_size - mixer->rp; |
235 | memcpy(r_buf, mixer->start_buf + mixer->rp, tail); |
236 | read_size -= tail; |
237 | memcpy((unsigned char*)r_buf + tail, mixer->start_buf, read_size); |
238 | mixer->rp = read_size; |
239 | } else { |
240 | memcpy(r_buf, mixer->start_buf + mixer->rp, read_size); |
241 | mixer->rp += read_size; |
242 | mixer->rp %= AML_HAL_MIXER_BUF_SIZE; |
243 | } |
244 | pthread_mutex_unlock(&mixer->lock); |
245 | return size; |
246 | } |
247 | // aml audio hal mixer code end |
248 | #if 0 |
249 | static inline bool hwsync_header_valid(uint8_t *header) |
250 | { |
251 | return (header[0] == 0x55) && |
252 | (header[1] == 0x55) && |
253 | (header[2] == 0x00) && |
254 | (header[3] == 0x01); |
255 | } |
256 | |
257 | static inline uint64_t hwsync_header_get_pts(uint8_t *header) |
258 | { |
259 | return (((uint64_t)header[8]) << 56) | |
260 | (((uint64_t)header[9]) << 48) | |
261 | (((uint64_t)header[10]) << 40) | |
262 | (((uint64_t)header[11]) << 32) | |
263 | (((uint64_t)header[12]) << 24) | |
264 | (((uint64_t)header[13]) << 16) | |
265 | (((uint64_t)header[14]) << 8) | |
266 | ((uint64_t)header[15]); |
267 | } |
268 | |
269 | static inline uint32_t hwsync_header_get_size(uint8_t *header) |
270 | { |
271 | return (((uint32_t)header[4]) << 24) | |
272 | (((uint32_t)header[5]) << 16) | |
273 | (((uint32_t)header[6]) << 8) | |
274 | ((uint32_t)header[7]); |
275 | } |
276 | #endif |
277 | |
278 | static void select_devices(struct aml_audio_device *adev) |
279 | { |
280 | LOGFUNC("%s(mode=%d, out_device=%#x)", __FUNCTION__, adev->mode, adev->out_device); |
281 | int headset_on; |
282 | int headphone_on; |
283 | int speaker_on; |
284 | int hdmi_on; |
285 | int earpiece; |
286 | int mic_in; |
287 | int headset_mic; |
288 | |
289 | headset_on = adev->out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET; |
290 | headphone_on = adev->out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; |
291 | speaker_on = adev->out_device & AUDIO_DEVICE_OUT_SPEAKER; |
292 | hdmi_on = adev->out_device & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
293 | earpiece = adev->out_device & AUDIO_DEVICE_OUT_EARPIECE; |
294 | mic_in = adev->in_device & (AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BACK_MIC); |
295 | headset_mic = adev->in_device & AUDIO_DEVICE_IN_WIRED_HEADSET; |
296 | |
297 | LOGFUNC("%s : hs=%d , hp=%d, sp=%d, hdmi=0x%x,earpiece=0x%x", __func__, |
298 | headset_on, headphone_on, speaker_on, hdmi_on, earpiece); |
299 | LOGFUNC("%s : in_device(%#x), mic_in(%#x), headset_mic(%#x)", __func__, |
300 | adev->in_device, mic_in, headset_mic); |
301 | audio_route_reset(adev->ar); |
302 | if (hdmi_on) { |
303 | audio_route_apply_path(adev->ar, "hdmi"); |
304 | } |
305 | if (headphone_on || headset_on) { |
306 | audio_route_apply_path(adev->ar, "headphone"); |
307 | } |
308 | if (speaker_on || earpiece) { |
309 | audio_route_apply_path(adev->ar, "speaker"); |
310 | } |
311 | if (mic_in) { |
312 | audio_route_apply_path(adev->ar, "main_mic"); |
313 | } |
314 | if (headset_mic) { |
315 | audio_route_apply_path(adev->ar, "headset-mic"); |
316 | } |
317 | |
318 | audio_route_update_mixer(adev->ar); |
319 | |
320 | } |
321 | |
322 | static void select_mode(struct aml_audio_device *adev) |
323 | { |
324 | LOGFUNC("%s(out_device=%#x)", __FUNCTION__, adev->out_device); |
325 | LOGFUNC("%s(in_device=%#x)", __FUNCTION__, adev->in_device); |
326 | return; |
327 | |
328 | /* force earpiece route for in call state if speaker is the |
329 | only currently selected route. This prevents having to tear |
330 | down the modem PCMs to change route from speaker to earpiece |
331 | after the ringtone is played, but doesn't cause a route |
332 | change if a headset or bt device is already connected. If |
333 | speaker is not the only thing active, just remove it from |
334 | the route. We'll assume it'll never be used initally during |
335 | a call. This works because we're sure that the audio policy |
336 | manager will update the output device after the audio mode |
337 | change, even if the device selection did not change. */ |
338 | if ((adev->out_device & AUDIO_DEVICE_OUT_ALL) == AUDIO_DEVICE_OUT_SPEAKER) { |
339 | adev->in_device = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN; |
340 | } else { |
341 | adev->out_device &= ~AUDIO_DEVICE_OUT_SPEAKER; |
342 | } |
343 | |
344 | return; |
345 | } |
346 | |
347 | static int get_aml_card(void) |
348 | { |
349 | int card = -1, err = 0; |
350 | int fd = -1; |
351 | unsigned fileSize = 512; |
352 | char *read_buf = NULL, *pd = NULL; |
353 | static const char *const SOUND_CARDS_PATH = "/proc/asound/cards"; |
354 | fd = open(SOUND_CARDS_PATH, O_RDONLY); |
355 | if (fd < 0) { |
356 | ALOGE("ERROR: failed to open config file %s error: %d\n", SOUND_CARDS_PATH, errno); |
357 | close(fd); |
358 | return -EINVAL; |
359 | } |
360 | |
361 | read_buf = (char *)malloc(fileSize); |
362 | if (!read_buf) { |
363 | ALOGE("Failed to malloc read_buf"); |
364 | close(fd); |
365 | return -ENOMEM; |
366 | } |
367 | memset(read_buf, 0x0, fileSize); |
368 | err = read(fd, read_buf, fileSize); |
369 | if (fd < 0) { |
370 | ALOGE("ERROR: failed to read config file %s error: %d\n", SOUND_CARDS_PATH, errno); |
371 | close(fd); |
372 | return -EINVAL; |
373 | } |
374 | pd = strstr(read_buf, "AML"); |
375 | card = *(pd - 3) - '0'; |
376 | |
377 | OUT: |
378 | free(read_buf); |
379 | close(fd); |
380 | return card; |
381 | } |
382 | |
383 | static int get_pcm_bt_port(void) |
384 | { |
385 | int port = -1, err = 0; |
386 | int fd = -1; |
387 | unsigned fileSize = 512; |
388 | char *read_buf = NULL, *pd = NULL; |
389 | static const char *const SOUND_PCM_PATH = "/proc/asound/pcm"; |
390 | fd = open(SOUND_PCM_PATH, O_RDONLY); |
391 | if (fd < 0) { |
392 | ALOGE("ERROR: failed to open config file %s error: %d\n", SOUND_PCM_PATH, errno); |
393 | close(fd); |
394 | return -EINVAL; |
395 | } |
396 | |
397 | read_buf = (char *)malloc(fileSize); |
398 | if (!read_buf) { |
399 | ALOGE("Failed to malloc read_buf"); |
400 | close(fd); |
401 | return -ENOMEM; |
402 | } |
403 | memset(read_buf, 0x0, fileSize); |
404 | err = read(fd, read_buf, fileSize); |
405 | if (fd < 0) { |
406 | ALOGE("ERROR: failed to read config file %s error: %d\n", SOUND_PCM_PATH, errno); |
407 | close(fd); |
408 | return -EINVAL; |
409 | } |
410 | pd = strstr(read_buf, "pcm2bt-pcm"); |
411 | port = *(pd + 11) - '0'; |
412 | |
413 | OUT: |
414 | free(read_buf); |
415 | close(fd); |
416 | return port; |
417 | } |
418 | |
419 | static int get_spdif_port() |
420 | { |
421 | int port = -1, err = 0; |
422 | int fd = -1; |
423 | unsigned fileSize = 512; |
424 | char *read_buf = NULL, *pd = NULL; |
425 | static const char *const SOUND_PCM_PATH = "/proc/asound/pcm"; |
426 | fd = open(SOUND_PCM_PATH, O_RDONLY); |
427 | if (fd < 0) { |
428 | ALOGE("ERROR: failed to open config file %s error: %d\n", SOUND_PCM_PATH, errno); |
429 | close(fd); |
430 | return -EINVAL; |
431 | } |
432 | |
433 | read_buf = (char *)malloc(fileSize); |
434 | if (!read_buf) { |
435 | ALOGE("Failed to malloc read_buf"); |
436 | close(fd); |
437 | return -ENOMEM; |
438 | } |
439 | memset(read_buf, 0x0, fileSize); |
440 | err = read(fd, read_buf, fileSize); |
441 | if (fd < 0) { |
442 | ALOGE("ERROR: failed to read config file %s error: %d\n", SOUND_PCM_PATH, errno); |
443 | close(fd); |
444 | return -EINVAL; |
445 | } |
446 | pd = strstr(read_buf, "SPDIF"); |
447 | port = *(pd - 3) - '0'; |
448 | |
449 | OUT: |
450 | free(read_buf); |
451 | close(fd); |
452 | return port; |
453 | } |
454 | /* must be called with hw device and output stream mutexes locked */ |
455 | static int start_output_stream(struct aml_stream_out *out) |
456 | { |
457 | struct aml_audio_device *adev = out->dev; |
458 | unsigned int card = CARD_AMLOGIC_BOARD; |
459 | unsigned int port = PORT_I2S; |
460 | int ret; |
461 | int i = 0; |
462 | struct aml_stream_out *out_removed = NULL; |
463 | bool hwsync_lpcm = (out->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC && audio_is_linear_pcm(out->hal_format)); |
464 | LOGFUNC("%s(adev->out_device=%#x, adev->mode=%d)", |
465 | __FUNCTION__, adev->out_device, adev->mode); |
466 | if (adev->mode != AUDIO_MODE_IN_CALL) { |
467 | /* FIXME: only works if only one output can be active at a time */ |
468 | select_devices(adev); |
469 | } |
470 | if (out->hw_sync_mode == true) { |
471 | adev->hwsync_output = out; |
472 | #if 0 |
473 | for (i = 0; i < MAX_STREAM_NUM; i++) { |
474 | if (adev->active_output[i]) { |
475 | out_removed = adev->active_output[i]; |
476 | pthread_mutex_lock(&out_removed->lock); |
477 | if (!out_removed->standby) { |
478 | ALOGI("hwsync start,force %p standby\n", out_removed); |
479 | do_output_standby(out_removed); |
480 | } |
481 | pthread_mutex_unlock(&out_removed->lock); |
482 | } |
483 | } |
484 | #endif |
485 | } |
486 | card = get_aml_card(); |
487 | if (adev->out_device & AUDIO_DEVICE_OUT_ALL_SCO) { |
488 | port = PORT_PCM; |
489 | out->config = pcm_config_bt; |
490 | } else if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT && !hwsync_lpcm) { |
491 | port = PORT_SPDIF; |
492 | } |
493 | |
494 | LOGFUNC("*%s, open card(%d) port(%d)", __FUNCTION__, card, port); |
495 | |
496 | /* default to low power: will be corrected in out_write if necessary before first write to |
497 | * tinyalsa. |
498 | */ |
499 | out->write_threshold = out->config.period_size * PLAYBACK_PERIOD_COUNT; |
500 | out->config.start_threshold = out->config.period_size * PLAYBACK_PERIOD_COUNT; |
501 | out->config.avail_min = 0;//SHORT_PERIOD_SIZE; |
502 | //added by xujian for NTS hwsync/system stream mix smooth playback. |
503 | //we need re-use the tinyalsa pcm handle by all the output stream, including |
504 | //hwsync direct output stream,system mixer output stream. |
505 | //TODO we need diff the code with AUDIO_DEVICE_OUT_ALL_SCO. |
506 | //as it share the same hal but with the different card id. |
507 | //TODO need reopen the tinyalsa card when sr/ch changed, |
508 | if (adev->pcm == NULL) { |
509 | out->pcm = pcm_open(card, port, PCM_OUT /*| PCM_MMAP | PCM_NOIRQ*/, &(out->config)); |
510 | if (!pcm_is_ready(out->pcm)) { |
511 | ALOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm)); |
512 | pcm_close(out->pcm); |
513 | return -ENOMEM; |
514 | } |
515 | if (out->config.rate != out_get_sample_rate(&out->stream.common)) { |
516 | LOGFUNC("%s(out->config.rate=%d, out->config.channels=%d)", |
517 | __FUNCTION__, out->config.rate, out->config.channels); |
518 | ret = create_resampler(out_get_sample_rate(&out->stream.common), |
519 | out->config.rate, |
520 | out->config.channels, |
521 | RESAMPLER_QUALITY_DEFAULT, |
522 | NULL, |
523 | &out->resampler); |
524 | if (ret != 0) { |
525 | ALOGE("cannot create resampler for output"); |
526 | return -ENOMEM; |
527 | } |
528 | out->buffer_frames = (out->config.period_size * out->config.rate) / |
529 | out_get_sample_rate(&out->stream.common) + 1; |
530 | out->buffer = malloc(pcm_frames_to_bytes(out->pcm, out->buffer_frames)); |
531 | if (out->buffer == NULL) { |
532 | ALOGE("cannot malloc memory for out->buffer"); |
533 | return -ENOMEM; |
534 | } |
535 | } |
536 | adev->pcm = out->pcm; |
537 | ALOGI("device pcm %p\n", adev->pcm); |
538 | } else { |
539 | ALOGI("stream %p share the pcm %p\n", out, adev->pcm); |
540 | out->pcm = adev->pcm; |
541 | } |
542 | LOGFUNC("channels=%d---format=%d---period_count%d---period_size%d---rate=%d---", |
543 | out->config.channels, out->config.format, out->config.period_count, |
544 | out->config.period_size, out->config.rate); |
545 | |
546 | if (adev->echo_reference != NULL) { |
547 | out->echo_reference = adev->echo_reference; |
548 | } |
549 | if (out->resampler) { |
550 | out->resampler->reset(out->resampler); |
551 | } |
552 | if (out->is_tv_platform == 1) { |
553 | sysfs_set_sysfs_str("/sys/class/amhdmitx/amhdmitx0/aud_output_chs", "2:2"); |
554 | } |
555 | //set_codec_type(0); |
556 | if (out->hw_sync_mode == 1) { |
557 | LOGFUNC("start_output_stream with hw sync enable %p\n", out); |
558 | } |
559 | for (i = 0; i < MAX_STREAM_NUM; i++) { |
560 | if (adev->active_output[i] == NULL) { |
561 | ALOGI("store out (%p) to index %d\n", out, i); |
562 | adev->active_output[i] = out; |
563 | adev->active_output_count++; |
564 | break; |
565 | } |
566 | } |
567 | if (i == MAX_STREAM_NUM) { |
568 | ALOGE("error,no space to store the dev stream \n"); |
569 | } |
570 | return 0; |
571 | } |
572 | |
573 | /* dircet stream mainly map to audio HDMI port */ |
574 | static int start_output_stream_direct(struct aml_stream_out *out) |
575 | { |
576 | struct aml_audio_device *adev = out->dev; |
577 | unsigned int card = CARD_AMLOGIC_BOARD; |
578 | unsigned int port = PORT_SPDIF; |
579 | int ret = 0; |
580 | |
581 | int codec_type = get_codec_type(out->hal_format); |
582 | if (codec_type == AUDIO_FORMAT_PCM && out->config.rate > 48000 && (out->flags & AUDIO_OUTPUT_FLAG_DIRECT)) { |
583 | ALOGI("start output stream for high sample rate pcm for direct mode\n"); |
584 | codec_type = TYPE_PCM_HIGH_SR; |
585 | } |
586 | if (codec_type == AUDIO_FORMAT_PCM && out->config.channels >= 6 && (out->flags & AUDIO_OUTPUT_FLAG_DIRECT)) { |
587 | ALOGI("start output stream for multi-channel pcm for direct mode\n"); |
588 | codec_type = TYPE_MULTI_PCM; |
589 | } |
590 | |
591 | card = get_aml_card(); |
592 | ALOGI("%s: hdmi sound card id %d,device id %d \n", __func__, card, port); |
593 | |
594 | if (out->config.channels == 6) { |
595 | ALOGI("round 6ch to 8 ch output \n"); |
596 | /* our hw only support 8 channel configure,so when 5.1,hw mask the last two channels*/ |
597 | sysfs_set_sysfs_str("/sys/class/amhdmitx/amhdmitx0/aud_output_chs", "6:7"); |
598 | out->config.channels = 8; |
599 | } |
600 | /* |
601 | * 8 channel audio only support 32 byte mode,so need convert them to |
602 | * PCM_FORMAT_S32_LE |
603 | */ |
604 | if (out->config.channels == 8) { |
605 | port = PORT_I2S; |
606 | out->config.format = PCM_FORMAT_S32_LE; |
607 | adev->out_device = AUDIO_DEVICE_OUT_SPEAKER; |
608 | ALOGI("[%s %d]8CH format output: set port/0 adev->out_device/%d\n", |
609 | __FUNCTION__, __LINE__, AUDIO_DEVICE_OUT_SPEAKER); |
610 | } |
611 | if (getprop_bool("media.libplayer.wfd")) { |
612 | out->config.period_size = PERIOD_SIZE; |
613 | } |
614 | switch (out->hal_format) { |
615 | case AUDIO_FORMAT_E_AC3: |
616 | out->config.period_size = PERIOD_SIZE * 2; |
617 | out->write_threshold = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE * 2; |
618 | out->config.start_threshold = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE * 2; |
619 | //as dd+ frame size = 1 and alsa sr as divide 16 |
620 | //out->raw_61937_frame_size = 16; |
621 | break; |
622 | case AUDIO_FORMAT_DTS_HD: |
623 | case AUDIO_FORMAT_TRUEHD: |
624 | out->config.period_size = PERIOD_SIZE * 4 * 2; |
625 | out->write_threshold = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE * 4 * 2; |
626 | out->config.start_threshold = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE * 4 * 2; |
627 | //out->raw_61937_frame_size = 16;//192k 2ch |
628 | break; |
629 | case AUDIO_FORMAT_PCM: |
630 | default: |
631 | out->config.period_size = PERIOD_SIZE; |
632 | out->write_threshold = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE; |
633 | out->config.start_threshold = PERIOD_SIZE * PLAYBACK_PERIOD_COUNT; |
634 | //out->raw_61937_frame_size = 4; |
635 | } |
636 | out->config.avail_min = 0; |
637 | set_codec_type(codec_type); |
638 | |
639 | if (out->config.channels == 6) { |
640 | ALOGI("round 6ch to 8 ch output \n"); |
641 | /* our hw only support 8 channel configure,so when 5.1,hw mask the last two channels*/ |
642 | sysfs_set_sysfs_str("/sys/class/amhdmitx/amhdmitx0/aud_output_chs", "6:7"); |
643 | out->config.channels = 8; |
644 | } |
645 | ALOGI("ALSA open configs: channels=%d, format=%d, period_count=%d, period_size=%d,,rate=%d", |
646 | out->config.channels, out->config.format, out->config.period_count, |
647 | out->config.period_size, out->config.rate); |
648 | |
649 | if (out->pcm == NULL) { |
650 | out->pcm = pcm_open(card, port, PCM_OUT, &out->config); |
651 | if (!pcm_is_ready(out->pcm)) { |
652 | ALOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm)); |
653 | pcm_close(out->pcm); |
654 | return -EINVAL; |
655 | } |
656 | } else { |
657 | ALOGE("stream %p share the pcm %p\n", out, out->pcm); |
658 | } |
659 | |
660 | if (codec_type_is_raw_data(codec_type) && !(out->flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)) { |
661 | spdifenc_init(out->pcm, out->hal_format); |
662 | out->spdif_enc_init_frame_write_sum = out->frame_write_sum; |
663 | } |
664 | out->codec_type = codec_type; |
665 | out->bytes_write_total = 0; |
666 | |
667 | if (out->hw_sync_mode == 1) { |
668 | LOGFUNC("start_output_stream with hw sync enable %p\n", out); |
669 | } |
670 | |
671 | return 0; |
672 | } |
673 | |
674 | static int check_input_parameters(uint32_t sample_rate, audio_format_t format, int channel_count) |
675 | { |
676 | LOGFUNC("%s(sample_rate=%d, format=%d, channel_count=%d)", __FUNCTION__, sample_rate, format, channel_count); |
677 | |
678 | if (format != AUDIO_FORMAT_PCM_16_BIT) { |
679 | return -EINVAL; |
680 | } |
681 | |
682 | if ((channel_count < 1) || (channel_count > 2)) { |
683 | return -EINVAL; |
684 | } |
685 | |
686 | switch (sample_rate) { |
687 | case 8000: |
688 | case 11025: |
689 | case 16000: |
690 | case 22050: |
691 | case 24000: |
692 | case 32000: |
693 | case 44100: |
694 | case 48000: |
695 | break; |
696 | default: |
697 | return -EINVAL; |
698 | } |
699 | |
700 | return 0; |
701 | } |
702 | |
703 | static size_t get_input_buffer_size(unsigned int period_size, uint32_t sample_rate, audio_format_t format, int channel_count) |
704 | { |
705 | size_t size; |
706 | |
707 | LOGFUNC("%s(sample_rate=%d, format=%d, channel_count=%d)", __FUNCTION__, sample_rate, format, channel_count); |
708 | |
709 | if (check_input_parameters(sample_rate, format, channel_count) != 0) { |
710 | return 0; |
711 | } |
712 | |
713 | /* take resampling into account and return the closest majoring |
714 | multiple of 16 frames, as audioflinger expects audio buffers to |
715 | be a multiple of 16 frames */ |
716 | if (period_size == 0) { |
717 | period_size = (pcm_config_in.period_size * sample_rate) / pcm_config_in.rate; |
718 | } |
719 | |
720 | size = period_size; |
721 | size = ((size + 15) / 16) * 16; |
722 | |
723 | return size * channel_count * sizeof(short); |
724 | } |
725 | |
726 | static void add_echo_reference(struct aml_stream_out *out, |
727 | struct echo_reference_itfe *reference) |
728 | { |
729 | pthread_mutex_lock(&out->lock); |
730 | out->echo_reference = reference; |
731 | pthread_mutex_unlock(&out->lock); |
732 | } |
733 | |
734 | static void remove_echo_reference(struct aml_stream_out *out, |
735 | struct echo_reference_itfe *reference) |
736 | { |
737 | pthread_mutex_lock(&out->lock); |
738 | if (out->echo_reference == reference) { |
739 | /* stop writing to echo reference */ |
740 | reference->write(reference, NULL); |
741 | out->echo_reference = NULL; |
742 | } |
743 | pthread_mutex_unlock(&out->lock); |
744 | } |
745 | |
746 | static void put_echo_reference(struct aml_audio_device *adev, |
747 | struct echo_reference_itfe *reference) |
748 | { |
749 | if (adev->echo_reference != NULL && |
750 | reference == adev->echo_reference) { |
751 | if (adev->active_output[0] != NULL) { |
752 | remove_echo_reference(adev->active_output[0], reference); |
753 | } |
754 | release_echo_reference(reference); |
755 | adev->echo_reference = NULL; |
756 | } |
757 | } |
758 | |
759 | static struct echo_reference_itfe *get_echo_reference(struct aml_audio_device *adev, |
760 | audio_format_t format __unused, |
761 | uint32_t channel_count, |
762 | uint32_t sampling_rate) |
763 | { |
764 | put_echo_reference(adev, adev->echo_reference); |
765 | if (adev->active_output[0] != NULL) { |
766 | struct audio_stream *stream = &adev->active_output[0]->stream.common; |
767 | uint32_t wr_channel_count = popcount(stream->get_channels(stream)); |
768 | uint32_t wr_sampling_rate = stream->get_sample_rate(stream); |
769 | |
770 | int status = create_echo_reference(AUDIO_FORMAT_PCM_16_BIT, |
771 | channel_count, |
772 | sampling_rate, |
773 | AUDIO_FORMAT_PCM_16_BIT, |
774 | wr_channel_count, |
775 | wr_sampling_rate, |
776 | &adev->echo_reference); |
777 | if (status == 0) { |
778 | add_echo_reference(adev->active_output[0], adev->echo_reference); |
779 | } |
780 | } |
781 | return adev->echo_reference; |
782 | } |
783 | |
784 | static int get_playback_delay(struct aml_stream_out *out, |
785 | size_t frames, |
786 | struct echo_reference_buffer *buffer) |
787 | { |
788 | |
789 | size_t kernel_frames; |
790 | int status; |
791 | status = pcm_get_htimestamp(out->pcm, &kernel_frames, &buffer->time_stamp); |
792 | if (status < 0) { |
793 | buffer->time_stamp.tv_sec = 0; |
794 | buffer->time_stamp.tv_nsec = 0; |
795 | buffer->delay_ns = 0; |
796 | ALOGV("get_playback_delay(): pcm_get_htimestamp error," |
797 | "setting playbackTimestamp to 0"); |
798 | return status; |
799 | } |
800 | kernel_frames = pcm_get_buffer_size(out->pcm) - kernel_frames; |
801 | ALOGV("~~pcm_get_buffer_size(out->pcm)=%d", pcm_get_buffer_size(out->pcm)); |
802 | /* adjust render time stamp with delay added by current driver buffer. |
803 | * Add the duration of current frame as we want the render time of the last |
804 | * sample being written. */ |
805 | buffer->delay_ns = (long)(((int64_t)(kernel_frames + frames) * 1000000000) / |
806 | out->config.rate); |
807 | |
808 | ALOGV("get_playback_delay time_stamp = [%ld].[%ld], delay_ns: [%d]," |
809 | "kernel_frames:[%d]", buffer->time_stamp.tv_sec , buffer->time_stamp.tv_nsec, |
810 | buffer->delay_ns, kernel_frames); |
811 | return 0; |
812 | } |
813 | |
814 | static uint32_t out_get_sample_rate(const struct audio_stream *stream) |
815 | { |
816 | const struct aml_stream_out *out = (const struct aml_stream_out *)stream; |
817 | unsigned int rate = out->hal_rate; |
818 | //ALOGV("out_get_sample_rate() = %d", rate); |
819 | return rate; |
820 | } |
821 | |
822 | static int out_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused) |
823 | { |
824 | return 0; |
825 | } |
826 | |
827 | static size_t out_get_buffer_size(const struct audio_stream *stream) |
828 | { |
829 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
830 | |
831 | //LOGFUNC("%s(out->config.rate=%d)", __FUNCTION__, out->config.rate); |
832 | |
833 | /* take resampling into account and return the closest majoring |
834 | * multiple of 16 frames, as audioflinger expects audio buffers to |
835 | * be a multiple of 16 frames |
836 | */ |
837 | size_t size = out->config.period_size; |
838 | switch (out->hal_format) { |
839 | case AUDIO_FORMAT_AC3: |
840 | case AUDIO_FORMAT_DTS: |
841 | if (out->flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO) { |
842 | size = 4 * PERIOD_SIZE * PLAYBACK_PERIOD_COUNT; |
843 | } else { |
844 | size = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE / 2; |
845 | } |
846 | break; |
847 | case AUDIO_FORMAT_E_AC3: |
848 | if (out->flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO) { |
849 | size = 16 * PERIOD_SIZE * PLAYBACK_PERIOD_COUNT; |
850 | } else { |
851 | size = PERIOD_SIZE; //2*PLAYBACK_PERIOD_COUNT*PERIOD_SIZE; |
852 | } |
853 | break; |
854 | case AUDIO_FORMAT_DTS_HD: |
855 | case AUDIO_FORMAT_TRUEHD: |
856 | if (out->flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO) { |
857 | size = 16 * PERIOD_SIZE * PLAYBACK_PERIOD_COUNT; |
858 | } else { |
859 | size = 4 * PLAYBACK_PERIOD_COUNT * PERIOD_SIZE; |
860 | } |
861 | break; |
862 | case AUDIO_FORMAT_PCM: |
863 | default: |
864 | size = PERIOD_SIZE; |
865 | } |
866 | size = ((size + 15) / 16) * 16; |
867 | return size * audio_stream_out_frame_size((struct audio_stream_out *)stream); |
868 | } |
869 | |
870 | static audio_channel_mask_t out_get_channels(const struct audio_stream *stream __unused) |
871 | { |
872 | //const struct aml_stream_out *out = (const struct aml_stream_out *)stream; |
873 | |
874 | return AUDIO_CHANNEL_OUT_STEREO; |
875 | } |
876 | |
877 | static audio_channel_mask_t out_get_channels_direct(const struct audio_stream *stream) |
878 | { |
879 | const struct aml_stream_out *out = (const struct aml_stream_out *)stream; |
880 | |
881 | return out->hal_channel_mask; |
882 | } |
883 | |
884 | static audio_format_t out_get_format(const struct audio_stream *stream __unused) |
885 | { |
886 | return AUDIO_FORMAT_PCM_16_BIT; |
887 | } |
888 | |
889 | static audio_format_t out_get_format_direct(const struct audio_stream *stream) |
890 | { |
891 | const struct aml_stream_out *out = (const struct aml_stream_out *)stream; |
892 | |
893 | return out->hal_format; |
894 | } |
895 | |
896 | static int out_set_format(struct audio_stream *stream __unused, audio_format_t format __unused) |
897 | { |
898 | return 0; |
899 | } |
900 | |
901 | /* must be called with hw device and output stream mutexes locked */ |
902 | static int do_output_standby(struct aml_stream_out *out) |
903 | { |
904 | struct aml_audio_device *adev = out->dev; |
905 | int i = 0; |
906 | |
907 | LOGFUNC("%s(%p)", __FUNCTION__, out); |
908 | |
909 | if (!out->standby) { |
910 | //commit here for hwsync/mix stream hal mixer |
911 | //pcm_close(out->pcm); |
912 | //out->pcm = NULL; |
913 | if (out->buffer) { |
914 | free(out->buffer); |
915 | out->buffer = NULL; |
916 | } |
917 | if (out->resampler) { |
918 | release_resampler(out->resampler); |
919 | out->resampler = NULL; |
920 | } |
921 | /* stop writing to echo reference */ |
922 | if (out->echo_reference != NULL) { |
923 | out->echo_reference->write(out->echo_reference, NULL); |
924 | out->echo_reference = NULL; |
925 | } |
926 | out->standby = 1; |
927 | for (i = 0; i < MAX_STREAM_NUM; i++) { |
928 | if (adev->active_output[i] == out) { |
929 | adev->active_output[i] = NULL; |
930 | adev->active_output_count--; |
931 | ALOGI("remove out (%p) from index %d\n", out, i); |
932 | break; |
933 | } |
934 | } |
935 | if (out->hw_sync_mode == 1 || adev->hwsync_output == out) { |
936 | #if 0 |
937 | //here to check if hwsync in pause status,if that,chear the status |
938 | //to release the sound card to other active output stream |
939 | if (out->pause_status == true && adev->active_output_count > 0) { |
940 | if (pcm_is_ready(out->pcm)) { |
941 | int r = pcm_ioctl(out->pcm, SNDRV_PCM_IOCTL_PAUSE, 0); |
942 | if (r < 0) { |
943 | ALOGE("here cannot resume channel\n"); |
944 | } else { |
945 | r = 0; |
946 | } |
947 | ALOGI("clear the hwsync output pause status.resume pcm\n"); |
948 | } |
949 | out->pause_status = false; |
950 | } |
951 | #endif |
952 | out->pause_status = false; |
953 | adev->hwsync_output = NULL; |
954 | ALOGI("clear hwsync_output when hwsync standby\n"); |
955 | } |
956 | if (i == MAX_STREAM_NUM) { |
957 | ALOGE("error, not found stream in dev stream list\n"); |
958 | } |
959 | /* no active output here,we can close the pcm to release the sound card now*/ |
960 | if (adev->active_output_count == 0) { |
961 | if (adev->pcm) { |
962 | ALOGI("close pcm %p\n", adev->pcm); |
963 | pcm_close(adev->pcm); |
964 | adev->pcm = NULL; |
965 | } |
966 | out->pause_status = false; |
967 | } |
968 | } |
969 | return 0; |
970 | } |
971 | /* must be called with hw device and output stream mutexes locked */ |
972 | static int do_output_standby_direct(struct aml_stream_out *out) |
973 | { |
974 | int status = 0; |
975 | |
976 | ALOGI("%s,out %p", __FUNCTION__, out); |
977 | |
978 | if (!out->standby) { |
979 | if (out->buffer) { |
980 | free(out->buffer); |
981 | out->buffer = NULL; |
982 | } |
983 | |
984 | out->standby = 1; |
985 | pcm_close(out->pcm); |
986 | out->pcm = NULL; |
987 | } |
988 | set_codec_type(TYPE_PCM); |
989 | /* clear the hdmitx channel config to default */ |
990 | if (out->multich == 6) { |
991 | sysfs_set_sysfs_str("/sys/class/amhdmitx/amhdmitx0/aud_output_chs", "0:0"); |
992 | } |
993 | return status; |
994 | } |
995 | static int out_standby(struct audio_stream *stream) |
996 | { |
997 | LOGFUNC("%s(%p)", __FUNCTION__, stream); |
998 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
999 | int status = 0; |
1000 | pthread_mutex_lock(&out->dev->lock); |
1001 | pthread_mutex_lock(&out->lock); |
1002 | status = do_output_standby(out); |
1003 | pthread_mutex_unlock(&out->lock); |
1004 | pthread_mutex_unlock(&out->dev->lock); |
1005 | return status; |
1006 | } |
1007 | |
1008 | static int out_standby_direct(struct audio_stream *stream) |
1009 | { |
1010 | struct aml_stream_out *out = (struct aml_stream_out *) stream; |
1011 | int status = 0; |
1012 | |
1013 | ALOGI("%s(%p),out %p", __FUNCTION__, stream, out); |
1014 | |
1015 | pthread_mutex_lock(&out->dev->lock); |
1016 | pthread_mutex_lock(&out->lock); |
1017 | if (!out->standby) { |
1018 | if (out->buffer) { |
1019 | free(out->buffer); |
1020 | out->buffer = NULL; |
1021 | } |
1022 | |
1023 | out->standby = 1; |
1024 | pcm_close(out->pcm); |
1025 | out->pcm = NULL; |
1026 | } |
1027 | set_codec_type(TYPE_PCM); |
1028 | /* clear the hdmitx channel config to default */ |
1029 | if (out->multich == 6) { |
1030 | sysfs_set_sysfs_str("/sys/class/amhdmitx/amhdmitx0/aud_output_chs", "0:0"); |
1031 | } |
1032 | pthread_mutex_unlock(&out->lock); |
1033 | pthread_mutex_unlock(&out->dev->lock); |
1034 | return status; |
1035 | } |
1036 | |
1037 | static int out_dump(const struct audio_stream *stream __unused, int fd __unused) |
1038 | { |
1039 | LOGFUNC("%s(%p, %d)", __FUNCTION__, stream, fd); |
1040 | return 0; |
1041 | } |
1042 | static int |
1043 | out_flush(struct audio_stream_out *stream) |
1044 | { |
1045 | LOGFUNC("%s(%p)", __FUNCTION__, stream); |
1046 | struct aml_stream_out *out = (struct aml_stream_out *) stream; |
1047 | struct aml_audio_device *adev = out->dev; |
1048 | int ret = 0; |
1049 | bool hwsync_lpcm = (out->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC && audio_is_linear_pcm(out->hal_format)); |
1050 | do_standby_func standy_func = NULL; |
1051 | if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT && !hwsync_lpcm) { |
1052 | standy_func = do_output_standby_direct; |
1053 | } else { |
1054 | standy_func = do_output_standby; |
1055 | } |
1056 | pthread_mutex_lock(&adev->lock); |
1057 | pthread_mutex_lock(&out->lock); |
1058 | if (out->pause_status == true) { |
1059 | // when pause status, set status prepare to avoid static pop sound |
1060 | ret = pcm_ioctl(out->pcm, SNDRV_PCM_IOCTL_PREPARE); |
1061 | if (ret < 0) { |
1062 | ALOGE("cannot prepare pcm!"); |
1063 | goto exit; |
1064 | } |
1065 | } |
1066 | standy_func(out); |
1067 | out->frame_write_sum = 0; |
1068 | out->last_frames_postion = 0; |
1069 | exit: |
1070 | pthread_mutex_unlock(&adev->lock); |
1071 | pthread_mutex_unlock(&out->lock); |
1072 | return 0; |
1073 | } |
1074 | static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) |
1075 | { |
1076 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
1077 | struct aml_audio_device *adev = out->dev; |
1078 | struct aml_stream_in *in; |
1079 | struct str_parms *parms; |
1080 | char *str; |
1081 | char value[32]; |
1082 | int ret; |
1083 | uint val = 0; |
1084 | bool force_input_standby = false; |
1085 | bool hwsync_lpcm = (out->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC && audio_is_linear_pcm(out->hal_format)); |
1086 | do_standby_func standy_func = NULL; |
1087 | do_startup_func startup_func = NULL; |
1088 | if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT && !hwsync_lpcm) { |
1089 | standy_func = do_output_standby_direct; |
1090 | startup_func = start_output_stream_direct; |
1091 | } else { |
1092 | standy_func = do_output_standby; |
1093 | startup_func = start_output_stream; |
1094 | } |
1095 | LOGFUNC("%s(kvpairs(%s), out_device=%#x)", __FUNCTION__, kvpairs, adev->out_device); |
1096 | parms = str_parms_create_str(kvpairs); |
1097 | |
1098 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); |
1099 | if (ret >= 0) { |
1100 | val = atoi(value); |
1101 | pthread_mutex_lock(&adev->lock); |
1102 | pthread_mutex_lock(&out->lock); |
1103 | if (((adev->out_device & AUDIO_DEVICE_OUT_ALL) != val) && (val != 0)) { |
1104 | if (1/* out == adev->active_output[0]*/) { |
1105 | ALOGI("audio hw select device!\n"); |
1106 | standy_func(out); |
1107 | /* a change in output device may change the microphone selection */ |
1108 | if (adev->active_input && |
1109 | adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) { |
1110 | force_input_standby = true; |
1111 | } |
1112 | /* force standby if moving to/from HDMI */ |
1113 | if (((val & AUDIO_DEVICE_OUT_AUX_DIGITAL) ^ |
1114 | (adev->out_device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) || |
1115 | ((val & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) ^ |
1116 | (adev->out_device & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET))) { |
1117 | standy_func(out); |
1118 | } |
1119 | } |
1120 | adev->out_device &= ~AUDIO_DEVICE_OUT_ALL; |
1121 | adev->out_device |= val; |
1122 | select_devices(adev); |
1123 | } |
1124 | pthread_mutex_unlock(&out->lock); |
1125 | if (force_input_standby) { |
1126 | in = adev->active_input; |
1127 | pthread_mutex_lock(&in->lock); |
1128 | do_input_standby(in); |
1129 | pthread_mutex_unlock(&in->lock); |
1130 | } |
1131 | pthread_mutex_unlock(&adev->lock); |
1132 | goto exit; |
1133 | } |
1134 | int sr = 0; |
1135 | ret = str_parms_get_int(parms, AUDIO_PARAMETER_STREAM_SAMPLING_RATE, &sr); |
1136 | if (ret >= 0) { |
1137 | if (sr > 0) { |
1138 | struct pcm_config *config = &out->config; |
1139 | ALOGI("audio hw sampling_rate change from %d to %d \n", config->rate, sr); |
1140 | config->rate = sr; |
1141 | pthread_mutex_lock(&adev->lock); |
1142 | pthread_mutex_lock(&out->lock); |
1143 | if (!out->standby) { |
1144 | standy_func(out); |
1145 | startup_func(out); |
1146 | out->standby = 0; |
1147 | } |
1148 | pthread_mutex_unlock(&adev->lock); |
1149 | pthread_mutex_unlock(&out->lock); |
1150 | } |
1151 | goto exit; |
1152 | } |
1153 | int frame_size = 0; |
1154 | ret = str_parms_get_int(parms, AUDIO_PARAMETER_STREAM_FRAME_COUNT, &frame_size); |
1155 | if (ret >= 0) { |
1156 | if (frame_size > 0) { |
1157 | struct pcm_config *config = &out->config; |
1158 | ALOGI("audio hw frame size change from %d to %d \n", config->period_size, frame_size); |
1159 | config->period_size = frame_size; |
1160 | pthread_mutex_lock(&adev->lock); |
1161 | pthread_mutex_lock(&out->lock); |
1162 | if (!out->standby) { |
1163 | standy_func(out); |
1164 | startup_func(out); |
1165 | out->standby = 0; |
1166 | } |
1167 | pthread_mutex_unlock(&adev->lock); |
1168 | pthread_mutex_unlock(&out->lock); |
1169 | } |
1170 | goto exit; |
1171 | } |
1172 | int EQ_parameters[5] = {0, 0, 0, 0, 0}; |
1173 | char tmp[2]; |
1174 | int data = 0, i = 0; |
1175 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_EQ, value, sizeof(value)); |
1176 | //ALOGI("audio effect EQ parameters are %s\n", value); |
1177 | if (ret >= 0) { |
1178 | for (i; i < 5; i++) { |
1179 | tmp[0] = value[2 * i]; |
1180 | tmp[1] = value[2 * i + 1]; |
1181 | data = atoi(tmp); |
1182 | EQ_parameters[i] = data - 10; |
1183 | } |
1184 | ALOGI("audio effect EQ parameters are %d,%d,%d,%d,%d\n", EQ_parameters[0], |
1185 | EQ_parameters[1], EQ_parameters[2], EQ_parameters[3], EQ_parameters[4]); |
1186 | ret = 0; |
1187 | HPEQ_setParameter(EQ_parameters[0], EQ_parameters[1], |
1188 | EQ_parameters[2], EQ_parameters[3], EQ_parameters[4]); |
1189 | goto exit; |
1190 | } |
1191 | int SRS_parameters[5] = {0, 0, 0, 0, 0}; |
1192 | char tmp1[3]; |
1193 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_SRS, value, sizeof(value)); |
1194 | //ALOGI("audio effect SRS parameters are %s\n", value); |
1195 | if (ret >= 0) { |
1196 | for (i; i < 5; i++) { |
1197 | tmp1[0] = value[3 * i]; |
1198 | tmp1[1] = value[3 * i + 1]; |
1199 | tmp1[2] = value[3 * i + 2]; |
1200 | SRS_parameters[i] = atoi(tmp1); |
1201 | } |
1202 | ALOGI("audio effect SRS parameters are %d,%d,%d,%d,%d\n", SRS_parameters[0], |
1203 | SRS_parameters[1], SRS_parameters[2], SRS_parameters[3], SRS_parameters[4]); |
1204 | ret = 0; |
1205 | srs_setParameter(SRS_parameters); |
1206 | goto exit; |
1207 | } |
1208 | int SRS_gain[2] = {0, 0}; |
1209 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_SRS_GAIN, value, sizeof(value)); |
1210 | if (ret >= 0) { |
1211 | for (i; i < 2; i++) { |
1212 | tmp1[0] = value[3 * i]; |
1213 | tmp1[1] = value[3 * i + 1]; |
1214 | tmp1[2] = value[3 * i + 2]; |
1215 | SRS_gain[i] = atoi(tmp1); |
1216 | } |
1217 | ALOGI("audio effect SRS input/output gain are %d,%d\n", SRS_gain[0], SRS_gain[1]); |
1218 | ret = 0; |
1219 | srs_set_gain(SRS_gain[0], SRS_gain[1]); |
1220 | goto exit; |
1221 | } |
1222 | int SRS_switch[3] = {0, 0, 0}; |
1223 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_SRS_SWITCH, value, sizeof(value)); |
1224 | if (ret >= 0) { |
1225 | for (i; i < 3; i++) { |
1226 | tmp[0] = value[2 * i]; |
1227 | tmp[1] = value[2 * i + 1]; |
1228 | SRS_switch[i] = atoi(tmp); |
1229 | } |
1230 | ALOGI("audio effect SRS switch %d, %d, %d\n", SRS_switch[0], SRS_switch[1], SRS_switch[2]); |
1231 | ret = 0; |
1232 | srs_surround_enable(SRS_switch[0]); |
1233 | srs_dialogclarity_enable(SRS_switch[1]); |
1234 | srs_truebass_enable(SRS_switch[2]); |
1235 | goto exit; |
1236 | } |
1237 | ret = str_parms_get_str(parms, "hw_av_sync", value, sizeof(value)); |
1238 | if (ret >= 0) { |
1239 | int hw_sync_id = atoi(value); |
1240 | unsigned char sync_enable = (hw_sync_id == 12345678) ? 1 : 0; |
1241 | ALOGI("(%p)set hw_sync_id %d,%s hw sync mode\n", |
1242 | out, hw_sync_id, sync_enable ? "enable" : "disable"); |
1243 | out->hw_sync_mode = sync_enable; |
1244 | out->first_apts_flag = false; |
1245 | pthread_mutex_lock(&adev->lock); |
1246 | pthread_mutex_lock(&out->lock); |
1247 | out->frame_write_sum = 0; |
1248 | out->last_frames_postion = 0; |
1249 | /* clear up previous playback output status */ |
1250 | if (!out->standby) { |
1251 | standy_func(out); |
1252 | } |
1253 | //adev->hwsync_output = sync_enable?out:NULL; |
1254 | if (sync_enable) { |
1255 | ALOGI("init hal mixer when hwsync\n"); |
1256 | aml_hal_mixer_init(&adev->hal_mixer); |
1257 | } |
1258 | pthread_mutex_unlock(&out->lock); |
1259 | pthread_mutex_unlock(&adev->lock); |
1260 | ret = 0; |
1261 | goto exit; |
1262 | } |
1263 | exit: |
1264 | str_parms_destroy(parms); |
1265 | return ret; |
1266 | } |
1267 | |
1268 | static char * out_get_parameters(const struct audio_stream *stream __unused, const char *keys __unused) |
1269 | { |
1270 | return strdup(""); |
1271 | } |
1272 | |
1273 | static uint32_t out_get_latency(const struct audio_stream_out *stream) |
1274 | { |
1275 | const struct aml_stream_out *out = (const struct aml_stream_out *)stream; |
1276 | uint32_t whole_latency; |
1277 | int ret = 0; |
1278 | snd_pcm_sframes_t frames = 0; |
1279 | whole_latency = (out->config.period_size * out->config.period_count * 1000) / out->config.rate; |
1280 | if (!out->pcm || !pcm_is_ready(out->pcm)) { |
1281 | return whole_latency; |
1282 | } |
1283 | ret = pcm_ioctl(out->pcm, SNDRV_PCM_IOCTL_DELAY, &frames); |
1284 | if (ret < 0) { |
1285 | return whole_latency; |
1286 | } |
1287 | return (frames * 1000) / out->config.rate/* (out->pcm->config.rate)*/; |
1288 | } |
1289 | |
1290 | static int out_set_volume(struct audio_stream_out *stream, float left, float right) |
1291 | { |
1292 | struct aml_stream_out *out = (struct aml_stream_out *) stream; |
1293 | out->volume_l = left; |
1294 | out->volume_r = right; |
1295 | return 0; |
1296 | } |
1297 | |
1298 | static int out_pause(struct audio_stream_out *stream) |
1299 | { |
1300 | LOGFUNC("out_pause(%p)\n", stream); |
1301 | |
1302 | struct aml_stream_out *out = (struct aml_stream_out *) stream; |
1303 | struct aml_audio_device *adev = out->dev; |
1304 | int r = 0; |
1305 | pthread_mutex_lock(&adev->lock); |
1306 | pthread_mutex_lock(&out->lock); |
1307 | if (out->standby || out->pause_status == true) { |
1308 | goto exit; |
1309 | } |
1310 | if (out->hw_sync_mode) { |
1311 | adev->hwsync_output = NULL; |
1312 | if (adev->active_output_count > 1) { |
1313 | ALOGI("more than one active stream,skip alsa hw pause\n"); |
1314 | goto exit1; |
1315 | } |
1316 | } |
1317 | if (pcm_is_ready(out->pcm)) { |
1318 | r = pcm_ioctl(out->pcm, SNDRV_PCM_IOCTL_PAUSE, 1); |
1319 | if (r < 0) { |
1320 | ALOGE("cannot pause channel\n"); |
1321 | } else { |
1322 | r = 0; |
1323 | } |
1324 | } |
1325 | exit1: |
1326 | if (out->hw_sync_mode) { |
1327 | sysfs_set_sysfs_str(TSYNC_EVENT, "AUDIO_PAUSE"); |
1328 | } |
1329 | out->pause_status = true; |
1330 | exit: |
1331 | pthread_mutex_unlock(&adev->lock); |
1332 | pthread_mutex_unlock(&out->lock); |
1333 | return r; |
1334 | } |
1335 | |
1336 | static int out_resume(struct audio_stream_out *stream) |
1337 | { |
1338 | LOGFUNC("out_resume (%p)\n", stream); |
1339 | struct aml_stream_out *out = (struct aml_stream_out *) stream; |
1340 | struct aml_audio_device *adev = out->dev; |
1341 | int r = 0; |
1342 | pthread_mutex_lock(&adev->lock); |
1343 | pthread_mutex_lock(&out->lock); |
1344 | if (out->standby || out->pause_status == false) { |
1345 | goto exit; |
1346 | } |
1347 | if (pcm_is_ready(out->pcm)) { |
1348 | r = pcm_ioctl(out->pcm, SNDRV_PCM_IOCTL_PAUSE, 0); |
1349 | if (r < 0) { |
1350 | ALOGE("cannot resume channel\n"); |
1351 | } else { |
1352 | r = 0; |
1353 | } |
1354 | } |
1355 | if (out->hw_sync_mode) { |
1356 | ALOGI("init hal mixer when hwsync resume\n"); |
1357 | adev->hwsync_output = out; |
1358 | aml_hal_mixer_init(&adev->hal_mixer); |
1359 | sysfs_set_sysfs_str(TSYNC_EVENT, "AUDIO_RESUME"); |
1360 | } |
1361 | out->pause_status = false; |
1362 | exit: |
1363 | pthread_mutex_unlock(&adev->lock); |
1364 | pthread_mutex_unlock(&out->lock); |
1365 | return r; |
1366 | } |
1367 | |
1368 | |
1369 | static int audio_effect_process(struct audio_stream_out *stream, |
1370 | short* buffer, int frame_size) |
1371 | { |
1372 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
1373 | int output_size = frame_size << 2; |
1374 | |
1375 | if (out->has_SRS_lib) { |
1376 | output_size = srs_process(buffer, buffer, frame_size); |
1377 | } |
1378 | if (out->has_EQ_lib) { |
1379 | HPEQ_process(buffer, buffer, frame_size); |
1380 | } |
1381 | if (out->has_aml_IIR_lib) { |
1382 | short *ptr = buffer; |
1383 | short data; |
1384 | int i; |
1385 | for (i = 0; i < frame_size; i++) { |
1386 | data = (short)aml_IIR_process((int)(*ptr), 0); |
1387 | *ptr++ = data; |
1388 | data = (short)aml_IIR_process((int)(*ptr), 1); |
1389 | *ptr++ = data; |
1390 | } |
1391 | } |
1392 | return output_size; |
1393 | } |
1394 | |
1395 | static ssize_t out_write_legacy(struct audio_stream_out *stream, const void* buffer, |
1396 | size_t bytes) |
1397 | { |
1398 | int ret = 0; |
1399 | size_t oldBytes = bytes; |
1400 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
1401 | struct aml_audio_device *adev = out->dev; |
1402 | size_t frame_size = audio_stream_out_frame_size(stream); |
1403 | size_t in_frames = bytes / frame_size; |
1404 | size_t out_frames; |
1405 | bool force_input_standby = false; |
1406 | int16_t *in_buffer = (int16_t *)buffer; |
1407 | int16_t *out_buffer = in_buffer; |
1408 | struct aml_stream_in *in; |
1409 | uint ouput_len; |
1410 | char *data, *data_dst; |
1411 | volatile char *data_src; |
1412 | uint i, total_len; |
1413 | int codec_type = 0; |
1414 | int samesource_flag = 0; |
1415 | uint32_t latency_frames = 0; |
1416 | int need_mix = 0; |
1417 | short *mix_buf = NULL; |
1418 | unsigned char enable_dump = getprop_bool("media.audiohal.outdump"); |
1419 | // limit HAL mixer buffer level within 200ms |
1420 | while ((adev->hwsync_output != NULL && adev->hwsync_output != out) && |
1421 | (aml_hal_mixer_get_content(&adev->hal_mixer) > 200 * 48 * 4)) { |
1422 | usleep(20000); |
1423 | } |
1424 | /* acquiring hw device mutex systematically is useful if a low priority thread is waiting |
1425 | * on the output stream mutex - e.g. executing select_mode() while holding the hw device |
1426 | * mutex |
1427 | */ |
1428 | pthread_mutex_lock(&adev->lock); |
1429 | pthread_mutex_lock(&out->lock); |
1430 | //here to check whether hwsync out stream and other stream are enabled at the same time. |
1431 | //if that we need do the hal mixer of the two out stream. |
1432 | if (out->hw_sync_mode == 1) { |
1433 | int content_size = aml_hal_mixer_get_content(&adev->hal_mixer); |
1434 | //ALOGI("content_size %d\n",content_size); |
1435 | if (content_size > 0) { |
1436 | if (adev->hal_mixer.need_cache_flag == 0) { |
1437 | //ALOGI("need do hal mixer\n"); |
1438 | need_mix = 1; |
1439 | } else if (content_size < 80 * 48 * 4) { //80 ms |
1440 | //ALOGI("hal mixed cached size %d\n", content_size); |
1441 | } else { |
1442 | ALOGI("start enable mix,cached size %d\n", content_size); |
1443 | adev->hal_mixer.need_cache_flag = 0; |
1444 | } |
1445 | |
1446 | } else { |
1447 | // ALOGI("content size %d,duration %d ms\n",content_size,content_size/48/4); |
1448 | } |
1449 | } |
1450 | /* if hwsync output stream are enabled,write other output to a mixe buffer and sleep for the pcm duration time */ |
1451 | if (adev->hwsync_output != NULL && adev->hwsync_output != out) { |
1452 | //ALOGI("dev hwsync enable,hwsync %p) cur (%p),size %d\n",adev->hwsync_output,out,bytes); |
1453 | // out->frame_write_sum += in_frames; |
1454 | #if 0 |
1455 | if (!out->standby) { |
1456 | do_output_standby(out); |
1457 | } |
1458 | #endif |
1459 | if (out->standby) { |
1460 | ret = start_output_stream(out); |
1461 | if (ret != 0) { |
1462 | pthread_mutex_unlock(&adev->lock); |
1463 | ALOGE("start_output_stream failed"); |
1464 | goto exit; |
1465 | } |
1466 | out->standby = false; |
1467 | } |
1468 | ret = -1; |
1469 | aml_hal_mixer_write(&adev->hal_mixer, buffer, bytes); |
1470 | pthread_mutex_unlock(&adev->lock); |
1471 | goto exit; |
1472 | } |
1473 | if (out->pause_status == true) { |
1474 | pthread_mutex_unlock(&adev->lock); |
1475 | pthread_mutex_unlock(&out->lock); |
1476 | ALOGI("call out_write when pause status (%p)\n", stream); |
1477 | return 0; |
1478 | } |
1479 | if ((out->standby) && (out->hw_sync_mode == 1)) { |
1480 | // todo: check timestamp header PTS discontinue for new sync point after seek |
1481 | out->first_apts_flag = false; |
1482 | out->hw_sync_state = HW_SYNC_STATE_HEADER; |
1483 | out->hw_sync_header_cnt = 0; |
1484 | } |
1485 | |
1486 | #if 1 |
1487 | if (enable_dump && out->hw_sync_mode == 0) { |
1488 | FILE *fp1 = fopen("/data/tmp/i2s_audio_out.pcm", "a+"); |
1489 | if (fp1) { |
1490 | int flen = fwrite((char *)buffer, 1, bytes, fp1); |
1491 | fclose(fp1); |
1492 | } |
1493 | } |
1494 | #endif |
1495 | |
1496 | if (out->hw_sync_mode == 1) { |
1497 | char buf[64] = {0}; |
1498 | unsigned char *header; |
1499 | |
1500 | if (out->hw_sync_state == HW_SYNC_STATE_RESYNC) { |
1501 | uint i = 0; |
1502 | uint8_t *p = (uint8_t *)buffer; |
1503 | while (i < bytes) { |
1504 | if (hwsync_header_valid(p)) { |
1505 | ALOGI("HWSYNC resync.%p", out); |
1506 | out->hw_sync_state = HW_SYNC_STATE_HEADER; |
1507 | out->hw_sync_header_cnt = 0; |
1508 | out->first_apts_flag = false; |
1509 | bytes -= i; |
1510 | buffer += i; |
1511 | in_frames = bytes / frame_size; |
1512 | ALOGI("in_frames = %d", in_frames); |
1513 | in_buffer = (int16_t *)buffer; |
1514 | break; |
1515 | } else { |
1516 | i += 4; |
1517 | p += 4; |
1518 | } |
1519 | } |
1520 | |
1521 | if (out->hw_sync_state == HW_SYNC_STATE_RESYNC) { |
1522 | ALOGI("Keep searching for HWSYNC header.%p", out); |
1523 | pthread_mutex_unlock(&adev->lock); |
1524 | goto exit; |
1525 | } |
1526 | } |
1527 | |
1528 | header = (unsigned char *)buffer; |
1529 | } |
1530 | if (out->standby) { |
1531 | ret = start_output_stream(out); |
1532 | if (ret != 0) { |
1533 | pthread_mutex_unlock(&adev->lock); |
1534 | ALOGE("start_output_stream failed"); |
1535 | goto exit; |
1536 | } |
1537 | out->standby = false; |
1538 | /* a change in output device may change the microphone selection */ |
1539 | if (adev->active_input && |
1540 | adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) { |
1541 | force_input_standby = true; |
1542 | } |
1543 | } |
1544 | pthread_mutex_unlock(&adev->lock); |
1545 | #if 1 |
1546 | /* Reduce number of channels, if necessary */ |
1547 | if (popcount(out_get_channels(&stream->common)) > |
1548 | (int)out->config.channels) { |
1549 | unsigned int i; |
1550 | |
1551 | /* Discard right channel */ |
1552 | for (i = 1; i < in_frames; i++) { |
1553 | in_buffer[i] = in_buffer[i * 2]; |
1554 | } |
1555 | |
1556 | /* The frame size is now half */ |
1557 | frame_size /= 2; |
1558 | } |
1559 | #endif |
1560 | /* only use resampler if required */ |
1561 | if (out->config.rate != out_get_sample_rate(&stream->common)) { |
1562 | out_frames = out->buffer_frames; |
1563 | out->resampler->resample_from_input(out->resampler, |
1564 | in_buffer, &in_frames, |
1565 | (int16_t*)out->buffer, &out_frames); |
1566 | in_buffer = (int16_t*)out->buffer; |
1567 | out_buffer = in_buffer; |
1568 | } else { |
1569 | out_frames = in_frames; |
1570 | } |
1571 | if (out->echo_reference != NULL) { |
1572 | |
1573 | struct echo_reference_buffer b; |
1574 | b.raw = (void *)buffer; |
1575 | b.frame_count = in_frames; |
1576 | get_playback_delay(out, out_frames, &b); |
1577 | out->echo_reference->write(out->echo_reference, &b); |
1578 | } |
1579 | |
1580 | #if 0 |
1581 | if (enable_dump && out->hw_sync_mode == 1) { |
1582 | FILE *fp1 = fopen("/data/tmp/i2s_audio_out.pcm", "a+"); |
1583 | if (fp1) { |
1584 | int flen = fwrite((char *)in_buffer, 1, out_frames * frame_size, fp1); |
1585 | LOGFUNC("flen = %d---outlen=%d ", flen, out_frames * frame_size); |
1586 | fclose(fp1); |
1587 | } else { |
1588 | LOGFUNC("could not open file:/data/i2s_audio_out.pcm"); |
1589 | } |
1590 | } |
1591 | #endif |
1592 | #if 1 |
1593 | if (!(adev->out_device & AUDIO_DEVICE_OUT_ALL_SCO)) { |
1594 | codec_type = get_sysfs_int("/sys/class/audiodsp/digital_codec"); |
1595 | //samesource_flag = get_sysfs_int("/sys/class/audiodsp/audio_samesource"); |
1596 | if (codec_type != out->last_codec_type/*samesource_flag == 0*/ && codec_type == 0) { |
1597 | ALOGI("to enable same source,need reset alsa,type %d,same source flag %d \n", codec_type, samesource_flag); |
1598 | pcm_stop(out->pcm); |
1599 | } |
1600 | out->last_codec_type = codec_type; |
1601 | } |
1602 | #endif |
1603 | if (out->is_tv_platform == 1) { |
1604 | int16_t *tmp_buffer = (int16_t *)out->audioeffect_tmp_buffer; |
1605 | memcpy((void *)tmp_buffer, (void *)in_buffer, out_frames * 4); |
1606 | audio_effect_process(stream, tmp_buffer, out_frames); |
1607 | for (i = 0; i < out_frames; i ++) { |
1608 | out->tmp_buffer_8ch[8 * i] = ((int32_t)(in_buffer[2 * i])) << 16; |
1609 | out->tmp_buffer_8ch[8 * i + 1] = ((int32_t)(in_buffer[2 * i + 1])) << 16; |
1610 | out->tmp_buffer_8ch[8 * i + 2] = ((int32_t)(tmp_buffer[2 * i])) << 16; |
1611 | out->tmp_buffer_8ch[8 * i + 3] = ((int32_t)(tmp_buffer[2 * i + 1])) << 16; |
1612 | out->tmp_buffer_8ch[8 * i + 4] = 0; |
1613 | out->tmp_buffer_8ch[8 * i + 5] = 0; |
1614 | out->tmp_buffer_8ch[8 * i + 6] = 0; |
1615 | out->tmp_buffer_8ch[8 * i + 7] = 0; |
1616 | } |
1617 | /*if (out->frame_count < 5*1024) { |
1618 | memset(out->tmp_buffer_8ch, 0, out_frames * frame_size * 8); |
1619 | }*/ |
1620 | ret = pcm_write(out->pcm, out->tmp_buffer_8ch, out_frames * frame_size * 8); |
1621 | out->frame_write_sum += out_frames; |
1622 | } else { |
1623 | if (out->hw_sync_mode) { |
1624 | |
1625 | int remain = out_frames * frame_size; |
1626 | uint8_t *p = buffer; |
1627 | |
1628 | //ALOGI(" --- out_write %d, cache cnt = %d, body = %d, hw_sync_state = %d", out_frames * frame_size, out->body_align_cnt, out->hw_sync_body_cnt, out->hw_sync_state); |
1629 | |
1630 | while (remain > 0) { |
1631 | if (out->hw_sync_state == HW_SYNC_STATE_HEADER) { |
1632 | //ALOGI("Add to header buffer [%d], 0x%x", out->hw_sync_header_cnt, *p); |
1633 | out->hw_sync_header[out->hw_sync_header_cnt++] = *p++; |
1634 | remain--; |
1635 | if (out->hw_sync_header_cnt == 16) { |
1636 | int64_t pts; |
1637 | if (!hwsync_header_valid(&out->hw_sync_header[0])) { |
1638 | ALOGE("hwsync header out of sync! Resync."); |
1639 | out->hw_sync_state = HW_SYNC_STATE_RESYNC; |
1640 | break; |
1641 | } |
1642 | out->hw_sync_state = HW_SYNC_STATE_BODY; |
1643 | out->hw_sync_body_cnt = hwsync_header_get_size(&out->hw_sync_header[0]); |
1644 | out->body_align_cnt = 0; |
1645 | pts = hwsync_header_get_pts(&out->hw_sync_header[0]); |
1646 | pts = pts * 90 / 1000000; |
1647 | #if 1 |
1648 | char buf[64] = {0}; |
1649 | if (out->first_apts_flag == false) { |
1650 | uint32_t apts_cal; |
1651 | ALOGI("HW SYNC new first APTS %lld,body size %d", pts, out->hw_sync_body_cnt); |
1652 | out->first_apts_flag = true; |
1653 | out->first_apts = pts; |
1654 | out->frame_write_sum = 0; |
1655 | out->last_apts_from_header = pts; |
1656 | sprintf(buf, "AUDIO_START:0x%llx", pts & 0xffffffff); |
1657 | ALOGI("tsync -> %s", buf); |
1658 | if (sysfs_set_sysfs_str(TSYNC_EVENT, buf) == -1) { |
1659 | ALOGE("set AUDIO_START failed \n"); |
1660 | } |
1661 | } else { |
1662 | int64_t apts; |
1663 | uint32_t latency = out_get_latency(out) * 90; |
1664 | apts = (uint64_t)out->frame_write_sum * 90000 / DEFAULT_OUT_SAMPLING_RATE; |
1665 | apts += out->first_apts; |
1666 | // check PTS discontinue, which may happen when audio track switching |
1667 | // discontinue means PTS calculated based on first_apts and frame_write_sum |
1668 | // does not match the timestamp of next audio samples |
1669 | apts -= latency; |
1670 | if (apts < 0) { |
1671 | apts = 0; |
1672 | } |
1673 | // here we use acutal audio frame gap,not use the differece of caculated current apts with the current frame pts, |
1674 | //as there is a offset of audio latency from alsa. |
1675 | // handle audio gap 0.5~5 s |
1676 | unsigned two_frame_gap = (unsigned)llabs(out->last_apts_from_header - pts); |
1677 | if (two_frame_gap > APTS_DISCONTINUE_THRESHOLD_MIN && two_frame_gap < APTS_DISCONTINUE_THRESHOLD_MAX) { |
1678 | /* if (abs(pts -apts) > APTS_DISCONTINUE_THRESHOLD_MIN && abs(pts -apts) < APTS_DISCONTINUE_THRESHOLD_MAX) { */ |
1679 | ALOGI("HW sync PTS discontinue, 0x%llx->0x%llx(from header) diff %d,last apts %llx(from header)", |
1680 | apts, pts, two_frame_gap, out->last_apts_from_header); |
1681 | //here handle the audio gap and insert zero to the alsa |
1682 | uint insert_size = 0; |
1683 | uint insert_size_total = 0; |
1684 | uint once_write_size = 0; |
1685 | insert_size = two_frame_gap/*abs(pts -apts) */ / 90 * 48 * 4; |
1686 | insert_size = insert_size & (~63); |
1687 | insert_size_total = insert_size; |
1688 | ALOGI("audio gap %d ms ,need insert pcm size %d\n", two_frame_gap/*abs(pts -apts) */ / 90, insert_size); |
1689 | char *insert_buf = (char*)malloc(8192); |
1690 | if (insert_buf == NULL) { |
1691 | ALOGE("malloc size failed \n"); |
1692 | pthread_mutex_unlock(&adev->lock); |
1693 | goto exit; |
1694 | } |
1695 | memset(insert_buf, 0, 8192); |
1696 | if (need_mix) { |
1697 | mix_buf = malloc(once_write_size); |
1698 | if (mix_buf == NULL) { |
1699 | ALOGE("mix_buf malloc failed\n"); |
1700 | free(insert_buf); |
1701 | pthread_mutex_unlock(&adev->lock); |
1702 | goto exit; |
1703 | } |
1704 | } |
1705 | while (insert_size > 0) { |
1706 | once_write_size = insert_size > 8192 ? 8192 : insert_size; |
1707 | if (need_mix) { |
1708 | pthread_mutex_lock(&adev->lock); |
1709 | aml_hal_mixer_read(&adev->hal_mixer, mix_buf, once_write_size); |
1710 | pthread_mutex_unlock(&adev->lock); |
1711 | memcpy(insert_buf, mix_buf, once_write_size); |
1712 | } |
1713 | #if 1 |
1714 | if (enable_dump) { |
1715 | FILE *fp1 = fopen("/data/tmp/i2s_audio_out.pcm", "a+"); |
1716 | if (fp1) { |
1717 | int flen = fwrite((char *)insert_buf, 1, once_write_size, fp1); |
1718 | fclose(fp1); |
1719 | } |
1720 | } |
1721 | #endif |
1722 | pthread_mutex_lock(&adev->pcm_write_lock); |
1723 | ret = pcm_write(out->pcm, (void *) insert_buf, once_write_size); |
1724 | pthread_mutex_unlock(&adev->pcm_write_lock); |
1725 | if (ret != 0) { |
1726 | ALOGE("pcm write failed\n"); |
1727 | free(insert_buf); |
1728 | if (mix_buf) { |
1729 | free(mix_buf); |
1730 | } |
1731 | pthread_mutex_unlock(&adev->lock); |
1732 | goto exit; |
1733 | } |
1734 | insert_size -= once_write_size; |
1735 | } |
1736 | if (mix_buf) { |
1737 | free(mix_buf); |
1738 | } |
1739 | mix_buf = NULL; |
1740 | free(insert_buf); |
1741 | // insert end |
1742 | //adev->first_apts = pts; |
1743 | out->frame_write_sum += insert_size_total / frame_size; |
1744 | #if 0 |
1745 | sprintf(buf, "AUDIO_TSTAMP_DISCONTINUITY:0x%lx", pts); |
1746 | if (sysfs_set_sysfs_str(TSYNC_EVENT, buf) == -1) { |
1747 | ALOGE("unable to open file %s,err: %s", TSYNC_EVENT, strerror(errno)); |
1748 | } |
1749 | #endif |
1750 | } else { |
1751 | int pcr = 0; |
1752 | if (get_sysfs_int16(TSYNC_PCRSCR, &pcr) == 0) { |
1753 | int32_t apts_cal = apts & 0xffffffff; |
1754 | if (abs(pcr - apts_cal) < SYSTIME_CORRECTION_THRESHOLD) { |
1755 | // do nothing |
1756 | } else { |
1757 | sprintf(buf, "0x%x", apts_cal); |
1758 | ALOGI("tsync -> reset pcrscr 0x%x -> 0x%x, diff %d ms,frame pts %llx,latency pts %d", pcr, apts_cal, (int)(apts_cal - pcr) / 90, pts, latency); |
1759 | int ret_val = sysfs_set_sysfs_str(TSYNC_APTS, buf); |
1760 | if (ret_val == -1) { |
1761 | ALOGE("unable to open file %s,err: %s", TSYNC_APTS, strerror(errno)); |
1762 | } |
1763 | } |
1764 | } |
1765 | } |
1766 | out->last_apts_from_header = pts; |
1767 | } |
1768 | #endif |
1769 | |
1770 | //ALOGI("get header body_cnt = %d, pts = %lld", out->hw_sync_body_cnt, pts); |
1771 | } |
1772 | continue; |
1773 | } else if (out->hw_sync_state == HW_SYNC_STATE_BODY) { |
1774 | uint align; |
1775 | uint m = (out->hw_sync_body_cnt < remain) ? out->hw_sync_body_cnt : remain; |
1776 | |
1777 | //ALOGI("m = %d", m); |
1778 | |
1779 | // process m bytes, upto end of hw_sync_body_cnt or end of remaining our_write bytes. |
1780 | // within m bytes, there is no hw_sync header and all are body bytes. |
1781 | if (out->body_align_cnt) { |
1782 | // clear fragment first for alignment limitation on ALSA driver, which |
1783 | // requires each pcm_writing aligned at 16 frame boundaries |
1784 | // assuming data are always PCM16 based, so aligned at 64 bytes unit. |
1785 | if ((m + out->body_align_cnt) < 64) { |
1786 | // merge only |
1787 | memcpy(&out->body_align[out->body_align_cnt], p, m); |
1788 | p += m; |
1789 | remain -= m; |
1790 | out->body_align_cnt += m; |
1791 | out->hw_sync_body_cnt -= m; |
1792 | if (out->hw_sync_body_cnt == 0) { |
1793 | // end of body, research for HW SYNC header |
1794 | out->hw_sync_state = HW_SYNC_STATE_HEADER; |
1795 | out->hw_sync_header_cnt = 0; |
1796 | continue; |
1797 | } |
1798 | //ALOGI("align cache add %d, cnt = %d", remain, out->body_align_cnt); |
1799 | break; |
1800 | } else { |
1801 | // merge-submit-continue |
1802 | memcpy(&out->body_align[out->body_align_cnt], p, 64 - out->body_align_cnt); |
1803 | p += 64 - out->body_align_cnt; |
1804 | remain -= 64 - out->body_align_cnt; |
1805 | //ALOGI("pcm_write 64, out remain %d", remain); |
1806 | |
1807 | short *w_buf = (short*)&out->body_align[0]; |
1808 | |
1809 | if (need_mix) { |
1810 | short mix_buf[32]; |
1811 | pthread_mutex_lock(&adev->lock); |
1812 | aml_hal_mixer_read(&adev->hal_mixer, mix_buf, 64); |
1813 | pthread_mutex_unlock(&adev->lock); |
1814 | |
1815 | for (i = 0; i < 64 / 2 / 2; i++) { |
1816 | int r; |
1817 | r = w_buf[2 * i] * out->volume_l + mix_buf[2 * i]; |
1818 | w_buf[2 * i] = CLIP(r); |
1819 | r = w_buf[2 * i + 1] * out->volume_r + mix_buf[2 * i + 1]; |
1820 | w_buf[2 * i + 1] = CLIP(r); |
1821 | } |
1822 | } else { |
1823 | for (i = 0; i < 64 / 2 / 2; i++) { |
1824 | int r; |
1825 | r = w_buf[2 * i] * out->volume_l; |
1826 | w_buf[2 * i] = CLIP(r); |
1827 | r = w_buf[2 * i + 1] * out->volume_r; |
1828 | w_buf[2 * i + 1] = CLIP(r); |
1829 | } |
1830 | } |
1831 | #if 1 |
1832 | if (enable_dump) { |
1833 | FILE *fp1 = fopen("/data/tmp/i2s_audio_out.pcm", "a+"); |
1834 | if (fp1) { |
1835 | int flen = fwrite((char *)w_buf, 1, 64, fp1); |
1836 | fclose(fp1); |
1837 | } |
1838 | } |
1839 | #endif |
1840 | pthread_mutex_lock(&adev->pcm_write_lock); |
1841 | ret = pcm_write(out->pcm, w_buf, 64); |
1842 | pthread_mutex_unlock(&adev->pcm_write_lock); |
1843 | out->frame_write_sum += 64 / frame_size; |
1844 | out->hw_sync_body_cnt -= 64 - out->body_align_cnt; |
1845 | out->body_align_cnt = 0; |
1846 | if (out->hw_sync_body_cnt == 0) { |
1847 | out->hw_sync_state = HW_SYNC_STATE_HEADER; |
1848 | out->hw_sync_header_cnt = 0; |
1849 | } |
1850 | continue; |
1851 | } |
1852 | } |
1853 | |
1854 | // process m bytes body with an empty fragment for alignment |
1855 | align = m & 63; |
1856 | if ((m - align) > 0) { |
1857 | short *w_buf = (short*)p; |
1858 | mix_buf = (short *)malloc(m - align); |
1859 | if (mix_buf == NULL) { |
1860 | ALOGE("!!!fatal err,malloc %d bytes fail\n", m - align); |
1861 | ret = -1; |
1862 | goto exit; |
1863 | } |
1864 | if (need_mix) { |
1865 | pthread_mutex_lock(&adev->lock); |
1866 | aml_hal_mixer_read(&adev->hal_mixer, mix_buf, m - align); |
1867 | pthread_mutex_unlock(&adev->lock); |
1868 | for (i = 0; i < (m - align) / 2 / 2; i++) { |
1869 | int r; |
1870 | r = w_buf[2 * i] * out->volume_l + mix_buf[2 * i]; |
1871 | mix_buf[2 * i] = CLIP(r); |
1872 | r = w_buf[2 * i + 1] * out->volume_r + mix_buf[2 * i + 1]; |
1873 | mix_buf[2 * i + 1] = CLIP(r); |
1874 | } |
1875 | } else { |
1876 | for (i = 0; i < (m - align) / 2 / 2; i++) { |
1877 | |
1878 | int r; |
1879 | r = w_buf[2 * i] * out->volume_l; |
1880 | mix_buf[2 * i] = CLIP(r); |
1881 | r = w_buf[2 * i + 1] * out->volume_r; |
1882 | mix_buf[2 * i + 1] = CLIP(r); |
1883 | } |
1884 | } |
1885 | #if 1 |
1886 | if (enable_dump) { |
1887 | FILE *fp1 = fopen("/data/tmp/i2s_audio_out.pcm", "a+"); |
1888 | if (fp1) { |
1889 | int flen = fwrite((char *)mix_buf, 1, m - align, fp1); |
1890 | fclose(fp1); |
1891 | } |
1892 | } |
1893 | #endif |
1894 | pthread_mutex_lock(&adev->pcm_write_lock); |
1895 | ret = pcm_write(out->pcm, mix_buf, m - align); |
1896 | pthread_mutex_unlock(&adev->pcm_write_lock); |
1897 | free(mix_buf); |
1898 | out->frame_write_sum += (m - align) / frame_size; |
1899 | |
1900 | p += m - align; |
1901 | remain -= m - align; |
1902 | //ALOGI("pcm_write %d, remain %d", m - align, remain); |
1903 | |
1904 | out->hw_sync_body_cnt -= (m - align); |
1905 | if (out->hw_sync_body_cnt == 0) { |
1906 | out->hw_sync_state = HW_SYNC_STATE_HEADER; |
1907 | out->hw_sync_header_cnt = 0; |
1908 | continue; |
1909 | } |
1910 | } |
1911 | |
1912 | if (align) { |
1913 | memcpy(&out->body_align[0], p, align); |
1914 | p += align; |
1915 | remain -= align; |
1916 | out->body_align_cnt = align; |
1917 | //ALOGI("align cache add %d, cnt = %d, remain = %d", align, out->body_align_cnt, remain); |
1918 | |
1919 | out->hw_sync_body_cnt -= align; |
1920 | if (out->hw_sync_body_cnt == 0) { |
1921 | out->hw_sync_state = HW_SYNC_STATE_HEADER; |
1922 | out->hw_sync_header_cnt = 0; |
1923 | continue; |
1924 | } |
1925 | } |
1926 | } |
1927 | } |
1928 | |
1929 | } else { |
1930 | struct aml_hal_mixer *mixer = &adev->hal_mixer; |
1931 | pthread_mutex_lock(&adev->pcm_write_lock); |
1932 | if (aml_hal_mixer_get_content(mixer) > 0) { |
1933 | pthread_mutex_lock(&mixer->lock); |
1934 | if (mixer->wp > mixer->rp) { |
1935 | pcm_write(out->pcm, mixer->start_buf + mixer->rp, mixer->wp - mixer->rp); |
1936 | } else { |
1937 | pcm_write(out->pcm, mixer->start_buf + mixer->wp, mixer->buf_size - mixer->rp); |
1938 | pcm_write(out->pcm, mixer->start_buf, mixer->wp); |
1939 | } |
1940 | mixer->rp = mixer->wp = 0; |
1941 | pthread_mutex_unlock(&mixer->lock); |
1942 | } |
1943 | ret = pcm_write(out->pcm, out_buffer, out_frames * frame_size); |
1944 | pthread_mutex_unlock(&adev->pcm_write_lock); |
1945 | out->frame_write_sum += out_frames; |
1946 | } |
1947 | } |
1948 | |
1949 | exit: |
1950 | latency_frames = out_get_latency(out) * out->config.rate / 1000; |
1951 | if (out->frame_write_sum >= latency_frames) { |
1952 | out->last_frames_postion = out->frame_write_sum - latency_frames; |
1953 | } else { |
1954 | out->last_frames_postion = out->frame_write_sum; |
1955 | } |
1956 | pthread_mutex_unlock(&out->lock); |
1957 | if (ret != 0) { |
1958 | usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) / |
1959 | out_get_sample_rate(&stream->common) * 15 / 16); |
1960 | } |
1961 | |
1962 | if (force_input_standby) { |
1963 | pthread_mutex_lock(&adev->lock); |
1964 | if (adev->active_input) { |
1965 | in = adev->active_input; |
1966 | pthread_mutex_lock(&in->lock); |
1967 | do_input_standby(in); |
1968 | pthread_mutex_unlock(&in->lock); |
1969 | } |
1970 | pthread_mutex_unlock(&adev->lock); |
1971 | } |
1972 | return oldBytes; |
1973 | } |
1974 | |
1975 | static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, |
1976 | size_t bytes) |
1977 | { |
1978 | int ret = 0; |
1979 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
1980 | struct aml_audio_device *adev = out->dev; |
1981 | size_t frame_size = audio_stream_out_frame_size(stream); |
1982 | size_t in_frames = bytes / frame_size; |
1983 | size_t out_frames; |
1984 | bool force_input_standby = false; |
1985 | int16_t *in_buffer = (int16_t *)buffer; |
1986 | struct aml_stream_in *in; |
1987 | uint ouput_len; |
1988 | char *data, *data_dst; |
1989 | volatile char *data_src; |
1990 | uint i, total_len; |
1991 | int codec_type = 0; |
1992 | int samesource_flag = 0; |
1993 | uint32_t latency_frames = 0; |
1994 | int need_mix = 0; |
1995 | short *mix_buf = NULL; |
1996 | unsigned char enable_dump = getprop_bool("media.audiohal.outdump"); |
1997 | |
1998 | /* acquiring hw device mutex systematically is useful if a low priority thread is waiting |
1999 | * on the output stream mutex - e.g. executing select_mode() while holding the hw device |
2000 | * mutex |
2001 | */ |
2002 | pthread_mutex_lock(&adev->lock); |
2003 | pthread_mutex_lock(&out->lock); |
2004 | |
2005 | #if 1 |
2006 | if (enable_dump && out->hw_sync_mode == 0) { |
2007 | FILE *fp1 = fopen("/data/tmp/i2s_audio_out.pcm", "a+"); |
2008 | if (fp1) { |
2009 | int flen = fwrite((char *)buffer, 1, bytes, fp1); |
2010 | fclose(fp1); |
2011 | } |
2012 | } |
2013 | #endif |
2014 | |
2015 | if (out->standby) { |
2016 | ret = start_output_stream(out); |
2017 | if (ret != 0) { |
2018 | pthread_mutex_unlock(&adev->lock); |
2019 | ALOGE("start_output_stream failed"); |
2020 | goto exit; |
2021 | } |
2022 | out->standby = false; |
2023 | /* a change in output device may change the microphone selection */ |
2024 | if (adev->active_input && |
2025 | adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) { |
2026 | force_input_standby = true; |
2027 | } |
2028 | } |
2029 | pthread_mutex_unlock(&adev->lock); |
2030 | #if 1 |
2031 | /* Reduce number of channels, if necessary */ |
2032 | if (popcount(out_get_channels(&stream->common)) > |
2033 | (int)out->config.channels) { |
2034 | unsigned int i; |
2035 | |
2036 | /* Discard right channel */ |
2037 | for (i = 1; i < in_frames; i++) { |
2038 | in_buffer[i] = in_buffer[i * 2]; |
2039 | } |
2040 | |
2041 | /* The frame size is now half */ |
2042 | frame_size /= 2; |
2043 | } |
2044 | #endif |
2045 | /* only use resampler if required */ |
2046 | if (out->config.rate != out_get_sample_rate(&stream->common)) { |
2047 | out_frames = out->buffer_frames; |
2048 | out->resampler->resample_from_input(out->resampler, |
2049 | in_buffer, &in_frames, |
2050 | (int16_t*)out->buffer, &out_frames); |
2051 | in_buffer = (int16_t*)out->buffer; |
2052 | } else { |
2053 | out_frames = in_frames; |
2054 | } |
2055 | if (out->echo_reference != NULL) { |
2056 | |
2057 | struct echo_reference_buffer b; |
2058 | b.raw = (void *)buffer; |
2059 | b.frame_count = in_frames; |
2060 | get_playback_delay(out, out_frames, &b); |
2061 | out->echo_reference->write(out->echo_reference, &b); |
2062 | } |
2063 | |
2064 | #if 1 |
2065 | if (!(adev->out_device & AUDIO_DEVICE_OUT_ALL_SCO)) { |
2066 | codec_type = get_sysfs_int("/sys/class/audiodsp/digital_codec"); |
2067 | samesource_flag = get_sysfs_int("/sys/class/audiodsp/audio_samesource"); |
2068 | if (samesource_flag == 0 && codec_type == 0) { |
2069 | ALOGI("to enable same source,need reset alsa,type %d,same source flag %d \n", |
2070 | codec_type, samesource_flag); |
2071 | pcm_stop(out->pcm); |
2072 | } |
2073 | } |
2074 | #endif |
2075 | |
2076 | struct aml_hal_mixer *mixer = &adev->hal_mixer; |
2077 | pthread_mutex_lock(&adev->pcm_write_lock); |
2078 | if (aml_hal_mixer_get_content(mixer) > 0) { |
2079 | pthread_mutex_lock(&mixer->lock); |
2080 | if (mixer->wp > mixer->rp) { |
2081 | pcm_write(out->pcm, mixer->start_buf + mixer->rp, mixer->wp - mixer->rp); |
2082 | } else { |
2083 | pcm_write(out->pcm, mixer->start_buf + mixer->wp, mixer->buf_size - mixer->rp); |
2084 | pcm_write(out->pcm, mixer->start_buf, mixer->wp); |
2085 | } |
2086 | mixer->rp = mixer->wp = 0; |
2087 | pthread_mutex_unlock(&mixer->lock); |
2088 | } |
2089 | ret = pcm_write(out->pcm, in_buffer, out_frames * frame_size); |
2090 | pthread_mutex_unlock(&adev->pcm_write_lock); |
2091 | out->frame_write_sum += out_frames; |
2092 | |
2093 | exit: |
2094 | latency_frames = out_get_latency(out) * out->config.rate / 1000; |
2095 | if (out->frame_write_sum >= latency_frames) { |
2096 | out->last_frames_postion = out->frame_write_sum - latency_frames; |
2097 | } else { |
2098 | out->last_frames_postion = out->frame_write_sum; |
2099 | } |
2100 | pthread_mutex_unlock(&out->lock); |
2101 | if (ret != 0) { |
2102 | usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) / |
2103 | out_get_sample_rate(&stream->common) * 15 / 16); |
2104 | } |
2105 | |
2106 | if (force_input_standby) { |
2107 | pthread_mutex_lock(&adev->lock); |
2108 | if (adev->active_input) { |
2109 | in = adev->active_input; |
2110 | pthread_mutex_lock(&in->lock); |
2111 | do_input_standby(in); |
2112 | pthread_mutex_unlock(&in->lock); |
2113 | } |
2114 | pthread_mutex_unlock(&adev->lock); |
2115 | } |
2116 | return bytes; |
2117 | } |
2118 | |
2119 | static ssize_t out_write_direct(struct audio_stream_out *stream, const void* buffer, |
2120 | size_t bytes) |
2121 | { |
2122 | int ret = 0; |
2123 | struct aml_stream_out *out = (struct aml_stream_out *) stream; |
2124 | struct aml_audio_device *adev = out->dev; |
2125 | size_t frame_size = audio_stream_out_frame_size(stream); |
2126 | size_t in_frames = bytes / frame_size; |
2127 | bool force_input_standby = false; |
2128 | size_t out_frames = 0; |
2129 | void *buf; |
2130 | uint i, total_len; |
2131 | char prop[PROPERTY_VALUE_MAX]; |
2132 | int codec_type = out->codec_type; |
2133 | int samesource_flag = 0; |
2134 | uint32_t latency_frames; |
2135 | uint64_t total_frame = 0; |
2136 | audio_hwsync_t *p_hwsync = &adev->hwsync; |
2137 | /* acquiring hw device mutex systematically is useful if a low priority thread is waiting |
2138 | * on the output stream mutex - e.g. executing select_mode() while holding the hw device |
2139 | * mutex |
2140 | */ |
2141 | out->bytes_write_total += bytes; |
2142 | ALOGV("out %p,dev %p out_write total size %lld\n", out, adev, out->bytes_write_total); |
2143 | pthread_mutex_lock(&adev->lock); |
2144 | pthread_mutex_lock(&out->lock); |
2145 | if (out->pause_status == true) { |
2146 | pthread_mutex_unlock(&adev->lock); |
2147 | pthread_mutex_unlock(&out->lock); |
2148 | ALOGI("call out_write when pause status,size %d,(%p)\n", bytes, out); |
2149 | return 0; |
2150 | } |
2151 | if ((out->standby) && adev->hw_sync_mode) { |
2152 | /* |
2153 | there are two types of raw data come to hdmi audio hal |
2154 | 1) compressed audio data without IEC61937 wrapped |
2155 | 2) compressed audio data with IEC61937 wrapped (typically from amlogic amadec source) |
2156 | we use the AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO to distiguwish the two cases. |
2157 | */ |
2158 | if ((codec_type == TYPE_AC3 || codec_type == TYPE_EAC3) && (out->flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)) { |
2159 | spdifenc_init(out->pcm, out->hal_format); |
2160 | out->spdif_enc_init_frame_write_sum = out->frame_write_sum; |
2161 | } |
2162 | // todo: check timestamp header PTS discontinue for new sync point after seek |
2163 | aml_audio_hwsync_clear_status(out); |
2164 | out->spdif_enc_init_frame_write_sum = out->frame_write_sum; |
2165 | } |
2166 | if (out->standby) { |
2167 | ret = start_output_stream_direct(out); |
2168 | if (ret != 0) { |
2169 | pthread_mutex_unlock(&adev->lock); |
2170 | goto exit; |
2171 | } |
2172 | out->standby = 0; |
2173 | /* a change in output device may change the microphone selection */ |
2174 | if (adev->active_input && |
2175 | adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) { |
2176 | force_input_standby = true; |
2177 | } |
2178 | } |
2179 | void *write_buf = NULL; |
2180 | int hwsync_cost_bytes = 0; |
2181 | if (adev->hw_sync_mode == 1) { |
2182 | int64_t cur_pts = 0xffffffff; |
2183 | int outsize = 0; |
2184 | char tempbuf[128]; |
2185 | ALOGV("before aml_audio_hwsync_find_frame bytes %d\n", bytes); |
2186 | hwsync_cost_bytes = aml_audio_hwsync_find_frame(out, buffer, bytes, &cur_pts, &outsize); |
2187 | ALOGV("after aml_audio_hwsync_find_frame bytes remain %d,cost %d,outsize %d,pts %llx\n", |
2188 | bytes - hwsync_cost_bytes, hwsync_cost_bytes, outsize, cur_pts); |
2189 | //TODO,skip 3 frames after flush, to tmp fix seek pts discontinue issue.need dig more |
2190 | // to find out why seek ppint pts frame is remained after flush.WTF. |
2191 | if (out->skip_frame > 0) { |
2192 | out->skip_frame--; |
2193 | ALOGI("skip pts@%llx,cur frame size %d,cost size %d\n", cur_pts, outsize, hwsync_cost_bytes); |
2194 | pthread_mutex_unlock(&adev->lock); |
2195 | pthread_mutex_unlock(&out->lock); |
2196 | return hwsync_cost_bytes; |
2197 | } |
2198 | if (cur_pts != 0xffffffff && outsize > 0) { |
2199 | // if we got the frame body,which means we get a complete frame. |
2200 | //we take this frame pts as the first apts. |
2201 | //this can fix the seek discontinue,we got a fake frame,which maybe cached before the seek |
2202 | if (p_hwsync->first_apts_flag == false) { |
2203 | p_hwsync->first_apts_flag = true; |
2204 | p_hwsync->first_apts = cur_pts; |
2205 | sprintf(tempbuf, "AUDIO_START:0x%llx", cur_pts & 0xffffffff); |
2206 | ALOGI("tsync -> %s,frame size %d", tempbuf, outsize); |
2207 | if (sysfs_set_sysfs_str(TSYNC_EVENT, tempbuf) == -1) { |
2208 | ALOGE("set AUDIO_START failed \n"); |
2209 | } |
2210 | } else { |
2211 | long apts; |
2212 | unsigned long latency = out_get_latency(out) * 90; |
2213 | // check PTS discontinue, which may happen when audio track switching |
2214 | // discontinue means PTS calculated based on first_apts and frame_write_sum |
2215 | // does not match the timestamp of next audio samples |
2216 | if (cur_pts > latency) { |
2217 | apts = (unsigned long)cur_pts - latency; |
2218 | } else { |
2219 | apts = 0; |
2220 | } |
2221 | if (0) { //abs(cur_pts -apts) > APTS_DISCONTINUE_THRESHOLD) { |
2222 | ALOGI("HW sync PTS discontinue, 0x%lx->0x%llx(from header) diff %llx,last apts %llx(from header)", |
2223 | apts, cur_pts, llabs(cur_pts - apts), p_hwsync->last_apts_from_header); |
2224 | p_hwsync->first_apts = cur_pts; |
2225 | sprintf(tempbuf, "AUDIO_TSTAMP_DISCONTINUITY:0x%llx", cur_pts); |
2226 | if (sysfs_set_sysfs_str(TSYNC_EVENT, tempbuf) == -1) { |
2227 | ALOGE("unable to open file %s,err: %s", TSYNC_EVENT, strerror(errno)); |
2228 | } |
2229 | } else { |
2230 | long pcr = 0; |
2231 | if (get_sysfs_int16(TSYNC_PCRSCR, &pcr) == 0) { |
2232 | uint32_t apts_cal = apts & 0xffffffff; |
2233 | if (abs(pcr - apts) < SYSTIME_CORRECTION_THRESHOLD) { |
2234 | // do nothing |
2235 | } |
2236 | // limit the gap handle to 0.5~5 s. |
2237 | else if ((apts - pcr) > APTS_DISCONTINUE_THRESHOLD_MIN && (apts - pcr) < APTS_DISCONTINUE_THRESHOLD_MAX) { |
2238 | int insert_size = 0; |
2239 | int once_write_size = 0; |
2240 | if (out->codec_type == TYPE_EAC3) { |
2241 | insert_size = abs(apts - pcr) / 90 * 48 * 4 * 4; |
2242 | } else { |
2243 | insert_size = abs(apts - pcr) / 90 * 48 * 4; |
2244 | } |
2245 | insert_size = insert_size & (~63); |
2246 | ALOGI("audio gap %d ms ,need insert data %d\n", abs(apts - pcr) / 90, insert_size); |
2247 | char *insert_buf = (char*)malloc(8192); |
2248 | if (insert_buf == NULL) { |
2249 | ALOGE("malloc size failed \n"); |
2250 | pthread_mutex_unlock(&adev->lock); |
2251 | goto exit; |
2252 | } |
2253 | memset(insert_buf, 0, 8192); |
2254 | while (insert_size > 0) { |
2255 | once_write_size = insert_size > 8192 ? 8192 : insert_size; |
2256 | ret = pcm_write(out->pcm, (void *) insert_buf, once_write_size); |
2257 | if (ret != 0) { |
2258 | ALOGE("pcm write failed\n"); |
2259 | free(insert_buf); |
2260 | pthread_mutex_unlock(&adev->lock); |
2261 | goto exit; |
2262 | } |
2263 | insert_size -= once_write_size; |
2264 | } |
2265 | free(insert_buf); |
2266 | } |
2267 | //audio pts smaller than pcr,need skip frame. |
2268 | else if ((pcr - apts) > APTS_DISCONTINUE_THRESHOLD_MIN && (pcr - apts) < APTS_DISCONTINUE_THRESHOLD_MAX) { |
2269 | //we assume one frame duration is 32 ms for DD+(6 blocks X 1536 frames,48K sample rate) |
2270 | if (out->codec_type == TYPE_EAC3 && outsize > 0) { |
2271 | ALOGI("audio slow 0x%lx,skip frame @pts 0x%llx,pcr 0x%lx,cur apts 0x%lx\n", (pcr - apts), cur_pts, pcr, apts); |
2272 | out->frame_skip_sum += 1536; |
2273 | bytes = outsize; |
2274 | pthread_mutex_unlock(&adev->lock); |
2275 | goto exit; |
2276 | } |
2277 | } else { |
2278 | sprintf(tempbuf, "0x%lx", apts); |
2279 | ALOGI("tsync -> reset pcrscr 0x%lx -> 0x%lx, %s big,diff %d ms", pcr, apts, apts > pcr ? "apts" : "pcr", abs(apts - pcr) / 90); |
2280 | #if 0 |
2281 | int ret_val = sysfs_set_sysfs_str(TSYNC_APTS, tempbuf); |
2282 | if (ret_val == -1) { |
2283 | ALOGE("unable to open file %s,err: %s", TSYNC_APTS, strerror(errno)); |
2284 | } |
2285 | #endif |
2286 | } |
2287 | } |
2288 | } |
2289 | } |
2290 | } |
2291 | if (outsize > 0) { |
2292 | in_frames = outsize / frame_size; |
2293 | write_buf = p_hwsync->hw_sync_body_buf; |
2294 | } else { |
2295 | bytes = hwsync_cost_bytes; |
2296 | pthread_mutex_unlock(&adev->lock); |
2297 | goto exit; |
2298 | } |
2299 | } else { |
2300 | write_buf = (void *) buffer; |
2301 | } |
2302 | pthread_mutex_unlock(&adev->lock); |
2303 | out_frames = in_frames; |
2304 | buf = (void *) write_buf; |
2305 | if (getprop_bool("media.hdmihal.outdump")) { |
2306 | FILE *fp1 = fopen("/data/tmp/hdmi_audio_out.pcm", "a+"); |
2307 | if (fp1) { |
2308 | int flen = fwrite((char *)buffer, 1, out_frames * frame_size, fp1); |
2309 | LOGFUNC("flen = %d---outlen=%d ", flen, out_frames * frame_size); |
2310 | fclose(fp1); |
2311 | } else { |
2312 | LOGFUNC("could not open file:/data/hdmi_audio_out.pcm"); |
2313 | } |
2314 | } |
2315 | if (codec_type_is_raw_data(out->codec_type) && !(out->flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)) { |
2316 | //here to do IEC61937 pack |
2317 | ALOGV("IEC61937 write size %d,hw_sync_mode %d,flag %x\n", out_frames * frame_size, adev->hw_sync_mode, out->flags); |
2318 | if (out->codec_type > 0) { |
2319 | // compressed audio DD/DD+ |
2320 | bytes = spdifenc_write((void *) buf, out_frames * frame_size); |
2321 | //need return actual size of this burst write |
2322 | if (adev->hw_sync_mode == 1) { |
2323 | bytes = hwsync_cost_bytes; |
2324 | } |
2325 | ALOGV("spdifenc_write return %d\n", bytes); |
2326 | if (out->codec_type == TYPE_EAC3) { |
2327 | out->frame_write_sum = spdifenc_get_total() / 16 + out->spdif_enc_init_frame_write_sum; |
2328 | } else { |
2329 | out->frame_write_sum = spdifenc_get_total() / 4 + out->spdif_enc_init_frame_write_sum; |
2330 | } |
2331 | ALOGV("out %p,spdifenc_get_total() / 4 %lld\n", out, spdifenc_get_total() / 16); |
2332 | } |
2333 | goto exit; |
2334 | } |
2335 | if (!out->standby) { |
2336 | if (out->multich == 8) { |
2337 | int *p32 = NULL; |
2338 | short *p16 = (short *) buf; |
2339 | short *p16_temp; |
2340 | int i, NumSamps; |
2341 | NumSamps = out_frames * frame_size / sizeof(short); |
2342 | p32 = malloc(NumSamps * sizeof(int)); |
2343 | if (p32 != NULL) { |
2344 | //here to swap the channnl data here |
2345 | //actual now:L,missing,R,RS,RRS,,LS,LRS,missing |
2346 | //expect L,C,R,RS,RRS,LRS,LS,LFE (LFE comes from to center) |
2347 | //actual audio data layout L,R,C,none/LFE,LRS,RRS,LS,RS |
2348 | p16_temp = (short *) p32; |
2349 | for (i = 0; i < NumSamps; i = i + 8) { |
2350 | p16_temp[0 + i]/*L*/ = p16[0 + i]; |
2351 | p16_temp[1 + i]/*R*/ = p16[1 + i]; |
2352 | p16_temp[2 + i] /*LFE*/ = p16[3 + i]; |
2353 | p16_temp[3 + i] /*C*/ = p16[2 + i]; |
2354 | p16_temp[4 + i] /*LS*/ = p16[6 + i]; |
2355 | p16_temp[5 + i] /*RS*/ = p16[7 + i]; |
2356 | p16_temp[6 + i] /*LRS*/ = p16[4 + i]; |
2357 | p16_temp[7 + i]/*RRS*/ = p16[5 + i]; |
2358 | } |
2359 | memcpy(p16, p16_temp, NumSamps * sizeof(short)); |
2360 | for (i = 0; i < NumSamps; i++) { //suppose 16bit/8ch PCM |
2361 | p32[i] = p16[i] << 16; |
2362 | } |
2363 | ret = pcm_write(out->pcm, (void *) p32, NumSamps * 4); |
2364 | free(p32); |
2365 | } |
2366 | } else if (out->multich == 6) { |
2367 | int *p32 = NULL; |
2368 | short *p16 = (short *) buf; |
2369 | short *p16_temp; |
2370 | int i, j, NumSamps, real_samples; |
2371 | real_samples = out_frames * frame_size / sizeof(short); |
2372 | NumSamps = real_samples * 8 / 6; |
2373 | //ALOGI("6ch to 8 ch real %d, to %d,bytes %d,frame size %d\n",real_samples,NumSamps,bytes,frame_size); |
2374 | p32 = malloc(NumSamps * sizeof(int)); |
2375 | if (p32 != NULL) { |
2376 | p16_temp = (short *) p32; |
2377 | for (i = 0; i < real_samples; i = i + 6) { |
2378 | p16_temp[0 + i]/*L*/ = p16[0 + i]; |
2379 | p16_temp[1 + i]/*R*/ = p16[1 + i]; |
2380 | p16_temp[2 + i] /*LFE*/ = p16[3 + i]; |
2381 | p16_temp[3 + i] /*C*/ = p16[2 + i]; |
2382 | p16_temp[4 + i] /*LS*/ = p16[4 + i]; |
2383 | p16_temp[5 + i] /*RS*/ = p16[5 + i]; |
2384 | } |
2385 | memcpy(p16, p16_temp, real_samples * sizeof(short)); |
2386 | memset(p32, 0, NumSamps * sizeof(int)); |
2387 | for (i = 0, j = 0; j < NumSamps; i = i + 6, j = j + 8) { //suppose 16bit/8ch PCM |
2388 | p32[j] = p16[i] << 16; |
2389 | p32[j + 1] = p16[i + 1] << 16; |
2390 | p32[j + 2] = p16[i + 2] << 16; |
2391 | p32[j + 3] = p16[i + 3] << 16; |
2392 | p32[j + 4] = p16[i + 4] << 16; |
2393 | p32[j + 5] = p16[i + 5] << 16; |
2394 | } |
2395 | ret = pcm_write(out->pcm, (void *) p32, NumSamps * 4); |
2396 | free(p32); |
2397 | } |
2398 | } else { |
2399 | #if 0 |
2400 | codec_type = |
2401 | get_sysfs_int("/sys/class/audiodsp/digital_codec"); |
2402 | samesource_flag = |
2403 | get_sysfs_int("/sys/class/audiodsp/audio_samesource"); |
2404 | if (out->last_codec_type > 0 && codec_type != out->last_codec_type) { |
2405 | samesource_flag = 1; |
2406 | } |
2407 | if (samesource_flag == 1 && codec_type) { |
2408 | ALOGI |
2409 | ("to disable same source,need reset alsa,last %d,type %d,same source flag %d ,\n", |
2410 | out->last_codec_type, codec_type, samesource_flag); |
2411 | out->last_codec_type = codec_type; |
2412 | pcm_stop(out->pcm); |
2413 | } |
2414 | #endif |
2415 | ALOGV("write size %d\n", out_frames * frame_size); |
2416 | ret = pcm_write(out->pcm, (void *) buf, out_frames * frame_size); |
2417 | if (ret == 0) { |
2418 | out->frame_write_sum += out_frames; |
2419 | } |
2420 | } |
2421 | } |
2422 | exit: |
2423 | total_frame = out->frame_write_sum + out->frame_skip_sum; |
2424 | latency_frames = out_get_latency(out) * out->config.rate / 1000; |
2425 | if (total_frame >= latency_frames) { |
2426 | out->last_frames_postion = total_frame - latency_frames; |
2427 | } else { |
2428 | out->last_frames_postion = total_frame; |
2429 | } |
2430 | pthread_mutex_unlock(&out->lock); |
2431 | if (ret != 0) { |
2432 | usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) / |
2433 | out_get_sample_rate(&stream->common)); |
2434 | } |
2435 | return bytes; |
2436 | } |
2437 | |
2438 | static ssize_t out_write_tv(struct audio_stream_out *stream, const void* buffer, |
2439 | size_t bytes) |
2440 | { |
2441 | // TV temporarily use legacy out write. |
2442 | /* TODO: add TV platform specific write here */ |
2443 | return out_write_legacy(stream, buffer, bytes); |
2444 | } |
2445 | |
2446 | static int out_get_render_position(const struct audio_stream_out *stream, |
2447 | uint32_t *dsp_frames) |
2448 | { |
2449 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
2450 | uint64_t dsp_frame_int64 = 0; |
2451 | *dsp_frames = out->last_frames_postion; |
2452 | if (out->flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO) { |
2453 | dsp_frame_int64 = out->last_frames_postion / out->raw_61937_frame_size; |
2454 | *dsp_frames = (uint32_t)(dsp_frame_int64 & 0xffffffff); |
2455 | if (out->last_dsp_frame > *dsp_frames) { |
2456 | ALOGI("maybe uint32_t wraparound,print something,last %u,now %u", out->last_dsp_frame, *dsp_frames); |
2457 | ALOGI("wraparound,out_get_render_position return %u,playback time %llu ms,sr %d\n", *dsp_frames, |
2458 | out->last_frames_postion * 1000 / out->raw_61937_frame_size / out->config.rate, out->config.rate); |
2459 | |
2460 | } |
2461 | } |
2462 | ALOGV("out_get_render_position %d\n", *dsp_frames); |
2463 | return 0; |
2464 | } |
2465 | |
2466 | static int out_add_audio_effect(const struct audio_stream *stream __unused, effect_handle_t effect __unused) |
2467 | { |
2468 | return 0; |
2469 | } |
2470 | |
2471 | static int out_remove_audio_effect(const struct audio_stream *stream __unused, effect_handle_t effect __unused) |
2472 | { |
2473 | return 0; |
2474 | } |
2475 | static int out_get_next_write_timestamp(const struct audio_stream_out *stream __unused, |
2476 | int64_t *timestamp __unused) |
2477 | { |
2478 | return -EINVAL; |
2479 | } |
2480 | |
2481 | //actually maybe it be not useful now except pass CTS_TEST: |
2482 | // run cts -c android.media.cts.AudioTrackTest -m testGetTimestamp |
2483 | static int out_get_presentation_position(const struct audio_stream_out *stream, uint64_t *frames, struct timespec *timestamp) |
2484 | { |
2485 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
2486 | if (frames != NULL) { |
2487 | *frames = out->last_frames_postion; |
2488 | } |
2489 | |
2490 | if (timestamp != NULL) { |
2491 | clock_gettime(CLOCK_MONOTONIC, timestamp); |
2492 | } |
2493 | if (out->flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO) { |
2494 | *frames = out->last_frames_postion / out->raw_61937_frame_size; |
2495 | } |
2496 | ALOGV("out_get_presentation_position %lld\n", *frames); |
2497 | |
2498 | return 0; |
2499 | } |
2500 | static int get_next_buffer(struct resampler_buffer_provider *buffer_provider, |
2501 | struct resampler_buffer* buffer); |
2502 | static void release_buffer(struct resampler_buffer_provider *buffer_provider, |
2503 | struct resampler_buffer* buffer); |
2504 | |
2505 | |
2506 | /** audio_stream_in implementation **/ |
2507 | |
2508 | /* must be called with hw device and input stream mutexes locked */ |
2509 | static int start_input_stream(struct aml_stream_in *in) |
2510 | { |
2511 | int ret = 0; |
2512 | unsigned int card = CARD_AMLOGIC_BOARD; |
2513 | unsigned int port = PORT_I2S; |
2514 | |
2515 | struct aml_audio_device *adev = in->dev; |
2516 | LOGFUNC("%s(need_echo_reference=%d, channels=%d, rate=%d, requested_rate=%d, mode= %d)", |
2517 | __FUNCTION__, in->need_echo_reference, in->config.channels, in->config.rate, in->requested_rate, adev->mode); |
2518 | adev->active_input = in; |
2519 | |
2520 | if (adev->mode != AUDIO_MODE_IN_CALL) { |
2521 | adev->in_device &= ~AUDIO_DEVICE_IN_ALL; |
2522 | adev->in_device |= in->device; |
2523 | select_devices(adev); |
2524 | } |
2525 | card = get_aml_card(); |
2526 | |
2527 | ALOGV("%s(in->requested_rate=%d, in->config.rate=%d)", |
2528 | __FUNCTION__, in->requested_rate, in->config.rate); |
2529 | if (adev->in_device & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { |
2530 | port = PORT_PCM; |
2531 | } else if (getprop_bool("sys.hdmiIn.Capture")) { |
2532 | port = PORT_SPDIF; |
2533 | } else { |
2534 | port = PORT_I2S; |
2535 | } |
2536 | LOGFUNC("*%s, open card(%d) port(%d)-------", __FUNCTION__, card, port); |
2537 | in->config.period_size = CAPTURE_PERIOD_SIZE; |
2538 | if (in->need_echo_reference && in->echo_reference == NULL) { |
2539 | in->echo_reference = get_echo_reference(adev, |
2540 | AUDIO_FORMAT_PCM_16_BIT, |
2541 | in->config.channels, |
2542 | in->requested_rate); |
2543 | LOGFUNC("%s(after get_echo_ref.... now in->echo_reference = %p)", __FUNCTION__, in->echo_reference); |
2544 | } |
2545 | /* this assumes routing is done previously */ |
2546 | in->pcm = pcm_open(card, port, PCM_IN, &in->config); |
2547 | if (!pcm_is_ready(in->pcm)) { |
2548 | ALOGE("cannot open pcm_in driver: %s", pcm_get_error(in->pcm)); |
2549 | pcm_close(in->pcm); |
2550 | adev->active_input = NULL; |
2551 | return -ENOMEM; |
2552 | } |
2553 | ALOGD("pcm_open in: card(%d), port(%d)", card, port); |
2554 | |
2555 | /* if no supported sample rate is available, use the resampler */ |
2556 | if (in->resampler) { |
2557 | in->resampler->reset(in->resampler); |
2558 | in->frames_in = 0; |
2559 | } |
2560 | return 0; |
2561 | } |
2562 | |
2563 | static uint32_t in_get_sample_rate(const struct audio_stream *stream) |
2564 | { |
2565 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
2566 | |
2567 | return in->requested_rate; |
2568 | } |
2569 | |
2570 | static int in_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused) |
2571 | { |
2572 | return 0; |
2573 | } |
2574 | |
2575 | static size_t in_get_buffer_size(const struct audio_stream *stream) |
2576 | { |
2577 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
2578 | |
2579 | return get_input_buffer_size(in->config.period_size, in->config.rate, |
2580 | AUDIO_FORMAT_PCM_16_BIT, |
2581 | in->config.channels); |
2582 | } |
2583 | |
2584 | static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) |
2585 | { |
2586 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
2587 | |
2588 | if (in->config.channels == 1) { |
2589 | return AUDIO_CHANNEL_IN_MONO; |
2590 | } else { |
2591 | return AUDIO_CHANNEL_IN_STEREO; |
2592 | } |
2593 | } |
2594 | |
2595 | static audio_format_t in_get_format(const struct audio_stream *stream __unused) |
2596 | { |
2597 | return AUDIO_FORMAT_PCM_16_BIT; |
2598 | } |
2599 | |
2600 | static int in_set_format(struct audio_stream *stream __unused, audio_format_t format __unused) |
2601 | { |
2602 | return 0; |
2603 | } |
2604 | |
2605 | /* must be called with hw device and input stream mutexes locked */ |
2606 | static int do_input_standby(struct aml_stream_in *in) |
2607 | { |
2608 | struct aml_audio_device *adev = in->dev; |
2609 | |
2610 | LOGFUNC("%s(%p)", __FUNCTION__, in); |
2611 | if (!in->standby) { |
2612 | pcm_close(in->pcm); |
2613 | in->pcm = NULL; |
2614 | |
2615 | adev->active_input = 0; |
2616 | if (adev->mode != AUDIO_MODE_IN_CALL) { |
2617 | adev->in_device &= ~AUDIO_DEVICE_IN_ALL; |
2618 | //select_input_device(adev); |
2619 | } |
2620 | |
2621 | if (in->echo_reference != NULL) { |
2622 | /* stop reading from echo reference */ |
2623 | in->echo_reference->read(in->echo_reference, NULL); |
2624 | put_echo_reference(adev, in->echo_reference); |
2625 | in->echo_reference = NULL; |
2626 | } |
2627 | |
2628 | in->standby = 1; |
2629 | #if 0 |
2630 | LOGFUNC("%s : output_standby=%d,input_standby=%d", |
2631 | __FUNCTION__, output_standby, input_standby); |
2632 | if (output_standby && input_standby) { |
2633 | reset_mixer_state(adev->ar); |
2634 | update_mixer_state(adev->ar); |
2635 | } |
2636 | #endif |
2637 | } |
2638 | return 0; |
2639 | } |
2640 | static int in_standby(struct audio_stream *stream) |
2641 | { |
2642 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
2643 | int status; |
2644 | LOGFUNC("%s(%p)", __FUNCTION__, stream); |
2645 | |
2646 | pthread_mutex_lock(&in->dev->lock); |
2647 | pthread_mutex_lock(&in->lock); |
2648 | status = do_input_standby(in); |
2649 | pthread_mutex_unlock(&in->lock); |
2650 | pthread_mutex_unlock(&in->dev->lock); |
2651 | return status; |
2652 | } |
2653 | |
2654 | static int in_dump(const struct audio_stream *stream __unused, int fd __unused) |
2655 | { |
2656 | return 0; |
2657 | } |
2658 | |
2659 | static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) |
2660 | { |
2661 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
2662 | struct aml_audio_device *adev = in->dev; |
2663 | struct str_parms *parms; |
2664 | char *str; |
2665 | char value[32]; |
2666 | int ret, val = 0; |
2667 | bool do_standby = false; |
2668 | |
2669 | LOGFUNC("%s(%p, %s)", __FUNCTION__, stream, kvpairs); |
2670 | parms = str_parms_create_str(kvpairs); |
2671 | |
2672 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value)); |
2673 | |
2674 | pthread_mutex_lock(&adev->lock); |
2675 | pthread_mutex_lock(&in->lock); |
2676 | if (ret >= 0) { |
2677 | val = atoi(value); |
2678 | /* no audio source uses val == 0 */ |
2679 | if ((in->source != val) && (val != 0)) { |
2680 | in->source = val; |
2681 | do_standby = true; |
2682 | } |
2683 | } |
2684 | |
2685 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); |
2686 | if (ret >= 0) { |
2687 | val = atoi(value) & ~AUDIO_DEVICE_BIT_IN; |
2688 | if ((in->device != val) && (val != 0)) { |
2689 | in->device = val; |
2690 | do_standby = true; |
2691 | } |
2692 | } |
2693 | |
2694 | if (do_standby) { |
2695 | do_input_standby(in); |
2696 | } |
2697 | pthread_mutex_unlock(&in->lock); |
2698 | pthread_mutex_unlock(&adev->lock); |
2699 | |
2700 | int framesize = 0; |
2701 | ret = str_parms_get_int(parms, AUDIO_PARAMETER_STREAM_FRAME_COUNT, &framesize); |
2702 | |
2703 | if (ret >= 0) { |
2704 | if (framesize > 0) { |
2705 | ALOGI("Reset audio input hw frame size from %d to %d\n", |
2706 | in->config.period_size * in->config.period_count, framesize); |
2707 | in->config.period_size = framesize / in->config.period_count; |
2708 | pthread_mutex_lock(&adev->lock); |
2709 | pthread_mutex_lock(&in->lock); |
2710 | |
2711 | if (!in->standby && (in == adev->active_input)) { |
2712 | do_input_standby(in); |
2713 | start_input_stream(in); |
2714 | in->standby = 0; |
2715 | } |
2716 | |
2717 | pthread_mutex_unlock(&in->lock); |
2718 | pthread_mutex_unlock(&adev->lock); |
2719 | } |
2720 | } |
2721 | |
2722 | str_parms_destroy(parms); |
2723 | return ret; |
2724 | } |
2725 | |
2726 | static char * in_get_parameters(const struct audio_stream *stream __unused, |
2727 | const char *keys __unused) |
2728 | { |
2729 | return strdup(""); |
2730 | } |
2731 | |
2732 | static int in_set_gain(struct audio_stream_in *stream __unused, float gain __unused) |
2733 | { |
2734 | return 0; |
2735 | } |
2736 | |
2737 | static void get_capture_delay(struct aml_stream_in *in, |
2738 | size_t frames __unused, |
2739 | struct echo_reference_buffer *buffer) |
2740 | { |
2741 | /* read frames available in kernel driver buffer */ |
2742 | size_t kernel_frames; |
2743 | struct timespec tstamp; |
2744 | long buf_delay; |
2745 | long rsmp_delay; |
2746 | long kernel_delay; |
2747 | long delay_ns; |
2748 | int rsmp_mul = in->config.rate / VX_NB_SAMPLING_RATE; |
2749 | if (pcm_get_htimestamp(in->pcm, &kernel_frames, &tstamp) < 0) { |
2750 | buffer->time_stamp.tv_sec = 0; |
2751 | buffer->time_stamp.tv_nsec = 0; |
2752 | buffer->delay_ns = 0; |
2753 | ALOGW("read get_capture_delay(): pcm_htimestamp error"); |
2754 | return; |
2755 | } |
2756 | |
2757 | /* read frames available in audio HAL input buffer |
2758 | * add number of frames being read as we want the capture time of first sample |
2759 | * in current buffer */ |
2760 | buf_delay = (long)(((int64_t)(in->frames_in + in->proc_frames_in * rsmp_mul) * 1000000000) |
2761 | / in->config.rate); |
2762 | /* add delay introduced by resampler */ |
2763 | rsmp_delay = 0; |
2764 | if (in->resampler) { |
2765 | rsmp_delay = in->resampler->delay_ns(in->resampler); |
2766 | } |
2767 | |
2768 | kernel_delay = (long)(((int64_t)kernel_frames * 1000000000) / in->config.rate); |
2769 | |
2770 | delay_ns = kernel_delay + buf_delay + rsmp_delay; |
2771 | |
2772 | buffer->time_stamp = tstamp; |
2773 | buffer->delay_ns = delay_ns; |
2774 | /*ALOGV("get_capture_delay time_stamp = [%ld].[%ld], delay_ns: [%d]," |
2775 | " kernel_delay:[%ld], buf_delay:[%ld], rsmp_delay:[%ld], kernel_frames:[%d], " |
2776 | "in->frames_in:[%d], in->proc_frames_in:[%d], frames:[%d]", |
2777 | buffer->time_stamp.tv_sec , buffer->time_stamp.tv_nsec, buffer->delay_ns, |
2778 | kernel_delay, buf_delay, rsmp_delay, kernel_frames, |
2779 | in->frames_in, in->proc_frames_in, frames);*/ |
2780 | |
2781 | } |
2782 | |
2783 | static int32_t update_echo_reference(struct aml_stream_in *in, size_t frames) |
2784 | { |
2785 | struct echo_reference_buffer b; |
2786 | b.delay_ns = 0; |
2787 | |
2788 | ALOGV("update_echo_reference, frames = [%d], in->ref_frames_in = [%d], " |
2789 | "b.frame_count = [%d]", frames, in->ref_frames_in, frames - in->ref_frames_in); |
2790 | if (in->ref_frames_in < frames) { |
2791 | if (in->ref_buf_size < frames) { |
2792 | in->ref_buf_size = frames; |
2793 | in->ref_buf = (int16_t *)realloc(in->ref_buf, |
2794 | in->ref_buf_size * in->config.channels * sizeof(int16_t)); |
2795 | } |
2796 | |
2797 | b.frame_count = frames - in->ref_frames_in; |
2798 | b.raw = (void *)(in->ref_buf + in->ref_frames_in * in->config.channels); |
2799 | |
2800 | get_capture_delay(in, frames, &b); |
2801 | LOGFUNC("update_echo_reference return ::b.delay_ns=%d", b.delay_ns); |
2802 | |
2803 | if (in->echo_reference->read(in->echo_reference, &b) == 0) { |
2804 | in->ref_frames_in += b.frame_count; |
2805 | ALOGV("update_echo_reference: in->ref_frames_in:[%d], " |
2806 | "in->ref_buf_size:[%d], frames:[%d], b.frame_count:[%d]", |
2807 | in->ref_frames_in, in->ref_buf_size, frames, b.frame_count); |
2808 | } |
2809 | } else { |
2810 | ALOGW("update_echo_reference: NOT enough frames to read ref buffer"); |
2811 | } |
2812 | return b.delay_ns; |
2813 | } |
2814 | |
2815 | static int set_preprocessor_param(effect_handle_t handle, |
2816 | effect_param_t *param) |
2817 | { |
2818 | uint32_t size = sizeof(int); |
2819 | uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + |
2820 | param->vsize; |
2821 | |
2822 | int status = (*handle)->command(handle, |
2823 | EFFECT_CMD_SET_PARAM, |
2824 | sizeof(effect_param_t) + psize, |
2825 | param, |
2826 | &size, |
2827 | ¶m->status); |
2828 | if (status == 0) { |
2829 | status = param->status; |
2830 | } |
2831 | |
2832 | return status; |
2833 | } |
2834 | |
2835 | static int set_preprocessor_echo_delay(effect_handle_t handle, |
2836 | int32_t delay_us) |
2837 | { |
2838 | uint32_t buf[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; |
2839 | effect_param_t *param = (effect_param_t *)buf; |
2840 | |
2841 | param->psize = sizeof(uint32_t); |
2842 | param->vsize = sizeof(uint32_t); |
2843 | *(uint32_t *)param->data = AEC_PARAM_ECHO_DELAY; |
2844 | *((int32_t *)param->data + 1) = delay_us; |
2845 | |
2846 | return set_preprocessor_param(handle, param); |
2847 | } |
2848 | |
2849 | static void push_echo_reference(struct aml_stream_in *in, size_t frames) |
2850 | { |
2851 | /* read frames from echo reference buffer and update echo delay |
2852 | * in->ref_frames_in is updated with frames available in in->ref_buf */ |
2853 | int32_t delay_us = update_echo_reference(in, frames) / 1000; |
2854 | int i; |
2855 | audio_buffer_t buf; |
2856 | |
2857 | if (in->ref_frames_in < frames) { |
2858 | frames = in->ref_frames_in; |
2859 | } |
2860 | |
2861 | buf.frameCount = frames; |
2862 | buf.raw = in->ref_buf; |
2863 | |
2864 | for (i = 0; i < in->num_preprocessors; i++) { |
2865 | if ((*in->preprocessors[i])->process_reverse == NULL) { |
2866 | continue; |
2867 | } |
2868 | |
2869 | (*in->preprocessors[i])->process_reverse(in->preprocessors[i], |
2870 | &buf, |
2871 | NULL); |
2872 | set_preprocessor_echo_delay(in->preprocessors[i], delay_us); |
2873 | } |
2874 | |
2875 | in->ref_frames_in -= buf.frameCount; |
2876 | if (in->ref_frames_in) { |
2877 | memcpy(in->ref_buf, |
2878 | in->ref_buf + buf.frameCount * in->config.channels, |
2879 | in->ref_frames_in * in->config.channels * sizeof(int16_t)); |
2880 | } |
2881 | } |
2882 | |
2883 | static int get_next_buffer(struct resampler_buffer_provider *buffer_provider, |
2884 | struct resampler_buffer* buffer) |
2885 | { |
2886 | struct aml_stream_in *in; |
2887 | |
2888 | if (buffer_provider == NULL || buffer == NULL) { |
2889 | return -EINVAL; |
2890 | } |
2891 | |
2892 | in = (struct aml_stream_in *)((char *)buffer_provider - |
2893 | offsetof(struct aml_stream_in, buf_provider)); |
2894 | |
2895 | if (in->pcm == NULL) { |
2896 | buffer->raw = NULL; |
2897 | buffer->frame_count = 0; |
2898 | in->read_status = -ENODEV; |
2899 | return -ENODEV; |
2900 | } |
2901 | |
2902 | if (in->frames_in == 0) { |
2903 | in->read_status = pcm_read(in->pcm, (void*)in->buffer, |
2904 | in->config.period_size * audio_stream_in_frame_size(&in->stream)); |
2905 | if (in->read_status != 0) { |
2906 | ALOGE("get_next_buffer() pcm_read error %d", in->read_status); |
2907 | buffer->raw = NULL; |
2908 | buffer->frame_count = 0; |
2909 | return in->read_status; |
2910 | } |
2911 | in->frames_in = in->config.period_size; |
2912 | } |
2913 | |
2914 | buffer->frame_count = (buffer->frame_count > in->frames_in) ? |
2915 | in->frames_in : buffer->frame_count; |
2916 | buffer->i16 = in->buffer + (in->config.period_size - in->frames_in) * |
2917 | in->config.channels; |
2918 | |
2919 | return in->read_status; |
2920 | |
2921 | } |
2922 | |
2923 | static void release_buffer(struct resampler_buffer_provider *buffer_provider, |
2924 | struct resampler_buffer* buffer) |
2925 | { |
2926 | struct aml_stream_in *in; |
2927 | |
2928 | if (buffer_provider == NULL || buffer == NULL) { |
2929 | return; |
2930 | } |
2931 | |
2932 | in = (struct aml_stream_in *)((char *)buffer_provider - |
2933 | offsetof(struct aml_stream_in, buf_provider)); |
2934 | |
2935 | in->frames_in -= buffer->frame_count; |
2936 | } |
2937 | |
2938 | /* read_frames() reads frames from kernel driver, down samples to capture rate |
2939 | * if necessary and output the number of frames requested to the buffer specified */ |
2940 | static ssize_t read_frames(struct aml_stream_in *in, void *buffer, ssize_t frames) |
2941 | { |
2942 | ssize_t frames_wr = 0; |
2943 | |
2944 | while (frames_wr < frames) { |
2945 | size_t frames_rd = frames - frames_wr; |
2946 | if (in->resampler != NULL) { |
2947 | in->resampler->resample_from_provider(in->resampler, |
2948 | (int16_t *)((char *)buffer + |
2949 | frames_wr * audio_stream_in_frame_size(&in->stream)), |
2950 | &frames_rd); |
2951 | } else { |
2952 | struct resampler_buffer buf = { |
2953 | { .raw = NULL, }, |
2954 | .frame_count = frames_rd, |
2955 | }; |
2956 | get_next_buffer(&in->buf_provider, &buf); |
2957 | if (buf.raw != NULL) { |
2958 | memcpy((char *)buffer + |
2959 | frames_wr * audio_stream_in_frame_size(&in->stream), |
2960 | buf.raw, |
2961 | buf.frame_count * audio_stream_in_frame_size(&in->stream)); |
2962 | frames_rd = buf.frame_count; |
2963 | } |
2964 | release_buffer(&in->buf_provider, &buf); |
2965 | } |
2966 | /* in->read_status is updated by getNextBuffer() also called by |
2967 | * in->resampler->resample_from_provider() */ |
2968 | if (in->read_status != 0) { |
2969 | return in->read_status; |
2970 | } |
2971 | |
2972 | frames_wr += frames_rd; |
2973 | } |
2974 | return frames_wr; |
2975 | } |
2976 | |
2977 | /* process_frames() reads frames from kernel driver (via read_frames()), |
2978 | * calls the active audio pre processings and output the number of frames requested |
2979 | * to the buffer specified */ |
2980 | static ssize_t process_frames(struct aml_stream_in *in, void* buffer, ssize_t frames) |
2981 | { |
2982 | ssize_t frames_wr = 0; |
2983 | audio_buffer_t in_buf; |
2984 | audio_buffer_t out_buf; |
2985 | int i; |
2986 | |
2987 | //LOGFUNC("%s(%d, %p, %ld)", __FUNCTION__, in->num_preprocessors, buffer, frames); |
2988 | while (frames_wr < frames) { |
2989 | /* first reload enough frames at the end of process input buffer */ |
2990 | if (in->proc_frames_in < (size_t)frames) { |
2991 | ssize_t frames_rd; |
2992 | |
2993 | if (in->proc_buf_size < (size_t)frames) { |
2994 | in->proc_buf_size = (size_t)frames; |
2995 | in->proc_buf = (int16_t *)realloc(in->proc_buf, |
2996 | in->proc_buf_size * |
2997 | in->config.channels * sizeof(int16_t)); |
2998 | ALOGV("process_frames(): in->proc_buf %p size extended to %d frames", |
2999 | in->proc_buf, in->proc_buf_size); |
3000 | } |
3001 | frames_rd = read_frames(in, |
3002 | in->proc_buf + |
3003 | in->proc_frames_in * in->config.channels, |
3004 | frames - in->proc_frames_in); |
3005 | if (frames_rd < 0) { |
3006 | frames_wr = frames_rd; |
3007 | break; |
3008 | } |
3009 | in->proc_frames_in += frames_rd; |
3010 | } |
3011 | |
3012 | if (in->echo_reference != NULL) { |
3013 | push_echo_reference(in, in->proc_frames_in); |
3014 | } |
3015 | |
3016 | /* in_buf.frameCount and out_buf.frameCount indicate respectively |
3017 | * the maximum number of frames to be consumed and produced by process() */ |
3018 | in_buf.frameCount = in->proc_frames_in; |
3019 | in_buf.s16 = in->proc_buf; |
3020 | out_buf.frameCount = frames - frames_wr; |
3021 | out_buf.s16 = (int16_t *)buffer + frames_wr * in->config.channels; |
3022 | |
3023 | for (i = 0; i < in->num_preprocessors; i++) |
3024 | (*in->preprocessors[i])->process(in->preprocessors[i], |
3025 | &in_buf, |
3026 | &out_buf); |
3027 | |
3028 | /* process() has updated the number of frames consumed and produced in |
3029 | * in_buf.frameCount and out_buf.frameCount respectively |
3030 | * move remaining frames to the beginning of in->proc_buf */ |
3031 | in->proc_frames_in -= in_buf.frameCount; |
3032 | if (in->proc_frames_in) { |
3033 | memcpy(in->proc_buf, |
3034 | in->proc_buf + in_buf.frameCount * in->config.channels, |
3035 | in->proc_frames_in * in->config.channels * sizeof(int16_t)); |
3036 | } |
3037 | |
3038 | /* if not enough frames were passed to process(), read more and retry. */ |
3039 | if (out_buf.frameCount == 0) { |
3040 | continue; |
3041 | } |
3042 | |
3043 | frames_wr += out_buf.frameCount; |
3044 | } |
3045 | return frames_wr; |
3046 | } |
3047 | |
3048 | static ssize_t in_read(struct audio_stream_in *stream, void* buffer, |
3049 | size_t bytes) |
3050 | { |
3051 | int ret = 0; |
3052 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
3053 | struct aml_audio_device *adev = in->dev; |
3054 | size_t frames_rq = bytes / audio_stream_in_frame_size(&in->stream); |
3055 | |
3056 | /* acquiring hw device mutex systematically is useful if a low priority thread is waiting |
3057 | * on the input stream mutex - e.g. executing select_mode() while holding the hw device |
3058 | * mutex |
3059 | */ |
3060 | pthread_mutex_lock(&adev->lock); |
3061 | pthread_mutex_lock(&in->lock); |
3062 | if (in->standby) { |
3063 | ret = start_input_stream(in); |
3064 | if (ret == 0) { |
3065 | in->standby = 0; |
3066 | } |
3067 | } |
3068 | pthread_mutex_unlock(&adev->lock); |
3069 | |
3070 | if (ret < 0) { |
3071 | goto exit; |
3072 | } |
3073 | |
3074 | if (in->num_preprocessors != 0) { |
3075 | ret = process_frames(in, buffer, frames_rq); |
3076 | } else if (in->resampler != NULL) { |
3077 | ret = read_frames(in, buffer, frames_rq); |
3078 | } else { |
3079 | ret = pcm_read(in->pcm, buffer, bytes); |
3080 | } |
3081 | |
3082 | if (ret > 0) { |
3083 | ret = 0; |
3084 | } |
3085 | |
3086 | if (ret == 0 && adev->mic_mute) { |
3087 | memset(buffer, 0, bytes); |
3088 | } |
3089 | |
3090 | #if 0 |
3091 | FILE *dump_fp = NULL; |
3092 | |
3093 | dump_fp = fopen("/data/audio_in.pcm", "a+"); |
3094 | if (dump_fp != NULL) { |
3095 | fwrite(buffer, bytes, 1, dump_fp); |
3096 | fclose(dump_fp); |
3097 | } else { |
3098 | ALOGW("[Error] Can't write to /data/dump_in.pcm"); |
3099 | } |
3100 | #endif |
3101 | |
3102 | exit: |
3103 | if (ret < 0) |
3104 | usleep(bytes * 1000000 / audio_stream_in_frame_size(stream) / |
3105 | in_get_sample_rate(&stream->common)); |
3106 | |
3107 | pthread_mutex_unlock(&in->lock); |
3108 | return bytes; |
3109 | |
3110 | } |
3111 | |
3112 | static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused) |
3113 | { |
3114 | return 0; |
3115 | } |
3116 | |
3117 | static int in_add_audio_effect(const struct audio_stream *stream, |
3118 | effect_handle_t effect) |
3119 | { |
3120 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
3121 | int status; |
3122 | effect_descriptor_t desc; |
3123 | |
3124 | pthread_mutex_lock(&in->dev->lock); |
3125 | pthread_mutex_lock(&in->lock); |
3126 | if (in->num_preprocessors >= MAX_PREPROCESSORS) { |
3127 | status = -ENOSYS; |
3128 | goto exit; |
3129 | } |
3130 | |
3131 | status = (*effect)->get_descriptor(effect, &desc); |
3132 | if (status != 0) { |
3133 | goto exit; |
3134 | } |
3135 | |
3136 | in->preprocessors[in->num_preprocessors++] = effect; |
3137 | |
3138 | if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) { |
3139 | in->need_echo_reference = true; |
3140 | do_input_standby(in); |
3141 | } |
3142 | |
3143 | exit: |
3144 | |
3145 | pthread_mutex_unlock(&in->lock); |
3146 | pthread_mutex_unlock(&in->dev->lock); |
3147 | return status; |
3148 | } |
3149 | |
3150 | static int in_remove_audio_effect(const struct audio_stream *stream, |
3151 | effect_handle_t effect) |
3152 | { |
3153 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
3154 | int i; |
3155 | int status = -EINVAL; |
3156 | bool found = false; |
3157 | effect_descriptor_t desc; |
3158 | |
3159 | pthread_mutex_lock(&in->dev->lock); |
3160 | pthread_mutex_lock(&in->lock); |
3161 | if (in->num_preprocessors <= 0) { |
3162 | status = -ENOSYS; |
3163 | goto exit; |
3164 | } |
3165 | |
3166 | for (i = 0; i < in->num_preprocessors; i++) { |
3167 | if (found) { |
3168 | in->preprocessors[i - 1] = in->preprocessors[i]; |
3169 | continue; |
3170 | } |
3171 | if (in->preprocessors[i] == effect) { |
3172 | in->preprocessors[i] = NULL; |
3173 | status = 0; |
3174 | found = true; |
3175 | } |
3176 | } |
3177 | |
3178 | if (status != 0) { |
3179 | goto exit; |
3180 | } |
3181 | |
3182 | in->num_preprocessors--; |
3183 | |
3184 | status = (*effect)->get_descriptor(effect, &desc); |
3185 | if (status != 0) { |
3186 | goto exit; |
3187 | } |
3188 | if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) { |
3189 | in->need_echo_reference = false; |
3190 | do_input_standby(in); |
3191 | } |
3192 | |
3193 | exit: |
3194 | |
3195 | pthread_mutex_unlock(&in->lock); |
3196 | pthread_mutex_unlock(&in->dev->lock); |
3197 | return status; |
3198 | } |
3199 | |
3200 | static int adev_open_output_stream(struct audio_hw_device *dev, |
3201 | audio_io_handle_t handle __unused, |
3202 | audio_devices_t devices, |
3203 | audio_output_flags_t flags, |
3204 | struct audio_config *config, |
3205 | struct audio_stream_out **stream_out, |
3206 | const char *address __unused) |
3207 | { |
3208 | struct aml_audio_device *ladev = (struct aml_audio_device *)dev; |
3209 | struct aml_stream_out *out; |
3210 | int channel_count = popcount(config->channel_mask); |
3211 | int digital_codec; |
3212 | bool direct = false; |
3213 | int ret; |
3214 | bool hwsync_lpcm = false; |
3215 | ALOGI("**enter %s(devices=0x%04x,format=%#x, ch=0x%04x, SR=%d, flags=0x%x)", __FUNCTION__, devices, |
3216 | config->format, config->channel_mask, config->sample_rate, flags); |
3217 | |
3218 | out = (struct aml_stream_out *)calloc(1, sizeof(struct aml_stream_out)); |
3219 | if (!out) { |
3220 | return -ENOMEM; |
3221 | } |
3222 | |
3223 | out->flags = flags; |
3224 | if (getprop_bool("ro.platform.has.tvuimode")) { |
3225 | out->is_tv_platform = 1; |
3226 | } |
3227 | out->config = pcm_config_out; |
3228 | //hwsync with LPCM still goes to out_write_legacy |
3229 | hwsync_lpcm = (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC && audio_is_linear_pcm(config->format)); |
3230 | ALOGI("hwsync_lpcm %d\n", hwsync_lpcm); |
3231 | if (flags & AUDIO_OUTPUT_FLAG_PRIMARY || hwsync_lpcm) { |
3232 | out->stream.common.get_channels = out_get_channels; |
3233 | out->stream.common.get_format = out_get_format; |
3234 | out->stream.write = out_write_legacy; |
3235 | out->stream.common.standby = out_standby; |
3236 | |
3237 | out->hal_rate = out->config.rate; |
3238 | out->hal_format = config->format; |
3239 | config->format = out_get_format(&out->stream.common); |
3240 | config->channel_mask = out_get_channels(&out->stream.common); |
3241 | config->sample_rate = out_get_sample_rate(&out->stream.common); |
3242 | } else if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { |
3243 | out->stream.common.get_channels = out_get_channels_direct; |
3244 | out->stream.common.get_format = out_get_format_direct; |
3245 | out->stream.write = out_write_direct; |
3246 | out->stream.common.standby = out_standby_direct; |
3247 | //out->config = pcm_config_out_direct; |
3248 | out->hal_channel_mask = config->channel_mask; |
3249 | if (config->sample_rate == 0) { |
3250 | config->sample_rate = 48000; |
3251 | } |
3252 | out->config.rate = out->hal_rate = config->sample_rate; |
3253 | out->hal_format = config->format; |
3254 | out->raw_61937_frame_size = 1; |
3255 | digital_codec = get_codec_type(config->format); |
3256 | if (digital_codec == TYPE_EAC3) { |
3257 | out->raw_61937_frame_size = 4; |
3258 | out->config.period_size = pcm_config_out.period_size * 2; |
3259 | } else if (digital_codec == TYPE_TRUE_HD || digital_codec == TYPE_DTS_HD) { |
3260 | out->config.period_size = pcm_config_out.period_size * 4 * 2; |
3261 | out->raw_61937_frame_size = 16; |
3262 | } |
3263 | else if (digital_codec == TYPE_AC3 || digital_codec == TYPE_DTS) |
3264 | out->raw_61937_frame_size = 4; |
3265 | |
3266 | if (channel_count > 2) { |
3267 | ALOGI("[adev_open_output_stream]: out/%p channel/%d\n", out, |
3268 | channel_count); |
3269 | out->multich = channel_count; |
3270 | out->config.channels = channel_count; |
3271 | } |
3272 | if (codec_type_is_raw_data(digital_codec)) { |
3273 | ALOGI("for raw audio output,force alsa stereo output\n"); |
3274 | out->config.channels = 2; |
3275 | out->multich = 2; |
3276 | } |
3277 | } else if (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { |
3278 | // TODO: add hwsync write here |
3279 | //out->stream.write = out_write_hwsync; |
3280 | } |
3281 | |
3282 | out->stream.common.get_sample_rate = out_get_sample_rate; |
3283 | out->stream.common.set_sample_rate = out_set_sample_rate; |
3284 | out->stream.common.get_buffer_size = out_get_buffer_size; |
3285 | out->stream.common.set_format = out_set_format; |
3286 | //out->stream.common.standby = out_standby; |
3287 | out->stream.common.dump = out_dump; |
3288 | out->stream.common.set_parameters = out_set_parameters; |
3289 | out->stream.common.get_parameters = out_get_parameters; |
3290 | out->stream.common.add_audio_effect = out_add_audio_effect; |
3291 | out->stream.common.remove_audio_effect = out_remove_audio_effect; |
3292 | out->stream.get_latency = out_get_latency; |
3293 | out->stream.set_volume = out_set_volume; |
3294 | out->stream.get_render_position = out_get_render_position; |
3295 | out->stream.get_next_write_timestamp = out_get_next_write_timestamp; |
3296 | out->stream.get_presentation_position = out_get_presentation_position; |
3297 | out->stream.pause = out_pause; |
3298 | out->stream.resume = out_resume; |
3299 | out->stream.flush = out_flush; |
3300 | out->volume_l = 1.0; |
3301 | out->volume_r = 1.0; |
3302 | out->dev = ladev; |
3303 | out->standby = true; |
3304 | out->frame_write_sum = 0; |
3305 | out->hw_sync_mode = false; |
3306 | out->first_apts_flag = false; |
3307 | //out->hal_rate = out->config.rate; |
3308 | if (0/*flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC*/) { |
3309 | out->hw_sync_mode = true; |
3310 | ALOGI("Output stream open with AUDIO_OUTPUT_FLAG_HW_AV_SYNC"); |
3311 | } |
3312 | /* FIXME: when we support multiple output devices, we will want to |
3313 | * do the following: |
3314 | * adev->devices &= ~AUDIO_DEVICE_OUT_ALL; |
3315 | * adev->devices |= out->device; |
3316 | * select_output_device(adev); |
3317 | * This is because out_set_parameters() with a route is not |
3318 | * guaranteed to be called after an output stream is opened. |
3319 | */ |
3320 | |
3321 | LOGFUNC("**leave %s(devices=0x%04x,format=%#x, ch=0x%04x, SR=%d)", __FUNCTION__, devices, |
3322 | config->format, config->channel_mask, config->sample_rate); |
3323 | |
3324 | *stream_out = &out->stream; |
3325 | |
3326 | if (out->is_tv_platform && !(flags & AUDIO_OUTPUT_FLAG_DIRECT)) { |
3327 | out->config.channels = 8; |
3328 | out->config.format = PCM_FORMAT_S32_LE; |
3329 | out->tmp_buffer_8ch = malloc(out->config.period_size * 4 * 8); |
3330 | if (out->tmp_buffer_8ch == NULL) { |
3331 | ALOGE("cannot malloc memory for out->tmp_buffer_8ch"); |
3332 | return -ENOMEM; |
3333 | } |
3334 | out->audioeffect_tmp_buffer = malloc(out->config.period_size * 6); |
3335 | if (out->audioeffect_tmp_buffer == NULL) { |
3336 | ALOGE("cannot malloc memory for audioeffect_tmp_buffer"); |
3337 | return -ENOMEM; |
3338 | } |
3339 | //EQ lib load and init EQ |
3340 | ret = load_EQ_lib(); |
3341 | if (ret < 0) { |
3342 | ALOGE("%s, Load EQ lib fail!\n", __FUNCTION__); |
3343 | out->has_EQ_lib = 0; |
3344 | } else { |
3345 | ret = HPEQ_init(); |
3346 | if (ret < 0) { |
3347 | out->has_EQ_lib = 0; |
3348 | } else { |
3349 | out->has_EQ_lib = 1; |
3350 | } |
3351 | HPEQ_enable(1); |
3352 | } |
3353 | //load srs lib and init it. |
3354 | ret = load_SRS_lib(); |
3355 | if (ret < 0) { |
3356 | ALOGE("%s, Load SRS lib fail!\n", __FUNCTION__); |
3357 | out->has_SRS_lib = 0; |
3358 | } else { |
3359 | ret = srs_init(48000); |
3360 | if (ret < 0) { |
3361 | out->has_SRS_lib = 0; |
3362 | } else { |
3363 | out->has_SRS_lib = 1; |
3364 | } |
3365 | } |
3366 | //load aml_IIR lib |
3367 | ret = load_aml_IIR_lib(); |
3368 | if (ret < 0) { |
3369 | ALOGE("%s, Load aml_IIR lib fail!\n", __FUNCTION__); |
3370 | out->has_aml_IIR_lib = 0; |
3371 | } else { |
3372 | char value[PROPERTY_VALUE_MAX]; |
3373 | int paramter = 0; |
3374 | if (property_get("media.audio.LFP.paramter", value, NULL) > 0) { |
3375 | paramter = atoi(value); |
3376 | } |
3377 | aml_IIR_init(paramter); |
3378 | out->has_aml_IIR_lib = 1; |
3379 | } |
3380 | } |
3381 | return 0; |
3382 | |
3383 | err_open: |
3384 | free(out); |
3385 | *stream_out = NULL; |
3386 | return ret; |
3387 | } |
3388 | |
3389 | static void adev_close_output_stream(struct audio_hw_device *dev, |
3390 | struct audio_stream_out *stream) |
3391 | { |
3392 | struct aml_stream_out *out = (struct aml_stream_out *)stream; |
3393 | struct aml_audio_device *adev = (struct aml_audio_device *)dev; |
3394 | bool hwsync_lpcm = false; |
3395 | LOGFUNC("%s(%p, %p)", __FUNCTION__, dev, stream); |
3396 | if (out->is_tv_platform == 1) { |
3397 | free(out->tmp_buffer_8ch); |
3398 | free(out->audioeffect_tmp_buffer); |
3399 | } |
3400 | |
3401 | hwsync_lpcm = (out->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC && audio_is_linear_pcm(out->hal_format)); |
3402 | if (out->flags & AUDIO_OUTPUT_FLAG_PRIMARY || hwsync_lpcm) { |
3403 | out_standby(&stream->common); |
3404 | } else if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT) { |
3405 | out_standby_direct(&stream->common); |
3406 | } |
3407 | if (adev->hwsync_output == out) { |
3408 | ALOGI("clear hwsync output when close stream\n"); |
3409 | adev->hwsync_output = NULL; |
3410 | } |
3411 | free(stream); |
3412 | } |
3413 | |
3414 | static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) |
3415 | { |
3416 | LOGFUNC("%s(%p, %s)", __FUNCTION__, dev, kvpairs); |
3417 | |
3418 | struct aml_audio_device *adev = (struct aml_audio_device *)dev; |
3419 | struct str_parms *parms; |
3420 | char *str; |
3421 | char value[32]; |
3422 | int ret; |
3423 | parms = str_parms_create_str(kvpairs); |
3424 | ret = str_parms_get_str(parms, "screen_state", value, sizeof(value)); |
3425 | if (ret >= 0) { |
3426 | if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) { |
3427 | adev->low_power = false; |
3428 | } else { |
3429 | adev->low_power = true; |
3430 | } |
3431 | } |
3432 | str_parms_destroy(parms); |
3433 | return ret; |
3434 | } |
3435 | |
3436 | static char * adev_get_parameters(const struct audio_hw_device *dev __unused, |
3437 | const char *keys __unused) |
3438 | { |
3439 | struct aml_audio_device *adev = (struct aml_audio_device *)dev; |
3440 | if (!strcmp(keys, AUDIO_PARAMETER_HW_AV_SYNC)) { |
3441 | ALOGI("get hwsync id\n"); |
3442 | return strdup("hw_av_sync=12345678"); |
3443 | } |
3444 | return strdup(""); |
3445 | } |
3446 | |
3447 | static int adev_init_check(const struct audio_hw_device *dev __unused) |
3448 | { |
3449 | return 0; |
3450 | } |
3451 | |
3452 | static int adev_set_voice_volume(struct audio_hw_device *dev __unused, float volume __unused) |
3453 | { |
3454 | return 0; |
3455 | } |
3456 | |
3457 | static int adev_set_master_volume(struct audio_hw_device *dev __unused, float volume __unused) |
3458 | { |
3459 | return -ENOSYS; |
3460 | } |
3461 | |
3462 | static int adev_get_master_volume(struct audio_hw_device *dev __unused, |
3463 | float *volume __unused) |
3464 | { |
3465 | return -ENOSYS; |
3466 | } |
3467 | |
3468 | static int adev_set_master_mute(struct audio_hw_device *dev __unused, bool muted __unused) |
3469 | { |
3470 | return -ENOSYS; |
3471 | } |
3472 | |
3473 | static int adev_get_master_mute(struct audio_hw_device *dev __unused, bool *muted __unused) |
3474 | { |
3475 | return -ENOSYS; |
3476 | } |
3477 | static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) |
3478 | { |
3479 | struct aml_audio_device *adev = (struct aml_audio_device *)dev; |
3480 | LOGFUNC("%s(%p, %d)", __FUNCTION__, dev, mode); |
3481 | |
3482 | pthread_mutex_lock(&adev->lock); |
3483 | if (adev->mode != mode) { |
3484 | adev->mode = mode; |
3485 | select_mode(adev); |
3486 | } |
3487 | pthread_mutex_unlock(&adev->lock); |
3488 | |
3489 | return 0; |
3490 | } |
3491 | |
3492 | static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) |
3493 | { |
3494 | struct aml_audio_device *adev = (struct aml_audio_device *)dev; |
3495 | |
3496 | adev->mic_mute = state; |
3497 | |
3498 | return 0; |
3499 | } |
3500 | |
3501 | static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) |
3502 | { |
3503 | struct aml_audio_device *adev = (struct aml_audio_device *)dev; |
3504 | |
3505 | *state = adev->mic_mute; |
3506 | |
3507 | return 0; |
3508 | |
3509 | } |
3510 | |
3511 | static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, |
3512 | const struct audio_config *config) |
3513 | { |
3514 | size_t size; |
3515 | int channel_count = popcount(config->channel_mask); |
3516 | |
3517 | LOGFUNC("%s(%p, %d, %d, %d)", __FUNCTION__, dev, config->sample_rate, |
3518 | config->format, channel_count); |
3519 | if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) { |
3520 | return 0; |
3521 | } |
3522 | |
3523 | return get_input_buffer_size(config->frame_count, config->sample_rate, |
3524 | config->format, channel_count); |
3525 | |
3526 | } |
3527 | |
3528 | static int adev_open_input_stream(struct audio_hw_device *dev, |
3529 | audio_io_handle_t handle __unused, |
3530 | audio_devices_t devices, |
3531 | struct audio_config *config, |
3532 | struct audio_stream_in **stream_in, |
3533 | audio_input_flags_t flags __unused, |
3534 | const char *address __unused, |
3535 | audio_source_t source __unused) |
3536 | { |
3537 | struct aml_audio_device *ladev = (struct aml_audio_device *)dev; |
3538 | struct aml_stream_in *in; |
3539 | int ret; |
3540 | int channel_count = popcount(config->channel_mask); |
3541 | LOGFUNC("%s(%#x, %d, 0x%04x, %d)", __FUNCTION__, |
3542 | devices, config->format, config->channel_mask, config->sample_rate); |
3543 | if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) { |
3544 | return -EINVAL; |
3545 | } |
3546 | |
3547 | in = (struct aml_stream_in *)calloc(1, sizeof(struct aml_stream_in)); |
3548 | if (!in) { |
3549 | return -ENOMEM; |
3550 | } |
3551 | |
3552 | in->stream.common.get_sample_rate = in_get_sample_rate; |
3553 | in->stream.common.set_sample_rate = in_set_sample_rate; |
3554 | in->stream.common.get_buffer_size = in_get_buffer_size; |
3555 | in->stream.common.get_channels = in_get_channels; |
3556 | in->stream.common.get_format = in_get_format; |
3557 | in->stream.common.set_format = in_set_format; |
3558 | in->stream.common.standby = in_standby; |
3559 | in->stream.common.dump = in_dump; |
3560 | in->stream.common.set_parameters = in_set_parameters; |
3561 | in->stream.common.get_parameters = in_get_parameters; |
3562 | in->stream.common.add_audio_effect = in_add_audio_effect; |
3563 | in->stream.common.remove_audio_effect = in_remove_audio_effect; |
3564 | in->stream.set_gain = in_set_gain; |
3565 | in->stream.read = in_read; |
3566 | in->stream.get_input_frames_lost = in_get_input_frames_lost; |
3567 | |
3568 | in->requested_rate = config->sample_rate; |
3569 | |
3570 | in->device = devices & ~AUDIO_DEVICE_BIT_IN; |
3571 | if (in->device & AUDIO_DEVICE_IN_ALL_SCO) { |
3572 | memcpy(&in->config, &pcm_config_bt, sizeof(pcm_config_bt)); |
3573 | } else { |
3574 | memcpy(&in->config, &pcm_config_in, sizeof(pcm_config_in)); |
3575 | } |
3576 | |
3577 | if (in->config.channels == 1) { |
3578 | config->channel_mask = AUDIO_CHANNEL_IN_MONO; |
3579 | } else if (in->config.channels == 2) { |
3580 | config->channel_mask = AUDIO_CHANNEL_IN_STEREO; |
3581 | } else { |
3582 | ALOGE("Bad value of channel count : %d", in->config.channels); |
3583 | } |
3584 | in->buffer = malloc(in->config.period_size * |
3585 | audio_stream_in_frame_size(&in->stream)); |
3586 | if (!in->buffer) { |
3587 | ret = -ENOMEM; |
3588 | goto err_open; |
3589 | } |
3590 | |
3591 | if (in->requested_rate != in->config.rate) { |
3592 | LOGFUNC("%s(in->requested_rate=%d, in->config.rate=%d)", |
3593 | __FUNCTION__, in->requested_rate, in->config.rate); |
3594 | in->buf_provider.get_next_buffer = get_next_buffer; |
3595 | in->buf_provider.release_buffer = release_buffer; |
3596 | ret = create_resampler(in->config.rate, |
3597 | in->requested_rate, |
3598 | in->config.channels, |
3599 | RESAMPLER_QUALITY_DEFAULT, |
3600 | &in->buf_provider, |
3601 | &in->resampler); |
3602 | |
3603 | if (ret != 0) { |
3604 | ret = -EINVAL; |
3605 | goto err_open; |
3606 | } |
3607 | } |
3608 | |
3609 | in->dev = ladev; |
3610 | in->standby = 1; |
3611 | *stream_in = &in->stream; |
3612 | return 0; |
3613 | |
3614 | err_open: |
3615 | if (in->resampler) { |
3616 | release_resampler(in->resampler); |
3617 | } |
3618 | |
3619 | free(in); |
3620 | *stream_in = NULL; |
3621 | return ret; |
3622 | } |
3623 | |
3624 | static void adev_close_input_stream(struct audio_hw_device *dev, |
3625 | struct audio_stream_in *stream) |
3626 | { |
3627 | struct aml_stream_in *in = (struct aml_stream_in *)stream; |
3628 | |
3629 | LOGFUNC("%s(%p, %p)", __FUNCTION__, dev, stream); |
3630 | in_standby(&stream->common); |
3631 | |
3632 | if (in->resampler) { |
3633 | free(in->buffer); |
3634 | release_resampler(in->resampler); |
3635 | } |
3636 | if (in->proc_buf) { |
3637 | free(in->proc_buf); |
3638 | } |
3639 | if (in->ref_buf) { |
3640 | free(in->ref_buf); |
3641 | } |
3642 | |
3643 | free(stream); |
3644 | |
3645 | return; |
3646 | } |
3647 | |
3648 | static int adev_dump(const audio_hw_device_t *device __unused, int fd __unused) |
3649 | { |
3650 | return 0; |
3651 | } |
3652 | |
3653 | static int adev_close(hw_device_t *device) |
3654 | { |
3655 | struct aml_audio_device *adev = (struct aml_audio_device *)device; |
3656 | |
3657 | audio_route_free(adev->ar); |
3658 | free(device); |
3659 | return 0; |
3660 | } |
3661 | |
3662 | static int adev_open(const hw_module_t* module, const char* name, |
3663 | hw_device_t** device) |
3664 | { |
3665 | struct aml_audio_device *adev; |
3666 | int card = CARD_AMLOGIC_BOARD; |
3667 | int ret; |
3668 | if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) { |
3669 | return -EINVAL; |
3670 | } |
3671 | |
3672 | adev = calloc(1, sizeof(struct aml_audio_device)); |
3673 | if (!adev) { |
3674 | return -ENOMEM; |
3675 | } |
3676 | |
3677 | adev->hw_device.common.tag = HARDWARE_DEVICE_TAG; |
3678 | adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0; |
3679 | adev->hw_device.common.module = (struct hw_module_t *) module; |
3680 | adev->hw_device.common.close = adev_close; |
3681 | |
3682 | adev->hw_device.init_check = adev_init_check; |
3683 | adev->hw_device.set_voice_volume = adev_set_voice_volume; |
3684 | adev->hw_device.set_master_volume = adev_set_master_volume; |
3685 | adev->hw_device.get_master_volume = adev_get_master_volume; |
3686 | adev->hw_device.set_master_mute = adev_set_master_mute; |
3687 | adev->hw_device.get_master_mute = adev_get_master_mute; |
3688 | adev->hw_device.set_mode = adev_set_mode; |
3689 | adev->hw_device.set_mic_mute = adev_set_mic_mute; |
3690 | adev->hw_device.get_mic_mute = adev_get_mic_mute; |
3691 | adev->hw_device.set_parameters = adev_set_parameters; |
3692 | adev->hw_device.get_parameters = adev_get_parameters; |
3693 | adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size; |
3694 | adev->hw_device.open_output_stream = adev_open_output_stream; |
3695 | adev->hw_device.close_output_stream = adev_close_output_stream; |
3696 | adev->hw_device.open_input_stream = adev_open_input_stream; |
3697 | adev->hw_device.close_input_stream = adev_close_input_stream; |
3698 | adev->hw_device.dump = adev_dump; |
3699 | card = get_aml_card(); |
3700 | if ((card < 0) || (card > 7)) { |
3701 | ALOGE("error to get audio card"); |
3702 | return -EINVAL; |
3703 | } |
3704 | |
3705 | adev->card = card; |
3706 | adev->ar = audio_route_init(adev->card, MIXER_XML_PATH); |
3707 | |
3708 | /* Set the default route before the PCM stream is opened */ |
3709 | adev->mode = AUDIO_MODE_NORMAL; |
3710 | adev->out_device = AUDIO_DEVICE_OUT_SPEAKER; |
3711 | adev->in_device = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN; |
3712 | |
3713 | select_devices(adev); |
3714 | |
3715 | *device = &adev->hw_device.common; |
3716 | return 0; |
3717 | } |
3718 | |
3719 | static struct hw_module_methods_t hal_module_methods = { |
3720 | .open = adev_open, |
3721 | }; |
3722 | |
3723 | struct audio_module HAL_MODULE_INFO_SYM = { |
3724 | .common = { |
3725 | .tag = HARDWARE_MODULE_TAG, |
3726 | .module_api_version = AUDIO_MODULE_API_VERSION_0_1, |
3727 | .hal_api_version = HARDWARE_HAL_API_VERSION, |
3728 | .id = AUDIO_HARDWARE_MODULE_ID, |
3729 | .name = "aml audio HW HAL", |
3730 | .author = "amlogic, Corp.", |
3731 | .methods = &hal_module_methods, |
3732 | }, |
3733 | }; |
3734 |