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Christopher N. Hesse297a6362017-01-28 12:40:45 +01001/*
2 * Copyright (C) 2013 The Android Open Source Project
Christopher N. Hesse2f6f8582017-01-28 12:46:15 +01003 * Copyright (C) 2017 Christopher N. Hesse <raymanfx@gmail.com>
Christopher N. Hesse297a6362017-01-28 12:40:45 +01004 *
5 * Licensed under the Apache License, Version 2.0 (the "License");
6 * you may not use this file except in compliance with the License.
7 * You may obtain a copy of the License at
8 *
9 * http://www.apache.org/licenses/LICENSE-2.0
10 *
11 * Unless required by applicable law or agreed to in writing, software
12 * distributed under the License is distributed on an "AS IS" BASIS,
13 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14 * See the License for the specific language governing permissions and
15 * limitations under the License.
16 */
17
Christopher N. Hesse0612a4e2017-01-28 14:05:39 +010018#ifndef SAMSUNG_AUDIO_HW_H
19#define SAMSUNG_AUDIO_HW_H
Christopher N. Hesse297a6362017-01-28 12:40:45 +010020
21#include <cutils/list.h>
22#include <hardware/audio.h>
23
24#include <tinyalsa/asoundlib.h>
25#include <tinycompress/tinycompress.h>
26/* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
27#include <audio_utils/resampler.h>
28#include <audio_route/audio_route.h>
29
30/* Retry for delay in FW loading*/
31#define RETRY_NUMBER 10
32#define RETRY_US 500000
33
34#ifdef __LP64__
35#define OFFLOAD_FX_LIBRARY_PATH "/system/lib64/soundfx/libnvvisualizer.so"
36#else
37#define OFFLOAD_FX_LIBRARY_PATH "/system/lib/soundfx/libnvvisualizer.so"
38#endif
39
Christopher N. Hesse297a6362017-01-28 12:40:45 +010040#ifdef PREPROCESSING_ENABLED
41#include <audio_utils/echo_reference.h>
42#define MAX_PREPROCESSORS 3
43struct effect_info_s {
44 effect_handle_t effect_itfe;
45 size_t num_channel_configs;
46 channel_config_t *channel_configs;
47};
48#endif
49
50#ifdef __LP64__
51#define SOUND_TRIGGER_HAL_LIBRARY_PATH "/system/lib64/hw/sound_trigger.primary.flounder.so"
52#else
53#define SOUND_TRIGGER_HAL_LIBRARY_PATH "/system/lib/hw/sound_trigger.primary.flounder.so"
54#endif
55
Christopher N. Hesse297a6362017-01-28 12:40:45 +010056/* Sound devices specific to the platform
57 * The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound
58 * devices to enable corresponding mixer paths
59 */
60enum {
61 SND_DEVICE_NONE = 0,
62
63 /* Playback devices */
64 SND_DEVICE_MIN,
65 SND_DEVICE_OUT_BEGIN = SND_DEVICE_MIN,
Christopher N. Hesse530cf0d2017-01-31 21:59:54 +010066 SND_DEVICE_OUT_EARPIECE = SND_DEVICE_OUT_BEGIN,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010067 SND_DEVICE_OUT_SPEAKER,
68 SND_DEVICE_OUT_HEADPHONES,
69 SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
Christopher N. Hesse530cf0d2017-01-31 21:59:54 +010070 SND_DEVICE_OUT_VOICE_EARPIECE,
Andreas Schneider59486fa2017-02-06 09:16:39 +010071 SND_DEVICE_OUT_VOICE_EARPIECE_WB,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010072 SND_DEVICE_OUT_VOICE_SPEAKER,
Andreas Schneider59486fa2017-02-06 09:16:39 +010073 SND_DEVICE_OUT_VOICE_SPEAKER_WB,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010074 SND_DEVICE_OUT_VOICE_HEADPHONES,
Andreas Schneider59486fa2017-02-06 09:16:39 +010075 SND_DEVICE_OUT_VOICE_HEADPHONES_WB,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010076 SND_DEVICE_OUT_HDMI,
77 SND_DEVICE_OUT_SPEAKER_AND_HDMI,
78 SND_DEVICE_OUT_BT_SCO,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010079 SND_DEVICE_OUT_END,
80
81 /*
82 * Note: IN_BEGIN should be same as OUT_END because total number of devices
83 * SND_DEVICES_MAX should not exceed MAX_RX + MAX_TX devices.
84 */
85 /* Capture devices */
86 SND_DEVICE_IN_BEGIN = SND_DEVICE_OUT_END,
Christopher N. Hesse530cf0d2017-01-31 21:59:54 +010087 SND_DEVICE_IN_EARPIECE_MIC = SND_DEVICE_IN_BEGIN,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010088 SND_DEVICE_IN_SPEAKER_MIC,
89 SND_DEVICE_IN_HEADSET_MIC,
Christopher N. Hesse530cf0d2017-01-31 21:59:54 +010090 SND_DEVICE_IN_EARPIECE_MIC_AEC,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010091 SND_DEVICE_IN_SPEAKER_MIC_AEC,
92 SND_DEVICE_IN_HEADSET_MIC_AEC,
Andreas Schneider82f32482017-02-06 09:00:48 +010093 SND_DEVICE_IN_VOICE_MIC,
94 SND_DEVICE_IN_VOICE_EARPIECE_MIC,
Andreas Schneider59486fa2017-02-06 09:16:39 +010095 SND_DEVICE_IN_VOICE_EARPIECE_MIC_WB,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010096 SND_DEVICE_IN_VOICE_SPEAKER_MIC,
Andreas Schneider59486fa2017-02-06 09:16:39 +010097 SND_DEVICE_IN_VOICE_SPEAKER_MIC_WB,
Christopher N. Hesse297a6362017-01-28 12:40:45 +010098 SND_DEVICE_IN_VOICE_HEADSET_MIC,
Andreas Schneider59486fa2017-02-06 09:16:39 +010099 SND_DEVICE_IN_VOICE_HEADSET_MIC_WB,
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100100 SND_DEVICE_IN_HDMI_MIC,
101 SND_DEVICE_IN_BT_SCO_MIC,
102 SND_DEVICE_IN_CAMCORDER_MIC,
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100103 SND_DEVICE_IN_VOICE_REC_HEADSET_MIC,
104 SND_DEVICE_IN_VOICE_REC_MIC,
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100105 SND_DEVICE_IN_LOOPBACK_AEC,
106 SND_DEVICE_IN_END,
107
108 SND_DEVICE_MAX = SND_DEVICE_IN_END,
109
110};
111
112
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100113/*
114 * tinyAlsa library interprets period size as number of frames
115 * one frame = channel_count * sizeof (pcm sample)
116 * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
117 * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
118 * We should take care of returning proper size when AudioFlinger queries for
119 * the buffer size of an input/output stream
120 */
121#define PLAYBACK_PERIOD_SIZE 256
122#define PLAYBACK_PERIOD_COUNT 2
123#define PLAYBACK_DEFAULT_CHANNEL_COUNT 2
124#define PLAYBACK_DEFAULT_SAMPLING_RATE 48000
125#define PLAYBACK_START_THRESHOLD(size, count) (((size) * (count)) - 1)
126#define PLAYBACK_STOP_THRESHOLD(size, count) ((size) * ((count) + 2))
127#define PLAYBACK_AVAILABLE_MIN 1
128
129
130#define SCO_PERIOD_SIZE 168
131#define SCO_PERIOD_COUNT 2
132#define SCO_DEFAULT_CHANNEL_COUNT 2
133#define SCO_DEFAULT_SAMPLING_RATE 8000
134#define SCO_START_THRESHOLD 335
135#define SCO_STOP_THRESHOLD 336
136#define SCO_AVAILABLE_MIN 1
137
138#define PLAYBACK_HDMI_MULTI_PERIOD_SIZE 1024
139#define PLAYBACK_HDMI_MULTI_PERIOD_COUNT 4
140#define PLAYBACK_HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
141#define PLAYBACK_HDMI_MULTI_PERIOD_BYTES \
142 (PLAYBACK_HDMI_MULTI_PERIOD_SIZE * PLAYBACK_HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)
143#define PLAYBACK_HDMI_MULTI_START_THRESHOLD 4095
144#define PLAYBACK_HDMI_MULTI_STOP_THRESHOLD 4096
145#define PLAYBACK_HDMI_MULTI_AVAILABLE_MIN 1
146
147#define PLAYBACK_HDMI_DEFAULT_CHANNEL_COUNT 2
148
149#define CAPTURE_PERIOD_SIZE 1024
150#define CAPTURE_PERIOD_SIZE_LOW_LATENCY 256
151#define CAPTURE_PERIOD_COUNT 2
152#define CAPTURE_PERIOD_COUNT_LOW_LATENCY 2
153#define CAPTURE_DEFAULT_CHANNEL_COUNT 2
154#define CAPTURE_DEFAULT_SAMPLING_RATE 48000
155#define CAPTURE_START_THRESHOLD 1
156
157#define COMPRESS_CARD 0
158#define COMPRESS_DEVICE 5
159#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
160#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
161/* ToDo: Check and update a proper value in msec */
162#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
163#define COMPRESS_PLAYBACK_VOLUME_MAX 0x10000 //NV suggested value
164
165#define DEEP_BUFFER_OUTPUT_SAMPLING_RATE 48000
166#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 480
167#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 8
168
169#define MAX_SUPPORTED_CHANNEL_MASKS 2
170
171typedef int snd_device_t;
172
173/* These are the supported use cases by the hardware.
174 * Each usecase is mapped to a specific PCM device.
175 * Refer to pcm_device_table[].
176 */
177typedef enum {
178 USECASE_INVALID = -1,
179 /* Playback usecases */
180 USECASE_AUDIO_PLAYBACK = 0,
181 USECASE_AUDIO_PLAYBACK_MULTI_CH,
182 USECASE_AUDIO_PLAYBACK_OFFLOAD,
183 USECASE_AUDIO_PLAYBACK_DEEP_BUFFER,
184
185 /* Capture usecases */
186 USECASE_AUDIO_CAPTURE,
187 USECASE_AUDIO_CAPTURE_HOTWORD,
188
189 USECASE_VOICE_CALL,
190 AUDIO_USECASE_MAX
191} audio_usecase_t;
192
193#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
194
195/*
196 * tinyAlsa library interprets period size as number of frames
197 * one frame = channel_count * sizeof (pcm sample)
198 * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
199 * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
200 * We should take care of returning proper size when AudioFlinger queries for
201 * the buffer size of an input/output stream
202 */
203
204enum {
205 OFFLOAD_CMD_EXIT, /* exit compress offload thread loop*/
206 OFFLOAD_CMD_DRAIN, /* send a full drain request to DSP */
207 OFFLOAD_CMD_PARTIAL_DRAIN, /* send a partial drain request to DSP */
208 OFFLOAD_CMD_WAIT_FOR_BUFFER, /* wait for buffer released by DSP */
209};
210
211enum {
212 OFFLOAD_STATE_IDLE,
213 OFFLOAD_STATE_PLAYING,
214 OFFLOAD_STATE_PAUSED,
215 OFFLOAD_STATE_PAUSED_FLUSHED,
216};
217
218typedef enum {
219 PCM_PLAYBACK = 0x1,
220 PCM_CAPTURE = 0x2,
221 VOICE_CALL = 0x4,
222 PCM_HOTWORD_STREAMING = 0x8,
223 PCM_CAPTURE_LOW_LATENCY = 0x10,
224} usecase_type_t;
225
226struct offload_cmd {
227 struct listnode node;
228 int cmd;
229 int data[];
230};
231
232struct pcm_device_profile {
233 struct pcm_config config;
234 int card;
235 int id;
236 usecase_type_t type;
237 audio_devices_t devices;
238};
239
240struct pcm_device {
241 struct listnode stream_list_node;
242 struct pcm_device_profile* pcm_profile;
243 struct pcm* pcm;
244 int status;
245 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
246 struct resampler_itfe* resampler;
247 int16_t* res_buffer;
248 size_t res_byte_count;
249 int sound_trigger_handle;
250};
251
252struct stream_out {
253 struct audio_stream_out stream;
254 pthread_mutex_t lock; /* see note below on mutex acquisition order */
255 pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
256 pthread_cond_t cond;
257 struct pcm_config config;
258 struct listnode pcm_dev_list;
259 struct compr_config compr_config;
260 struct compress* compr;
261 int standby;
262 unsigned int sample_rate;
263 audio_channel_mask_t channel_mask;
264 audio_format_t format;
265 audio_devices_t devices;
266 audio_output_flags_t flags;
267 audio_usecase_t usecase;
268 /* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
269 audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
270 bool muted;
271 /* total frames written, not cleared when entering standby */
272 uint64_t written;
273 audio_io_handle_t handle;
274
275 int non_blocking;
276 int offload_state;
277 pthread_cond_t offload_cond;
278 pthread_t offload_thread;
279 struct listnode offload_cmd_list;
280 bool offload_thread_blocked;
281
282 stream_callback_t offload_callback;
283 void* offload_cookie;
284 struct compr_gapless_mdata gapless_mdata;
285 int send_new_metadata;
286
287 struct audio_device* dev;
288
289#ifdef PREPROCESSING_ENABLED
290 struct echo_reference_itfe *echo_reference;
291 // echo_reference_generation indicates if the echo reference used by the output stream is
292 // in sync with the one known by the audio_device. When different from the generation stored
293 // in the audio_device the output stream must release the echo reference.
294 // always modified with audio device and stream mutex locked.
295 int32_t echo_reference_generation;
296#endif
297
298 bool is_fastmixer_affinity_set;
Christopher N. Hessee6b3a3e2017-01-08 00:03:23 +0100299
300 int64_t last_write_time_us;
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100301};
302
303struct stream_in {
304 struct audio_stream_in stream;
305 pthread_mutex_t lock; /* see note below on mutex acquisition order */
306 pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by
307 capture thread */
308 struct pcm_config config;
309 struct listnode pcm_dev_list;
310 int standby;
311 audio_source_t source;
312 audio_devices_t devices;
313 uint32_t main_channels;
314 audio_usecase_t usecase;
315 usecase_type_t usecase_type;
316 bool enable_aec;
317 audio_input_flags_t input_flags;
318
319 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
320 unsigned int requested_rate;
321 struct resampler_itfe* resampler;
322 struct resampler_buffer_provider buf_provider;
323 int read_status;
324 int16_t* read_buf;
325 size_t read_buf_size;
326 size_t read_buf_frames;
327
328 int16_t *proc_buf_in;
329 int16_t *proc_buf_out;
330 size_t proc_buf_size;
331 size_t proc_buf_frames;
332
333#ifdef PREPROCESSING_ENABLED
334 struct echo_reference_itfe *echo_reference;
335 int16_t *ref_buf;
336 size_t ref_buf_size;
337 size_t ref_buf_frames;
338
339#ifdef HW_AEC_LOOPBACK
340 bool hw_echo_reference;
341 int16_t* hw_ref_buf;
342 size_t hw_ref_buf_size;
343#endif
344
345 int num_preprocessors;
346 struct effect_info_s preprocessors[MAX_PREPROCESSORS];
347
348 bool aux_channels_changed;
349 uint32_t aux_channels;
350#endif
351
352 struct audio_device* dev;
353 bool is_fastcapture_affinity_set;
Christopher N. Hessee6b3a3e2017-01-08 00:03:23 +0100354
355 int64_t last_read_time_us;
Christopher N. Hessece6d5af2017-01-12 11:40:30 +0100356 int64_t frames_read; /* total frames read, not cleared when
357 entering standby */
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100358};
359
360struct mixer_card {
361 struct listnode adev_list_node;
362 struct listnode uc_list_node[AUDIO_USECASE_MAX];
363 int card;
364 struct mixer* mixer;
365 struct audio_route* audio_route;
Andreas Schneider759368f2017-02-02 16:11:14 +0100366 struct timespec dsp_poweroff_time;
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100367};
368
369struct audio_usecase {
370 struct listnode adev_list_node;
371 audio_usecase_t id;
372 usecase_type_t type;
373 audio_devices_t devices;
374 snd_device_t out_snd_device;
375 snd_device_t in_snd_device;
376 struct audio_stream* stream;
377 struct listnode mixer_list;
378};
379
Andreas Schneider74ef3a12017-02-02 18:29:12 +0100380struct voice_data {
381 bool in_call;
382 float volume;
383 bool bluetooth_nrec;
Christopher N. Hesse41c9f3d2017-02-02 20:48:56 +0100384 void *session;
Andreas Schneider74ef3a12017-02-02 18:29:12 +0100385};
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100386
387struct audio_device {
388 struct audio_hw_device device;
389 pthread_mutex_t lock; /* see note below on mutex acquisition order */
390 struct listnode mixer_list;
391 audio_mode_t mode;
392 struct stream_in* active_input;
393 struct stream_out* primary_output;
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100394 bool mic_mute;
Andreas Schneiderecd17ce2017-02-09 10:45:21 +0100395 bool screen_off;
Andreas Schneider74ef3a12017-02-02 18:29:12 +0100396
397 struct voice_data voice;
398
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100399 int* snd_dev_ref_cnt;
400 struct listnode usecase_list;
401 bool speaker_lr_swap;
402 unsigned int cur_hdmi_channels;
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100403 bool ns_in_voice_rec;
404
405 void* offload_fx_lib;
406 int (*offload_fx_start_output)(audio_io_handle_t);
407 int (*offload_fx_stop_output)(audio_io_handle_t);
408
409#ifdef PREPROCESSING_ENABLED
410 struct echo_reference_itfe* echo_reference;
411 // echo_reference_generation indicates if the echo reference used by the output stream is
412 // in sync with the one known by the audio_device.
413 // incremented atomically with a memory barrier and audio device mutex locked but WITHOUT
414 // stream mutex locked: the stream will load it atomically with a barrier and re-read it
415 // with audio device mutex if needed
416 volatile int32_t echo_reference_generation;
417#endif
418
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100419 void* sound_trigger_lib;
420 int (*sound_trigger_open_for_streaming)();
421 size_t (*sound_trigger_read_samples)(int, void*, size_t);
422 int (*sound_trigger_close_for_streaming)(int);
423
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100424 pthread_mutex_t lock_inputs; /* see note below on mutex acquisition order */
425};
426
427/*
428 * NOTE: when multiple mutexes have to be acquired, always take the
Christopher N. Hesse2f6f8582017-01-28 12:46:15 +0100429 * lock_inputs, stream_in, stream_out, then audio_device mutex.
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100430 * stream_in mutex must always be before stream_out mutex
Christopher N. Hesse297a6362017-01-28 12:40:45 +0100431 * lock_inputs must be held in order to either close the input stream, or prevent closure.
432 */
433
Christopher N. Hesse0612a4e2017-01-28 14:05:39 +0100434#endif // SAMSUNG_AUDIO_HW_H