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1 /*
2  * Copyright (C) 2013-2014 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_NDEBUG 0*/
19 /*#define VERY_VERY_VERBOSE_LOGGING*/
20 #ifdef VERY_VERY_VERBOSE_LOGGING
21 #define ALOGVV ALOGV
22 #else
23 #define ALOGVV(a...) do { } while(0)
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 <math.h>
32 #include <dlfcn.h>
33 #include <sys/resource.h>
34 #include <sys/prctl.h>
35 
36 #include <cutils/log.h>
37 #include <cutils/str_parms.h>
38 #include <cutils/properties.h>
39 #include <cutils/atomic.h>
40 #include <cutils/sched_policy.h>
41 
42 #include <hardware/audio_effect.h>
43 #include <hardware/audio_alsaops.h>
44 #include <system/thread_defs.h>
45 #include <audio_effects/effect_aec.h>
46 #include <audio_effects/effect_ns.h>
47 #include "audio_hw.h"
48 #include "audio_extn.h"
49 #include "platform_api.h"
50 #include <platform.h>
51 #include "voice_extn.h"
52 
53 #include "sound/compress_params.h"
54 
55 #define COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
56 // 2 buffers causes problems with high bitrate files
57 #define COMPRESS_OFFLOAD_NUM_FRAGMENTS 3
58 /* ToDo: Check and update a proper value in msec */
59 #define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
60 #define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
61 
62 #define PROXY_OPEN_RETRY_COUNT           100
63 #define PROXY_OPEN_WAIT_TIME             20
64 
65 #define MIN_CHANNEL_COUNT                1
66 #define DEFAULT_CHANNEL_COUNT            2
67 
68 #ifndef MAX_TARGET_SPECIFIC_CHANNEL_CNT
69 #define MAX_CHANNEL_COUNT 1
70 #else
71 #define MAX_CHANNEL_COUNT atoi(XSTR(MAX_TARGET_SPECIFIC_CHANNEL_CNT))
72 #define XSTR(x) STR(x)
73 #define STR(x) #x
74 #endif
75 
76 static unsigned int configured_low_latency_capture_period_size =
77         LOW_LATENCY_CAPTURE_PERIOD_SIZE;
78 
79 /* This constant enables extended precision handling.
80  * TODO The flag is off until more testing is done.
81  */
82 static const bool k_enable_extended_precision = false;
83 
84 struct pcm_config pcm_config_deep_buffer = {
85     .channels = DEFAULT_CHANNEL_COUNT,
86     .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
87     .period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE,
88     .period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT,
89     .format = PCM_FORMAT_S16_LE,
90     .start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
91     .stop_threshold = INT_MAX,
92     .avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
93 };
94 
95 struct pcm_config pcm_config_low_latency = {
96     .channels = DEFAULT_CHANNEL_COUNT,
97     .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
98     .period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE,
99     .period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT,
100     .format = PCM_FORMAT_S16_LE,
101     .start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
102     .stop_threshold = INT_MAX,
103     .avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
104 };
105 
106 struct pcm_config pcm_config_hdmi_multi = {
107     .channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
108     .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
109     .period_size = HDMI_MULTI_PERIOD_SIZE,
110     .period_count = HDMI_MULTI_PERIOD_COUNT,
111     .format = PCM_FORMAT_S16_LE,
112     .start_threshold = 0,
113     .stop_threshold = INT_MAX,
114     .avail_min = 0,
115 };
116 
117 struct pcm_config pcm_config_audio_capture = {
118     .channels = DEFAULT_CHANNEL_COUNT,
119     .period_count = AUDIO_CAPTURE_PERIOD_COUNT,
120     .format = PCM_FORMAT_S16_LE,
121     .stop_threshold = INT_MAX,
122     .avail_min = 0,
123 };
124 
125 #define AFE_PROXY_CHANNEL_COUNT 2
126 #define AFE_PROXY_SAMPLING_RATE 48000
127 
128 #define AFE_PROXY_PLAYBACK_PERIOD_SIZE  768
129 #define AFE_PROXY_PLAYBACK_PERIOD_COUNT 4
130 
131 struct pcm_config pcm_config_afe_proxy_playback = {
132     .channels = AFE_PROXY_CHANNEL_COUNT,
133     .rate = AFE_PROXY_SAMPLING_RATE,
134     .period_size = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
135     .period_count = AFE_PROXY_PLAYBACK_PERIOD_COUNT,
136     .format = PCM_FORMAT_S16_LE,
137     .start_threshold = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
138     .stop_threshold = INT_MAX,
139     .avail_min = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
140 };
141 
142 #define AFE_PROXY_RECORD_PERIOD_SIZE  768
143 #define AFE_PROXY_RECORD_PERIOD_COUNT 4
144 
145 struct pcm_config pcm_config_afe_proxy_record = {
146     .channels = AFE_PROXY_CHANNEL_COUNT,
147     .rate = AFE_PROXY_SAMPLING_RATE,
148     .period_size = AFE_PROXY_RECORD_PERIOD_SIZE,
149     .period_count = AFE_PROXY_RECORD_PERIOD_COUNT,
150     .format = PCM_FORMAT_S16_LE,
151     .start_threshold = AFE_PROXY_RECORD_PERIOD_SIZE,
152     .stop_threshold = INT_MAX,
153     .avail_min = AFE_PROXY_RECORD_PERIOD_SIZE,
154 };
155 
156 const char * const use_case_table[AUDIO_USECASE_MAX] = {
157     [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback",
158     [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback",
159     [USECASE_AUDIO_PLAYBACK_MULTI_CH] = "multi-channel-playback",
160     [USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback",
161     [USECASE_AUDIO_PLAYBACK_TTS] = "audio-tts-playback",
162     [USECASE_AUDIO_PLAYBACK_ULL] = "audio-ull-playback",
163 
164     [USECASE_AUDIO_RECORD] = "audio-record",
165     [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
166 
167     [USECASE_AUDIO_HFP_SCO] = "hfp-sco",
168     [USECASE_AUDIO_HFP_SCO_WB] = "hfp-sco-wb",
169 
170     [USECASE_VOICE_CALL] = "voice-call",
171     [USECASE_VOICE2_CALL] = "voice2-call",
172     [USECASE_VOLTE_CALL] = "volte-call",
173     [USECASE_QCHAT_CALL] = "qchat-call",
174     [USECASE_VOWLAN_CALL] = "vowlan-call",
175 
176     [USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib",
177     [USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record",
178 
179     [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback",
180     [USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record",
181 };
182 
183 
184 #define STRING_TO_ENUM(string) { #string, string }
185 
186 struct string_to_enum {
187     const char *name;
188     uint32_t value;
189 };
190 
191 static const struct string_to_enum out_channels_name_to_enum_table[] = {
192     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
193     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
194     STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
195 };
196 
197 static int set_voice_volume_l(struct audio_device *adev, float volume);
198 static struct audio_device *adev = NULL;
199 static pthread_mutex_t adev_init_lock;
200 static unsigned int audio_device_ref_count;
201 
202 __attribute__ ((visibility ("default")))
audio_hw_send_gain_dep_calibration(int level)203 bool audio_hw_send_gain_dep_calibration(int level) {
204     bool ret_val = false;
205     ALOGV("%s: enter ... ", __func__);
206 
207     pthread_mutex_lock(&adev_init_lock);
208 
209     if (adev != NULL && adev->platform != NULL) {
210         pthread_mutex_lock(&adev->lock);
211         ret_val = platform_send_gain_dep_cal(adev->platform, level);
212         pthread_mutex_unlock(&adev->lock);
213     } else {
214         ALOGE("%s: %s is NULL", __func__, adev == NULL ? "adev" : "adev->platform");
215     }
216 
217     pthread_mutex_unlock(&adev_init_lock);
218 
219     ALOGV("%s: exit with ret_val %d ", __func__, ret_val);
220     return ret_val;
221 }
222 
is_supported_format(audio_format_t format)223 static bool is_supported_format(audio_format_t format)
224 {
225     switch (format) {
226         case AUDIO_FORMAT_MP3:
227         case AUDIO_FORMAT_AAC_LC:
228         case AUDIO_FORMAT_AAC_HE_V1:
229         case AUDIO_FORMAT_AAC_HE_V2:
230             return true;
231         default:
232             break;
233     }
234     return false;
235 }
236 
get_snd_codec_id(audio_format_t format)237 static int get_snd_codec_id(audio_format_t format)
238 {
239     int id = 0;
240 
241     switch (format & AUDIO_FORMAT_MAIN_MASK) {
242     case AUDIO_FORMAT_MP3:
243         id = SND_AUDIOCODEC_MP3;
244         break;
245     case AUDIO_FORMAT_AAC:
246         id = SND_AUDIOCODEC_AAC;
247         break;
248     default:
249         ALOGE("%s: Unsupported audio format", __func__);
250     }
251 
252     return id;
253 }
254 
enable_audio_route(struct audio_device * adev,struct audio_usecase * usecase)255 int enable_audio_route(struct audio_device *adev,
256                        struct audio_usecase *usecase)
257 {
258     snd_device_t snd_device;
259     char mixer_path[50];
260 
261     if (usecase == NULL)
262         return -EINVAL;
263 
264     ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
265 
266     if (usecase->type == PCM_CAPTURE)
267         snd_device = usecase->in_snd_device;
268     else
269         snd_device = usecase->out_snd_device;
270 
271     strcpy(mixer_path, use_case_table[usecase->id]);
272     platform_add_backend_name(adev->platform, mixer_path, snd_device);
273     ALOGD("%s: apply and update mixer path: %s", __func__, mixer_path);
274     audio_route_apply_and_update_path(adev->audio_route, mixer_path);
275 
276     ALOGV("%s: exit", __func__);
277     return 0;
278 }
279 
disable_audio_route(struct audio_device * adev,struct audio_usecase * usecase)280 int disable_audio_route(struct audio_device *adev,
281                         struct audio_usecase *usecase)
282 {
283     snd_device_t snd_device;
284     char mixer_path[50];
285 
286     if (usecase == NULL)
287         return -EINVAL;
288 
289     ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
290     if (usecase->type == PCM_CAPTURE)
291         snd_device = usecase->in_snd_device;
292     else
293         snd_device = usecase->out_snd_device;
294     strcpy(mixer_path, use_case_table[usecase->id]);
295     platform_add_backend_name(adev->platform, mixer_path, snd_device);
296     ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path);
297     audio_route_reset_and_update_path(adev->audio_route, mixer_path);
298 
299     ALOGV("%s: exit", __func__);
300     return 0;
301 }
302 
enable_snd_device(struct audio_device * adev,snd_device_t snd_device)303 int enable_snd_device(struct audio_device *adev,
304                       snd_device_t snd_device)
305 {
306     int i, num_devices = 0;
307     snd_device_t new_snd_devices[2];
308 
309     if (snd_device < SND_DEVICE_MIN ||
310         snd_device >= SND_DEVICE_MAX) {
311         ALOGE("%s: Invalid sound device %d", __func__, snd_device);
312         return -EINVAL;
313     }
314 
315     platform_send_audio_calibration(adev->platform, snd_device);
316 
317     adev->snd_dev_ref_cnt[snd_device]++;
318     if (adev->snd_dev_ref_cnt[snd_device] > 1) {
319         ALOGV("%s: snd_device(%d: %s) is already active",
320               __func__, snd_device, platform_get_snd_device_name(snd_device));
321         return 0;
322     }
323 
324     /* due to the possibility of calibration overwrite between listen
325         and audio, notify sound trigger hal before audio calibration is sent */
326     audio_extn_sound_trigger_update_device_status(snd_device,
327                                     ST_EVENT_SND_DEVICE_BUSY);
328 
329     if (audio_extn_spkr_prot_is_enabled())
330          audio_extn_spkr_prot_calib_cancel(adev);
331 
332     audio_extn_dsm_feedback_enable(adev, snd_device, true);
333 
334     if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
335         snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
336         audio_extn_spkr_prot_is_enabled()) {
337         if (audio_extn_spkr_prot_get_acdb_id(snd_device) < 0) {
338             adev->snd_dev_ref_cnt[snd_device]--;
339             return -EINVAL;
340         }
341         if (audio_extn_spkr_prot_start_processing(snd_device)) {
342             ALOGE("%s: spkr_start_processing failed", __func__);
343             return -EINVAL;
344         }
345     } else if (platform_can_split_snd_device(snd_device, &num_devices, new_snd_devices)) {
346         for (i = 0; i < num_devices; i++) {
347             enable_snd_device(adev, new_snd_devices[i]);
348         }
349         platform_set_speaker_gain_in_combo(adev, snd_device, true);
350     } else {
351         const char * dev_path = platform_get_snd_device_name(snd_device);
352         ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, dev_path);
353         audio_route_apply_and_update_path(adev->audio_route, dev_path);
354     }
355 
356     return 0;
357 }
358 
disable_snd_device(struct audio_device * adev,snd_device_t snd_device)359 int disable_snd_device(struct audio_device *adev,
360                        snd_device_t snd_device)
361 {
362     int i, num_devices = 0;
363     snd_device_t new_snd_devices[2];
364 
365     if (snd_device < SND_DEVICE_MIN ||
366         snd_device >= SND_DEVICE_MAX) {
367         ALOGE("%s: Invalid sound device %d", __func__, snd_device);
368         return -EINVAL;
369     }
370     if (adev->snd_dev_ref_cnt[snd_device] <= 0) {
371         ALOGE("%s: device ref cnt is already 0", __func__);
372         return -EINVAL;
373     }
374     adev->snd_dev_ref_cnt[snd_device]--;
375     if (adev->snd_dev_ref_cnt[snd_device] == 0) {
376         const char * dev_path = platform_get_snd_device_name(snd_device);
377         ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, dev_path);
378 
379         audio_extn_dsm_feedback_enable(adev, snd_device, false);
380         if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
381             snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
382             audio_extn_spkr_prot_is_enabled()) {
383             audio_extn_spkr_prot_stop_processing(snd_device);
384         } else if (platform_can_split_snd_device(snd_device, &num_devices, new_snd_devices)) {
385             for (i = 0; i < num_devices; i++) {
386                 disable_snd_device(adev, new_snd_devices[i]);
387             }
388             platform_set_speaker_gain_in_combo(adev, snd_device, false);
389         } else {
390             audio_route_reset_and_update_path(adev->audio_route, dev_path);
391         }
392         audio_extn_sound_trigger_update_device_status(snd_device,
393                                         ST_EVENT_SND_DEVICE_FREE);
394     }
395 
396     return 0;
397 }
398 
check_and_route_playback_usecases(struct audio_device * adev,struct audio_usecase * uc_info,snd_device_t snd_device)399 static void check_and_route_playback_usecases(struct audio_device *adev,
400                                               struct audio_usecase *uc_info,
401                                               snd_device_t snd_device)
402 {
403     struct listnode *node;
404     struct audio_usecase *usecase;
405     bool switch_device[AUDIO_USECASE_MAX];
406     int i, num_uc_to_switch = 0;
407 
408     /*
409      * This function is to make sure that all the usecases that are active on
410      * the hardware codec backend are always routed to any one device that is
411      * handled by the hardware codec.
412      * For example, if low-latency and deep-buffer usecases are currently active
413      * on speaker and out_set_parameters(headset) is received on low-latency
414      * output, then we have to make sure deep-buffer is also switched to headset,
415      * because of the limitation that both the devices cannot be enabled
416      * at the same time as they share the same backend.
417      */
418     /* Disable all the usecases on the shared backend other than the
419        specified usecase */
420     for (i = 0; i < AUDIO_USECASE_MAX; i++)
421         switch_device[i] = false;
422 
423     list_for_each(node, &adev->usecase_list) {
424         usecase = node_to_item(node, struct audio_usecase, list);
425         if (usecase->type != PCM_CAPTURE &&
426                 usecase != uc_info &&
427                 usecase->out_snd_device != snd_device &&
428                 usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND &&
429                 platform_check_backends_match(snd_device, usecase->out_snd_device)) {
430             ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
431                   __func__, use_case_table[usecase->id],
432                   platform_get_snd_device_name(usecase->out_snd_device));
433             disable_audio_route(adev, usecase);
434             switch_device[usecase->id] = true;
435             num_uc_to_switch++;
436         }
437     }
438 
439     if (num_uc_to_switch) {
440         list_for_each(node, &adev->usecase_list) {
441             usecase = node_to_item(node, struct audio_usecase, list);
442             if (switch_device[usecase->id]) {
443                 disable_snd_device(adev, usecase->out_snd_device);
444             }
445         }
446 
447         list_for_each(node, &adev->usecase_list) {
448             usecase = node_to_item(node, struct audio_usecase, list);
449             if (switch_device[usecase->id]) {
450                 enable_snd_device(adev, snd_device);
451             }
452         }
453 
454         /* Re-route all the usecases on the shared backend other than the
455            specified usecase to new snd devices */
456         list_for_each(node, &adev->usecase_list) {
457             usecase = node_to_item(node, struct audio_usecase, list);
458             /* Update the out_snd_device only before enabling the audio route */
459             if (switch_device[usecase->id] ) {
460                 usecase->out_snd_device = snd_device;
461                 enable_audio_route(adev, usecase);
462             }
463         }
464     }
465 }
466 
check_and_route_capture_usecases(struct audio_device * adev,struct audio_usecase * uc_info,snd_device_t snd_device)467 static void check_and_route_capture_usecases(struct audio_device *adev,
468                                              struct audio_usecase *uc_info,
469                                              snd_device_t snd_device)
470 {
471     struct listnode *node;
472     struct audio_usecase *usecase;
473     bool switch_device[AUDIO_USECASE_MAX];
474     int i, num_uc_to_switch = 0;
475 
476     /*
477      * This function is to make sure that all the active capture usecases
478      * are always routed to the same input sound device.
479      * For example, if audio-record and voice-call usecases are currently
480      * active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece)
481      * is received for voice call then we have to make sure that audio-record
482      * usecase is also switched to earpiece i.e. voice-dmic-ef,
483      * because of the limitation that two devices cannot be enabled
484      * at the same time if they share the same backend.
485      */
486     for (i = 0; i < AUDIO_USECASE_MAX; i++)
487         switch_device[i] = false;
488 
489     list_for_each(node, &adev->usecase_list) {
490         usecase = node_to_item(node, struct audio_usecase, list);
491         if (usecase->type != PCM_PLAYBACK &&
492                 usecase != uc_info &&
493                 usecase->in_snd_device != snd_device &&
494                 (usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) {
495             ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
496                   __func__, use_case_table[usecase->id],
497                   platform_get_snd_device_name(usecase->in_snd_device));
498             disable_audio_route(adev, usecase);
499             switch_device[usecase->id] = true;
500             num_uc_to_switch++;
501         }
502     }
503 
504     if (num_uc_to_switch) {
505         list_for_each(node, &adev->usecase_list) {
506             usecase = node_to_item(node, struct audio_usecase, list);
507             if (switch_device[usecase->id]) {
508                 disable_snd_device(adev, usecase->in_snd_device);
509             }
510         }
511 
512         list_for_each(node, &adev->usecase_list) {
513             usecase = node_to_item(node, struct audio_usecase, list);
514             if (switch_device[usecase->id]) {
515                 enable_snd_device(adev, snd_device);
516             }
517         }
518 
519         /* Re-route all the usecases on the shared backend other than the
520            specified usecase to new snd devices */
521         list_for_each(node, &adev->usecase_list) {
522             usecase = node_to_item(node, struct audio_usecase, list);
523             /* Update the in_snd_device only before enabling the audio route */
524             if (switch_device[usecase->id] ) {
525                 usecase->in_snd_device = snd_device;
526                 enable_audio_route(adev, usecase);
527             }
528         }
529     }
530 }
531 
532 /* must be called with hw device mutex locked */
read_hdmi_channel_masks(struct stream_out * out)533 static int read_hdmi_channel_masks(struct stream_out *out)
534 {
535     int ret = 0;
536     int channels = platform_edid_get_max_channels(out->dev->platform);
537 
538     switch (channels) {
539         /*
540          * Do not handle stereo output in Multi-channel cases
541          * Stereo case is handled in normal playback path
542          */
543     case 6:
544         ALOGV("%s: HDMI supports 5.1", __func__);
545         out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
546         break;
547     case 8:
548         ALOGV("%s: HDMI supports 5.1 and 7.1 channels", __func__);
549         out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
550         out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1;
551         break;
552     default:
553         ALOGE("HDMI does not support multi channel playback");
554         ret = -ENOSYS;
555         break;
556     }
557     return ret;
558 }
559 
get_voice_usecase_id_from_list(struct audio_device * adev)560 static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev)
561 {
562     struct audio_usecase *usecase;
563     struct listnode *node;
564 
565     list_for_each(node, &adev->usecase_list) {
566         usecase = node_to_item(node, struct audio_usecase, list);
567         if (usecase->type == VOICE_CALL) {
568             ALOGV("%s: usecase id %d", __func__, usecase->id);
569             return usecase->id;
570         }
571     }
572     return USECASE_INVALID;
573 }
574 
get_usecase_from_list(struct audio_device * adev,audio_usecase_t uc_id)575 struct audio_usecase *get_usecase_from_list(struct audio_device *adev,
576                                             audio_usecase_t uc_id)
577 {
578     struct audio_usecase *usecase;
579     struct listnode *node;
580 
581     list_for_each(node, &adev->usecase_list) {
582         usecase = node_to_item(node, struct audio_usecase, list);
583         if (usecase->id == uc_id)
584             return usecase;
585     }
586     return NULL;
587 }
588 
select_devices(struct audio_device * adev,audio_usecase_t uc_id)589 int select_devices(struct audio_device *adev,
590                    audio_usecase_t uc_id)
591 {
592     snd_device_t out_snd_device = SND_DEVICE_NONE;
593     snd_device_t in_snd_device = SND_DEVICE_NONE;
594     struct audio_usecase *usecase = NULL;
595     struct audio_usecase *vc_usecase = NULL;
596     struct audio_usecase *hfp_usecase = NULL;
597     audio_usecase_t hfp_ucid;
598     struct listnode *node;
599     int status = 0;
600 
601     usecase = get_usecase_from_list(adev, uc_id);
602     if (usecase == NULL) {
603         ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id);
604         return -EINVAL;
605     }
606 
607     if ((usecase->type == VOICE_CALL) ||
608         (usecase->type == PCM_HFP_CALL)) {
609         out_snd_device = platform_get_output_snd_device(adev->platform,
610                                                         usecase->stream.out->devices);
611         in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices);
612         usecase->devices = usecase->stream.out->devices;
613     } else {
614         /*
615          * If the voice call is active, use the sound devices of voice call usecase
616          * so that it would not result any device switch. All the usecases will
617          * be switched to new device when select_devices() is called for voice call
618          * usecase. This is to avoid switching devices for voice call when
619          * check_and_route_playback_usecases() is called below.
620          */
621         if (voice_is_in_call(adev)) {
622             vc_usecase = get_usecase_from_list(adev,
623                                                get_voice_usecase_id_from_list(adev));
624             if ((vc_usecase != NULL) &&
625                 ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
626                 (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
627                 in_snd_device = vc_usecase->in_snd_device;
628                 out_snd_device = vc_usecase->out_snd_device;
629             }
630         } else if (audio_extn_hfp_is_active(adev)) {
631             hfp_ucid = audio_extn_hfp_get_usecase();
632             hfp_usecase = get_usecase_from_list(adev, hfp_ucid);
633             if (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
634                    in_snd_device = hfp_usecase->in_snd_device;
635                    out_snd_device = hfp_usecase->out_snd_device;
636             }
637         }
638         if (usecase->type == PCM_PLAYBACK) {
639             usecase->devices = usecase->stream.out->devices;
640             in_snd_device = SND_DEVICE_NONE;
641             if (out_snd_device == SND_DEVICE_NONE) {
642                 out_snd_device = platform_get_output_snd_device(adev->platform,
643                                             usecase->stream.out->devices);
644                 if (usecase->stream.out == adev->primary_output &&
645                         adev->active_input &&
646                         (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
647                             adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
648                         out_snd_device != usecase->out_snd_device) {
649                     select_devices(adev, adev->active_input->usecase);
650                 }
651             }
652         } else if (usecase->type == PCM_CAPTURE) {
653             usecase->devices = usecase->stream.in->device;
654             out_snd_device = SND_DEVICE_NONE;
655             if (in_snd_device == SND_DEVICE_NONE) {
656                 audio_devices_t out_device = AUDIO_DEVICE_NONE;
657                 if (adev->active_input &&
658                         (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
659                             adev->mode == AUDIO_MODE_IN_COMMUNICATION)) {
660                     platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
661                     if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
662                         out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
663                     } else if (adev->primary_output) {
664                         out_device = adev->primary_output->devices;
665                     }
666                 }
667                 in_snd_device = platform_get_input_snd_device(adev->platform, out_device);
668             }
669         }
670     }
671 
672     if (out_snd_device == usecase->out_snd_device &&
673         in_snd_device == usecase->in_snd_device) {
674         return 0;
675     }
676 
677     ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__,
678           out_snd_device, platform_get_snd_device_name(out_snd_device),
679           in_snd_device,  platform_get_snd_device_name(in_snd_device));
680 
681     /*
682      * Limitation: While in call, to do a device switch we need to disable
683      * and enable both RX and TX devices though one of them is same as current
684      * device.
685      */
686     if ((usecase->type == VOICE_CALL) &&
687         (usecase->in_snd_device != SND_DEVICE_NONE) &&
688         (usecase->out_snd_device != SND_DEVICE_NONE)) {
689         status = platform_switch_voice_call_device_pre(adev->platform);
690         /* Disable sidetone only if voice call already exists */
691         if (voice_is_call_state_active(adev))
692             voice_set_sidetone(adev, usecase->out_snd_device, false);
693     }
694 
695     /* Disable current sound devices */
696     if (usecase->out_snd_device != SND_DEVICE_NONE) {
697         disable_audio_route(adev, usecase);
698         disable_snd_device(adev, usecase->out_snd_device);
699     }
700 
701     if (usecase->in_snd_device != SND_DEVICE_NONE) {
702         disable_audio_route(adev, usecase);
703         disable_snd_device(adev, usecase->in_snd_device);
704     }
705 
706     /* Applicable only on the targets that has external modem.
707      * New device information should be sent to modem before enabling
708      * the devices to reduce in-call device switch time.
709      */
710     if ((usecase->type == VOICE_CALL) &&
711         (usecase->in_snd_device != SND_DEVICE_NONE) &&
712         (usecase->out_snd_device != SND_DEVICE_NONE)) {
713         status = platform_switch_voice_call_enable_device_config(adev->platform,
714                                                                  out_snd_device,
715                                                                  in_snd_device);
716     }
717 
718     /* Enable new sound devices */
719     if (out_snd_device != SND_DEVICE_NONE) {
720         if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)
721             check_and_route_playback_usecases(adev, usecase, out_snd_device);
722         enable_snd_device(adev, out_snd_device);
723     }
724 
725     if (in_snd_device != SND_DEVICE_NONE) {
726         check_and_route_capture_usecases(adev, usecase, in_snd_device);
727         enable_snd_device(adev, in_snd_device);
728     }
729 
730     if (usecase->type == VOICE_CALL)
731         status = platform_switch_voice_call_device_post(adev->platform,
732                                                         out_snd_device,
733                                                         in_snd_device);
734 
735     usecase->in_snd_device = in_snd_device;
736     usecase->out_snd_device = out_snd_device;
737 
738     enable_audio_route(adev, usecase);
739 
740     /* Applicable only on the targets that has external modem.
741      * Enable device command should be sent to modem only after
742      * enabling voice call mixer controls
743      */
744     if (usecase->type == VOICE_CALL) {
745         status = platform_switch_voice_call_usecase_route_post(adev->platform,
746                                                                out_snd_device,
747                                                                in_snd_device);
748          /* Enable sidetone only if voice call already exists */
749         if (voice_is_call_state_active(adev))
750             voice_set_sidetone(adev, out_snd_device, true);
751     }
752 
753     return status;
754 }
755 
stop_input_stream(struct stream_in * in)756 static int stop_input_stream(struct stream_in *in)
757 {
758     int i, ret = 0;
759     struct audio_usecase *uc_info;
760     struct audio_device *adev = in->dev;
761 
762     adev->active_input = NULL;
763 
764     ALOGV("%s: enter: usecase(%d: %s)", __func__,
765           in->usecase, use_case_table[in->usecase]);
766     uc_info = get_usecase_from_list(adev, in->usecase);
767     if (uc_info == NULL) {
768         ALOGE("%s: Could not find the usecase (%d) in the list",
769               __func__, in->usecase);
770         return -EINVAL;
771     }
772 
773     /* 1. Disable stream specific mixer controls */
774     disable_audio_route(adev, uc_info);
775 
776     /* 2. Disable the tx device */
777     disable_snd_device(adev, uc_info->in_snd_device);
778 
779     list_remove(&uc_info->list);
780     free(uc_info);
781 
782     ALOGV("%s: exit: status(%d)", __func__, ret);
783     return ret;
784 }
785 
start_input_stream(struct stream_in * in)786 int start_input_stream(struct stream_in *in)
787 {
788     /* 1. Enable output device and stream routing controls */
789     int ret = 0;
790     struct audio_usecase *uc_info;
791     struct audio_device *adev = in->dev;
792 
793     ALOGV("%s: enter: usecase(%d)", __func__, in->usecase);
794     in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE);
795     if (in->pcm_device_id < 0) {
796         ALOGE("%s: Could not find PCM device id for the usecase(%d)",
797               __func__, in->usecase);
798         ret = -EINVAL;
799         goto error_config;
800     }
801 
802     adev->active_input = in;
803     uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
804     uc_info->id = in->usecase;
805     uc_info->type = PCM_CAPTURE;
806     uc_info->stream.in = in;
807     uc_info->devices = in->device;
808     uc_info->in_snd_device = SND_DEVICE_NONE;
809     uc_info->out_snd_device = SND_DEVICE_NONE;
810 
811     list_add_tail(&adev->usecase_list, &uc_info->list);
812 
813     audio_extn_perf_lock_acquire();
814 
815     select_devices(adev, in->usecase);
816 
817     ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
818           __func__, adev->snd_card, in->pcm_device_id, in->config.channels);
819 
820     unsigned int flags = PCM_IN;
821     unsigned int pcm_open_retry_count = 0;
822 
823     if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
824         flags |= PCM_MMAP | PCM_NOIRQ;
825         pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
826     }
827 
828     while (1) {
829         in->pcm = pcm_open(adev->snd_card, in->pcm_device_id,
830                            flags, &in->config);
831         if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
832             ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
833             if (in->pcm != NULL) {
834                 pcm_close(in->pcm);
835                 in->pcm = NULL;
836             }
837             if (pcm_open_retry_count-- == 0) {
838                 ret = -EIO;
839                 goto error_open;
840             }
841             usleep(PROXY_OPEN_WAIT_TIME * 1000);
842             continue;
843         }
844         break;
845     }
846 
847     ALOGV("%s: pcm_prepare start", __func__);
848     pcm_prepare(in->pcm);
849 
850     audio_extn_perf_lock_release();
851 
852     ALOGV("%s: exit", __func__);
853 
854     return ret;
855 
856 error_open:
857     stop_input_stream(in);
858     audio_extn_perf_lock_release();
859 
860 error_config:
861     adev->active_input = NULL;
862     ALOGD("%s: exit: status(%d)", __func__, ret);
863 
864     return ret;
865 }
866 
lock_input_stream(struct stream_in * in)867 void lock_input_stream(struct stream_in *in)
868 {
869     pthread_mutex_lock(&in->pre_lock);
870     pthread_mutex_lock(&in->lock);
871     pthread_mutex_unlock(&in->pre_lock);
872 }
873 
lock_output_stream(struct stream_out * out)874 void lock_output_stream(struct stream_out *out)
875 {
876     pthread_mutex_lock(&out->pre_lock);
877     pthread_mutex_lock(&out->lock);
878     pthread_mutex_unlock(&out->pre_lock);
879 }
880 
881 /* must be called with out->lock locked */
send_offload_cmd_l(struct stream_out * out,int command)882 static int send_offload_cmd_l(struct stream_out* out, int command)
883 {
884     struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));
885 
886     ALOGVV("%s %d", __func__, command);
887 
888     cmd->cmd = command;
889     list_add_tail(&out->offload_cmd_list, &cmd->node);
890     pthread_cond_signal(&out->offload_cond);
891     return 0;
892 }
893 
894 /* must be called iwth out->lock locked */
stop_compressed_output_l(struct stream_out * out)895 static void stop_compressed_output_l(struct stream_out *out)
896 {
897     out->offload_state = OFFLOAD_STATE_IDLE;
898     out->playback_started = 0;
899     out->send_new_metadata = 1;
900     if (out->compr != NULL) {
901         compress_stop(out->compr);
902         while (out->offload_thread_blocked) {
903             pthread_cond_wait(&out->cond, &out->lock);
904         }
905     }
906 }
907 
offload_thread_loop(void * context)908 static void *offload_thread_loop(void *context)
909 {
910     struct stream_out *out = (struct stream_out *) context;
911     struct listnode *item;
912 
913     out->offload_state = OFFLOAD_STATE_IDLE;
914     out->playback_started = 0;
915 
916     setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
917     set_sched_policy(0, SP_FOREGROUND);
918     prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);
919 
920     ALOGV("%s", __func__);
921     lock_output_stream(out);
922     for (;;) {
923         struct offload_cmd *cmd = NULL;
924         stream_callback_event_t event;
925         bool send_callback = false;
926 
927         ALOGVV("%s offload_cmd_list %d out->offload_state %d",
928               __func__, list_empty(&out->offload_cmd_list),
929               out->offload_state);
930         if (list_empty(&out->offload_cmd_list)) {
931             ALOGV("%s SLEEPING", __func__);
932             pthread_cond_wait(&out->offload_cond, &out->lock);
933             ALOGV("%s RUNNING", __func__);
934             continue;
935         }
936 
937         item = list_head(&out->offload_cmd_list);
938         cmd = node_to_item(item, struct offload_cmd, node);
939         list_remove(item);
940 
941         ALOGVV("%s STATE %d CMD %d out->compr %p",
942                __func__, out->offload_state, cmd->cmd, out->compr);
943 
944         if (cmd->cmd == OFFLOAD_CMD_EXIT) {
945             free(cmd);
946             break;
947         }
948 
949         if (out->compr == NULL) {
950             ALOGE("%s: Compress handle is NULL", __func__);
951             pthread_cond_signal(&out->cond);
952             continue;
953         }
954         out->offload_thread_blocked = true;
955         pthread_mutex_unlock(&out->lock);
956         send_callback = false;
957         switch(cmd->cmd) {
958         case OFFLOAD_CMD_WAIT_FOR_BUFFER:
959             compress_wait(out->compr, -1);
960             send_callback = true;
961             event = STREAM_CBK_EVENT_WRITE_READY;
962             break;
963         case OFFLOAD_CMD_PARTIAL_DRAIN:
964             compress_next_track(out->compr);
965             compress_partial_drain(out->compr);
966             send_callback = true;
967             event = STREAM_CBK_EVENT_DRAIN_READY;
968             /* Resend the metadata for next iteration */
969             out->send_new_metadata = 1;
970             break;
971         case OFFLOAD_CMD_DRAIN:
972             compress_drain(out->compr);
973             send_callback = true;
974             event = STREAM_CBK_EVENT_DRAIN_READY;
975             break;
976         default:
977             ALOGE("%s unknown command received: %d", __func__, cmd->cmd);
978             break;
979         }
980         lock_output_stream(out);
981         out->offload_thread_blocked = false;
982         pthread_cond_signal(&out->cond);
983         if (send_callback) {
984             ALOGVV("%s: sending offload_callback event %d", __func__, event);
985             out->offload_callback(event, NULL, out->offload_cookie);
986         }
987         free(cmd);
988     }
989 
990     pthread_cond_signal(&out->cond);
991     while (!list_empty(&out->offload_cmd_list)) {
992         item = list_head(&out->offload_cmd_list);
993         list_remove(item);
994         free(node_to_item(item, struct offload_cmd, node));
995     }
996     pthread_mutex_unlock(&out->lock);
997 
998     return NULL;
999 }
1000 
create_offload_callback_thread(struct stream_out * out)1001 static int create_offload_callback_thread(struct stream_out *out)
1002 {
1003     pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL);
1004     list_init(&out->offload_cmd_list);
1005     pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL,
1006                     offload_thread_loop, out);
1007     return 0;
1008 }
1009 
destroy_offload_callback_thread(struct stream_out * out)1010 static int destroy_offload_callback_thread(struct stream_out *out)
1011 {
1012     lock_output_stream(out);
1013     stop_compressed_output_l(out);
1014     send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
1015 
1016     pthread_mutex_unlock(&out->lock);
1017     pthread_join(out->offload_thread, (void **) NULL);
1018     pthread_cond_destroy(&out->offload_cond);
1019 
1020     return 0;
1021 }
1022 
allow_hdmi_channel_config(struct audio_device * adev)1023 static bool allow_hdmi_channel_config(struct audio_device *adev)
1024 {
1025     struct listnode *node;
1026     struct audio_usecase *usecase;
1027     bool ret = true;
1028 
1029     list_for_each(node, &adev->usecase_list) {
1030         usecase = node_to_item(node, struct audio_usecase, list);
1031         if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
1032             /*
1033              * If voice call is already existing, do not proceed further to avoid
1034              * disabling/enabling both RX and TX devices, CSD calls, etc.
1035              * Once the voice call done, the HDMI channels can be configured to
1036              * max channels of remaining use cases.
1037              */
1038             if (usecase->id == USECASE_VOICE_CALL) {
1039                 ALOGD("%s: voice call is active, no change in HDMI channels",
1040                       __func__);
1041                 ret = false;
1042                 break;
1043             } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
1044                 ALOGD("%s: multi channel playback is active, "
1045                       "no change in HDMI channels", __func__);
1046                 ret = false;
1047                 break;
1048             }
1049         }
1050     }
1051     return ret;
1052 }
1053 
check_and_set_hdmi_channels(struct audio_device * adev,unsigned int channels)1054 static int check_and_set_hdmi_channels(struct audio_device *adev,
1055                                        unsigned int channels)
1056 {
1057     struct listnode *node;
1058     struct audio_usecase *usecase;
1059 
1060     /* Check if change in HDMI channel config is allowed */
1061     if (!allow_hdmi_channel_config(adev))
1062         return 0;
1063 
1064     if (channels == adev->cur_hdmi_channels) {
1065         ALOGD("%s: Requested channels are same as current", __func__);
1066         return 0;
1067     }
1068 
1069     platform_set_hdmi_channels(adev->platform, channels);
1070     adev->cur_hdmi_channels = channels;
1071 
1072     /*
1073      * Deroute all the playback streams routed to HDMI so that
1074      * the back end is deactivated. Note that backend will not
1075      * be deactivated if any one stream is connected to it.
1076      */
1077     list_for_each(node, &adev->usecase_list) {
1078         usecase = node_to_item(node, struct audio_usecase, list);
1079         if (usecase->type == PCM_PLAYBACK &&
1080                 usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
1081             disable_audio_route(adev, usecase);
1082         }
1083     }
1084 
1085     /*
1086      * Enable all the streams disabled above. Now the HDMI backend
1087      * will be activated with new channel configuration
1088      */
1089     list_for_each(node, &adev->usecase_list) {
1090         usecase = node_to_item(node, struct audio_usecase, list);
1091         if (usecase->type == PCM_PLAYBACK &&
1092                 usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
1093             enable_audio_route(adev, usecase);
1094         }
1095     }
1096 
1097     return 0;
1098 }
1099 
stop_output_stream(struct stream_out * out)1100 static int stop_output_stream(struct stream_out *out)
1101 {
1102     int i, ret = 0;
1103     struct audio_usecase *uc_info;
1104     struct audio_device *adev = out->dev;
1105 
1106     ALOGV("%s: enter: usecase(%d: %s)", __func__,
1107           out->usecase, use_case_table[out->usecase]);
1108     uc_info = get_usecase_from_list(adev, out->usecase);
1109     if (uc_info == NULL) {
1110         ALOGE("%s: Could not find the usecase (%d) in the list",
1111               __func__, out->usecase);
1112         return -EINVAL;
1113     }
1114 
1115     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1116         if (adev->visualizer_stop_output != NULL)
1117             adev->visualizer_stop_output(out->handle, out->pcm_device_id);
1118         if (adev->offload_effects_stop_output != NULL)
1119             adev->offload_effects_stop_output(out->handle, out->pcm_device_id);
1120     }
1121 
1122     /* 1. Get and set stream specific mixer controls */
1123     disable_audio_route(adev, uc_info);
1124 
1125     /* 2. Disable the rx device */
1126     disable_snd_device(adev, uc_info->out_snd_device);
1127 
1128     list_remove(&uc_info->list);
1129     free(uc_info);
1130 
1131     audio_extn_extspk_update(adev->extspk);
1132 
1133     /* Must be called after removing the usecase from list */
1134     if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
1135         check_and_set_hdmi_channels(adev, DEFAULT_HDMI_OUT_CHANNELS);
1136 
1137     ALOGV("%s: exit: status(%d)", __func__, ret);
1138     return ret;
1139 }
1140 
start_output_stream(struct stream_out * out)1141 int start_output_stream(struct stream_out *out)
1142 {
1143     int ret = 0;
1144     struct audio_usecase *uc_info;
1145     struct audio_device *adev = out->dev;
1146 
1147     ALOGV("%s: enter: usecase(%d: %s) devices(%#x)",
1148           __func__, out->usecase, use_case_table[out->usecase], out->devices);
1149     out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
1150     if (out->pcm_device_id < 0) {
1151         ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)",
1152               __func__, out->pcm_device_id, out->usecase);
1153         ret = -EINVAL;
1154         goto error_config;
1155     }
1156 
1157     uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
1158     uc_info->id = out->usecase;
1159     uc_info->type = PCM_PLAYBACK;
1160     uc_info->stream.out = out;
1161     uc_info->devices = out->devices;
1162     uc_info->in_snd_device = SND_DEVICE_NONE;
1163     uc_info->out_snd_device = SND_DEVICE_NONE;
1164 
1165     /* This must be called before adding this usecase to the list */
1166     if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
1167         check_and_set_hdmi_channels(adev, out->config.channels);
1168 
1169     list_add_tail(&adev->usecase_list, &uc_info->list);
1170 
1171     audio_extn_perf_lock_acquire();
1172 
1173     select_devices(adev, out->usecase);
1174 
1175     audio_extn_extspk_update(adev->extspk);
1176 
1177     ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
1178           __func__, adev->snd_card, out->pcm_device_id, out->config.format);
1179     if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1180         unsigned int flags = PCM_OUT;
1181         unsigned int pcm_open_retry_count = 0;
1182         if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
1183             flags |= PCM_MMAP | PCM_NOIRQ;
1184             pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
1185         } else
1186             flags |= PCM_MONOTONIC;
1187 
1188         while (1) {
1189             out->pcm = pcm_open(adev->snd_card, out->pcm_device_id,
1190                                flags, &out->config);
1191             if (out->pcm == NULL || !pcm_is_ready(out->pcm)) {
1192                 ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
1193                 if (out->pcm != NULL) {
1194                     pcm_close(out->pcm);
1195                     out->pcm = NULL;
1196                 }
1197                 if (pcm_open_retry_count-- == 0) {
1198                     ret = -EIO;
1199                     goto error_open;
1200                 }
1201                 usleep(PROXY_OPEN_WAIT_TIME * 1000);
1202                 continue;
1203             }
1204             break;
1205         }
1206         ALOGV("%s: pcm_prepare start", __func__);
1207         if (pcm_is_ready(out->pcm))
1208             pcm_prepare(out->pcm);
1209 
1210     } else {
1211         out->pcm = NULL;
1212         out->compr = compress_open(adev->snd_card, out->pcm_device_id,
1213                                    COMPRESS_IN, &out->compr_config);
1214         if (out->compr && !is_compress_ready(out->compr)) {
1215             ALOGE("%s: %s", __func__, compress_get_error(out->compr));
1216             compress_close(out->compr);
1217             out->compr = NULL;
1218             ret = -EIO;
1219             goto error_open;
1220         }
1221         if (out->offload_callback)
1222             compress_nonblock(out->compr, out->non_blocking);
1223 
1224         if (adev->visualizer_start_output != NULL)
1225             adev->visualizer_start_output(out->handle, out->pcm_device_id);
1226         if (adev->offload_effects_start_output != NULL)
1227             adev->offload_effects_start_output(out->handle, out->pcm_device_id);
1228     }
1229     audio_extn_perf_lock_release();
1230     ALOGV("%s: exit", __func__);
1231     return 0;
1232 error_open:
1233     audio_extn_perf_lock_release();
1234     stop_output_stream(out);
1235 error_config:
1236     return ret;
1237 }
1238 
check_input_parameters(uint32_t sample_rate,audio_format_t format,int channel_count)1239 static int check_input_parameters(uint32_t sample_rate,
1240                                   audio_format_t format,
1241                                   int channel_count)
1242 {
1243     if (format != AUDIO_FORMAT_PCM_16_BIT) {
1244         ALOGE("%s: unsupported AUDIO FORMAT (%d) ", __func__, format);
1245         return -EINVAL;
1246     }
1247 
1248     if ((channel_count < MIN_CHANNEL_COUNT) || (channel_count > MAX_CHANNEL_COUNT)) {
1249         ALOGE("%s: unsupported channel count (%d) passed  Min / Max (%d / %d)", __func__,
1250                channel_count, MIN_CHANNEL_COUNT, MAX_CHANNEL_COUNT);
1251         return -EINVAL;
1252     }
1253 
1254     switch (sample_rate) {
1255     case 8000:
1256     case 11025:
1257     case 12000:
1258     case 16000:
1259     case 22050:
1260     case 24000:
1261     case 32000:
1262     case 44100:
1263     case 48000:
1264         break;
1265     default:
1266         ALOGE("%s: unsupported (%d) samplerate passed ", __func__, sample_rate);
1267         return -EINVAL;
1268     }
1269 
1270     return 0;
1271 }
1272 
get_input_buffer_size(uint32_t sample_rate,audio_format_t format,int channel_count,bool is_low_latency)1273 static size_t get_input_buffer_size(uint32_t sample_rate,
1274                                     audio_format_t format,
1275                                     int channel_count,
1276                                     bool is_low_latency)
1277 {
1278     size_t size = 0;
1279 
1280     if (check_input_parameters(sample_rate, format, channel_count) != 0)
1281         return 0;
1282 
1283     size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
1284     if (is_low_latency)
1285         size = configured_low_latency_capture_period_size;
1286     /* ToDo: should use frame_size computed based on the format and
1287        channel_count here. */
1288     size *= sizeof(short) * channel_count;
1289 
1290     /* make sure the size is multiple of 32 bytes
1291      * At 48 kHz mono 16-bit PCM:
1292      *  5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
1293      *  3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
1294      */
1295     size += 0x1f;
1296     size &= ~0x1f;
1297 
1298     return size;
1299 }
1300 
out_get_sample_rate(const struct audio_stream * stream)1301 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
1302 {
1303     struct stream_out *out = (struct stream_out *)stream;
1304 
1305     return out->sample_rate;
1306 }
1307 
out_set_sample_rate(struct audio_stream * stream __unused,uint32_t rate __unused)1308 static int out_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused)
1309 {
1310     return -ENOSYS;
1311 }
1312 
out_get_buffer_size(const struct audio_stream * stream)1313 static size_t out_get_buffer_size(const struct audio_stream *stream)
1314 {
1315     struct stream_out *out = (struct stream_out *)stream;
1316 
1317     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1318         return out->compr_config.fragment_size;
1319     }
1320     return out->config.period_size *
1321                 audio_stream_out_frame_size((const struct audio_stream_out *)stream);
1322 }
1323 
out_get_channels(const struct audio_stream * stream)1324 static uint32_t out_get_channels(const struct audio_stream *stream)
1325 {
1326     struct stream_out *out = (struct stream_out *)stream;
1327 
1328     return out->channel_mask;
1329 }
1330 
out_get_format(const struct audio_stream * stream)1331 static audio_format_t out_get_format(const struct audio_stream *stream)
1332 {
1333     struct stream_out *out = (struct stream_out *)stream;
1334 
1335     return out->format;
1336 }
1337 
out_set_format(struct audio_stream * stream __unused,audio_format_t format __unused)1338 static int out_set_format(struct audio_stream *stream __unused, audio_format_t format __unused)
1339 {
1340     return -ENOSYS;
1341 }
1342 
out_standby(struct audio_stream * stream)1343 static int out_standby(struct audio_stream *stream)
1344 {
1345     struct stream_out *out = (struct stream_out *)stream;
1346     struct audio_device *adev = out->dev;
1347 
1348     ALOGV("%s: enter: usecase(%d: %s)", __func__,
1349           out->usecase, use_case_table[out->usecase]);
1350 
1351     lock_output_stream(out);
1352     if (!out->standby) {
1353         if (adev->adm_deregister_stream)
1354             adev->adm_deregister_stream(adev->adm_data, out->handle);
1355 
1356         pthread_mutex_lock(&adev->lock);
1357         out->standby = true;
1358         if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1359             if (out->pcm) {
1360                 pcm_close(out->pcm);
1361                 out->pcm = NULL;
1362             }
1363         } else {
1364             stop_compressed_output_l(out);
1365             out->gapless_mdata.encoder_delay = 0;
1366             out->gapless_mdata.encoder_padding = 0;
1367             if (out->compr != NULL) {
1368                 compress_close(out->compr);
1369                 out->compr = NULL;
1370             }
1371         }
1372         stop_output_stream(out);
1373         pthread_mutex_unlock(&adev->lock);
1374     }
1375     pthread_mutex_unlock(&out->lock);
1376     ALOGV("%s: exit", __func__);
1377     return 0;
1378 }
1379 
out_dump(const struct audio_stream * stream __unused,int fd __unused)1380 static int out_dump(const struct audio_stream *stream __unused, int fd __unused)
1381 {
1382     return 0;
1383 }
1384 
parse_compress_metadata(struct stream_out * out,struct str_parms * parms)1385 static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms)
1386 {
1387     int ret = 0;
1388     char value[32];
1389     struct compr_gapless_mdata tmp_mdata;
1390 
1391     if (!out || !parms) {
1392         return -EINVAL;
1393     }
1394 
1395     ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value));
1396     if (ret >= 0) {
1397         tmp_mdata.encoder_delay = atoi(value); //whats a good limit check?
1398     } else {
1399         return -EINVAL;
1400     }
1401 
1402     ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value));
1403     if (ret >= 0) {
1404         tmp_mdata.encoder_padding = atoi(value);
1405     } else {
1406         return -EINVAL;
1407     }
1408 
1409     out->gapless_mdata = tmp_mdata;
1410     out->send_new_metadata = 1;
1411     ALOGV("%s new encoder delay %u and padding %u", __func__,
1412           out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding);
1413 
1414     return 0;
1415 }
1416 
output_drives_call(struct audio_device * adev,struct stream_out * out)1417 static bool output_drives_call(struct audio_device *adev, struct stream_out *out)
1418 {
1419     return out == adev->primary_output || out == adev->voice_tx_output;
1420 }
1421 
out_set_parameters(struct audio_stream * stream,const char * kvpairs)1422 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
1423 {
1424     struct stream_out *out = (struct stream_out *)stream;
1425     struct audio_device *adev = out->dev;
1426     struct audio_usecase *usecase;
1427     struct listnode *node;
1428     struct str_parms *parms;
1429     char value[32];
1430     int ret, val = 0;
1431     bool select_new_device = false;
1432     int status = 0;
1433 
1434     ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
1435           __func__, out->usecase, use_case_table[out->usecase], kvpairs);
1436     parms = str_parms_create_str(kvpairs);
1437     ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
1438     if (ret >= 0) {
1439         val = atoi(value);
1440         lock_output_stream(out);
1441         pthread_mutex_lock(&adev->lock);
1442 
1443         /*
1444          * When HDMI cable is unplugged the music playback is paused and
1445          * the policy manager sends routing=0. But the audioflinger
1446          * continues to write data until standby time (3sec).
1447          * As the HDMI core is turned off, the write gets blocked.
1448          * Avoid this by routing audio to speaker until standby.
1449          */
1450         if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL &&
1451                 val == AUDIO_DEVICE_NONE) {
1452             val = AUDIO_DEVICE_OUT_SPEAKER;
1453         }
1454 
1455         /*
1456          * select_devices() call below switches all the usecases on the same
1457          * backend to the new device. Refer to check_and_route_playback_usecases() in
1458          * the select_devices(). But how do we undo this?
1459          *
1460          * For example, music playback is active on headset (deep-buffer usecase)
1461          * and if we go to ringtones and select a ringtone, low-latency usecase
1462          * will be started on headset+speaker. As we can't enable headset+speaker
1463          * and headset devices at the same time, select_devices() switches the music
1464          * playback to headset+speaker while starting low-lateny usecase for ringtone.
1465          * So when the ringtone playback is completed, how do we undo the same?
1466          *
1467          * We are relying on the out_set_parameters() call on deep-buffer output,
1468          * once the ringtone playback is ended.
1469          * NOTE: We should not check if the current devices are same as new devices.
1470          *       Because select_devices() must be called to switch back the music
1471          *       playback to headset.
1472          */
1473         if (val != 0) {
1474             out->devices = val;
1475 
1476             if (!out->standby)
1477                 select_devices(adev, out->usecase);
1478 
1479             if (output_drives_call(adev, out)) {
1480                 if (!voice_is_in_call(adev)) {
1481                     if (adev->mode == AUDIO_MODE_IN_CALL) {
1482                         adev->current_call_output = out;
1483                         ret = voice_start_call(adev);
1484                     }
1485                 } else {
1486                     adev->current_call_output = out;
1487                     voice_update_devices_for_all_voice_usecases(adev);
1488                 }
1489             }
1490         }
1491 
1492         pthread_mutex_unlock(&adev->lock);
1493         pthread_mutex_unlock(&out->lock);
1494 
1495         /*handles device and call state changes*/
1496         audio_extn_extspk_update(adev->extspk);
1497     }
1498 
1499     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1500         parse_compress_metadata(out, parms);
1501     }
1502 
1503     str_parms_destroy(parms);
1504     ALOGV("%s: exit: code(%d)", __func__, status);
1505     return status;
1506 }
1507 
out_get_parameters(const struct audio_stream * stream,const char * keys)1508 static char* out_get_parameters(const struct audio_stream *stream, const char *keys)
1509 {
1510     struct stream_out *out = (struct stream_out *)stream;
1511     struct str_parms *query = str_parms_create_str(keys);
1512     char *str;
1513     char value[256];
1514     struct str_parms *reply = str_parms_create();
1515     size_t i, j;
1516     int ret;
1517     bool first = true;
1518     ALOGV("%s: enter: keys - %s", __func__, keys);
1519     ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value));
1520     if (ret >= 0) {
1521         value[0] = '\0';
1522         i = 0;
1523         while (out->supported_channel_masks[i] != 0) {
1524             for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) {
1525                 if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) {
1526                     if (!first) {
1527                         strcat(value, "|");
1528                     }
1529                     strcat(value, out_channels_name_to_enum_table[j].name);
1530                     first = false;
1531                     break;
1532                 }
1533             }
1534             i++;
1535         }
1536         str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
1537         str = str_parms_to_str(reply);
1538     } else {
1539         str = strdup(keys);
1540     }
1541     str_parms_destroy(query);
1542     str_parms_destroy(reply);
1543     ALOGV("%s: exit: returns - %s", __func__, str);
1544     return str;
1545 }
1546 
out_get_latency(const struct audio_stream_out * stream)1547 static uint32_t out_get_latency(const struct audio_stream_out *stream)
1548 {
1549     struct stream_out *out = (struct stream_out *)stream;
1550 
1551     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD)
1552         return COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
1553 
1554     return (out->config.period_count * out->config.period_size * 1000) /
1555            (out->config.rate);
1556 }
1557 
out_set_volume(struct audio_stream_out * stream,float left,float right)1558 static int out_set_volume(struct audio_stream_out *stream, float left,
1559                           float right)
1560 {
1561     struct stream_out *out = (struct stream_out *)stream;
1562     int volume[2];
1563 
1564     if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
1565         /* only take left channel into account: the API is for stereo anyway */
1566         out->muted = (left == 0.0f);
1567         return 0;
1568     } else if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1569         const char *mixer_ctl_name = "Compress Playback Volume";
1570         struct audio_device *adev = out->dev;
1571         struct mixer_ctl *ctl;
1572         ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
1573         if (!ctl) {
1574             /* try with the control based on device id */
1575             int pcm_device_id = platform_get_pcm_device_id(out->usecase,
1576                                                        PCM_PLAYBACK);
1577             char ctl_name[128] = {0};
1578             snprintf(ctl_name, sizeof(ctl_name),
1579                      "Compress Playback %d Volume", pcm_device_id);
1580             ctl = mixer_get_ctl_by_name(adev->mixer, ctl_name);
1581             if (!ctl) {
1582                 ALOGE("%s: Could not get volume ctl mixer cmd", __func__);
1583                 return -EINVAL;
1584             }
1585         }
1586         volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX);
1587         volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX);
1588         mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
1589         return 0;
1590     }
1591 
1592     return -ENOSYS;
1593 }
1594 
1595 #ifdef NO_AUDIO_OUT
out_write_for_no_output(struct audio_stream_out * stream,const void * buffer,size_t bytes)1596 static ssize_t out_write_for_no_output(struct audio_stream_out *stream,
1597                                        const void *buffer, size_t bytes)
1598 {
1599     struct stream_out *out = (struct stream_out *)stream;
1600 
1601     /* No Output device supported other than BT for playback.
1602      * Sleep for the amount of buffer duration
1603      */
1604     lock_output_stream(out);
1605     usleep(bytes * 1000000 / audio_stream_frame_size(&out->stream.common) /
1606             out_get_sample_rate(&out->stream.common));
1607     pthread_mutex_unlock(&out->lock);
1608     return bytes;
1609 }
1610 #endif
1611 
out_write(struct audio_stream_out * stream,const void * buffer,size_t bytes)1612 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
1613                          size_t bytes)
1614 {
1615     struct stream_out *out = (struct stream_out *)stream;
1616     struct audio_device *adev = out->dev;
1617     ssize_t ret = 0;
1618 
1619     lock_output_stream(out);
1620     if (out->standby) {
1621         out->standby = false;
1622         pthread_mutex_lock(&adev->lock);
1623         ret = start_output_stream(out);
1624         pthread_mutex_unlock(&adev->lock);
1625         /* ToDo: If use case is compress offload should return 0 */
1626         if (ret != 0) {
1627             out->standby = true;
1628             goto exit;
1629         }
1630         if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD && adev->adm_register_output_stream)
1631             adev->adm_register_output_stream(adev->adm_data, out->handle, out->flags);
1632     }
1633 
1634     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1635         ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes);
1636         if (out->send_new_metadata) {
1637             ALOGVV("send new gapless metadata");
1638             compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
1639             out->send_new_metadata = 0;
1640         }
1641 
1642         ret = compress_write(out->compr, buffer, bytes);
1643         ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret);
1644         if (ret >= 0 && ret < (ssize_t)bytes) {
1645             send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
1646         }
1647         if (!out->playback_started) {
1648             compress_start(out->compr);
1649             out->playback_started = 1;
1650             out->offload_state = OFFLOAD_STATE_PLAYING;
1651         }
1652         pthread_mutex_unlock(&out->lock);
1653         return ret;
1654     } else {
1655         if (out->pcm) {
1656             if (out->muted)
1657                 memset((void *)buffer, 0, bytes);
1658 
1659             ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
1660             if (adev->adm_request_focus)
1661                 adev->adm_request_focus(adev->adm_data, out->handle);
1662 
1663             if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
1664                 ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
1665             }
1666             else
1667                 ret = pcm_write(out->pcm, (void *)buffer, bytes);
1668 
1669             if (ret == 0)
1670                 out->written += bytes / (out->config.channels * sizeof(short));
1671 
1672             if (adev->adm_abandon_focus)
1673                 adev->adm_abandon_focus(adev->adm_data, out->handle);
1674         }
1675     }
1676 
1677 exit:
1678     pthread_mutex_unlock(&out->lock);
1679 
1680     if (ret != 0) {
1681         if (out->pcm)
1682             ALOGE("%s: error %zu - %s", __func__, ret, pcm_get_error(out->pcm));
1683         out_standby(&out->stream.common);
1684         usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
1685                out_get_sample_rate(&out->stream.common));
1686     }
1687     return bytes;
1688 }
1689 
out_get_render_position(const struct audio_stream_out * stream,uint32_t * dsp_frames)1690 static int out_get_render_position(const struct audio_stream_out *stream,
1691                                    uint32_t *dsp_frames)
1692 {
1693     struct stream_out *out = (struct stream_out *)stream;
1694     *dsp_frames = 0;
1695     if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) {
1696         lock_output_stream(out);
1697         if (out->compr != NULL) {
1698             compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
1699                     &out->sample_rate);
1700             ALOGVV("%s rendered frames %d sample_rate %d",
1701                    __func__, *dsp_frames, out->sample_rate);
1702         }
1703         pthread_mutex_unlock(&out->lock);
1704         return 0;
1705     } else
1706         return -EINVAL;
1707 }
1708 
out_add_audio_effect(const struct audio_stream * stream __unused,effect_handle_t effect __unused)1709 static int out_add_audio_effect(const struct audio_stream *stream __unused,
1710                                 effect_handle_t effect __unused)
1711 {
1712     return 0;
1713 }
1714 
out_remove_audio_effect(const struct audio_stream * stream __unused,effect_handle_t effect __unused)1715 static int out_remove_audio_effect(const struct audio_stream *stream __unused,
1716                                    effect_handle_t effect __unused)
1717 {
1718     return 0;
1719 }
1720 
out_get_next_write_timestamp(const struct audio_stream_out * stream __unused,int64_t * timestamp __unused)1721 static int out_get_next_write_timestamp(const struct audio_stream_out *stream __unused,
1722                                         int64_t *timestamp __unused)
1723 {
1724     return -EINVAL;
1725 }
1726 
out_get_presentation_position(const struct audio_stream_out * stream,uint64_t * frames,struct timespec * timestamp)1727 static int out_get_presentation_position(const struct audio_stream_out *stream,
1728                                    uint64_t *frames, struct timespec *timestamp)
1729 {
1730     struct stream_out *out = (struct stream_out *)stream;
1731     int ret = -1;
1732     unsigned long dsp_frames;
1733 
1734     lock_output_stream(out);
1735 
1736     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1737         if (out->compr != NULL) {
1738             compress_get_tstamp(out->compr, &dsp_frames,
1739                     &out->sample_rate);
1740             ALOGVV("%s rendered frames %ld sample_rate %d",
1741                    __func__, dsp_frames, out->sample_rate);
1742             *frames = dsp_frames;
1743             ret = 0;
1744             /* this is the best we can do */
1745             clock_gettime(CLOCK_MONOTONIC, timestamp);
1746         }
1747     } else {
1748         if (out->pcm) {
1749             unsigned int avail;
1750             if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
1751                 size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
1752                 int64_t signed_frames = out->written - kernel_buffer_size + avail;
1753                 // This adjustment accounts for buffering after app processor.
1754                 // It is based on estimated DSP latency per use case, rather than exact.
1755                 signed_frames -=
1756                     (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
1757 
1758                 // It would be unusual for this value to be negative, but check just in case ...
1759                 if (signed_frames >= 0) {
1760                     *frames = signed_frames;
1761                     ret = 0;
1762                 }
1763             }
1764         }
1765     }
1766 
1767     pthread_mutex_unlock(&out->lock);
1768 
1769     return ret;
1770 }
1771 
out_set_callback(struct audio_stream_out * stream,stream_callback_t callback,void * cookie)1772 static int out_set_callback(struct audio_stream_out *stream,
1773             stream_callback_t callback, void *cookie)
1774 {
1775     struct stream_out *out = (struct stream_out *)stream;
1776 
1777     ALOGV("%s", __func__);
1778     lock_output_stream(out);
1779     out->offload_callback = callback;
1780     out->offload_cookie = cookie;
1781     pthread_mutex_unlock(&out->lock);
1782     return 0;
1783 }
1784 
out_pause(struct audio_stream_out * stream)1785 static int out_pause(struct audio_stream_out* stream)
1786 {
1787     struct stream_out *out = (struct stream_out *)stream;
1788     int status = -ENOSYS;
1789     ALOGV("%s", __func__);
1790     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1791         lock_output_stream(out);
1792         if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
1793             status = compress_pause(out->compr);
1794             out->offload_state = OFFLOAD_STATE_PAUSED;
1795         }
1796         pthread_mutex_unlock(&out->lock);
1797     }
1798     return status;
1799 }
1800 
out_resume(struct audio_stream_out * stream)1801 static int out_resume(struct audio_stream_out* stream)
1802 {
1803     struct stream_out *out = (struct stream_out *)stream;
1804     int status = -ENOSYS;
1805     ALOGV("%s", __func__);
1806     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1807         status = 0;
1808         lock_output_stream(out);
1809         if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
1810             status = compress_resume(out->compr);
1811             out->offload_state = OFFLOAD_STATE_PLAYING;
1812         }
1813         pthread_mutex_unlock(&out->lock);
1814     }
1815     return status;
1816 }
1817 
out_drain(struct audio_stream_out * stream,audio_drain_type_t type)1818 static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type )
1819 {
1820     struct stream_out *out = (struct stream_out *)stream;
1821     int status = -ENOSYS;
1822     ALOGV("%s", __func__);
1823     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1824         lock_output_stream(out);
1825         if (type == AUDIO_DRAIN_EARLY_NOTIFY)
1826             status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN);
1827         else
1828             status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN);
1829         pthread_mutex_unlock(&out->lock);
1830     }
1831     return status;
1832 }
1833 
out_flush(struct audio_stream_out * stream)1834 static int out_flush(struct audio_stream_out* stream)
1835 {
1836     struct stream_out *out = (struct stream_out *)stream;
1837     ALOGV("%s", __func__);
1838     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
1839         lock_output_stream(out);
1840         stop_compressed_output_l(out);
1841         pthread_mutex_unlock(&out->lock);
1842         return 0;
1843     }
1844     return -ENOSYS;
1845 }
1846 
1847 /** audio_stream_in implementation **/
in_get_sample_rate(const struct audio_stream * stream)1848 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1849 {
1850     struct stream_in *in = (struct stream_in *)stream;
1851 
1852     return in->config.rate;
1853 }
1854 
in_set_sample_rate(struct audio_stream * stream __unused,uint32_t rate __unused)1855 static int in_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused)
1856 {
1857     return -ENOSYS;
1858 }
1859 
in_get_buffer_size(const struct audio_stream * stream)1860 static size_t in_get_buffer_size(const struct audio_stream *stream)
1861 {
1862     struct stream_in *in = (struct stream_in *)stream;
1863 
1864     return in->config.period_size *
1865                 audio_stream_in_frame_size((const struct audio_stream_in *)stream);
1866 }
1867 
in_get_channels(const struct audio_stream * stream)1868 static uint32_t in_get_channels(const struct audio_stream *stream)
1869 {
1870     struct stream_in *in = (struct stream_in *)stream;
1871 
1872     return in->channel_mask;
1873 }
1874 
in_get_format(const struct audio_stream * stream __unused)1875 static audio_format_t in_get_format(const struct audio_stream *stream __unused)
1876 {
1877     return AUDIO_FORMAT_PCM_16_BIT;
1878 }
1879 
in_set_format(struct audio_stream * stream __unused,audio_format_t format __unused)1880 static int in_set_format(struct audio_stream *stream __unused, audio_format_t format __unused)
1881 {
1882     return -ENOSYS;
1883 }
1884 
in_standby(struct audio_stream * stream)1885 static int in_standby(struct audio_stream *stream)
1886 {
1887     struct stream_in *in = (struct stream_in *)stream;
1888     struct audio_device *adev = in->dev;
1889     int status = 0;
1890     ALOGV("%s: enter", __func__);
1891 
1892     lock_input_stream(in);
1893 
1894     if (!in->standby && in->is_st_session) {
1895         ALOGD("%s: sound trigger pcm stop lab", __func__);
1896         audio_extn_sound_trigger_stop_lab(in);
1897         in->standby = true;
1898     }
1899 
1900     if (!in->standby) {
1901         if (adev->adm_deregister_stream)
1902             adev->adm_deregister_stream(adev->adm_data, in->capture_handle);
1903 
1904         pthread_mutex_lock(&adev->lock);
1905         in->standby = true;
1906         if (in->pcm) {
1907             pcm_close(in->pcm);
1908             in->pcm = NULL;
1909         }
1910         adev->enable_voicerx = false;
1911         platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE );
1912         status = stop_input_stream(in);
1913         pthread_mutex_unlock(&adev->lock);
1914     }
1915     pthread_mutex_unlock(&in->lock);
1916     ALOGV("%s: exit:  status(%d)", __func__, status);
1917     return status;
1918 }
1919 
in_dump(const struct audio_stream * stream __unused,int fd __unused)1920 static int in_dump(const struct audio_stream *stream __unused, int fd __unused)
1921 {
1922     return 0;
1923 }
1924 
in_set_parameters(struct audio_stream * stream,const char * kvpairs)1925 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1926 {
1927     struct stream_in *in = (struct stream_in *)stream;
1928     struct audio_device *adev = in->dev;
1929     struct str_parms *parms;
1930     char *str;
1931     char value[32];
1932     int ret, val = 0;
1933     int status = 0;
1934 
1935     ALOGV("%s: enter: kvpairs=%s", __func__, kvpairs);
1936     parms = str_parms_create_str(kvpairs);
1937 
1938     ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
1939 
1940     lock_input_stream(in);
1941 
1942     pthread_mutex_lock(&adev->lock);
1943     if (ret >= 0) {
1944         val = atoi(value);
1945         /* no audio source uses val == 0 */
1946         if ((in->source != val) && (val != 0)) {
1947             in->source = val;
1948         }
1949     }
1950 
1951     ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
1952 
1953     if (ret >= 0) {
1954         val = atoi(value);
1955         if (((int)in->device != val) && (val != 0)) {
1956             in->device = val;
1957             /* If recording is in progress, change the tx device to new device */
1958             if (!in->standby)
1959                 status = select_devices(adev, in->usecase);
1960         }
1961     }
1962 
1963     pthread_mutex_unlock(&adev->lock);
1964     pthread_mutex_unlock(&in->lock);
1965 
1966     str_parms_destroy(parms);
1967     ALOGV("%s: exit: status(%d)", __func__, status);
1968     return status;
1969 }
1970 
in_get_parameters(const struct audio_stream * stream __unused,const char * keys __unused)1971 static char* in_get_parameters(const struct audio_stream *stream __unused,
1972                                const char *keys __unused)
1973 {
1974     return strdup("");
1975 }
1976 
in_set_gain(struct audio_stream_in * stream __unused,float gain __unused)1977 static int in_set_gain(struct audio_stream_in *stream __unused, float gain __unused)
1978 {
1979     return 0;
1980 }
1981 
in_read(struct audio_stream_in * stream,void * buffer,size_t bytes)1982 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1983                        size_t bytes)
1984 {
1985     struct stream_in *in = (struct stream_in *)stream;
1986     struct audio_device *adev = in->dev;
1987     int i, ret = -1;
1988 
1989     lock_input_stream(in);
1990 
1991     if (in->is_st_session) {
1992         ALOGVV(" %s: reading on st session bytes=%d", __func__, bytes);
1993         /* Read from sound trigger HAL */
1994         audio_extn_sound_trigger_read(in, buffer, bytes);
1995         pthread_mutex_unlock(&in->lock);
1996         return bytes;
1997     }
1998 
1999     if (in->standby) {
2000         pthread_mutex_lock(&adev->lock);
2001         ret = start_input_stream(in);
2002         pthread_mutex_unlock(&adev->lock);
2003         if (ret != 0) {
2004             goto exit;
2005         }
2006         in->standby = 0;
2007         if (adev->adm_register_input_stream)
2008             adev->adm_register_input_stream(adev->adm_data, in->capture_handle, in->flags);
2009     }
2010 
2011     if (adev->adm_request_focus)
2012         adev->adm_request_focus(adev->adm_data, in->capture_handle);
2013 
2014     if (in->pcm) {
2015         if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
2016             ret = pcm_mmap_read(in->pcm, buffer, bytes);
2017         } else
2018             ret = pcm_read(in->pcm, buffer, bytes);
2019     }
2020 
2021     if (adev->adm_abandon_focus)
2022         adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
2023 
2024     /*
2025      * Instead of writing zeroes here, we could trust the hardware
2026      * to always provide zeroes when muted.
2027      * No need to acquire adev->lock to read mic_muted here as we don't change its state.
2028      */
2029     if (ret == 0 && adev->mic_muted && in->usecase != USECASE_AUDIO_RECORD_AFE_PROXY)
2030         memset(buffer, 0, bytes);
2031 
2032 exit:
2033     pthread_mutex_unlock(&in->lock);
2034 
2035     if (ret != 0) {
2036         in_standby(&in->stream.common);
2037         ALOGV("%s: read failed - sleeping for buffer duration", __func__);
2038         usleep(bytes * 1000000 / audio_stream_in_frame_size(stream) /
2039                in_get_sample_rate(&in->stream.common));
2040     }
2041     return bytes;
2042 }
2043 
in_get_input_frames_lost(struct audio_stream_in * stream __unused)2044 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused)
2045 {
2046     return 0;
2047 }
2048 
add_remove_audio_effect(const struct audio_stream * stream,effect_handle_t effect,bool enable)2049 static int add_remove_audio_effect(const struct audio_stream *stream,
2050                                    effect_handle_t effect,
2051                                    bool enable)
2052 {
2053     struct stream_in *in = (struct stream_in *)stream;
2054     struct audio_device *adev = in->dev;
2055     int status = 0;
2056     effect_descriptor_t desc;
2057 
2058     status = (*effect)->get_descriptor(effect, &desc);
2059     if (status != 0)
2060         return status;
2061 
2062     lock_input_stream(in);
2063     pthread_mutex_lock(&in->dev->lock);
2064     if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
2065             adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
2066             in->enable_aec != enable &&
2067             (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) {
2068         in->enable_aec = enable;
2069         if (!enable)
2070             platform_set_echo_reference(in->dev, enable, AUDIO_DEVICE_NONE);
2071         adev->enable_voicerx = enable;
2072         struct audio_usecase *usecase;
2073         struct listnode *node;
2074         list_for_each(node, &adev->usecase_list) {
2075             usecase = node_to_item(node, struct audio_usecase, list);
2076             if (usecase->type == PCM_PLAYBACK) {
2077                 select_devices(adev, usecase->id);
2078             break;
2079             }
2080         }
2081         if (!in->standby)
2082             select_devices(in->dev, in->usecase);
2083     }
2084     if (in->enable_ns != enable &&
2085             (memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0)) {
2086         in->enable_ns = enable;
2087         if (!in->standby)
2088             select_devices(in->dev, in->usecase);
2089     }
2090     pthread_mutex_unlock(&in->dev->lock);
2091     pthread_mutex_unlock(&in->lock);
2092 
2093     return 0;
2094 }
2095 
in_add_audio_effect(const struct audio_stream * stream,effect_handle_t effect)2096 static int in_add_audio_effect(const struct audio_stream *stream,
2097                                effect_handle_t effect)
2098 {
2099     ALOGV("%s: effect %p", __func__, effect);
2100     return add_remove_audio_effect(stream, effect, true);
2101 }
2102 
in_remove_audio_effect(const struct audio_stream * stream,effect_handle_t effect)2103 static int in_remove_audio_effect(const struct audio_stream *stream,
2104                                   effect_handle_t effect)
2105 {
2106     ALOGV("%s: effect %p", __func__, effect);
2107     return add_remove_audio_effect(stream, effect, false);
2108 }
2109 
adev_open_output_stream(struct audio_hw_device * dev,audio_io_handle_t handle,audio_devices_t devices,audio_output_flags_t flags,struct audio_config * config,struct audio_stream_out ** stream_out,const char * address __unused)2110 static int adev_open_output_stream(struct audio_hw_device *dev,
2111                                    audio_io_handle_t handle,
2112                                    audio_devices_t devices,
2113                                    audio_output_flags_t flags,
2114                                    struct audio_config *config,
2115                                    struct audio_stream_out **stream_out,
2116                                    const char *address __unused)
2117 {
2118     struct audio_device *adev = (struct audio_device *)dev;
2119     struct stream_out *out;
2120     int i, ret;
2121 
2122     ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)",
2123           __func__, config->sample_rate, config->channel_mask, devices, flags);
2124     *stream_out = NULL;
2125     out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
2126 
2127     if (devices == AUDIO_DEVICE_NONE)
2128         devices = AUDIO_DEVICE_OUT_SPEAKER;
2129 
2130     out->flags = flags;
2131     out->devices = devices;
2132     out->dev = adev;
2133     out->format = config->format;
2134     out->sample_rate = config->sample_rate;
2135     out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
2136     out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
2137     out->handle = handle;
2138 
2139     /* Init use case and pcm_config */
2140     if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT &&
2141             !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
2142         out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
2143         pthread_mutex_lock(&adev->lock);
2144         ret = read_hdmi_channel_masks(out);
2145         pthread_mutex_unlock(&adev->lock);
2146         if (ret != 0)
2147             goto error_open;
2148 
2149         if (config->sample_rate == 0)
2150             config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
2151         if (config->channel_mask == 0)
2152             config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
2153 
2154         out->channel_mask = config->channel_mask;
2155         out->sample_rate = config->sample_rate;
2156         out->usecase = USECASE_AUDIO_PLAYBACK_MULTI_CH;
2157         out->config = pcm_config_hdmi_multi;
2158         out->config.rate = config->sample_rate;
2159         out->config.channels = audio_channel_count_from_out_mask(out->channel_mask);
2160         out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2);
2161     } else if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
2162         if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version ||
2163             config->offload_info.size != AUDIO_INFO_INITIALIZER.size) {
2164             ALOGE("%s: Unsupported Offload information", __func__);
2165             ret = -EINVAL;
2166             goto error_open;
2167         }
2168         if (!is_supported_format(config->offload_info.format)) {
2169             ALOGE("%s: Unsupported audio format", __func__);
2170             ret = -EINVAL;
2171             goto error_open;
2172         }
2173 
2174         out->compr_config.codec = (struct snd_codec *)
2175                                     calloc(1, sizeof(struct snd_codec));
2176 
2177         out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD;
2178         if (config->offload_info.channel_mask)
2179             out->channel_mask = config->offload_info.channel_mask;
2180         else if (config->channel_mask)
2181             out->channel_mask = config->channel_mask;
2182         out->format = config->offload_info.format;
2183         out->sample_rate = config->offload_info.sample_rate;
2184 
2185         out->stream.set_callback = out_set_callback;
2186         out->stream.pause = out_pause;
2187         out->stream.resume = out_resume;
2188         out->stream.drain = out_drain;
2189         out->stream.flush = out_flush;
2190 
2191         out->compr_config.codec->id =
2192                 get_snd_codec_id(config->offload_info.format);
2193         out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
2194         out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
2195         out->compr_config.codec->sample_rate = config->offload_info.sample_rate;
2196         out->compr_config.codec->bit_rate =
2197                     config->offload_info.bit_rate;
2198         out->compr_config.codec->ch_in =
2199                 audio_channel_count_from_out_mask(config->channel_mask);
2200         out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
2201 
2202         if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
2203             out->non_blocking = 1;
2204 
2205         out->send_new_metadata = 1;
2206         create_offload_callback_thread(out);
2207         ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
2208                 __func__, config->offload_info.version,
2209                 config->offload_info.bit_rate);
2210     } else  if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
2211         if (config->sample_rate == 0)
2212             config->sample_rate = AFE_PROXY_SAMPLING_RATE;
2213         if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
2214                 config->sample_rate != 8000) {
2215             config->sample_rate = AFE_PROXY_SAMPLING_RATE;
2216             ret = -EINVAL;
2217             goto error_open;
2218         }
2219         out->sample_rate = config->sample_rate;
2220         out->config.rate = config->sample_rate;
2221         if (config->format == AUDIO_FORMAT_DEFAULT)
2222             config->format = AUDIO_FORMAT_PCM_16_BIT;
2223         if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
2224             config->format = AUDIO_FORMAT_PCM_16_BIT;
2225             ret = -EINVAL;
2226             goto error_open;
2227         }
2228         out->format = config->format;
2229         out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
2230         out->config = pcm_config_afe_proxy_playback;
2231         adev->voice_tx_output = out;
2232     } else {
2233         if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
2234             out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
2235             out->config = pcm_config_deep_buffer;
2236         } else if (out->flags & AUDIO_OUTPUT_FLAG_TTS) {
2237             out->usecase = USECASE_AUDIO_PLAYBACK_TTS;
2238             out->config = pcm_config_deep_buffer;
2239         } else if (out->flags & AUDIO_OUTPUT_FLAG_RAW) {
2240             out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
2241             out->config = pcm_config_low_latency;
2242         } else {
2243             out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
2244             out->config = pcm_config_low_latency;
2245         }
2246         if (config->format != audio_format_from_pcm_format(out->config.format)) {
2247             if (k_enable_extended_precision
2248                     && pcm_params_format_test(adev->use_case_table[out->usecase],
2249                             pcm_format_from_audio_format(config->format))) {
2250                 out->config.format = pcm_format_from_audio_format(config->format);
2251                 /* out->format already set to config->format */
2252             } else {
2253                 /* deny the externally proposed config format
2254                  * and use the one specified in audio_hw layer configuration.
2255                  * Note: out->format is returned by out->stream.common.get_format()
2256                  * and is used to set config->format in the code several lines below.
2257                  */
2258                 out->format = audio_format_from_pcm_format(out->config.format);
2259             }
2260         }
2261         out->sample_rate = out->config.rate;
2262     }
2263     ALOGV("%s: Usecase(%s) config->format %#x  out->config.format %#x\n",
2264             __func__, use_case_table[out->usecase], config->format, out->config.format);
2265 
2266     if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) {
2267         if (adev->primary_output == NULL)
2268             adev->primary_output = out;
2269         else {
2270             ALOGE("%s: Primary output is already opened", __func__);
2271             ret = -EEXIST;
2272             goto error_open;
2273         }
2274     }
2275 
2276     /* Check if this usecase is already existing */
2277     pthread_mutex_lock(&adev->lock);
2278     if (get_usecase_from_list(adev, out->usecase) != NULL) {
2279         ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase);
2280         pthread_mutex_unlock(&adev->lock);
2281         ret = -EEXIST;
2282         goto error_open;
2283     }
2284     pthread_mutex_unlock(&adev->lock);
2285 
2286     out->stream.common.get_sample_rate = out_get_sample_rate;
2287     out->stream.common.set_sample_rate = out_set_sample_rate;
2288     out->stream.common.get_buffer_size = out_get_buffer_size;
2289     out->stream.common.get_channels = out_get_channels;
2290     out->stream.common.get_format = out_get_format;
2291     out->stream.common.set_format = out_set_format;
2292     out->stream.common.standby = out_standby;
2293     out->stream.common.dump = out_dump;
2294     out->stream.common.set_parameters = out_set_parameters;
2295     out->stream.common.get_parameters = out_get_parameters;
2296     out->stream.common.add_audio_effect = out_add_audio_effect;
2297     out->stream.common.remove_audio_effect = out_remove_audio_effect;
2298     out->stream.get_latency = out_get_latency;
2299     out->stream.set_volume = out_set_volume;
2300 #ifdef NO_AUDIO_OUT
2301     out->stream.write = out_write_for_no_output;
2302 #else
2303     out->stream.write = out_write;
2304 #endif
2305     out->stream.get_render_position = out_get_render_position;
2306     out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
2307     out->stream.get_presentation_position = out_get_presentation_position;
2308 
2309     out->standby = 1;
2310     /* out->muted = false; by calloc() */
2311     /* out->written = 0; by calloc() */
2312 
2313     pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
2314     pthread_mutex_init(&out->pre_lock, (const pthread_mutexattr_t *) NULL);
2315     pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL);
2316 
2317     config->format = out->stream.common.get_format(&out->stream.common);
2318     config->channel_mask = out->stream.common.get_channels(&out->stream.common);
2319     config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common);
2320 
2321     *stream_out = &out->stream;
2322     ALOGV("%s: exit", __func__);
2323     return 0;
2324 
2325 error_open:
2326     free(out);
2327     *stream_out = NULL;
2328     ALOGD("%s: exit: ret %d", __func__, ret);
2329     return ret;
2330 }
2331 
adev_close_output_stream(struct audio_hw_device * dev __unused,struct audio_stream_out * stream)2332 static void adev_close_output_stream(struct audio_hw_device *dev __unused,
2333                                      struct audio_stream_out *stream)
2334 {
2335     struct stream_out *out = (struct stream_out *)stream;
2336     struct audio_device *adev = out->dev;
2337 
2338     ALOGV("%s: enter", __func__);
2339     out_standby(&stream->common);
2340     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
2341         destroy_offload_callback_thread(out);
2342 
2343         if (out->compr_config.codec != NULL)
2344             free(out->compr_config.codec);
2345     }
2346 
2347     if (adev->voice_tx_output == out)
2348         adev->voice_tx_output = NULL;
2349 
2350     pthread_cond_destroy(&out->cond);
2351     pthread_mutex_destroy(&out->lock);
2352     free(stream);
2353     ALOGV("%s: exit", __func__);
2354 }
2355 
adev_set_parameters(struct audio_hw_device * dev,const char * kvpairs)2356 static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
2357 {
2358     struct audio_device *adev = (struct audio_device *)dev;
2359     struct str_parms *parms;
2360     char *str;
2361     char value[32];
2362     int val;
2363     int ret;
2364     int status = 0;
2365 
2366     ALOGD("%s: enter: %s", __func__, kvpairs);
2367 
2368     pthread_mutex_lock(&adev->lock);
2369 
2370     parms = str_parms_create_str(kvpairs);
2371     status = voice_set_parameters(adev, parms);
2372     if (status != 0) {
2373         goto done;
2374     }
2375 
2376     ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value));
2377     if (ret >= 0) {
2378         /* When set to false, HAL should disable EC and NS */
2379         if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
2380             adev->bluetooth_nrec = true;
2381         else
2382             adev->bluetooth_nrec = false;
2383     }
2384 
2385     ret = str_parms_get_str(parms, "screen_state", value, sizeof(value));
2386     if (ret >= 0) {
2387         if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
2388             adev->screen_off = false;
2389         else
2390             adev->screen_off = true;
2391     }
2392 
2393     ret = str_parms_get_int(parms, "rotation", &val);
2394     if (ret >= 0) {
2395         bool reverse_speakers = false;
2396         switch(val) {
2397         // FIXME: note that the code below assumes that the speakers are in the correct placement
2398         //   relative to the user when the device is rotated 90deg from its default rotation. This
2399         //   assumption is device-specific, not platform-specific like this code.
2400         case 270:
2401             reverse_speakers = true;
2402             break;
2403         case 0:
2404         case 90:
2405         case 180:
2406             break;
2407         default:
2408             ALOGE("%s: unexpected rotation of %d", __func__, val);
2409             status = -EINVAL;
2410         }
2411         if (status == 0) {
2412             platform_swap_lr_channels(adev, reverse_speakers);
2413         }
2414     }
2415 
2416     ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value));
2417     if (ret >= 0) {
2418         adev->bt_wb_speech_enabled = !strcmp(value, AUDIO_PARAMETER_VALUE_ON);
2419     }
2420 
2421     audio_extn_hfp_set_parameters(adev, parms);
2422 done:
2423     str_parms_destroy(parms);
2424     pthread_mutex_unlock(&adev->lock);
2425     ALOGV("%s: exit with code(%d)", __func__, status);
2426     return status;
2427 }
2428 
adev_get_parameters(const struct audio_hw_device * dev,const char * keys)2429 static char* adev_get_parameters(const struct audio_hw_device *dev,
2430                                  const char *keys)
2431 {
2432     struct audio_device *adev = (struct audio_device *)dev;
2433     struct str_parms *reply = str_parms_create();
2434     struct str_parms *query = str_parms_create_str(keys);
2435     char *str;
2436 
2437     pthread_mutex_lock(&adev->lock);
2438 
2439     voice_get_parameters(adev, query, reply);
2440     str = str_parms_to_str(reply);
2441     str_parms_destroy(query);
2442     str_parms_destroy(reply);
2443 
2444     pthread_mutex_unlock(&adev->lock);
2445     ALOGV("%s: exit: returns - %s", __func__, str);
2446     return str;
2447 }
2448 
adev_init_check(const struct audio_hw_device * dev __unused)2449 static int adev_init_check(const struct audio_hw_device *dev __unused)
2450 {
2451     return 0;
2452 }
2453 
adev_set_voice_volume(struct audio_hw_device * dev,float volume)2454 static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
2455 {
2456     int ret;
2457     struct audio_device *adev = (struct audio_device *)dev;
2458 
2459     audio_extn_extspk_set_voice_vol(adev->extspk, volume);
2460 
2461     pthread_mutex_lock(&adev->lock);
2462     ret = voice_set_volume(adev, volume);
2463     pthread_mutex_unlock(&adev->lock);
2464 
2465     return ret;
2466 }
2467 
adev_set_master_volume(struct audio_hw_device * dev __unused,float volume __unused)2468 static int adev_set_master_volume(struct audio_hw_device *dev __unused, float volume __unused)
2469 {
2470     return -ENOSYS;
2471 }
2472 
adev_get_master_volume(struct audio_hw_device * dev __unused,float * volume __unused)2473 static int adev_get_master_volume(struct audio_hw_device *dev __unused,
2474                                   float *volume __unused)
2475 {
2476     return -ENOSYS;
2477 }
2478 
adev_set_master_mute(struct audio_hw_device * dev __unused,bool muted __unused)2479 static int adev_set_master_mute(struct audio_hw_device *dev __unused, bool muted __unused)
2480 {
2481     return -ENOSYS;
2482 }
2483 
adev_get_master_mute(struct audio_hw_device * dev __unused,bool * muted __unused)2484 static int adev_get_master_mute(struct audio_hw_device *dev __unused, bool *muted __unused)
2485 {
2486     return -ENOSYS;
2487 }
2488 
adev_set_mode(struct audio_hw_device * dev,audio_mode_t mode)2489 static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
2490 {
2491     struct audio_device *adev = (struct audio_device *)dev;
2492 
2493     pthread_mutex_lock(&adev->lock);
2494     if (adev->mode != mode) {
2495         ALOGD("%s: mode %d\n", __func__, mode);
2496         adev->mode = mode;
2497         if ((mode == AUDIO_MODE_NORMAL || mode == AUDIO_MODE_IN_COMMUNICATION) &&
2498                 voice_is_in_call(adev)) {
2499             voice_stop_call(adev);
2500             adev->current_call_output = NULL;
2501         }
2502     }
2503     pthread_mutex_unlock(&adev->lock);
2504 
2505     audio_extn_extspk_set_mode(adev->extspk, mode);
2506 
2507     return 0;
2508 }
2509 
adev_set_mic_mute(struct audio_hw_device * dev,bool state)2510 static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
2511 {
2512     int ret;
2513     struct audio_device *adev = (struct audio_device *)dev;
2514 
2515     ALOGD("%s: state %d\n", __func__, state);
2516     pthread_mutex_lock(&adev->lock);
2517     ret = voice_set_mic_mute(adev, state);
2518     adev->mic_muted = state;
2519     pthread_mutex_unlock(&adev->lock);
2520 
2521     return ret;
2522 }
2523 
adev_get_mic_mute(const struct audio_hw_device * dev,bool * state)2524 static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
2525 {
2526     *state = voice_get_mic_mute((struct audio_device *)dev);
2527     return 0;
2528 }
2529 
adev_get_input_buffer_size(const struct audio_hw_device * dev __unused,const struct audio_config * config)2530 static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev __unused,
2531                                          const struct audio_config *config)
2532 {
2533     int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
2534 
2535     return get_input_buffer_size(config->sample_rate, config->format, channel_count,
2536             false /* is_low_latency: since we don't know, be conservative */);
2537 }
2538 
adev_open_input_stream(struct audio_hw_device * dev,audio_io_handle_t handle,audio_devices_t devices,struct audio_config * config,struct audio_stream_in ** stream_in,audio_input_flags_t flags,const char * address __unused,audio_source_t source)2539 static int adev_open_input_stream(struct audio_hw_device *dev,
2540                                   audio_io_handle_t handle,
2541                                   audio_devices_t devices,
2542                                   struct audio_config *config,
2543                                   struct audio_stream_in **stream_in,
2544                                   audio_input_flags_t flags,
2545                                   const char *address __unused,
2546                                   audio_source_t source )
2547 {
2548     struct audio_device *adev = (struct audio_device *)dev;
2549     struct stream_in *in;
2550     int ret = 0, buffer_size, frame_size;
2551     int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
2552     bool is_low_latency = false;
2553 
2554     ALOGV("%s: enter", __func__);
2555     *stream_in = NULL;
2556     if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
2557         return -EINVAL;
2558 
2559     in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
2560 
2561     pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
2562     pthread_mutex_init(&in->pre_lock, (const pthread_mutexattr_t *) NULL);
2563 
2564     in->stream.common.get_sample_rate = in_get_sample_rate;
2565     in->stream.common.set_sample_rate = in_set_sample_rate;
2566     in->stream.common.get_buffer_size = in_get_buffer_size;
2567     in->stream.common.get_channels = in_get_channels;
2568     in->stream.common.get_format = in_get_format;
2569     in->stream.common.set_format = in_set_format;
2570     in->stream.common.standby = in_standby;
2571     in->stream.common.dump = in_dump;
2572     in->stream.common.set_parameters = in_set_parameters;
2573     in->stream.common.get_parameters = in_get_parameters;
2574     in->stream.common.add_audio_effect = in_add_audio_effect;
2575     in->stream.common.remove_audio_effect = in_remove_audio_effect;
2576     in->stream.set_gain = in_set_gain;
2577     in->stream.read = in_read;
2578     in->stream.get_input_frames_lost = in_get_input_frames_lost;
2579 
2580     in->device = devices;
2581     in->source = source;
2582     in->dev = adev;
2583     in->standby = 1;
2584     in->channel_mask = config->channel_mask;
2585     in->capture_handle = handle;
2586     in->flags = flags;
2587 
2588     /* Update config params with the requested sample rate and channels */
2589     if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
2590         if (config->sample_rate == 0)
2591             config->sample_rate = AFE_PROXY_SAMPLING_RATE;
2592         if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
2593                 config->sample_rate != 8000) {
2594             config->sample_rate = AFE_PROXY_SAMPLING_RATE;
2595             ret = -EINVAL;
2596             goto err_open;
2597         }
2598         if (config->format == AUDIO_FORMAT_DEFAULT)
2599             config->format = AUDIO_FORMAT_PCM_16_BIT;
2600         if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
2601             config->format = AUDIO_FORMAT_PCM_16_BIT;
2602             ret = -EINVAL;
2603             goto err_open;
2604         }
2605 
2606         in->usecase = USECASE_AUDIO_RECORD_AFE_PROXY;
2607         in->config = pcm_config_afe_proxy_record;
2608     } else {
2609         in->usecase = USECASE_AUDIO_RECORD;
2610         if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
2611                 (flags & AUDIO_INPUT_FLAG_FAST) != 0) {
2612             is_low_latency = true;
2613 #if LOW_LATENCY_CAPTURE_USE_CASE
2614             in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
2615 #endif
2616         }
2617         in->config = pcm_config_audio_capture;
2618 
2619         frame_size = audio_stream_in_frame_size(&in->stream);
2620         buffer_size = get_input_buffer_size(config->sample_rate,
2621                                             config->format,
2622                                             channel_count,
2623                                             is_low_latency);
2624         in->config.period_size = buffer_size / frame_size;
2625     }
2626     in->config.channels = channel_count;
2627     in->config.rate = config->sample_rate;
2628 
2629     /* This stream could be for sound trigger lab,
2630        get sound trigger pcm if present */
2631     audio_extn_sound_trigger_check_and_get_session(in);
2632 
2633     *stream_in = &in->stream;
2634     ALOGV("%s: exit", __func__);
2635     return 0;
2636 
2637 err_open:
2638     free(in);
2639     *stream_in = NULL;
2640     return ret;
2641 }
2642 
adev_close_input_stream(struct audio_hw_device * dev __unused,struct audio_stream_in * stream)2643 static void adev_close_input_stream(struct audio_hw_device *dev __unused,
2644                                     struct audio_stream_in *stream)
2645 {
2646     ALOGV("%s", __func__);
2647 
2648     in_standby(&stream->common);
2649     free(stream);
2650 
2651     return;
2652 }
2653 
adev_dump(const audio_hw_device_t * device __unused,int fd __unused)2654 static int adev_dump(const audio_hw_device_t *device __unused, int fd __unused)
2655 {
2656     return 0;
2657 }
2658 
2659 /* verifies input and output devices and their capabilities.
2660  *
2661  * This verification is required when enabling extended bit-depth or
2662  * sampling rates, as not all qcom products support it.
2663  *
2664  * Suitable for calling only on initialization such as adev_open().
2665  * It fills the audio_device use_case_table[] array.
2666  *
2667  * Has a side-effect that it needs to configure audio routing / devices
2668  * in order to power up the devices and read the device parameters.
2669  * It does not acquire any hw device lock. Should restore the devices
2670  * back to "normal state" upon completion.
2671  */
adev_verify_devices(struct audio_device * adev)2672 static int adev_verify_devices(struct audio_device *adev)
2673 {
2674     /* enumeration is a bit difficult because one really wants to pull
2675      * the use_case, device id, etc from the hidden pcm_device_table[].
2676      * In this case there are the following use cases and device ids.
2677      *
2678      * [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {0, 0},
2679      * [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = {15, 15},
2680      * [USECASE_AUDIO_PLAYBACK_MULTI_CH] = {1, 1},
2681      * [USECASE_AUDIO_PLAYBACK_OFFLOAD] = {9, 9},
2682      * [USECASE_AUDIO_RECORD] = {0, 0},
2683      * [USECASE_AUDIO_RECORD_LOW_LATENCY] = {15, 15},
2684      * [USECASE_VOICE_CALL] = {2, 2},
2685      *
2686      * USECASE_AUDIO_PLAYBACK_OFFLOAD, USECASE_AUDIO_PLAYBACK_MULTI_CH omitted.
2687      * USECASE_VOICE_CALL omitted, but possible for either input or output.
2688      */
2689 
2690     /* should be the usecases enabled in adev_open_input_stream() */
2691     static const int test_in_usecases[] = {
2692              USECASE_AUDIO_RECORD,
2693              USECASE_AUDIO_RECORD_LOW_LATENCY, /* does not appear to be used */
2694     };
2695     /* should be the usecases enabled in adev_open_output_stream()*/
2696     static const int test_out_usecases[] = {
2697             USECASE_AUDIO_PLAYBACK_DEEP_BUFFER,
2698             USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
2699     };
2700     static const usecase_type_t usecase_type_by_dir[] = {
2701             PCM_PLAYBACK,
2702             PCM_CAPTURE,
2703     };
2704     static const unsigned flags_by_dir[] = {
2705             PCM_OUT,
2706             PCM_IN,
2707     };
2708 
2709     size_t i;
2710     unsigned dir;
2711     const unsigned card_id = adev->snd_card;
2712     char info[512]; /* for possible debug info */
2713 
2714     for (dir = 0; dir < 2; ++dir) {
2715         const usecase_type_t usecase_type = usecase_type_by_dir[dir];
2716         const unsigned flags_dir = flags_by_dir[dir];
2717         const size_t testsize =
2718                 dir ? ARRAY_SIZE(test_in_usecases) : ARRAY_SIZE(test_out_usecases);
2719         const int *testcases =
2720                 dir ? test_in_usecases : test_out_usecases;
2721         const audio_devices_t audio_device =
2722                 dir ? AUDIO_DEVICE_IN_BUILTIN_MIC : AUDIO_DEVICE_OUT_SPEAKER;
2723 
2724         for (i = 0; i < testsize; ++i) {
2725             const audio_usecase_t audio_usecase = testcases[i];
2726             int device_id;
2727             snd_device_t snd_device;
2728             struct pcm_params **pparams;
2729             struct stream_out out;
2730             struct stream_in in;
2731             struct audio_usecase uc_info;
2732             int retval;
2733 
2734             pparams = &adev->use_case_table[audio_usecase];
2735             pcm_params_free(*pparams); /* can accept null input */
2736             *pparams = NULL;
2737 
2738             /* find the device ID for the use case (signed, for error) */
2739             device_id = platform_get_pcm_device_id(audio_usecase, usecase_type);
2740             if (device_id < 0)
2741                 continue;
2742 
2743             /* prepare structures for device probing */
2744             memset(&uc_info, 0, sizeof(uc_info));
2745             uc_info.id = audio_usecase;
2746             uc_info.type = usecase_type;
2747             if (dir) {
2748                 adev->active_input = &in;
2749                 memset(&in, 0, sizeof(in));
2750                 in.device = audio_device;
2751                 in.source = AUDIO_SOURCE_VOICE_COMMUNICATION;
2752                 uc_info.stream.in = &in;
2753             }  else {
2754                 adev->active_input = NULL;
2755             }
2756             memset(&out, 0, sizeof(out));
2757             out.devices = audio_device; /* only field needed in select_devices */
2758             uc_info.stream.out = &out;
2759             uc_info.devices = audio_device;
2760             uc_info.in_snd_device = SND_DEVICE_NONE;
2761             uc_info.out_snd_device = SND_DEVICE_NONE;
2762             list_add_tail(&adev->usecase_list, &uc_info.list);
2763 
2764             /* select device - similar to start_(in/out)put_stream() */
2765             retval = select_devices(adev, audio_usecase);
2766             if (retval >= 0) {
2767                 *pparams = pcm_params_get(card_id, device_id, flags_dir);
2768 #if LOG_NDEBUG == 0
2769                 if (*pparams) {
2770                     ALOGV("%s: (%s) card %d  device %d", __func__,
2771                             dir ? "input" : "output", card_id, device_id);
2772                     pcm_params_to_string(*pparams, info, ARRAY_SIZE(info));
2773                     ALOGV(info); /* print parameters */
2774                 } else {
2775                     ALOGV("%s: cannot locate card %d  device %d", __func__, card_id, device_id);
2776                 }
2777 #endif
2778             }
2779 
2780             /* deselect device - similar to stop_(in/out)put_stream() */
2781             /* 1. Get and set stream specific mixer controls */
2782             retval = disable_audio_route(adev, &uc_info);
2783             /* 2. Disable the rx device */
2784             retval = disable_snd_device(adev,
2785                     dir ? uc_info.in_snd_device : uc_info.out_snd_device);
2786             list_remove(&uc_info.list);
2787         }
2788     }
2789     adev->active_input = NULL; /* restore adev state */
2790     return 0;
2791 }
2792 
adev_close(hw_device_t * device)2793 static int adev_close(hw_device_t *device)
2794 {
2795     size_t i;
2796     struct audio_device *adev = (struct audio_device *)device;
2797 
2798     if (!adev)
2799         return 0;
2800 
2801     pthread_mutex_lock(&adev_init_lock);
2802 
2803     if ((--audio_device_ref_count) == 0) {
2804         audio_route_free(adev->audio_route);
2805         free(adev->snd_dev_ref_cnt);
2806         platform_deinit(adev->platform);
2807         audio_extn_extspk_deinit(adev->extspk);
2808         audio_extn_sound_trigger_deinit(adev);
2809         for (i = 0; i < ARRAY_SIZE(adev->use_case_table); ++i) {
2810             pcm_params_free(adev->use_case_table[i]);
2811         }
2812         if (adev->adm_deinit)
2813             adev->adm_deinit(adev->adm_data);
2814         free(device);
2815     }
2816 
2817     pthread_mutex_unlock(&adev_init_lock);
2818 
2819     return 0;
2820 }
2821 
2822 /* This returns 1 if the input parameter looks at all plausible as a low latency period size,
2823  * or 0 otherwise.  A return value of 1 doesn't mean the value is guaranteed to work,
2824  * just that it _might_ work.
2825  */
period_size_is_plausible_for_low_latency(int period_size)2826 static int period_size_is_plausible_for_low_latency(int period_size)
2827 {
2828     switch (period_size) {
2829     case 48:
2830     case 96:
2831     case 144:
2832     case 160:
2833     case 192:
2834     case 240:
2835     case 320:
2836     case 480:
2837         return 1;
2838     default:
2839         return 0;
2840     }
2841 }
2842 
adev_open(const hw_module_t * module,const char * name,hw_device_t ** device)2843 static int adev_open(const hw_module_t *module, const char *name,
2844                      hw_device_t **device)
2845 {
2846     int i, ret;
2847 
2848     ALOGD("%s: enter", __func__);
2849     if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
2850     pthread_mutex_lock(&adev_init_lock);
2851     if (audio_device_ref_count != 0) {
2852         *device = &adev->device.common;
2853         audio_device_ref_count++;
2854         ALOGV("%s: returning existing instance of adev", __func__);
2855         ALOGV("%s: exit", __func__);
2856         pthread_mutex_unlock(&adev_init_lock);
2857         return 0;
2858     }
2859     adev = calloc(1, sizeof(struct audio_device));
2860 
2861     pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
2862 
2863     adev->device.common.tag = HARDWARE_DEVICE_TAG;
2864     adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
2865     adev->device.common.module = (struct hw_module_t *)module;
2866     adev->device.common.close = adev_close;
2867 
2868     adev->device.init_check = adev_init_check;
2869     adev->device.set_voice_volume = adev_set_voice_volume;
2870     adev->device.set_master_volume = adev_set_master_volume;
2871     adev->device.get_master_volume = adev_get_master_volume;
2872     adev->device.set_master_mute = adev_set_master_mute;
2873     adev->device.get_master_mute = adev_get_master_mute;
2874     adev->device.set_mode = adev_set_mode;
2875     adev->device.set_mic_mute = adev_set_mic_mute;
2876     adev->device.get_mic_mute = adev_get_mic_mute;
2877     adev->device.set_parameters = adev_set_parameters;
2878     adev->device.get_parameters = adev_get_parameters;
2879     adev->device.get_input_buffer_size = adev_get_input_buffer_size;
2880     adev->device.open_output_stream = adev_open_output_stream;
2881     adev->device.close_output_stream = adev_close_output_stream;
2882     adev->device.open_input_stream = adev_open_input_stream;
2883     adev->device.close_input_stream = adev_close_input_stream;
2884     adev->device.dump = adev_dump;
2885 
2886     /* Set the default route before the PCM stream is opened */
2887     pthread_mutex_lock(&adev->lock);
2888     adev->mode = AUDIO_MODE_NORMAL;
2889     adev->active_input = NULL;
2890     adev->primary_output = NULL;
2891     adev->bluetooth_nrec = true;
2892     adev->acdb_settings = TTY_MODE_OFF;
2893     /* adev->cur_hdmi_channels = 0;  by calloc() */
2894     adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
2895     voice_init(adev);
2896     list_init(&adev->usecase_list);
2897     pthread_mutex_unlock(&adev->lock);
2898 
2899     /* Loads platform specific libraries dynamically */
2900     adev->platform = platform_init(adev);
2901     if (!adev->platform) {
2902         free(adev->snd_dev_ref_cnt);
2903         free(adev);
2904         ALOGE("%s: Failed to init platform data, aborting.", __func__);
2905         *device = NULL;
2906         pthread_mutex_unlock(&adev_init_lock);
2907         return -EINVAL;
2908     }
2909 
2910     adev->extspk = audio_extn_extspk_init(adev);
2911     audio_extn_sound_trigger_init(adev);
2912 
2913     if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) {
2914         adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW);
2915         if (adev->visualizer_lib == NULL) {
2916             ALOGE("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH);
2917         } else {
2918             ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH);
2919             adev->visualizer_start_output =
2920                         (int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib,
2921                                                         "visualizer_hal_start_output");
2922             adev->visualizer_stop_output =
2923                         (int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib,
2924                                                         "visualizer_hal_stop_output");
2925         }
2926     }
2927 
2928     if (access(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, R_OK) == 0) {
2929         adev->offload_effects_lib = dlopen(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, RTLD_NOW);
2930         if (adev->offload_effects_lib == NULL) {
2931             ALOGE("%s: DLOPEN failed for %s", __func__,
2932                   OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
2933         } else {
2934             ALOGV("%s: DLOPEN successful for %s", __func__,
2935                   OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
2936             adev->offload_effects_start_output =
2937                         (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
2938                                          "offload_effects_bundle_hal_start_output");
2939             adev->offload_effects_stop_output =
2940                         (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
2941                                          "offload_effects_bundle_hal_stop_output");
2942         }
2943     }
2944 
2945     if (access(ADM_LIBRARY_PATH, R_OK) == 0) {
2946         adev->adm_lib = dlopen(ADM_LIBRARY_PATH, RTLD_NOW);
2947         if (adev->adm_lib == NULL) {
2948             ALOGE("%s: DLOPEN failed for %s", __func__, ADM_LIBRARY_PATH);
2949         } else {
2950             ALOGV("%s: DLOPEN successful for %s", __func__, ADM_LIBRARY_PATH);
2951             adev->adm_init = (adm_init_t)
2952                                     dlsym(adev->adm_lib, "adm_init");
2953             adev->adm_deinit = (adm_deinit_t)
2954                                     dlsym(adev->adm_lib, "adm_deinit");
2955             adev->adm_register_input_stream = (adm_register_input_stream_t)
2956                                     dlsym(adev->adm_lib, "adm_register_input_stream");
2957             adev->adm_register_output_stream = (adm_register_output_stream_t)
2958                                     dlsym(adev->adm_lib, "adm_register_output_stream");
2959             adev->adm_deregister_stream = (adm_deregister_stream_t)
2960                                     dlsym(adev->adm_lib, "adm_deregister_stream");
2961             adev->adm_request_focus = (adm_request_focus_t)
2962                                     dlsym(adev->adm_lib, "adm_request_focus");
2963             adev->adm_abandon_focus = (adm_abandon_focus_t)
2964                                     dlsym(adev->adm_lib, "adm_abandon_focus");
2965         }
2966     }
2967 
2968     adev->bt_wb_speech_enabled = false;
2969     adev->enable_voicerx = false;
2970 
2971     *device = &adev->device.common;
2972 
2973     if (k_enable_extended_precision)
2974         adev_verify_devices(adev);
2975 
2976     char value[PROPERTY_VALUE_MAX];
2977     int trial;
2978     if (property_get("audio_hal.period_size", value, NULL) > 0) {
2979         trial = atoi(value);
2980         if (period_size_is_plausible_for_low_latency(trial)) {
2981             pcm_config_low_latency.period_size = trial;
2982             pcm_config_low_latency.start_threshold = trial / 4;
2983             pcm_config_low_latency.avail_min = trial / 4;
2984             configured_low_latency_capture_period_size = trial;
2985         }
2986     }
2987     if (property_get("audio_hal.in_period_size", value, NULL) > 0) {
2988         trial = atoi(value);
2989         if (period_size_is_plausible_for_low_latency(trial)) {
2990             configured_low_latency_capture_period_size = trial;
2991         }
2992     }
2993 
2994     audio_device_ref_count++;
2995     pthread_mutex_unlock(&adev_init_lock);
2996 
2997     if (adev->adm_init)
2998         adev->adm_data = adev->adm_init();
2999 
3000     audio_extn_perf_lock_init();
3001 
3002     ALOGV("%s: exit", __func__);
3003     return 0;
3004 }
3005 
3006 static struct hw_module_methods_t hal_module_methods = {
3007     .open = adev_open,
3008 };
3009 
3010 struct audio_module HAL_MODULE_INFO_SYM = {
3011     .common = {
3012         .tag = HARDWARE_MODULE_TAG,
3013         .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
3014         .hal_api_version = HARDWARE_HAL_API_VERSION,
3015         .id = AUDIO_HARDWARE_MODULE_ID,
3016         .name = "QCOM Audio HAL",
3017         .author = "Code Aurora Forum",
3018         .methods = &hal_module_methods,
3019     },
3020 };
3021