Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1 | /* audio_stream_out.c |
| 2 | ** |
| 3 | ** Copyright 2008-2009 Wind River Systems |
| 4 | ** Copyright (c) 2011-2013, The Linux Foundation. All rights reserved |
| 5 | ** Not a Contribution, Apache license notifications and license are retained |
| 6 | ** for attribution purposes only. |
| 7 | ** |
| 8 | ** Licensed under the Apache License, Version 2.0 (the "License"); |
| 9 | ** you may not use this file except in compliance with the License. |
| 10 | ** You may obtain a copy of the License at |
| 11 | ** |
| 12 | ** http://www.apache.org/licenses/LICENSE-2.0 |
| 13 | ** |
| 14 | ** Unless required by applicable law or agreed to in writing, software |
| 15 | ** distributed under the License is distributed on an "AS IS" BASIS, |
| 16 | ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| 17 | ** See the License for the specific language governing permissions and |
| 18 | ** limitations under the License. |
| 19 | */ |
| 20 | |
| 21 | #define LOG_TAG "audio_stream_out" |
| 22 | /*#define LOG_NDEBUG 0*/ |
| 23 | /*#define VERY_VERY_VERBOSE_LOGGING*/ |
| 24 | #ifdef VERY_VERY_VERBOSE_LOGGING |
| 25 | #define ALOGVV ALOGV |
| 26 | #else |
| 27 | #define ALOGVV(a...) do { } while(0) |
| 28 | #endif |
| 29 | |
| 30 | #include <errno.h> |
| 31 | #include <pthread.h> |
| 32 | #include <stdint.h> |
| 33 | #include <sys/time.h> |
| 34 | #include <stdlib.h> |
| 35 | #include <math.h> |
| 36 | #include <dlfcn.h> |
| 37 | #include <sys/resource.h> |
| 38 | #include <sys/prctl.h> |
| 39 | |
| 40 | #include <cutils/log.h> |
| 41 | #include <cutils/str_parms.h> |
| 42 | #include <cutils/properties.h> |
| 43 | #include <cutils/atomic.h> |
| 44 | #include <cutils/sched_policy.h> |
| 45 | |
| 46 | #include <system/thread_defs.h> |
| 47 | #include "audio_hw.h" |
| 48 | #include "platform_api.h" |
| 49 | #include <platform.h> |
| 50 | |
| 51 | #include "sound/compress_params.h" |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 52 | #include "audio_bitstream_sm.h" |
| 53 | |
| 54 | //TODO: enable sw_decode if required |
| 55 | #define USE_SWDECODE 0 |
| 56 | |
| 57 | #if USE_SWDECODE |
| 58 | #include "SoftMS11.h" |
| 59 | #endif |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 60 | |
| 61 | #define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024) |
| 62 | #define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4 |
| 63 | /* ToDo: Check and update a proper value in msec */ |
| 64 | #define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96 |
| 65 | #define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000 |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 66 | #define STRING_LENGTH_OF_INTEGER 12 |
| 67 | |
| 68 | static int send_offload_cmd_l(struct stream_out* out, int command); |
| 69 | static int get_snd_codec_id(audio_format_t format); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 70 | |
| 71 | struct pcm_config pcm_config_deep_buffer = { |
| 72 | .channels = 2, |
| 73 | .rate = DEFAULT_OUTPUT_SAMPLING_RATE, |
| 74 | .period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE, |
| 75 | .period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT, |
| 76 | .format = PCM_FORMAT_S16_LE, |
| 77 | .start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4, |
| 78 | .stop_threshold = INT_MAX, |
| 79 | .avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4, |
| 80 | }; |
| 81 | |
| 82 | struct pcm_config pcm_config_low_latency = { |
| 83 | .channels = 2, |
| 84 | .rate = DEFAULT_OUTPUT_SAMPLING_RATE, |
| 85 | .period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE, |
| 86 | .period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT, |
| 87 | .format = PCM_FORMAT_S16_LE, |
| 88 | .start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4, |
| 89 | .stop_threshold = INT_MAX, |
| 90 | .avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4, |
| 91 | }; |
| 92 | |
| 93 | struct pcm_config pcm_config_hdmi_multi = { |
| 94 | .channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */ |
| 95 | .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */ |
| 96 | .period_size = HDMI_MULTI_PERIOD_SIZE, |
| 97 | .period_count = HDMI_MULTI_PERIOD_COUNT, |
| 98 | .format = PCM_FORMAT_S16_LE, |
| 99 | .start_threshold = 0, |
| 100 | .stop_threshold = INT_MAX, |
| 101 | .avail_min = 0, |
| 102 | }; |
| 103 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 104 | inline int nextMultiple(int n, int m) { |
| 105 | return ((n/m) + 1) * m; |
| 106 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 107 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 108 | /******************************************************************************* |
| 109 | Description: check for MS11 supported formats |
| 110 | *******************************************************************************/ |
| 111 | //TODO: enable sw_decode if required |
| 112 | #if USE_SWDECODE |
| 113 | int is_ms11_supported_fromats(int format) |
| 114 | { |
| 115 | ALOGVV("is_ms11_supported_fromats"); |
| 116 | int main_format = format & AUDIO_FORMAT_MAIN_MASK; |
| 117 | if(((main_format == AUDIO_FORMAT_AAC) || |
| 118 | (main_format == AUDIO_FORMAT_HE_AAC_V1) || |
| 119 | (main_format == AUDIO_FORMAT_HE_AAC_V2) || |
| 120 | (main_format == AUDIO_FORMAT_AC3) || |
| 121 | (main_format == AUDIO_FORMAT_AC3_PLUS) || |
| 122 | (main_format == AUDIO_FORMAT_EAC3))) { |
| 123 | return 1; |
| 124 | } else { |
| 125 | return 0; |
| 126 | } |
| 127 | } |
| 128 | #endif |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 129 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 130 | /******************************************************************************* |
| 131 | Description: check if ac3 can played as pass through without MS11 decoder |
| 132 | *******************************************************************************/ |
| 133 | //TODO: enable sw_decode if required |
| 134 | #if USE_SWDECODE |
| 135 | int can_ac3_passthrough_without_ms11(struct stream_out *out, int format) |
| 136 | { |
| 137 | ALOGVV("can_ac3_passthrough_without_ms11"); |
| 138 | int main_format = format & AUDIO_FORMAT_MAIN_MASK; |
| 139 | if(main_format == AUDIO_FORMAT_AC3) { |
| 140 | if(((out->hdmi_format == COMPRESSED) || |
| 141 | (out->hdmi_format == AUTO_DEVICE_FORMAT) || |
| 142 | (out->hdmi_format == COMPRESSED_CONVERT_EAC3_AC3) || |
| 143 | (out->hdmi_format == COMPRESSED_CONVERT_ANY_AC3)) && |
| 144 | ((out->spdif_format == COMPRESSED) || |
| 145 | (out->spdif_format == AUTO_DEVICE_FORMAT) || |
| 146 | (out->spdif_format == COMPRESSED_CONVERT_EAC3_AC3) || |
| 147 | (out->spdif_format == COMPRESSED_CONVERT_ANY_AC3))) { |
| 148 | return 1; |
| 149 | } |
| 150 | } |
| 151 | return 0; |
| 152 | } |
| 153 | #endif |
| 154 | |
| 155 | /******************************************************************************* |
| 156 | Description: get levels of buffering, interms of number of buffers |
| 157 | *******************************************************************************/ |
| 158 | int get_buffering_factor(struct stream_out *out) |
| 159 | { |
| 160 | ALOGVV("get_buffering_factor"); |
| 161 | if((out->format == AUDIO_FORMAT_PCM_16_BIT) || |
| 162 | (out->format == AUDIO_FORMAT_PCM_24_BIT)) |
| 163 | return 1; |
| 164 | else |
| 165 | return NUM_OF_PERIODS; |
| 166 | } |
| 167 | |
| 168 | /******************************************************************************* |
| 169 | Description: get the buffer size based on format and device format type |
| 170 | *******************************************************************************/ |
| 171 | void get_fragment_size_and_format(struct stream_out *out, int routeFormat, int *fragment_size, |
| 172 | int *fragment_count, int *format) |
| 173 | { |
| 174 | ALOGV("get_fragment_size_and_format"); |
| 175 | |
| 176 | int frame_size = 0; |
| 177 | *format = out->format; |
| 178 | *fragment_count = NUM_OF_PERIODS; |
| 179 | switch(out->format) { |
| 180 | case AUDIO_FORMAT_PCM_16_BIT: |
| 181 | frame_size = PCM_16_BITS_PER_SAMPLE * out->channels; |
| 182 | /*TODO: do we need below calculation */ |
| 183 | *fragment_size = nextMultiple(((frame_size * out->sample_rate * TIME_PER_BUFFER)/1000) + MIN_SIZE_FOR_METADATA , frame_size * 32); |
| 184 | break; |
| 185 | case AUDIO_FORMAT_PCM_24_BIT: |
| 186 | frame_size = PCM_24_BITS_PER_SAMPLE * out->channels; |
| 187 | *fragment_size = nextMultiple(((frame_size * out->sample_rate * TIME_PER_BUFFER)/1000) + MIN_SIZE_FOR_METADATA, frame_size * 32); |
| 188 | break; |
| 189 | case AUDIO_FORMAT_AAC: |
| 190 | case AUDIO_FORMAT_HE_AAC_V1: |
| 191 | case AUDIO_FORMAT_HE_AAC_V2: |
| 192 | case AUDIO_FORMAT_AAC_ADIF: |
| 193 | case AUDIO_FORMAT_AC3: |
| 194 | case AUDIO_FORMAT_AC3_DM: |
| 195 | case AUDIO_FORMAT_EAC3: |
| 196 | case AUDIO_FORMAT_EAC3_DM: |
| 197 | if(routeFormat == ROUTE_UNCOMPRESSED_MCH) { |
| 198 | frame_size = PCM_16_BITS_PER_SAMPLE * out->channels; |
| 199 | *fragment_size = nextMultiple(AC3_PERIOD_SIZE * out->channels + MIN_SIZE_FOR_METADATA, frame_size * 32); |
| 200 | *format = AUDIO_FORMAT_PCM_16_BIT; |
| 201 | } else if(routeFormat == ROUTE_UNCOMPRESSED) { |
| 202 | frame_size = PCM_16_BITS_PER_SAMPLE * 2; |
| 203 | *fragment_size = nextMultiple(AC3_PERIOD_SIZE * 2 + MIN_SIZE_FOR_METADATA, frame_size * 32); |
| 204 | *format = AUDIO_FORMAT_PCM_16_BIT; |
| 205 | } else { |
| 206 | *fragment_size = PERIOD_SIZE_COMPR; |
| 207 | } |
| 208 | break; |
| 209 | case AUDIO_FORMAT_DTS: |
| 210 | case AUDIO_FORMAT_DTS_LBR: |
| 211 | case AUDIO_FORMAT_MP3: |
| 212 | case AUDIO_FORMAT_WMA: |
| 213 | case AUDIO_FORMAT_WMA_PRO: |
| 214 | case AUDIO_FORMAT_MP2: |
| 215 | *fragment_size = PERIOD_SIZE_COMPR; |
| 216 | break; |
| 217 | default: |
| 218 | *fragment_size = PERIOD_SIZE_COMPR; |
| 219 | *format = out->format; |
| 220 | } |
| 221 | |
| 222 | /*TODO: remove this if fragement count needs to be decided based on the format*/ |
| 223 | *fragment_count = COMPRESS_OFFLOAD_NUM_FRAGMENTS; |
| 224 | fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; |
| 225 | |
| 226 | ALOGV("fragment_size: %d, fragment_count: %d", *fragment_size, *fragment_count); |
| 227 | return; |
| 228 | } |
| 229 | |
| 230 | /******************************************************************************* |
| 231 | Description: buffer length updated to player |
| 232 | *******************************************************************************/ |
| 233 | int get_buffer_length(struct stream_out *out) |
| 234 | { |
| 235 | /* TODO: Do we need below */ |
| 236 | ALOGV("get_buffer_length"); |
| 237 | int buffer_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; |
| 238 | switch(out->format) { |
| 239 | case AUDIO_FORMAT_PCM_16_BIT: |
| 240 | buffer_size = ((PCM_16_BITS_PER_SAMPLE * out->channels * out->sample_rate * TIME_PER_BUFFER)/1000); |
| 241 | break; |
| 242 | case AUDIO_FORMAT_PCM_24_BIT: |
| 243 | buffer_size = ((PCM_24_BITS_PER_SAMPLE * out->channels * out->sample_rate * TIME_PER_BUFFER)/1000); |
| 244 | break; |
| 245 | case AUDIO_FORMAT_AAC: |
| 246 | case AUDIO_FORMAT_HE_AAC_V1: |
| 247 | case AUDIO_FORMAT_HE_AAC_V2: |
| 248 | case AUDIO_FORMAT_AAC_ADIF: |
| 249 | buffer_size = AAC_BLOCK_PER_CHANNEL_MS11 * out->channels; |
| 250 | break; |
| 251 | case AUDIO_FORMAT_AC3: |
| 252 | case AUDIO_FORMAT_AC3_DM: |
| 253 | case AUDIO_FORMAT_EAC3: |
| 254 | case AUDIO_FORMAT_EAC3_DM: |
| 255 | buffer_size = AC3_BUFFER_SIZE; |
| 256 | break; |
| 257 | case AUDIO_FORMAT_DTS: |
| 258 | case AUDIO_FORMAT_DTS_LBR: |
| 259 | case AUDIO_FORMAT_MP3: |
| 260 | case AUDIO_FORMAT_WMA: |
| 261 | case AUDIO_FORMAT_WMA_PRO: |
| 262 | case AUDIO_FORMAT_MP2: |
| 263 | buffer_size = COMPR_INPUT_BUFFER_SIZE; |
| 264 | break; |
| 265 | default: |
| 266 | buffer_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; |
| 267 | } |
| 268 | |
| 269 | /*TODO: remove this if fragement count needs to be decided based on the format*/ |
| 270 | buffer_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; |
| 271 | return buffer_size; |
| 272 | } |
| 273 | |
| 274 | /* TODO: Uncomment this when enabling A2DP |
| 275 | TODO: add support for the 24 bit playback*/ |
| 276 | #if 0 |
| 277 | /******************************************************************************* |
| 278 | Description: fix up devices for supporting A2DP playback |
| 279 | *******************************************************************************/ |
| 280 | void fixUpDevicesForA2DPPlayback(struct stream_out *out) |
| 281 | { |
| 282 | ALOGVV("fixUpDevicesForA2DPPlayback"); |
| 283 | if(out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) { |
| 284 | out->route_audio_to_a2dp = 1; |
| 285 | out->devices &= ~AUDIO_DEVICE_OUT_ALL_A2DP; |
| 286 | //TODO: add spdif and proxy |
| 287 | //out->devices &= ~AUDIO_DEVICE_OUT_SPDIF; |
| 288 | //out->devices |= AudioSystem::DEVICE_OUT_PROXY; |
| 289 | } |
| 290 | } |
| 291 | #endif |
| 292 | |
| 293 | /******************************************************************************* |
| 294 | Description: open temp buffer so that meta data mode can be updated properly |
| 295 | *******************************************************************************/ |
| 296 | int open_temp_buf_for_metadata(struct stream_out *out) |
| 297 | { |
| 298 | ALOGV("%s", __func__); |
| 299 | if (out->write_temp_buf == NULL) { |
| 300 | /*Max Period size which is exposed by the compr driver |
| 301 | The value needs to be modified when the period size is modified*/ |
| 302 | out->write_temp_buf = (char *) malloc(PLAYBACK_MAX_PERIOD_SIZE); |
| 303 | if (out->write_temp_buf == NULL) { |
| 304 | ALOGE("Memory allocation of temp buffer to write pcm to driver failed"); |
| 305 | return -EINVAL; |
| 306 | } |
| 307 | } |
| 308 | return 0; |
| 309 | } |
| 310 | |
| 311 | /******************************************************************************* |
| 312 | Description: get index of handle based on device handle device |
| 313 | *******************************************************************************/ |
| 314 | struct alsa_handle * get_handle_based_on_devices(struct stream_out *out, int handleDevices) |
| 315 | { |
| 316 | ALOGVV("get_handle_based_on_devices"); |
| 317 | struct listnode *node; |
| 318 | struct alsa_handle *handle = NULL; |
| 319 | |
| 320 | list_for_each(node, &out->session_list) { |
| 321 | handle = node_to_item(node, struct alsa_handle, list); |
| 322 | if(handle->devices & handleDevices) |
| 323 | break; |
| 324 | } |
| 325 | return handle; |
| 326 | } |
| 327 | |
| 328 | void reset_out_parameters(struct stream_out *out) { |
| 329 | |
| 330 | out->hdmi_format = UNCOMPRESSED; |
| 331 | out->spdif_format = UNCOMPRESSED; |
| 332 | out->decoder_type = UNCOMPRESSED ; |
| 333 | out->dec_conf_set = false; |
| 334 | out->min_bytes_req_to_dec = 0; |
| 335 | out->is_m11_file_mode = false; |
| 336 | out->dec_conf_bufLength = 0; |
| 337 | out->first_bitstrm_buf = false; |
| 338 | out->open_dec_route = false; |
| 339 | out->dec_format_devices = AUDIO_DEVICE_NONE; |
| 340 | out->open_dec_mch_route = false; |
| 341 | out->dec_mch_format_devices =AUDIO_DEVICE_NONE; |
| 342 | out->open_passt_route = false; |
| 343 | out->passt_format_devices = AUDIO_DEVICE_NONE; |
| 344 | out->sw_open_trans_route = false; |
| 345 | out->sw_trans_format_devices = AUDIO_DEVICE_NONE; |
| 346 | out->hw_open_trans_route =false ; |
| 347 | out->hw_trans_format_devices = AUDIO_DEVICE_NONE; |
| 348 | out->channel_status_set = false; |
| 349 | out->route_audio_to_a2dp = false; |
| 350 | out->is_ms11_file_playback_mode = false; |
| 351 | out->write_temp_buf = NULL; |
| 352 | return; |
| 353 | } |
| 354 | |
| 355 | struct alsa_handle *get_alsa_handle() { |
| 356 | |
| 357 | struct alsa_handle *handle; |
| 358 | handle = (struct alsa_handle *)calloc(1, sizeof(struct alsa_handle)); |
| 359 | if(handle == NULL) { |
| 360 | ALOGE("%s calloc failed for handle", __func__); |
| 361 | } else { |
| 362 | ALOGE("%s handle is 0x%x", __func__,(uint32_t)handle); |
| 363 | } |
| 364 | |
| 365 | return handle; |
| 366 | } |
| 367 | |
| 368 | void free_alsa_handle(struct alsa_handle *handle) { |
| 369 | |
| 370 | if(handle == NULL) { |
| 371 | ALOGE("%s Invalid handle", __func__); |
| 372 | } |
| 373 | free(handle); |
| 374 | |
| 375 | return; |
| 376 | } |
| 377 | |
| 378 | |
| 379 | struct alsa_handle *get_handle_by_route_format(struct stream_out *out, |
| 380 | int route_format) |
| 381 | { |
| 382 | struct listnode *node; |
| 383 | struct alsa_handle *handle = NULL; |
| 384 | ALOGV("%s",__func__); |
| 385 | list_for_each(node, &out->session_list) { |
| 386 | handle = node_to_item(node, struct alsa_handle, list); |
| 387 | if(handle->route_format & route_format) { |
| 388 | ALOGV("%s found handle %x",__func__,(uint32_t)handle); |
| 389 | break; |
| 390 | } |
| 391 | } |
| 392 | |
| 393 | return handle; |
| 394 | } |
| 395 | |
| 396 | /******************************************************************************* |
| 397 | Description: get the format index |
| 398 | *******************************************************************************/ |
| 399 | int get_format_index(int format) |
| 400 | { |
| 401 | ALOGVV("get_format_index"); |
| 402 | int idx = 0,i; |
| 403 | for(i=0; i<NUM_SUPPORTED_CODECS; i++) { |
| 404 | if(format == format_index[i][0]) { |
| 405 | idx = format_index[i][1]; |
| 406 | break; |
| 407 | } |
| 408 | } |
| 409 | return idx; |
| 410 | } |
| 411 | |
| 412 | int get_compress_available_space(struct alsa_handle *handle) |
| 413 | { |
| 414 | uint32_t ret; |
| 415 | size_t avail = 0; |
| 416 | struct timespec tstamp; |
| 417 | ret = compress_get_hpointer(handle->compr,&avail, &tstamp); |
| 418 | if(ret!=0) { |
| 419 | ALOGE("cannot get available space\n"); |
| 420 | } else |
| 421 | ret = (int)avail; |
| 422 | return ret; |
| 423 | } |
| 424 | |
| 425 | |
| 426 | /******************************************************************************* |
| 427 | Description: validate if the decoder requires configuration to be set as first |
| 428 | buffer |
| 429 | *******************************************************************************/ |
| 430 | int is_decoder_config_required(struct stream_out *out) |
| 431 | { |
| 432 | ALOGVV("is_decoder_config_required"); |
| 433 | int main_format = out->format & AUDIO_FORMAT_MAIN_MASK; |
| 434 | uint32_t i; |
| 435 | if(!out->is_ms11_file_playback_mode) |
| 436 | return 0; |
| 437 | for(i=0; i<sizeof(decodersRequireConfig)/sizeof(int); i++) |
| 438 | if(main_format == decodersRequireConfig[i]) |
| 439 | return 1; |
| 440 | return 0; |
| 441 | } |
| 442 | |
| 443 | /******************************************************************************* |
| 444 | Description: query if input buffering mode require |
| 445 | *******************************************************************************/ |
| 446 | int is_input_buffering_mode_reqd(struct stream_out *out) |
| 447 | { |
| 448 | ALOGVV("is_input_buffering_mode_reqd"); |
| 449 | if((out->decoder_type == SW_PASSTHROUGH) || |
| 450 | (out->decoder_type == DSP_PASSTHROUGH)) |
| 451 | return 1; |
| 452 | else |
| 453 | return 0; |
| 454 | } |
| 455 | |
| 456 | |
| 457 | |
| 458 | /******************************************************************************* |
| 459 | Description: update use case and routing flags |
| 460 | *******************************************************************************/ |
| 461 | void update_decode_type_and_routing_states(struct stream_out *out) |
| 462 | { |
| 463 | ALOGV("%s", __func__); |
| 464 | |
| 465 | int format_index = get_format_index(out->format); |
| 466 | int decodeType, idx; |
| 467 | |
| 468 | out->open_dec_route = false; |
| 469 | out->open_dec_mch_route = false; |
| 470 | out->open_passt_route = false; |
| 471 | out->sw_open_trans_route = false; |
| 472 | out->hw_open_trans_route = false; |
| 473 | out->dec_format_devices = out->devices; |
| 474 | out->dec_mch_format_devices = AUDIO_DEVICE_NONE; |
| 475 | out->passt_format_devices = AUDIO_DEVICE_NONE; |
| 476 | out->sw_trans_format_devices = AUDIO_DEVICE_NONE; |
| 477 | out->hw_trans_format_devices = AUDIO_DEVICE_NONE; |
| 478 | out->decoder_type = 0; |
| 479 | |
| 480 | //TODO: enable sw_decode if required |
| 481 | #if USE_SWDECODE |
| 482 | if(is_ms11_supported_fromats(out->format)) |
| 483 | out->use_ms11_decoder = true; |
| 484 | #endif |
| 485 | |
| 486 | ALOGV("format_index: %d devices %x", format_index,out->devices); |
| 487 | if(out->devices & AUDIO_DEVICE_OUT_SPDIF) { |
| 488 | decodeType = usecase_docode_hdmi_spdif[NUM_STATES_FOR_EACH_DEVICE_FMT*format_index] |
| 489 | [out->spdif_format]; |
| 490 | ALOGV("SPDIF: decoderType: %d", decodeType); |
| 491 | out->decoder_type = decodeType; |
| 492 | for(idx=0; idx<NUM_DECODE_PATH; idx++) { |
| 493 | if(route_to_driver[idx][DECODER_TYPE_IDX] == decodeType) { |
| 494 | switch(route_to_driver[idx][ROUTE_FORMAT_IDX]) { |
| 495 | case ROUTE_UNCOMPRESSED: |
| 496 | ALOGVV("ROUTE_UNCOMPRESSED"); |
| 497 | ALOGVV("SPDIF opened with stereo decode"); |
| 498 | out->open_dec_route = true; |
| 499 | break; |
| 500 | case ROUTE_UNCOMPRESSED_MCH: |
| 501 | ALOGVV("ROUTE_UNCOMPRESSED_MCH"); |
| 502 | ALOGVV("SPDIF opened with multichannel decode"); |
| 503 | out->open_dec_mch_route = true; |
| 504 | out->dec_format_devices &= ~AUDIO_DEVICE_OUT_SPDIF; |
| 505 | out->dec_mch_format_devices |= AUDIO_DEVICE_OUT_SPDIF; |
| 506 | break; |
| 507 | case ROUTE_COMPRESSED: |
| 508 | ALOGVV("ROUTE_COMPRESSED"); |
| 509 | out->open_passt_route = true; |
| 510 | out->dec_format_devices &= ~AUDIO_DEVICE_OUT_SPDIF; |
| 511 | out->passt_format_devices = AUDIO_DEVICE_OUT_SPDIF; |
| 512 | break; |
| 513 | case ROUTE_DSP_TRANSCODED_COMPRESSED: |
| 514 | ALOGVV("ROUTE_DSP_TRANSCODED_COMPRESSED"); |
| 515 | out->hw_open_trans_route = true; |
| 516 | out->hw_trans_format_devices = AUDIO_DEVICE_OUT_SPDIF; |
| 517 | break; |
| 518 | case ROUTE_SW_TRANSCODED_COMPRESSED: |
| 519 | ALOGVV("ROUTE_SW_TRANSCODED_COMPRESSED"); |
| 520 | out->sw_open_trans_route = true; |
| 521 | out->dec_format_devices &= ~AUDIO_DEVICE_OUT_SPDIF; |
| 522 | out->sw_trans_format_devices = AUDIO_DEVICE_OUT_SPDIF; |
| 523 | break; |
| 524 | default: |
| 525 | ALOGW("INVALID ROUTE for SPDIF, decoderType %d, routeFormat %d", |
| 526 | decodeType, route_to_driver[idx][ROUTE_FORMAT_IDX]); |
| 527 | break; |
| 528 | } |
| 529 | } |
| 530 | } |
| 531 | } |
| 532 | if(out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { |
| 533 | decodeType = usecase_docode_hdmi_spdif[NUM_STATES_FOR_EACH_DEVICE_FMT*format_index] |
| 534 | [out->hdmi_format]; |
| 535 | ALOGV("HDMI: decoderType: %d", decodeType); |
| 536 | out->decoder_type |= decodeType; |
| 537 | for(idx=0; idx<NUM_DECODE_PATH; idx++) { |
| 538 | if(route_to_driver[idx][DECODER_TYPE_IDX] == decodeType) { |
| 539 | switch(route_to_driver[idx][ROUTE_FORMAT_IDX]) { |
| 540 | case ROUTE_UNCOMPRESSED: |
| 541 | ALOGVV("ROUTE_UNCOMPRESSED"); |
| 542 | ALOGVV("HDMI opened with stereo decode"); |
| 543 | out->open_dec_route = true; |
| 544 | break; |
| 545 | case ROUTE_UNCOMPRESSED_MCH: |
| 546 | ALOGVV("ROUTE_UNCOMPRESSED_MCH"); |
| 547 | ALOGVV("HDMI opened with multichannel decode"); |
| 548 | out->open_dec_mch_route = true; |
| 549 | out->dec_format_devices &= ~AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| 550 | out->dec_mch_format_devices |= AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| 551 | break; |
| 552 | case ROUTE_COMPRESSED: |
| 553 | ALOGVV("ROUTE_COMPRESSED"); |
| 554 | out->open_passt_route = true; |
| 555 | out->dec_format_devices &= ~AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| 556 | out->passt_format_devices |= AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| 557 | break; |
| 558 | case ROUTE_DSP_TRANSCODED_COMPRESSED: |
| 559 | ALOGVV("ROUTE_DSP_TRANSCODED_COMPRESSED"); |
| 560 | out->hw_open_trans_route = true; |
| 561 | out->hw_trans_format_devices |= AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| 562 | break; |
| 563 | case ROUTE_SW_TRANSCODED_COMPRESSED: |
| 564 | ALOGVV("ROUTE_SW_TRANSCODED_COMPRESSED"); |
| 565 | out->sw_open_trans_route = true; |
| 566 | out->dec_format_devices &= ~AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| 567 | out->sw_trans_format_devices |= AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| 568 | break; |
| 569 | default: |
| 570 | ALOGW("INVALID ROUTE for HDMI, decoderType %d, routeFormat %d", |
| 571 | decodeType, route_to_driver[idx][ROUTE_FORMAT_IDX]); |
| 572 | break; |
| 573 | } |
| 574 | } |
| 575 | } |
| 576 | } |
| 577 | if(out->devices & ~(AUDIO_DEVICE_OUT_AUX_DIGITAL | |
| 578 | AUDIO_DEVICE_OUT_SPDIF)) { |
| 579 | decodeType = usecase_decode_format[NUM_STATES_FOR_EACH_DEVICE_FMT*format_index]; |
| 580 | ALOGV("Other Devices: decoderType: %d", decodeType); |
| 581 | out->decoder_type |= decodeType; |
| 582 | for(idx=0; idx<NUM_DECODE_PATH; idx++) { |
| 583 | if(route_to_driver[idx][DECODER_TYPE_IDX] == decodeType) { |
| 584 | switch(route_to_driver[idx][ROUTE_FORMAT_IDX]) { |
| 585 | case ROUTE_UNCOMPRESSED: |
| 586 | ALOGVV("ROUTE_UNCOMPRESSED"); |
| 587 | ALOGVV("Other Devices opened with stereo decode"); |
| 588 | out->open_dec_route = true; |
| 589 | break; |
| 590 | case ROUTE_UNCOMPRESSED_MCH: |
| 591 | ALOGVV("ROUTE_UNCOMPRESSED_MCH"); |
| 592 | ALOGVV("Other Devices opened with multichannel decode"); |
| 593 | out->open_dec_mch_route = true; |
| 594 | out->dec_format_devices &= ~(out->devices & |
| 595 | ~(AUDIO_DEVICE_OUT_SPDIF | |
| 596 | AUDIO_DEVICE_OUT_AUX_DIGITAL)); |
| 597 | out->dec_mch_format_devices |= (out->devices & |
| 598 | ~(AUDIO_DEVICE_OUT_SPDIF | |
| 599 | AUDIO_DEVICE_OUT_AUX_DIGITAL)); |
| 600 | break; |
| 601 | default: |
| 602 | ALOGW("INVALID ROUTE for Other Devices, decoderType %d, routeFormat %d", |
| 603 | decodeType, route_to_driver[idx][ROUTE_FORMAT_IDX]); |
| 604 | break; |
| 605 | } |
| 606 | } |
| 607 | } |
| 608 | } |
| 609 | } |
| 610 | |
| 611 | /******************************************************************************* |
| 612 | Description: update handle states |
| 613 | *******************************************************************************/ |
| 614 | int update_alsa_handle_state(struct stream_out *out) |
| 615 | { |
| 616 | ALOGV("%s", __func__); |
| 617 | |
| 618 | struct alsa_handle *handle = NULL; |
| 619 | struct listnode *node; |
| 620 | |
| 621 | if(out->open_dec_route) { |
| 622 | if((handle = get_alsa_handle())== NULL) |
| 623 | goto error; |
| 624 | list_add_tail(&out->session_list, &handle->list); |
| 625 | handle->route_format = ROUTE_UNCOMPRESSED; |
| 626 | handle->devices = out->dec_format_devices; |
| 627 | handle->usecase = platform_get_usecase(USECASE_AUDIO_PLAYBACK_OFFLOAD); |
| 628 | handle->out = out; |
| 629 | handle->cmd_pending = false; |
| 630 | ALOGD("open_dec_route: routeformat: %d, devices: 0x%x: " |
| 631 | ,handle->route_format, handle->devices); |
| 632 | } |
| 633 | if(out->open_dec_mch_route) { |
| 634 | if((handle = get_alsa_handle())== NULL) |
| 635 | goto error; |
| 636 | list_add_tail(&out->session_list, &handle->list); |
| 637 | handle->route_format = ROUTE_UNCOMPRESSED_MCH; |
| 638 | handle->devices = out->dec_mch_format_devices; |
| 639 | handle->usecase = platform_get_usecase(USECASE_AUDIO_PLAYBACK_OFFLOAD); |
| 640 | handle->out = out; |
| 641 | handle->cmd_pending = false; |
| 642 | ALOGD("OpenMCHDecodeRoute: routeformat: %d, devices: 0x%x: " |
| 643 | ,handle->route_format, handle->devices); |
| 644 | } |
| 645 | if(out->open_passt_route) { |
| 646 | if((handle = get_alsa_handle())== NULL) |
| 647 | goto error; |
| 648 | list_add_tail(&out->session_list, &handle->list); |
| 649 | handle->route_format = ROUTE_COMPRESSED; |
| 650 | handle->devices = out->passt_format_devices; |
| 651 | handle->usecase = platform_get_usecase(USECASE_AUDIO_PLAYBACK_OFFLOAD); |
| 652 | handle->out = out; |
| 653 | handle->cmd_pending = false; |
| 654 | ALOGD("open_passt_route: routeformat: %d, devices: 0x%x: " |
| 655 | ,handle->route_format, handle->devices); |
| 656 | } |
| 657 | if(out->sw_open_trans_route) { |
| 658 | if((handle = get_alsa_handle())== NULL) |
| 659 | goto error; |
| 660 | handle->route_format = ROUTE_SW_TRANSCODED_COMPRESSED; |
| 661 | handle->devices = out->sw_trans_format_devices; |
| 662 | handle->usecase = platform_get_usecase(USECASE_AUDIO_PLAYBACK_OFFLOAD); |
| 663 | handle->out = out; |
| 664 | handle->cmd_pending = false; |
| 665 | ALOGD("OpenTranscodeRoute: routeformat: %d, devices: 0x%x: " |
| 666 | ,handle->route_format, handle->devices); |
| 667 | } |
| 668 | if(out->hw_open_trans_route) { |
| 669 | if((handle = get_alsa_handle())== NULL) |
| 670 | goto error; |
| 671 | handle->route_format = ROUTE_DSP_TRANSCODED_COMPRESSED; |
| 672 | handle->devices = out->hw_trans_format_devices; |
| 673 | handle->usecase = platform_get_usecase(USECASE_AUDIO_PLAYBACK_OFFLOAD); |
| 674 | handle->out = out; |
| 675 | handle->cmd_pending = false; |
| 676 | ALOGD("OpenTranscodeRoute: routeformat: %d, devices: 0x%x: " |
| 677 | ,handle->route_format, handle->devices); |
| 678 | } |
| 679 | |
| 680 | return 0; |
| 681 | |
| 682 | error: |
| 683 | list_for_each(node, &out->session_list) { |
| 684 | handle = node_to_item(node, struct alsa_handle, list); |
| 685 | free_alsa_handle(handle); |
| 686 | } |
| 687 | |
| 688 | return -ENOMEM; |
| 689 | } |
| 690 | |
| 691 | /******************************************************************************* |
| 692 | Description: setup input path |
| 693 | *******************************************************************************/ |
| 694 | int allocate_internal_buffers(struct stream_out *out) |
| 695 | { |
| 696 | ALOGV("%s",__func__); |
| 697 | int ret = 0; |
| 698 | int main_format = out->format & AUDIO_FORMAT_MAIN_MASK; |
| 699 | |
| 700 | /* |
| 701 | setup the bitstream state machine |
| 702 | */ |
| 703 | out->bitstrm = ( struct audio_bitstream_sm *)calloc(1, |
| 704 | sizeof(struct audio_bitstream_sm)); |
| 705 | if(!audio_bitstream_init(out->bitstrm, get_buffering_factor(out))) { |
| 706 | ALOGE("%s Unable to allocate bitstream buffering for MS11",__func__); |
| 707 | free(out->bitstrm); |
| 708 | out->bitstrm = NULL; |
| 709 | return -EINVAL; |
| 710 | } |
| 711 | |
| 712 | if(is_input_buffering_mode_reqd(out)) |
| 713 | audio_bitstream_start_input_buffering_mode(out->bitstrm); |
| 714 | |
| 715 | /* |
| 716 | setup the buffering data required for decode to start |
| 717 | AAC_ADIF would require worst case frame size before decode starts |
| 718 | other decoder formats handles the partial data, hence threshold is zero. |
| 719 | */ |
| 720 | |
| 721 | if(main_format == AUDIO_FORMAT_AAC_ADIF) |
| 722 | out->min_bytes_req_to_dec = AAC_BLOCK_PER_CHANNEL_MS11*out->channels-1; |
| 723 | else |
| 724 | out->min_bytes_req_to_dec = 0; |
| 725 | |
| 726 | ret = open_temp_buf_for_metadata(out); |
| 727 | if(ret < 0) { |
| 728 | free(out->bitstrm); |
| 729 | out->bitstrm = NULL; |
| 730 | } |
| 731 | out->buffer_size = get_buffer_length(out); |
| 732 | |
| 733 | return ret; |
| 734 | } |
| 735 | |
| 736 | /******************************************************************************* |
| 737 | Description: setup input path |
| 738 | *******************************************************************************/ |
| 739 | int free_internal_buffers(struct stream_out *out) |
| 740 | { |
| 741 | if(out->bitstrm) { |
| 742 | free(out->bitstrm); |
| 743 | out->bitstrm = NULL; |
| 744 | } |
| 745 | |
| 746 | if(out->write_temp_buf) { |
| 747 | free(out->write_temp_buf); |
| 748 | out->write_temp_buf = NULL; |
| 749 | } |
| 750 | |
| 751 | if(out->dec_conf_buf) { |
| 752 | free(out->dec_conf_buf); |
| 753 | out->dec_conf_buf = NULL; |
| 754 | } |
| 755 | return 0; |
| 756 | } |
| 757 | |
| 758 | /******************************************************************************* |
| 759 | Description: open MS11 instance |
| 760 | *******************************************************************************/ |
| 761 | //TODO: enable sw_decode if required |
| 762 | #if USE_SWDECODE |
| 763 | static int open_ms11_instance(struct stream_out *out) |
| 764 | { |
| 765 | ALOGV("openMS11Instance"); |
| 766 | int32_t formatMS11; |
| 767 | int main_format = out->format & AUDIO_FORMAT_MAIN_MASK; |
| 768 | out->ms11_decoder = get_soft_ms11(); |
| 769 | if(!out->ms11_decoder) { |
| 770 | ALOGE("Could not resolve all symbols Required for MS11"); |
| 771 | return -EINVAL; |
| 772 | } |
| 773 | /* |
| 774 | MS11 created |
| 775 | */ |
| 776 | if(initialize_ms11_function_pointers(out->ms11_decoder) == false) { |
| 777 | ALOGE("Could not resolve all symbols Required for MS11"); |
| 778 | free_soft_ms11(out->ms11_decoder); |
| 779 | return -EINVAL; |
| 780 | } |
| 781 | /* |
| 782 | update format |
| 783 | */ |
| 784 | if((main_format == AUDIO_FORMAT_AC3) || |
| 785 | (main_format == AUDIO_FORMAT_EAC3)) { |
| 786 | /*TODO: who wil setCOMPRESSED_CONVERT_AC3_ASSOC */ |
| 787 | if (out->spdif_format == COMPRESSED_CONVERT_AC3_ASSOC) |
| 788 | formatMS11 = FORMAT_DOLBY_DIGITAL_PLUS_MAIN_ASSOC; |
| 789 | else |
| 790 | formatMS11 = FORMAT_DOLBY_DIGITAL_PLUS_MAIN; |
| 791 | } else |
| 792 | formatMS11 = FORMAT_DOLBY_PULSE_MAIN; |
| 793 | /* |
| 794 | set the use case to the MS11 decoder and open the stream for decoding |
| 795 | */ |
| 796 | if(ms11_set_usecase_and_open_stream_with_mode(out->ms11_decoder, |
| 797 | formatMS11, out->channels, out->sample_rate, |
| 798 | out->is_m11_file_mode)) { |
| 799 | ALOGE("SetUseCaseAndOpen MS11 failed"); |
| 800 | free_soft_ms11(out->ms11_decoder); |
| 801 | return EINVAL; |
| 802 | } |
| 803 | if(is_decoder_config_required(out) && out->dec_conf_buf && out->dec_conf_bufLength) { |
| 804 | if(ms11_set_aac_config(out->ms11_decoder, (unsigned char *)out->dec_conf_buf, |
| 805 | out->dec_conf_bufLength) == true) { |
| 806 | out->dec_conf_set = true; |
| 807 | } |
| 808 | } |
| 809 | |
| 810 | return 0; |
| 811 | } |
| 812 | #endif |
| 813 | /******************************************************************************* |
| 814 | Description: copy input to internal buffer |
| 815 | *******************************************************************************/ |
| 816 | void copy_bitstream_internal_buffer(struct audio_bitstream_sm *bitstrm, |
| 817 | char *buffer, size_t bytes) |
| 818 | { |
| 819 | // copy bitstream to internal buffer |
| 820 | audio_bitstream_copy_to_internal_buffer(bitstrm, (char *)buffer, bytes); |
| 821 | #ifdef DEBUG |
| 822 | dumpInputOutput(INPUT, buffer, bytes, 0); |
| 823 | #endif |
| 824 | } |
| 825 | |
| 826 | /******************************************************************************* |
| 827 | Description: set decoder config |
| 828 | *******************************************************************************/ |
| 829 | //TODO: enable sw_decode if required |
| 830 | #if USE_SWDECODE |
| 831 | int setDecodeConfig(struct stream_out *out, char *buffer, size_t bytes) |
| 832 | { |
| 833 | ALOGV("%s ", __func__); |
| 834 | |
| 835 | int main_format = out->format & AUDIO_FORMAT_MAIN_MASK; |
| 836 | if(!out->dec_conf_set) { |
| 837 | if(main_format == AUDIO_FORMAT_AAC || |
| 838 | main_format == AUDIO_FORMAT_HE_AAC_V1 || |
| 839 | main_format == AUDIO_FORMAT_AAC_ADIF || |
| 840 | main_format == AUDIO_FORMAT_HE_AAC_V2) { |
| 841 | if(out->ms11_decoder != NULL) { |
| 842 | if(ms11_set_aac_config(out->ms11_decoder,(unsigned char *)buffer, |
| 843 | bytes) == false) { |
| 844 | ALOGE("AAC decoder config fail"); |
| 845 | return 0; |
| 846 | } |
| 847 | } |
| 848 | } |
| 849 | |
| 850 | out->dec_conf_bufLength = bytes; |
| 851 | if(out->dec_conf_buf) |
| 852 | free(out->dec_conf_buf); |
| 853 | |
| 854 | out->dec_conf_buf = malloc(out->dec_conf_bufLength); |
| 855 | memcpy(out->dec_conf_buf, |
| 856 | buffer, |
| 857 | out->dec_conf_bufLength); |
| 858 | out->dec_conf_set = true; |
| 859 | } |
| 860 | out->dec_conf_set = true; |
| 861 | return bytes; |
| 862 | } |
| 863 | #endif |
| 864 | |
| 865 | //TODO: enable sw_decode if required |
| 866 | #if USE_SWDECODE |
| 867 | int validate_sw_free_space(struct stream_out* out, int bytes_consumed_in_decode, int *pcm_2ch_len, |
| 868 | int *pcm_mch_len, int *passthru_len, int *transcode_len, bool *wait_for_write_done) { |
| 869 | |
| 870 | struct alsa_handle *handle = NULL; |
| 871 | char *bufPtr; |
| 872 | int copy_output_buffer_size; |
| 873 | |
| 874 | *pcm_2ch_len = *pcm_mch_len = *passthru_len = *transcode_len = *wait_for_write_done = 0; |
| 875 | |
| 876 | if(out->decoder_type & SW_DECODE) { |
| 877 | bufPtr = audio_bitstream_get_output_buffer_write_ptr(out->bitstrm, |
| 878 | PCM_2CH_OUT); |
| 879 | /*TODO: there is chance of illegale access if ms11 output exceeds bitstream |
| 880 | output buffer boudary */ |
| 881 | copy_output_buffer_size = ms11_copy_output_from_ms11buf(out->ms11_decoder, |
| 882 | PCM_2CH_OUT, |
| 883 | bufPtr); |
| 884 | handle = get_handle_by_route_format(out, ROUTE_UNCOMPRESSED); |
| 885 | if(handle == NULL) { |
| 886 | ALOGE("%s Invalid handle", __func__); |
| 887 | return -EINVAL; |
| 888 | } |
| 889 | if(get_compress_available_space(handle) < copy_output_buffer_size) { |
| 890 | handle->cmd_pending = true; |
| 891 | *wait_for_write_done = true; |
| 892 | } |
| 893 | *pcm_2ch_len = copy_output_buffer_size; |
| 894 | |
| 895 | } |
| 896 | if(out->decoder_type & SW_DECODE_MCH) { |
| 897 | bufPtr=audio_bitstream_get_output_buffer_write_ptr(out->bitstrm, |
| 898 | PCM_MCH_OUT); |
| 899 | copy_output_buffer_size = ms11_copy_output_from_ms11buf(out->ms11_decoder, |
| 900 | PCM_MCH_OUT, |
| 901 | bufPtr); |
| 902 | handle = get_handle_by_route_format(out, ROUTE_UNCOMPRESSED_MCH); |
| 903 | if(handle == NULL) { |
| 904 | ALOGE("%s Invalid handle", __func__); |
| 905 | return -EINVAL; |
| 906 | } |
| 907 | |
| 908 | if(get_compress_available_space(handle) < copy_output_buffer_size) { |
| 909 | handle->cmd_pending = true; |
| 910 | *wait_for_write_done = true; |
| 911 | } |
| 912 | *pcm_mch_len = copy_output_buffer_size; |
| 913 | } |
| 914 | if(out->decoder_type & SW_PASSTHROUGH) { |
| 915 | bufPtr = audio_bitstream_get_output_buffer_write_ptr(out->bitstrm, COMPRESSED_OUT); |
| 916 | copy_output_buffer_size = bytes_consumed_in_decode; |
| 917 | memcpy(bufPtr, audio_bitstream_get_input_buffer_ptr(out->bitstrm), copy_output_buffer_size); |
| 918 | |
| 919 | handle = get_handle_by_route_format(out, ROUTE_COMPRESSED); |
| 920 | if(handle == NULL) { |
| 921 | ALOGE("%s Invalid handle", __func__); |
| 922 | return -EINVAL; |
| 923 | } |
| 924 | |
| 925 | if(get_compress_available_space(handle) < copy_output_buffer_size) { |
| 926 | handle->cmd_pending = true; |
| 927 | *wait_for_write_done = true; |
| 928 | } |
| 929 | *passthru_len = copy_output_buffer_size; |
| 930 | } |
| 931 | if(out->decoder_type & SW_TRANSCODE) { |
| 932 | bufPtr = audio_bitstream_get_output_buffer_write_ptr(out->bitstrm, |
| 933 | TRANSCODE_OUT); |
| 934 | copy_output_buffer_size = ms11_copy_output_from_ms11buf(out->bitstrm, |
| 935 | COMPRESSED_OUT, |
| 936 | bufPtr); |
| 937 | handle = get_handle_by_route_format(out, ROUTE_SW_TRANSCODED_COMPRESSED); |
| 938 | if(handle == NULL) { |
| 939 | ALOGE("%s Invalid handle", __func__); |
| 940 | return -EINVAL; |
| 941 | } |
| 942 | if(get_compress_available_space(handle) < copy_output_buffer_size) { |
| 943 | handle->cmd_pending = true; |
| 944 | *wait_for_write_done = true; |
| 945 | } |
| 946 | *transcode_len = copy_output_buffer_size; |
| 947 | } |
| 948 | return 0; |
| 949 | } |
| 950 | #endif |
| 951 | |
| 952 | int validate_hw_free_space(struct stream_out *out, int bytes_consumed_in_decode, int *pcm_2ch_len, |
| 953 | int *pcm_mch_len, int *passthru_len, int *transcode_len, bool *wait_for_write_done) { |
| 954 | |
| 955 | struct alsa_handle *handle = NULL; |
| 956 | char *bufPtr; |
| 957 | int copy_output_buffer_size; |
| 958 | *pcm_2ch_len = *pcm_mch_len = *passthru_len = *transcode_len = *wait_for_write_done = 0; |
| 959 | if(out->decoder_type & DSP_DECODE) { |
| 960 | ALOGVV("DSP_DECODE"); |
| 961 | bufPtr = audio_bitstream_get_output_buffer_write_ptr(out->bitstrm, |
| 962 | PCM_MCH_OUT); |
| 963 | copy_output_buffer_size = bytes_consumed_in_decode; |
| 964 | memcpy(bufPtr, audio_bitstream_get_input_buffer_ptr(out->bitstrm), |
| 965 | copy_output_buffer_size); |
| 966 | ALOGVV("%s bytes_consumed %d out bufPtr %x, pcm_mch_out_buf_size%d", |
| 967 | __func__,bytes_consumed_in_decode,bufPtr, |
| 968 | out->bitstrm->pcm_mch_out_buf_size); |
| 969 | handle = get_handle_by_route_format(out, ROUTE_UNCOMPRESSED);/*TODO: revisit */ |
| 970 | if(handle == NULL) { |
| 971 | ALOGE("%s Invalid handle", __func__); |
| 972 | return -EINVAL; |
| 973 | } |
| 974 | if(get_compress_available_space(handle) < copy_output_buffer_size) { |
| 975 | handle->cmd_pending = true; |
| 976 | *wait_for_write_done = true; |
| 977 | /*reset input buffer pointer as flinger will resend the data back */ |
| 978 | audio_bitstream_set_input_buffer_write_ptr(out->bitstrm, |
| 979 | -copy_output_buffer_size); |
| 980 | *pcm_mch_len = copy_output_buffer_size; |
| 981 | } |
| 982 | else |
| 983 | *pcm_mch_len = copy_output_buffer_size; |
| 984 | } |
| 985 | if(out->decoder_type & DSP_PASSTHROUGH) { |
| 986 | ALOGVV("DSP_PASSTHROUGH"); |
| 987 | bufPtr = audio_bitstream_get_output_buffer_write_ptr(out->bitstrm, COMPRESSED_OUT); |
| 988 | copy_output_buffer_size = bytes_consumed_in_decode; |
| 989 | memcpy(bufPtr, audio_bitstream_get_input_buffer_ptr(out->bitstrm), copy_output_buffer_size); |
| 990 | handle = get_handle_by_route_format(out, ROUTE_COMPRESSED); |
| 991 | if(handle == NULL) { |
| 992 | ALOGE("%s Invalid handle", __func__); |
| 993 | return -EINVAL; |
| 994 | } |
| 995 | if(get_compress_available_space(handle) < copy_output_buffer_size) { |
| 996 | handle->cmd_pending = true; |
| 997 | *wait_for_write_done = true; |
| 998 | *passthru_len = copy_output_buffer_size; |
| 999 | /*reset input buffer pointer as flinger will resend the data back */ |
| 1000 | audio_bitstream_set_input_buffer_ptr(out->bitstrm, -copy_output_buffer_size); |
| 1001 | } |
| 1002 | else |
| 1003 | *passthru_len = copy_output_buffer_size; |
| 1004 | } |
| 1005 | /*TODO: handle DSP Transcode usecase */ |
| 1006 | return 0; |
| 1007 | } |
| 1008 | |
| 1009 | int update_bitstrm_pointers(struct stream_out *out, int pcm_2ch_len, |
| 1010 | int pcm_mch_len, int passthru_len, int transcode_len) { |
| 1011 | |
| 1012 | if(out->decoder_type & SW_DECODE) { |
| 1013 | audio_bitstream_set_output_buffer_write_ptr(out->bitstrm, PCM_2CH_OUT, |
| 1014 | pcm_2ch_len); |
| 1015 | |
| 1016 | } |
| 1017 | if(out->decoder_type & SW_DECODE_MCH || out->decoder_type & DSP_DECODE) { |
| 1018 | audio_bitstream_set_output_buffer_write_ptr(out->bitstrm, PCM_MCH_OUT, pcm_mch_len); |
| 1019 | } |
| 1020 | if(out->decoder_type & SW_PASSTHROUGH || out->decoder_type & DSP_PASSTHROUGH) { |
| 1021 | audio_bitstream_set_output_buffer_write_ptr(out->bitstrm, COMPRESSED_OUT, passthru_len); |
| 1022 | } |
| 1023 | if(out->decoder_type & SW_TRANSCODE) { |
| 1024 | audio_bitstream_set_output_buffer_write_ptr(out->bitstrm, |
| 1025 | TRANSCODE_OUT, |
| 1026 | transcode_len); |
| 1027 | } |
| 1028 | return 0; |
| 1029 | } |
| 1030 | |
| 1031 | /*TODO correct it */ |
| 1032 | static int configure_compr(struct stream_out *out, |
| 1033 | struct alsa_handle *handle) { |
| 1034 | handle->compr_config.codec = (struct snd_codec *) |
| 1035 | calloc(1, sizeof(struct snd_codec)); |
| 1036 | handle->compr_config.codec->id = |
| 1037 | get_snd_codec_id(out->format); /*TODO: correct this based on format*/ |
| 1038 | handle->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; |
| 1039 | handle->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS; |
| 1040 | handle->compr_config.codec->sample_rate = |
| 1041 | compress_get_alsa_rate(out->sample_rate); |
| 1042 | handle->compr_config.codec->bit_rate = out->compr_config.codec->bit_rate; |
| 1043 | handle->compr_config.codec->ch_in = |
| 1044 | popcount(out->channel_mask); |
| 1045 | handle->compr_config.codec->ch_out = handle->compr_config.codec->ch_in; |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1046 | handle->compr_config.codec->format = out->compr_config.codec->format; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1047 | memcpy(&handle->compr_config.codec->options, |
| 1048 | &out->compr_config.codec->options, |
| 1049 | sizeof(union snd_codec_options)); |
| 1050 | return 0; |
| 1051 | } |
| 1052 | |
| 1053 | /*TODO: do we need to apply volume at the session open*/ |
| 1054 | static int set_compress_volume(struct alsa_handle *handle, int left, int right) |
| 1055 | { |
| 1056 | |
| 1057 | struct audio_device *adev = handle->out->dev; |
| 1058 | struct mixer_ctl *ctl; |
| 1059 | int volume[2]; |
| 1060 | |
| 1061 | char mixer_ctl_name[44]; // max length of name is 44 as defined |
| 1062 | char device_id[STRING_LENGTH_OF_INTEGER+1]; |
| 1063 | |
| 1064 | memset(mixer_ctl_name, 0, sizeof(mixer_ctl_name)); |
| 1065 | strlcpy(mixer_ctl_name, "Compress Playback Volume", sizeof(mixer_ctl_name)); |
| 1066 | |
| 1067 | memset(device_id, 0, sizeof(device_id)); |
| 1068 | snprintf(device_id, "%d", handle->device_id, sizeof(device_id)); |
| 1069 | |
| 1070 | strlcat(mixer_ctl_name, device_id, sizeof(mixer_ctl_name)); |
| 1071 | |
| 1072 | ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); |
| 1073 | if (!ctl) { |
| 1074 | ALOGE("%s: Could not get ctl for mixer cmd - %s", |
| 1075 | __func__, mixer_ctl_name); |
| 1076 | return -EINVAL; |
| 1077 | } |
| 1078 | volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX); |
| 1079 | volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX); |
| 1080 | mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0])); |
| 1081 | |
| 1082 | return 0; |
| 1083 | |
| 1084 | } |
| 1085 | |
| 1086 | /******************************************************************************* |
| 1087 | Description: software decode handling |
| 1088 | *******************************************************************************/ |
| 1089 | //TODO: enable sw_decode if required |
| 1090 | #if USE_SWDECODE |
| 1091 | static int sw_decode(struct stream_out *out, |
| 1092 | char *buffer, |
| 1093 | size_t bytes, |
| 1094 | size_t *bytes_consumed, |
| 1095 | bool *continueDecode) |
| 1096 | { |
| 1097 | /* bytes pending to be decoded in current buffer*/ |
| 1098 | bool wait_for_write_done = false; |
| 1099 | int bytes_pending_for_decode = 0; |
| 1100 | /* bytes consumed in current write buffer */ |
| 1101 | int total_bytes_consumed = 0; |
| 1102 | size_t copyBytesMS11 = 0; |
| 1103 | size_t bytes_consumed_in_decode = 0; |
| 1104 | size_t copy_output_buffer_size = 0; |
| 1105 | uint32_t outSampleRate = out->sample_rate; |
| 1106 | uint32_t outChannels = out->channels; |
| 1107 | char * bufPtr; |
| 1108 | int pcm_2ch_len, pcm_mch_len, passthru_len, transcode_len; |
| 1109 | struct alsa_handle *handle = NULL; |
| 1110 | |
| 1111 | ALOGVV("sw Decode"); |
| 1112 | // eos handling |
| 1113 | if(bytes == 0) { |
| 1114 | if(out->format == AUDIO_FORMAT_AAC_ADIF) |
| 1115 | audio_bitstream_append_silence_internal_buffer(out->bitstrm, |
| 1116 | out->min_bytes_req_to_dec,0x0); |
| 1117 | else |
| 1118 | return false; |
| 1119 | } |
| 1120 | /* |
| 1121 | check for sync word, if present then configure MS11 for fileplayback mode |
| 1122 | OFF. This is specifically done to handle Widevine usecase, in which the |
| 1123 | ADTS HEADER is not stripped off by the Widevine parser |
| 1124 | */ |
| 1125 | if(out->first_bitstrm_buf == true) { |
| 1126 | uint16_t uData = (*((char *)buffer) << 8) + *((char *)buffer + 1) ; |
| 1127 | if(ADTS_HEADER_SYNC_RESULT == (uData & ADTS_HEADER_SYNC_MASK)) { |
| 1128 | ALOGD("Sync word found hence configure MS11 in file_playback Mode OFF"); |
| 1129 | free_soft_ms11(out->ms11_decoder); |
| 1130 | out->is_m11_file_mode = false; |
| 1131 | open_ms11_instance(out); |
| 1132 | } |
| 1133 | out->first_bitstrm_buf = false; |
| 1134 | } |
| 1135 | //decode |
| 1136 | if(out->decoder_type == SW_PASSTHROUGH) { |
| 1137 | /*TODO: check if correct */ |
| 1138 | bytes_consumed_in_decode = audio_bitstream_get_size(out->bitstrm); |
| 1139 | } else { |
| 1140 | if(audio_bitstream_sufficient_buffer_to_decode(out->bitstrm, |
| 1141 | out->min_bytes_req_to_dec) == true) { |
| 1142 | bufPtr = audio_bitstream_get_input_buffer_ptr(out->bitstrm); |
| 1143 | copyBytesMS11 = audio_bitstream_get_size(out->bitstrm); |
| 1144 | ms11_copy_bitstream_to_ms11_inpbuf(out->ms11_decoder, bufPtr,copyBytesMS11); |
| 1145 | bytes_consumed_in_decode = ms11_stream_decode(out->ms11_decoder, |
| 1146 | &outSampleRate, &outChannels); |
| 1147 | } |
| 1148 | } |
| 1149 | |
| 1150 | if((out->sample_rate != outSampleRate) || (out->channels != outChannels)) { |
| 1151 | ALOGD("Change in sample rate. New sample rate: %d", outSampleRate); |
| 1152 | out->sample_rate = outSampleRate; |
| 1153 | out->channels = outChannels; |
| 1154 | handle = get_handle_by_route_format(out, ROUTE_UNCOMPRESSED); |
| 1155 | if(handle !=NULL) { |
| 1156 | configure_compr(out, handle); |
| 1157 | handle->compr = compress_open(SOUND_CARD, handle->device_id, |
| 1158 | COMPRESS_IN, &handle->compr_config); |
| 1159 | if (handle->compr && !is_compress_ready(handle->compr)) { |
| 1160 | ALOGE("%s: %s", __func__, compress_get_error(handle->compr)); |
| 1161 | compress_close(handle->compr); |
| 1162 | handle->compr = NULL; |
| 1163 | } |
| 1164 | if (out->offload_callback) |
| 1165 | compress_nonblock(handle->compr, out->non_blocking); |
| 1166 | |
| 1167 | set_compress_volume(handle, out->left_volume, out->right_volume); |
| 1168 | } |
| 1169 | |
| 1170 | handle = get_handle_by_route_format(out, ROUTE_UNCOMPRESSED_MCH); |
| 1171 | if(handle !=NULL) { |
| 1172 | configure_compr(out, handle); |
| 1173 | handle->compr = compress_open(SOUND_CARD, handle->device_id, |
| 1174 | COMPRESS_IN, &handle->compr_config); |
| 1175 | if (handle->compr && !is_compress_ready(handle->compr)) { |
| 1176 | ALOGE("%s: %s", __func__, compress_get_error(handle->compr)); |
| 1177 | compress_close(handle->compr); |
| 1178 | handle->compr = NULL; |
| 1179 | } |
| 1180 | if (out->offload_callback) |
| 1181 | compress_nonblock(handle->compr, out->non_blocking); |
| 1182 | set_compress_volume(handle, out->left_volume, out->right_volume); |
| 1183 | out->channel_status_set = false; |
| 1184 | } |
| 1185 | } |
| 1186 | |
| 1187 | |
| 1188 | validate_sw_free_space(out, bytes_consumed_in_decode, &pcm_2ch_len, &pcm_mch_len, |
| 1189 | &passthru_len, &transcode_len, &wait_for_write_done); |
| 1190 | |
| 1191 | if(wait_for_write_done && out->non_blocking) { |
| 1192 | send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); |
| 1193 | *continueDecode = false; |
| 1194 | *bytes_consumed = 0; |
| 1195 | return 0; |
| 1196 | } else { |
| 1197 | update_bitstrm_pointers(out, pcm_2ch_len, pcm_mch_len, |
| 1198 | passthru_len, transcode_len); |
| 1199 | audio_bitstream_copy_residue_to_start(out->bitstrm, bytes_consumed_in_decode); |
| 1200 | *bytes_consumed = bytes_consumed_in_decode; |
| 1201 | } |
| 1202 | |
| 1203 | copy_output_buffer_size = pcm_2ch_len + pcm_mch_len + passthru_len + transcode_len; |
| 1204 | if(copy_output_buffer_size && |
| 1205 | audio_bitstream_sufficient_buffer_to_decode(out->bitstrm, out->min_bytes_req_to_dec) == true) { |
| 1206 | *continueDecode = true; |
| 1207 | return 0; |
| 1208 | } |
| 1209 | return 0; |
| 1210 | } |
| 1211 | #endif |
| 1212 | |
| 1213 | /******************************************************************************* |
| 1214 | Description: dsp decode handling |
| 1215 | *******************************************************************************/ |
| 1216 | static bool dsp_decode(struct stream_out *out, char *buffer, size_t bytes, |
| 1217 | size_t *bytes_consumed, bool *continueDecode) |
| 1218 | { |
| 1219 | char *bufPtr; |
| 1220 | size_t bytes_consumed_in_decode = 0; |
| 1221 | |
| 1222 | bool wait_for_write_done = false; |
| 1223 | int pcm_2ch_len, pcm_mch_len, passthru_len, transcode_len; |
| 1224 | |
| 1225 | ALOGVV("dsp_decode"); |
| 1226 | // decode |
| 1227 | { |
| 1228 | bytes_consumed_in_decode = audio_bitstream_get_size(out->bitstrm); |
| 1229 | } |
| 1230 | // handle change in sample rate |
| 1231 | { |
| 1232 | } |
| 1233 | //TODO: check if the copy of the buffers can be avoided |
| 1234 | /* can be removed as its not required for dsp decode usecase */ |
| 1235 | *continueDecode = false; |
| 1236 | validate_hw_free_space(out, bytes_consumed_in_decode, &pcm_2ch_len, &pcm_mch_len, |
| 1237 | &passthru_len, &transcode_len, &wait_for_write_done); |
| 1238 | |
| 1239 | if(wait_for_write_done && out->non_blocking) { |
| 1240 | send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); |
| 1241 | *bytes_consumed = 0; |
| 1242 | return 0; |
| 1243 | } else { |
| 1244 | update_bitstrm_pointers(out, pcm_2ch_len, pcm_mch_len, |
| 1245 | passthru_len, transcode_len); |
| 1246 | audio_bitstream_copy_residue_to_start(out->bitstrm, bytes_consumed_in_decode); |
| 1247 | *bytes_consumed = bytes_consumed_in_decode; |
| 1248 | ALOGV("%s bytes_consumed_in_decode =%d",__func__,bytes_consumed_in_decode); |
| 1249 | } |
| 1250 | |
| 1251 | return 0; |
| 1252 | } |
| 1253 | |
| 1254 | static bool decode(struct stream_out *out, char * buffer, size_t bytes, |
| 1255 | size_t *bytes_consumed, bool *continuedecode) |
| 1256 | { |
| 1257 | ALOGV("decode"); |
| 1258 | bool continueDecode = false; |
| 1259 | int ret = 0; |
| 1260 | |
| 1261 | // TODO: enable software decode if required |
| 1262 | /*if (out->use_ms11_decoder) { |
| 1263 | ret = sw_decode(out, buffer, bytes, |
| 1264 | bytes_consumed, continuedecode); |
| 1265 | |
| 1266 | // set channel status |
| 1267 | // Set the channel status after first frame decode/transcode |
| 1268 | //TODO: set the SPDIF channel status bits |
| 1269 | if(out->channel_status_set == false) |
| 1270 | setSpdifchannel_status( |
| 1271 | audio_bitstream_get_output_buffer_ptr(out->bitstrm, COMPRESSED_OUT), |
| 1272 | bytes, AUDIO_PARSER_CODEC_AC3); |
| 1273 | |
| 1274 | } else */{ |
| 1275 | ret = dsp_decode(out, buffer, bytes, |
| 1276 | bytes_consumed, continuedecode); |
| 1277 | // set channel status |
| 1278 | // Set the channel status after first frame decode/transcode |
| 1279 | //TODO: set the SPDIF channel status bits |
| 1280 | /* if(out->channel_status_set == false) |
| 1281 | setSpdifchannel_status( |
| 1282 | audio_bitstream_get_output_buffer_ptr(out->bitstrm, COMPRESSED_OUT), |
| 1283 | bytes, AUDIO_PARSER_CODEC_DTS); |
| 1284 | */ |
| 1285 | } |
| 1286 | return ret; |
| 1287 | } |
| 1288 | |
| 1289 | /******************************************************************************* |
| 1290 | Description: fixup sample rate and channel info based on format |
| 1291 | *******************************************************************************/ |
| 1292 | void fixupSampleRateChannelModeMS11Formats(struct stream_out *out) |
| 1293 | { |
| 1294 | ALOGV("fixupSampleRateChannelModeMS11Formats"); |
| 1295 | int main_format = out->format & AUDIO_FORMAT_MAIN_MASK; |
| 1296 | int subFormat = out->format & AUDIO_FORMAT_SUB_MASK; |
| 1297 | /* |
| 1298 | NOTE: For AAC, the output of MS11 is 48000 for the sample rates greater than |
| 1299 | 24000. The samples rates <= 24000 will be at their native sample rate |
| 1300 | For AC3, the PCM output is at its native sample rate if the decoding is |
| 1301 | single decode usecase for MS11. |
| 1302 | */ |
| 1303 | if(main_format == AUDIO_FORMAT_AAC || |
| 1304 | main_format == AUDIO_FORMAT_HE_AAC_V1 || |
| 1305 | main_format == AUDIO_FORMAT_HE_AAC_V2 || |
| 1306 | main_format == AUDIO_FORMAT_AAC_ADIF) { |
| 1307 | out->sample_rate = out->sample_rate > 24000 ? 48000 : out->sample_rate; |
| 1308 | out->channels = 6; |
| 1309 | } else if (main_format == AUDIO_FORMAT_AC3 || |
| 1310 | main_format == AUDIO_FORMAT_EAC3) { |
| 1311 | /* transcode AC3/EAC3 44.1K to 48K AC3 for non dual-mono clips */ |
| 1312 | if (out->sample_rate == 44100 && |
| 1313 | (subFormat != AUDIO_FORMAT_DOLBY_SUB_DM) && |
| 1314 | (out->spdif_format == COMPRESSED || |
| 1315 | out->spdif_format == AUTO_DEVICE_FORMAT || |
| 1316 | out->spdif_format == COMPRESSED_CONVERT_EAC3_AC3) && |
| 1317 | (out->hdmi_format == UNCOMPRESSED || |
| 1318 | out->hdmi_format == UNCOMPRESSED_MCH)) { |
| 1319 | out->sample_rate = 48000; |
| 1320 | out->spdif_format = COMPRESSED_CONVERT_AC3_ASSOC; |
| 1321 | } else if (out->sample_rate == 44100) { |
| 1322 | out->spdif_format = UNCOMPRESSED; |
| 1323 | } |
| 1324 | out->channels = 6; |
| 1325 | } |
| 1326 | ALOGD("ms11 format fixup: out->spdif_format %d, out->hdmi_format %d", |
| 1327 | out->spdif_format, out->hdmi_format); |
| 1328 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1329 | |
| 1330 | static bool is_supported_format(audio_format_t format) |
| 1331 | { |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1332 | switch (format) { |
| 1333 | case AUDIO_FORMAT_PCM_16_BIT: |
| 1334 | case AUDIO_FORMAT_MP3: |
| 1335 | case AUDIO_FORMAT_AAC: |
| 1336 | case AUDIO_FORMAT_WMA: |
| 1337 | case AUDIO_FORMAT_WMA_PRO: |
| 1338 | case AUDIO_FORMAT_MP2: |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1339 | return true; |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1340 | default: |
| 1341 | ALOGE("%s: Unsupported audio format: %x", __func__, format); |
| 1342 | break; |
| 1343 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1344 | |
| 1345 | return false; |
| 1346 | } |
| 1347 | |
| 1348 | static int get_snd_codec_id(audio_format_t format) |
| 1349 | { |
| 1350 | int id = 0; |
| 1351 | |
| 1352 | switch (format) { |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1353 | case AUDIO_FORMAT_PCM_16_BIT: |
| 1354 | id = SND_AUDIOCODEC_PCM; |
| 1355 | break; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1356 | case AUDIO_FORMAT_MP3: |
| 1357 | id = SND_AUDIOCODEC_MP3; |
| 1358 | break; |
| 1359 | case AUDIO_FORMAT_AAC: |
| 1360 | id = SND_AUDIOCODEC_AAC; |
| 1361 | break; |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1362 | case AUDIO_FORMAT_WMA: |
| 1363 | id = SND_AUDIOCODEC_WMA; |
| 1364 | break; |
| 1365 | case AUDIO_FORMAT_WMA_PRO: |
| 1366 | id = SND_AUDIOCODEC_WMA_PRO; |
| 1367 | break; |
| 1368 | case AUDIO_FORMAT_MP2: |
| 1369 | id = SND_AUDIOCODEC_MP2; |
| 1370 | break; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1371 | default: |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1372 | ALOGE("%s: Unsupported audio format %x", __func__, format); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1373 | } |
| 1374 | |
| 1375 | return id; |
| 1376 | } |
| 1377 | |
| 1378 | /* must be called with hw device mutex locked */ |
| 1379 | static int read_hdmi_channel_masks(struct stream_out *out) |
| 1380 | { |
| 1381 | int ret = 0; |
| 1382 | int channels = platform_edid_get_max_channels(out->dev->platform); |
| 1383 | |
| 1384 | switch (channels) { |
| 1385 | /* |
| 1386 | * Do not handle stereo output in Multi-channel cases |
| 1387 | * Stereo case is handled in normal playback path |
| 1388 | */ |
| 1389 | case 6: |
| 1390 | ALOGV("%s: HDMI supports 5.1", __func__); |
| 1391 | out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1; |
| 1392 | break; |
| 1393 | case 8: |
| 1394 | ALOGV("%s: HDMI supports 5.1 and 7.1 channels", __func__); |
| 1395 | out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1; |
| 1396 | out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1; |
| 1397 | break; |
| 1398 | default: |
| 1399 | ALOGE("HDMI does not support multi channel playback"); |
| 1400 | ret = -ENOSYS; |
| 1401 | break; |
| 1402 | } |
| 1403 | return ret; |
| 1404 | } |
| 1405 | |
| 1406 | /* must be called with out->lock locked */ |
| 1407 | static int send_offload_cmd_l(struct stream_out* out, int command) |
| 1408 | { |
| 1409 | struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd)); |
| 1410 | |
| 1411 | ALOGVV("%s %d", __func__, command); |
| 1412 | |
| 1413 | cmd->cmd = command; |
| 1414 | list_add_tail(&out->offload_cmd_list, &cmd->node); |
| 1415 | pthread_cond_signal(&out->offload_cond); |
| 1416 | return 0; |
| 1417 | } |
| 1418 | |
| 1419 | /* must be called iwth out->lock locked */ |
| 1420 | static void stop_compressed_output_l(struct stream_out *out) |
| 1421 | { |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1422 | struct listnode *node; |
| 1423 | struct alsa_handle *handle; |
| 1424 | bool is_compr_out = false; |
| 1425 | |
| 1426 | ALOGV("%s", __func__); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1427 | out->offload_state = OFFLOAD_STATE_IDLE; |
| 1428 | out->playback_started = 0; |
| 1429 | out->send_new_metadata = 1; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1430 | list_for_each(node, &out->session_list) { |
| 1431 | handle = node_to_item(node, struct alsa_handle, list); |
| 1432 | if (handle->compr != NULL) { |
| 1433 | compress_stop(handle->compr); |
| 1434 | is_compr_out = true; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1435 | } |
| 1436 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1437 | if (is_compr_out) { |
| 1438 | while (out->offload_thread_blocked) |
| 1439 | pthread_cond_wait(&out->cond, &out->lock); |
| 1440 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1441 | } |
| 1442 | |
| 1443 | static void *offload_thread_loop(void *context) |
| 1444 | { |
| 1445 | struct stream_out *out = (struct stream_out *) context; |
| 1446 | struct listnode *item; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1447 | struct listnode *node; |
| 1448 | struct alsa_handle *handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1449 | |
| 1450 | out->offload_state = OFFLOAD_STATE_IDLE; |
| 1451 | out->playback_started = 0; |
| 1452 | |
| 1453 | setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO); |
| 1454 | set_sched_policy(0, SP_FOREGROUND); |
| 1455 | prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0); |
| 1456 | |
| 1457 | ALOGV("%s", __func__); |
| 1458 | pthread_mutex_lock(&out->lock); |
| 1459 | for (;;) { |
| 1460 | struct offload_cmd *cmd = NULL; |
| 1461 | stream_callback_event_t event; |
| 1462 | bool send_callback = false; |
| 1463 | |
| 1464 | ALOGVV("%s offload_cmd_list %d out->offload_state %d", |
| 1465 | __func__, list_empty(&out->offload_cmd_list), |
| 1466 | out->offload_state); |
| 1467 | if (list_empty(&out->offload_cmd_list)) { |
| 1468 | ALOGV("%s SLEEPING", __func__); |
| 1469 | pthread_cond_wait(&out->offload_cond, &out->lock); |
| 1470 | ALOGV("%s RUNNING", __func__); |
| 1471 | continue; |
| 1472 | } |
| 1473 | |
| 1474 | item = list_head(&out->offload_cmd_list); |
| 1475 | cmd = node_to_item(item, struct offload_cmd, node); |
| 1476 | list_remove(item); |
| 1477 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1478 | ALOGVV("%s STATE %d CMD %d", |
| 1479 | __func__, out->offload_state, cmd->cmd); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1480 | |
| 1481 | if (cmd->cmd == OFFLOAD_CMD_EXIT) { |
| 1482 | free(cmd); |
| 1483 | break; |
| 1484 | } |
| 1485 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1486 | if (list_empty(&out->session_list)) { |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1487 | ALOGE("%s: Compress handle is NULL", __func__); |
| 1488 | pthread_cond_signal(&out->cond); |
| 1489 | continue; |
| 1490 | } |
| 1491 | out->offload_thread_blocked = true; |
| 1492 | pthread_mutex_unlock(&out->lock); |
| 1493 | send_callback = false; |
| 1494 | switch(cmd->cmd) { |
| 1495 | case OFFLOAD_CMD_WAIT_FOR_BUFFER: |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1496 | list_for_each(node, &out->session_list) { |
| 1497 | handle = node_to_item(node, struct alsa_handle, list); |
| 1498 | if (handle->compr && handle->cmd_pending) { |
| 1499 | compress_wait(handle->compr, -1); |
| 1500 | handle->cmd_pending = false; |
| 1501 | } |
| 1502 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1503 | send_callback = true; |
| 1504 | event = STREAM_CBK_EVENT_WRITE_READY; |
| 1505 | break; |
| 1506 | case OFFLOAD_CMD_PARTIAL_DRAIN: |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1507 | list_for_each(node, &out->session_list) { |
| 1508 | handle = node_to_item(node, struct alsa_handle, list); |
| 1509 | if (handle->compr) { |
| 1510 | compress_next_track(handle->compr); |
| 1511 | compress_partial_drain(handle->compr); |
| 1512 | } |
| 1513 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1514 | send_callback = true; |
| 1515 | event = STREAM_CBK_EVENT_DRAIN_READY; |
| 1516 | break; |
| 1517 | case OFFLOAD_CMD_DRAIN: |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1518 | list_for_each(node, &out->session_list) { |
| 1519 | handle = node_to_item(node, struct alsa_handle, list); |
| 1520 | if (handle->compr) { |
| 1521 | compress_drain(handle->compr); |
| 1522 | } |
| 1523 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1524 | send_callback = true; |
| 1525 | event = STREAM_CBK_EVENT_DRAIN_READY; |
| 1526 | break; |
| 1527 | default: |
| 1528 | ALOGE("%s unknown command received: %d", __func__, cmd->cmd); |
| 1529 | break; |
| 1530 | } |
| 1531 | pthread_mutex_lock(&out->lock); |
| 1532 | out->offload_thread_blocked = false; |
| 1533 | pthread_cond_signal(&out->cond); |
| 1534 | if (send_callback) { |
| 1535 | out->offload_callback(event, NULL, out->offload_cookie); |
| 1536 | } |
| 1537 | free(cmd); |
| 1538 | } |
| 1539 | |
| 1540 | pthread_cond_signal(&out->cond); |
| 1541 | while (!list_empty(&out->offload_cmd_list)) { |
| 1542 | item = list_head(&out->offload_cmd_list); |
| 1543 | list_remove(item); |
| 1544 | free(node_to_item(item, struct offload_cmd, node)); |
| 1545 | } |
| 1546 | pthread_mutex_unlock(&out->lock); |
| 1547 | |
| 1548 | return NULL; |
| 1549 | } |
| 1550 | |
| 1551 | static int create_offload_callback_thread(struct stream_out *out) |
| 1552 | { |
| 1553 | pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL); |
| 1554 | list_init(&out->offload_cmd_list); |
| 1555 | pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL, |
| 1556 | offload_thread_loop, out); |
| 1557 | return 0; |
| 1558 | } |
| 1559 | |
| 1560 | static int destroy_offload_callback_thread(struct stream_out *out) |
| 1561 | { |
| 1562 | pthread_mutex_lock(&out->lock); |
| 1563 | stop_compressed_output_l(out); |
| 1564 | send_offload_cmd_l(out, OFFLOAD_CMD_EXIT); |
| 1565 | |
| 1566 | pthread_mutex_unlock(&out->lock); |
| 1567 | pthread_join(out->offload_thread, (void **) NULL); |
| 1568 | pthread_cond_destroy(&out->offload_cond); |
| 1569 | |
| 1570 | return 0; |
| 1571 | } |
| 1572 | |
| 1573 | static bool allow_hdmi_channel_config(struct audio_device *adev) |
| 1574 | { |
| 1575 | struct listnode *node; |
| 1576 | struct audio_usecase *usecase; |
| 1577 | bool ret = true; |
| 1578 | |
| 1579 | list_for_each(node, &adev->usecase_list) { |
| 1580 | usecase = node_to_item(node, struct audio_usecase, list); |
| 1581 | if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { |
| 1582 | /* |
| 1583 | * If voice call is already existing, do not proceed further to avoid |
| 1584 | * disabling/enabling both RX and TX devices, CSD calls, etc. |
| 1585 | * Once the voice call done, the HDMI channels can be configured to |
| 1586 | * max channels of remaining use cases. |
| 1587 | */ |
| 1588 | if (usecase->id == USECASE_VOICE_CALL) { |
| 1589 | ALOGD("%s: voice call is active, no change in HDMI channels", |
| 1590 | __func__); |
| 1591 | ret = false; |
| 1592 | break; |
| 1593 | } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) { |
| 1594 | ALOGD("%s: multi channel playback is active, " |
| 1595 | "no change in HDMI channels", __func__); |
| 1596 | ret = false; |
| 1597 | break; |
| 1598 | } |
| 1599 | } |
| 1600 | } |
| 1601 | return ret; |
| 1602 | } |
| 1603 | |
| 1604 | static int check_and_set_hdmi_channels(struct audio_device *adev, |
| 1605 | unsigned int channels) |
| 1606 | { |
| 1607 | struct listnode *node; |
| 1608 | struct audio_usecase *usecase; |
| 1609 | |
| 1610 | /* Check if change in HDMI channel config is allowed */ |
| 1611 | if (!allow_hdmi_channel_config(adev)) |
| 1612 | return 0; |
| 1613 | |
| 1614 | if (channels == adev->cur_hdmi_channels) { |
| 1615 | ALOGD("%s: Requested channels are same as current", __func__); |
| 1616 | return 0; |
| 1617 | } |
| 1618 | |
| 1619 | platform_set_hdmi_channels(adev->platform, channels); |
| 1620 | adev->cur_hdmi_channels = channels; |
| 1621 | |
| 1622 | /* |
| 1623 | * Deroute all the playback streams routed to HDMI so that |
| 1624 | * the back end is deactivated. Note that backend will not |
| 1625 | * be deactivated if any one stream is connected to it. |
| 1626 | */ |
| 1627 | list_for_each(node, &adev->usecase_list) { |
| 1628 | usecase = node_to_item(node, struct audio_usecase, list); |
| 1629 | if (usecase->type == PCM_PLAYBACK && |
| 1630 | usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { |
| 1631 | disable_audio_route(adev, usecase, true); |
| 1632 | } |
| 1633 | } |
| 1634 | |
| 1635 | /* |
| 1636 | * Enable all the streams disabled above. Now the HDMI backend |
| 1637 | * will be activated with new channel configuration |
| 1638 | */ |
| 1639 | list_for_each(node, &adev->usecase_list) { |
| 1640 | usecase = node_to_item(node, struct audio_usecase, list); |
| 1641 | if (usecase->type == PCM_PLAYBACK && |
| 1642 | usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { |
| 1643 | enable_audio_route(adev, usecase, true); |
| 1644 | } |
| 1645 | } |
| 1646 | |
| 1647 | return 0; |
| 1648 | } |
| 1649 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1650 | static int stop_output_stream(struct stream_out *out, struct alsa_handle *handle) |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1651 | { |
| 1652 | int i, ret = 0; |
| 1653 | struct audio_usecase *uc_info; |
| 1654 | struct audio_device *adev = out->dev; |
| 1655 | |
| 1656 | ALOGV("%s: enter: usecase(%d: %s)", __func__, |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1657 | handle->usecase, use_case_table[handle->usecase]); |
| 1658 | uc_info = get_usecase_from_list(adev, handle->usecase); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1659 | if (uc_info == NULL) { |
| 1660 | ALOGE("%s: Could not find the usecase (%d) in the list", |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1661 | __func__, handle->usecase); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1662 | return -EINVAL; |
| 1663 | } |
| 1664 | |
| 1665 | if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD && |
| 1666 | adev->visualizer_stop_output != NULL) |
| 1667 | adev->visualizer_stop_output(out->handle); |
| 1668 | |
| 1669 | /* 1. Get and set stream specific mixer controls */ |
| 1670 | disable_audio_route(adev, uc_info, true); |
| 1671 | |
| 1672 | /* 2. Disable the rx device */ |
| 1673 | disable_snd_device(adev, uc_info->out_snd_device, true); |
| 1674 | |
| 1675 | list_remove(&uc_info->list); |
| 1676 | free(uc_info); |
| 1677 | |
| 1678 | /* Must be called after removing the usecase from list */ |
| 1679 | if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) |
| 1680 | check_and_set_hdmi_channels(adev, DEFAULT_HDMI_OUT_CHANNELS); |
| 1681 | |
| 1682 | ALOGV("%s: exit: status(%d)", __func__, ret); |
| 1683 | return ret; |
| 1684 | } |
| 1685 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1686 | int start_output_stream(struct stream_out *out, struct alsa_handle *handle) |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1687 | { |
| 1688 | int ret = 0; |
| 1689 | struct audio_usecase *uc_info; |
| 1690 | struct audio_device *adev = out->dev; |
| 1691 | |
| 1692 | ALOGV("%s: enter: usecase(%d: %s) devices(%#x)", |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1693 | __func__, handle->usecase, use_case_table[handle->usecase], handle->devices); |
| 1694 | handle->device_id = platform_get_pcm_device_id(handle->usecase, PCM_PLAYBACK); |
| 1695 | if (handle->device_id < 0) { |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1696 | ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)", |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1697 | __func__, handle->device_id, handle->usecase); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1698 | ret = -EINVAL; |
| 1699 | goto error_config; |
| 1700 | } |
| 1701 | |
| 1702 | uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase)); |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1703 | uc_info->id = handle->usecase; |
| 1704 | uc_info->handle = handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1705 | uc_info->type = PCM_PLAYBACK; |
| 1706 | uc_info->stream.out = out; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1707 | uc_info->devices = handle->devices; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1708 | uc_info->in_snd_device = SND_DEVICE_NONE; |
| 1709 | uc_info->out_snd_device = SND_DEVICE_NONE; |
| 1710 | |
| 1711 | /* This must be called before adding this usecase to the list */ |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1712 | //if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) |
| 1713 | // check_and_set_hdmi_channels(adev, out->config.channels); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1714 | |
| 1715 | list_add_tail(&adev->usecase_list, &uc_info->list); |
| 1716 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1717 | select_devices(adev, handle->usecase); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1718 | |
| 1719 | ALOGV("%s: Opening PCM device card_id(%d) device_id(%d)", |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1720 | __func__, 0, handle->device_id); |
| 1721 | if (out->uc_strm_type != OFFLOAD_PLAYBACK_STREAM) { |
| 1722 | handle->compr = NULL; |
| 1723 | handle->pcm = pcm_open(SOUND_CARD, handle->device_id, |
| 1724 | PCM_OUT | PCM_MONOTONIC, &handle->config); |
| 1725 | if (handle->pcm && !pcm_is_ready(handle->pcm)) { |
| 1726 | ALOGE("%s: %s", __func__, pcm_get_error(handle->pcm)); |
| 1727 | pcm_close(handle->pcm); |
| 1728 | handle->pcm = NULL; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1729 | ret = -EIO; |
| 1730 | goto error_open; |
| 1731 | } |
| 1732 | } else { |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1733 | handle->pcm = NULL; |
| 1734 | configure_compr(out, handle); |
| 1735 | handle->compr = compress_open(SOUND_CARD, handle->device_id, |
| 1736 | COMPRESS_IN, &handle->compr_config); |
| 1737 | if (handle->compr && !is_compress_ready(handle->compr)) { |
| 1738 | ALOGE("%s: %s", __func__, compress_get_error(handle->compr)); |
| 1739 | compress_close(handle->compr); |
| 1740 | handle->compr = NULL; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1741 | ret = -EIO; |
| 1742 | goto error_open; |
| 1743 | } |
| 1744 | if (out->offload_callback) |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1745 | compress_nonblock(handle->compr, out->non_blocking); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1746 | |
| 1747 | if (adev->visualizer_start_output != NULL) |
| 1748 | adev->visualizer_start_output(out->handle); |
| 1749 | } |
| 1750 | ALOGV("%s: exit", __func__); |
| 1751 | return 0; |
| 1752 | error_open: |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1753 | stop_output_stream(out, handle); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1754 | error_config: |
| 1755 | return ret; |
| 1756 | } |
| 1757 | |
| 1758 | static uint32_t out_get_sample_rate(const struct audio_stream *stream) |
| 1759 | { |
| 1760 | struct stream_out *out = (struct stream_out *)stream; |
| 1761 | |
| 1762 | return out->sample_rate; |
| 1763 | } |
| 1764 | |
| 1765 | static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) |
| 1766 | { |
| 1767 | return -ENOSYS; |
| 1768 | } |
| 1769 | |
| 1770 | static size_t out_get_buffer_size(const struct audio_stream *stream) |
| 1771 | { |
| 1772 | struct stream_out *out = (struct stream_out *)stream; |
| 1773 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1774 | /*if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1775 | return out->compr_config.fragment_size; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1776 | */ |
| 1777 | return (size_t)out->buffer_size; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1778 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1779 | //return out->config.period_size * audio_stream_frame_size(stream); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1780 | } |
| 1781 | |
| 1782 | static uint32_t out_get_channels(const struct audio_stream *stream) |
| 1783 | { |
| 1784 | struct stream_out *out = (struct stream_out *)stream; |
| 1785 | |
| 1786 | return out->channel_mask; |
| 1787 | } |
| 1788 | |
| 1789 | static audio_format_t out_get_format(const struct audio_stream *stream) |
| 1790 | { |
| 1791 | struct stream_out *out = (struct stream_out *)stream; |
| 1792 | |
| 1793 | return out->format; |
| 1794 | } |
| 1795 | |
| 1796 | static int out_set_format(struct audio_stream *stream, audio_format_t format) |
| 1797 | { |
| 1798 | return -ENOSYS; |
| 1799 | } |
| 1800 | |
| 1801 | static int out_standby(struct audio_stream *stream) |
| 1802 | { |
| 1803 | struct stream_out *out = (struct stream_out *)stream; |
| 1804 | struct audio_device *adev = out->dev; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1805 | struct listnode *node; |
| 1806 | struct alsa_handle *handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1807 | |
| 1808 | ALOGV("%s: enter: usecase(%d: %s)", __func__, |
| 1809 | out->usecase, use_case_table[out->usecase]); |
| 1810 | if (out->usecase == USECASE_COMPRESS_VOIP_CALL) { |
| 1811 | /* Ignore standby in case of voip call because the voip output |
| 1812 | * stream is closed in adev_close_output_stream() |
| 1813 | */ |
| 1814 | ALOGV("%s: Ignore Standby in VOIP call", __func__); |
| 1815 | return 0; |
| 1816 | } |
| 1817 | |
| 1818 | pthread_mutex_lock(&out->lock); |
| 1819 | pthread_mutex_lock(&adev->lock); |
| 1820 | if (!out->standby) { |
| 1821 | out->standby = true; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1822 | stop_compressed_output_l(out); |
| 1823 | out->gapless_mdata.encoder_delay = 0; |
| 1824 | out->gapless_mdata.encoder_padding = 0; |
| 1825 | |
| 1826 | list_for_each(node, &out->session_list) { |
| 1827 | handle = node_to_item(node, struct alsa_handle, list); |
| 1828 | if (handle->compr != NULL) { |
| 1829 | compress_close(handle->compr); |
| 1830 | handle->compr = NULL; |
| 1831 | } else if (handle->pcm) { |
| 1832 | pcm_close(handle->pcm); |
| 1833 | handle->pcm = NULL; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1834 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1835 | stop_output_stream(out, handle); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1836 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1837 | } |
| 1838 | pthread_mutex_unlock(&adev->lock); |
| 1839 | pthread_mutex_unlock(&out->lock); |
| 1840 | ALOGV("%s: exit", __func__); |
| 1841 | return 0; |
| 1842 | } |
| 1843 | |
| 1844 | static int out_dump(const struct audio_stream *stream, int fd) |
| 1845 | { |
| 1846 | return 0; |
| 1847 | } |
| 1848 | |
| 1849 | static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms) |
| 1850 | { |
| 1851 | int ret = 0; |
| 1852 | char value[32]; |
| 1853 | struct compr_gapless_mdata tmp_mdata; |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1854 | bool gapless_meta_set = true; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1855 | |
| 1856 | if (!out || !parms) { |
| 1857 | return -EINVAL; |
| 1858 | } |
| 1859 | |
| 1860 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value)); |
| 1861 | if (ret >= 0) { |
| 1862 | tmp_mdata.encoder_delay = atoi(value); //whats a good limit check? |
| 1863 | } else { |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1864 | gapless_meta_set = false; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1865 | } |
| 1866 | |
| 1867 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value)); |
| 1868 | if (ret >= 0) { |
| 1869 | tmp_mdata.encoder_padding = atoi(value); |
| 1870 | } else { |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1871 | gapless_meta_set = false; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1872 | } |
| 1873 | |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1874 | if (gapless_meta_set) { |
| 1875 | out->gapless_mdata = tmp_mdata; |
| 1876 | out->send_new_metadata = 1; |
| 1877 | ALOGV("%s new encoder delay %u and padding %u", __func__, |
| 1878 | out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding); |
| 1879 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1880 | |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 1881 | if(out->format == AUDIO_FORMAT_WMA || out->format == AUDIO_FORMAT_WMA_PRO) { |
| 1882 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_FORMAT_TAG, value, sizeof(value)); |
| 1883 | if (ret >= 0) { |
| 1884 | out->compr_config.codec->format = atoi(value); |
| 1885 | } |
| 1886 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_BLOCK_ALIGN, value, sizeof(value)); |
| 1887 | if (ret >= 0) { |
| 1888 | out->compr_config.codec->options.wma.super_block_align = atoi(value); |
| 1889 | } |
| 1890 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_BIT_PER_SAMPLE, value, sizeof(value)); |
| 1891 | if (ret >= 0) { |
| 1892 | out->compr_config.codec->options.wma.bits_per_sample = atoi(value); |
| 1893 | } |
| 1894 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_CHANNEL_MASK, value, sizeof(value)); |
| 1895 | if (ret >= 0) { |
| 1896 | out->compr_config.codec->options.wma.channelmask = atoi(value); |
| 1897 | } |
| 1898 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION, value, sizeof(value)); |
| 1899 | if (ret >= 0) { |
| 1900 | out->compr_config.codec->options.wma.encodeopt = atoi(value); |
| 1901 | } |
| 1902 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION1, value, sizeof(value)); |
| 1903 | if (ret >= 0) { |
| 1904 | out->compr_config.codec->options.wma.encodeopt1 = atoi(value); |
| 1905 | } |
| 1906 | ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_WMA_ENCODE_OPTION2, value, sizeof(value)); |
| 1907 | if (ret >= 0) { |
| 1908 | out->compr_config.codec->options.wma.encodeopt2 = atoi(value); |
| 1909 | } |
| 1910 | ALOGV("WMA params: fmt %x, balgn %x, sr %d, chmsk %x, encop %x, op1 %x, op2 %x", |
| 1911 | out->compr_config.codec->format, |
| 1912 | out->compr_config.codec->options.wma.super_block_align, |
| 1913 | out->compr_config.codec->options.wma.bits_per_sample, |
| 1914 | out->compr_config.codec->options.wma.channelmask, |
| 1915 | out->compr_config.codec->options.wma.encodeopt, |
| 1916 | out->compr_config.codec->options.wma.encodeopt1, |
| 1917 | out->compr_config.codec->options.wma.encodeopt2); |
| 1918 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1919 | return 0; |
| 1920 | } |
| 1921 | |
| 1922 | static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) |
| 1923 | { |
| 1924 | struct stream_out *out = (struct stream_out *)stream; |
| 1925 | struct audio_device *adev = out->dev; |
| 1926 | struct audio_usecase *usecase; |
| 1927 | struct listnode *node; |
| 1928 | struct str_parms *parms; |
| 1929 | char value[32]; |
| 1930 | int ret, val = 0; |
| 1931 | bool select_new_device = false; |
| 1932 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1933 | ALOGD("%s: enter: kvpairs: %s", __func__, kvpairs); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 1934 | parms = str_parms_create_str(kvpairs); |
| 1935 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); |
| 1936 | if (ret >= 0) { |
| 1937 | val = atoi(value); |
| 1938 | pthread_mutex_lock(&out->lock); |
| 1939 | pthread_mutex_lock(&adev->lock); |
| 1940 | |
| 1941 | /* |
| 1942 | * When HDMI cable is unplugged the music playback is paused and |
| 1943 | * the policy manager sends routing=0. But the audioflinger |
| 1944 | * continues to write data until standby time (3sec). |
| 1945 | * As the HDMI core is turned off, the write gets blocked. |
| 1946 | * Avoid this by routing audio to speaker until standby. |
| 1947 | */ |
| 1948 | if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL && |
| 1949 | val == AUDIO_DEVICE_NONE) { |
| 1950 | val = AUDIO_DEVICE_OUT_SPEAKER; |
| 1951 | } |
| 1952 | |
| 1953 | /* |
| 1954 | * select_devices() call below switches all the usecases on the same |
| 1955 | * backend to the new device. Refer to check_usecases_codec_backend() in |
| 1956 | * the select_devices(). But how do we undo this? |
| 1957 | * |
| 1958 | * For example, music playback is active on headset (deep-buffer usecase) |
| 1959 | * and if we go to ringtones and select a ringtone, low-latency usecase |
| 1960 | * will be started on headset+speaker. As we can't enable headset+speaker |
| 1961 | * and headset devices at the same time, select_devices() switches the music |
| 1962 | * playback to headset+speaker while starting low-lateny usecase for ringtone. |
| 1963 | * So when the ringtone playback is completed, how do we undo the same? |
| 1964 | * |
| 1965 | * We are relying on the out_set_parameters() call on deep-buffer output, |
| 1966 | * once the ringtone playback is ended. |
| 1967 | * NOTE: We should not check if the current devices are same as new devices. |
| 1968 | * Because select_devices() must be called to switch back the music |
| 1969 | * playback to headset. |
| 1970 | */ |
| 1971 | if (val != 0) { |
| 1972 | out->devices = val; |
| 1973 | |
| 1974 | if (!out->standby) |
| 1975 | select_devices(adev, out->usecase); |
| 1976 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 1977 | //TODO: |
| 1978 | //Get the device and device format mapping from the RoutingManager. |
| 1979 | //Decide which streams need to be derouted and which need to opened/closed |
| 1980 | //Update the respective device in each of the handles |
| 1981 | #if 0 |
| 1982 | if (out->uc_strm_type == OFFLOAD_PLAYBACK_STREAM) { |
| 1983 | |
| 1984 | /* TODO get format form routing manager */ |
| 1985 | update_decode_type_and_routing_states(out); |
| 1986 | |
| 1987 | if(is_input_buffering_mode_reqd(out)) |
| 1988 | audio_bitstream_start_input_buffering_mode(out->bitstrm); |
| 1989 | else |
| 1990 | audio_bitstream_stop_input_buffering_mode(out->bitstrm); |
| 1991 | /* |
| 1992 | For the runtime format change, close the device first to avoid any |
| 1993 | concurrent PCM + Compressed sessions on the same device. |
| 1994 | */ |
| 1995 | close_handles_for_device_switch(out); |
| 1996 | if(!out->mopen_dec_route) |
| 1997 | handleCloseForDeviceSwitch(ROUTE_UNCOMPRESSED); |
| 1998 | |
| 1999 | if(!out->mopen_dec_mch_route) |
| 2000 | handleCloseForDeviceSwitch(ROUTE_UNCOMPRESSED_MCH); |
| 2001 | |
| 2002 | if(!out->mopen_passt_route) |
| 2003 | handleCloseForDeviceSwitch(ROUTE_COMPRESSED); |
| 2004 | |
| 2005 | if(!msw_open_trans_route) |
| 2006 | handleCloseForDeviceSwitch(ROUTE_SW_TRANSCODED_COMPRESSED); |
| 2007 | |
| 2008 | if(!mhw_open_trans_route) |
| 2009 | handleCloseForDeviceSwitch(ROUTE_DSP_TRANSCODED_COMPRESSED); |
| 2010 | |
| 2011 | if(out->mopen_dec_route) |
| 2012 | handleSwitchAndOpenForDeviceSwitch(mdec_format_devices, |
| 2013 | ROUTE_UNCOMPRESSED); |
| 2014 | if(out->mopen_dec_mch_route) |
| 2015 | handleSwitchAndOpenForDeviceSwitch(mdec_mch_format_devices, |
| 2016 | ROUTE_UNCOMPRESSED_MCH); |
| 2017 | if(out->mopen_passt_route) |
| 2018 | handleSwitchAndOpenForDeviceSwitch(mpasst_format_devices, |
| 2019 | ROUTE_COMPRESSED); |
| 2020 | if(out->msw_open_trans_route) |
| 2021 | handleSwitchAndOpenForDeviceSwitch(msw_trans_format_devices, |
| 2022 | ROUTE_SW_TRANSCODED_COMPRESSED); |
| 2023 | if(out->mhw_open_trans_route) |
| 2024 | handleSwitchAndOpenForDeviceSwitch(mhw_trans_format_devices, |
| 2025 | ROUTE_DSP_TRANSCODED_COMPRESSED); |
| 2026 | } |
| 2027 | #endif |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2028 | |
| 2029 | pthread_mutex_unlock(&adev->lock); |
| 2030 | pthread_mutex_unlock(&out->lock); |
| 2031 | } |
| 2032 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2033 | if (out->uc_strm_type == OFFLOAD_PLAYBACK_STREAM) { |
Dhananjay Kumar | 5a553e4 | 2013-12-03 23:06:49 +0530 | [diff] [blame] | 2034 | ret = parse_compress_metadata(out, parms); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2035 | } |
| 2036 | |
| 2037 | str_parms_destroy(parms); |
| 2038 | ALOGV("%s: exit: code(%d)", __func__, ret); |
| 2039 | return ret; |
| 2040 | } |
| 2041 | |
| 2042 | static char* out_get_parameters(const struct audio_stream *stream, const char *keys) |
| 2043 | { |
| 2044 | struct stream_out *out = (struct stream_out *)stream; |
| 2045 | struct str_parms *query = str_parms_create_str(keys); |
| 2046 | char *str; |
| 2047 | char value[256]; |
| 2048 | struct str_parms *reply = str_parms_create(); |
| 2049 | size_t i, j; |
| 2050 | int ret; |
| 2051 | bool first = true; |
| 2052 | ALOGV("%s: enter: keys - %s", __func__, keys); |
| 2053 | ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value)); |
| 2054 | if (ret >= 0) { |
| 2055 | value[0] = '\0'; |
| 2056 | i = 0; |
| 2057 | while (out->supported_channel_masks[i] != 0) { |
| 2058 | for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) { |
| 2059 | if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) { |
| 2060 | if (!first) { |
| 2061 | strlcat(value, "|", sizeof(value)); |
| 2062 | } |
| 2063 | strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value)); |
| 2064 | first = false; |
| 2065 | break; |
| 2066 | } |
| 2067 | } |
| 2068 | i++; |
| 2069 | } |
| 2070 | str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value); |
| 2071 | str = str_parms_to_str(reply); |
| 2072 | } |
| 2073 | str_parms_destroy(query); |
| 2074 | str_parms_destroy(reply); |
| 2075 | ALOGV("%s: exit: returns - %s", __func__, str); |
| 2076 | return str; |
| 2077 | } |
| 2078 | |
| 2079 | static uint32_t out_get_latency(const struct audio_stream_out *stream) |
| 2080 | { |
| 2081 | struct stream_out *out = (struct stream_out *)stream; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2082 | struct listnode *item; |
| 2083 | struct alsa_handle *handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2084 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2085 | //TODO: decide based on the clip properties |
| 2086 | if (out->uc_strm_type == OFFLOAD_PLAYBACK_STREAM) |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2087 | return COMPRESS_OFFLOAD_PLAYBACK_LATENCY; |
| 2088 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2089 | item = list_head(&out->session_list); |
| 2090 | handle = node_to_item(item, struct alsa_handle, list); |
| 2091 | if(!handle) { |
| 2092 | ALOGE("%s: error pcm handle NULL", __func__); |
| 2093 | return -EINVAL; |
| 2094 | } |
| 2095 | |
| 2096 | return (handle->config.period_count * handle->config.period_size * 1000) / |
| 2097 | (handle->config.rate); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2098 | } |
| 2099 | |
| 2100 | static int out_set_volume(struct audio_stream_out *stream, float left, |
| 2101 | float right) |
| 2102 | { |
| 2103 | struct stream_out *out = (struct stream_out *)stream; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2104 | struct listnode *node; |
| 2105 | struct alsa_handle *handle; |
| 2106 | struct audio_device *adev = out->dev; |
| 2107 | int ret = -ENOSYS; |
| 2108 | ALOGV("%s", __func__); |
| 2109 | pthread_mutex_lock(&out->lock); |
| 2110 | list_for_each(node, &out->session_list) { |
| 2111 | handle = node_to_item(node, struct alsa_handle, list); |
| 2112 | if (handle->pcm && (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH)){ |
| 2113 | /* only take left channel into account: the API is for stereo anyway */ |
| 2114 | out->muted = (left == 0.0f); |
| 2115 | ret = 0; |
| 2116 | } else if (handle->compr) { |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2117 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2118 | out->left_volume = left; |
| 2119 | out->right_volume = right; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2120 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2121 | //ret = set_compress_volume(handle, left, right); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2122 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2123 | } |
| 2124 | pthread_mutex_unlock(&out->lock); |
| 2125 | |
| 2126 | return ret; |
| 2127 | } |
| 2128 | |
| 2129 | static int write_data(struct stream_out *out, struct alsa_handle *handle, |
| 2130 | const void *buffer, int bytes) { |
| 2131 | |
| 2132 | int ret = 0; |
| 2133 | if (out->uc_strm_type == OFFLOAD_PLAYBACK_STREAM) { |
| 2134 | ALOGV("%s: writing buffer (%d bytes) to compress device", __func__, bytes); |
| 2135 | |
| 2136 | ret = compress_write(handle->compr, buffer, bytes); |
| 2137 | ALOGV("%s: writing buffer (%d bytes) to compress device returned %d", |
| 2138 | __func__, bytes, ret); |
| 2139 | /* TODO:disnable this if ms12 */ |
| 2140 | |
| 2141 | if (ret >= 0 && ret < (ssize_t)bytes) { |
Aviral Gupta | fdb08f4 | 2013-12-21 00:33:47 -0800 | [diff] [blame] | 2142 | handle->cmd_pending = true; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2143 | send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); |
| 2144 | } |
| 2145 | return ret; |
| 2146 | } else { |
| 2147 | if (handle->pcm) { |
| 2148 | if (out->muted) |
| 2149 | memset((void *)buffer, 0, bytes); |
| 2150 | ALOGV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes); |
| 2151 | ret = pcm_write(handle->pcm, (void *)buffer, bytes); |
| 2152 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2153 | } |
| 2154 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2155 | if (ret != 0) { |
| 2156 | if ((handle && handle->pcm)) |
| 2157 | ALOGE("%s: error %d - %s", __func__, ret, pcm_get_error(handle->pcm)); |
| 2158 | out_standby(&out->stream.common); |
| 2159 | usleep(bytes * 1000000 / audio_stream_frame_size(&out->stream.common) / |
| 2160 | out_get_sample_rate(&out->stream.common)); |
| 2161 | } |
| 2162 | return bytes; |
| 2163 | } |
| 2164 | |
| 2165 | /******************************************************************************* |
| 2166 | Description: render |
| 2167 | *******************************************************************************/ |
| 2168 | size_t render_offload_data(struct stream_out *out, const void *buffer, size_t bytes) |
| 2169 | { |
| 2170 | int ret =0; |
| 2171 | uint32_t renderedPcmBytes = 0; |
| 2172 | int fragment_size; |
| 2173 | uint32_t availableSize; |
| 2174 | int bytes_to_write = bytes; |
| 2175 | int renderType; |
| 2176 | /*int metadataLength = sizeof(out->output_meta_data);*/ |
| 2177 | struct listnode *node; |
| 2178 | struct alsa_handle *handle; |
| 2179 | |
| 2180 | ALOGV("%s", __func__); |
| 2181 | |
| 2182 | list_for_each(node, &out->session_list) { |
| 2183 | handle = node_to_item(node, struct alsa_handle, list); |
| 2184 | if (out->send_new_metadata) { |
| 2185 | ALOGVV("send new gapless metadata"); |
| 2186 | compress_set_gapless_metadata(handle->compr, &out->gapless_mdata); |
| 2187 | } |
| 2188 | |
| 2189 | switch(handle->route_format) { |
| 2190 | case ROUTE_UNCOMPRESSED: |
| 2191 | ALOGVV("ROUTE_UNCOMPRESSED"); |
| 2192 | renderType = PCM_2CH_OUT; |
| 2193 | break; |
| 2194 | case ROUTE_UNCOMPRESSED_MCH: |
| 2195 | ALOGVV("ROUTE_UNCOMPRESSED_MCH"); |
| 2196 | renderType = PCM_MCH_OUT; |
| 2197 | break; |
| 2198 | case ROUTE_COMPRESSED: |
| 2199 | ALOGVV("ROUTE_COMPRESSED"); |
| 2200 | renderType = COMPRESSED_OUT; |
| 2201 | break; |
| 2202 | case ROUTE_SW_TRANSCODED_COMPRESSED: |
| 2203 | ALOGVV("ROUTE_SW_TRANSCODED_COMPRESSED"); |
| 2204 | renderType = TRANSCODE_OUT; |
| 2205 | break; |
| 2206 | case ROUTE_DSP_TRANSCODED_COMPRESSED: |
| 2207 | ALOGVV("ROUTE_DSP_TRANSCODED_COMPRESSED"); |
| 2208 | continue; |
| 2209 | default: |
| 2210 | continue; |
| 2211 | }; |
| 2212 | |
| 2213 | fragment_size = handle->compr_config.fragment_size; |
| 2214 | /*TODO handle timestamp case */ |
| 2215 | #if USE_SWDECODE |
| 2216 | while(audio_bitstream_sufficient_sample_to_render(out->bitstrm, |
| 2217 | renderType, 1) == true) { |
| 2218 | availableSize = audio_bitstream_get_output_buffer_write_ptr(out->bitstrm, renderType) - |
| 2219 | audio_bitstream_get_output_buffer_ptr(out->bitstrm, renderType); |
| 2220 | buffer = audio_bitstream_get_output_buffer_ptr(out->bitstrm, renderType); |
| 2221 | bytes_to_write = availableSize; |
| 2222 | |
| 2223 | TODO: meta data is only neded for TS mode |
| 2224 | out->output_meta_data.metadataLength = metadataLength; |
| 2225 | out->output_meta_data.bufferLength = (availableSize >= |
| 2226 | (fragment_size - metadataLength)) ? |
| 2227 | fragment_size - metadataLength : |
| 2228 | availableSize; |
| 2229 | bytes_to_write = metadataLength +out->output_meta_data.bufferLength; |
| 2230 | out->output_meta_data.timestamp = 0; |
| 2231 | memcpy(out->write_temp_buf, &out->output_meta_data, metadataLength); |
| 2232 | memcpy(out->write_temp_buf+metadataLength, |
| 2233 | audio_bitstream_get_output_buffer_ptr(out->bitstrm, renderType), |
| 2234 | out->output_meta_data.bufferLength); |
| 2235 | ret = write_data(out, handle, out->write_temp_buf, bytes_to_write); |
| 2236 | #endif |
| 2237 | |
| 2238 | ret = write_data(out, handle, buffer, bytes_to_write); |
| 2239 | ALOGD("write_data returned with %d", ret); |
| 2240 | if(ret < 0) { |
| 2241 | ALOGE("write_data returned ret < 0"); |
| 2242 | return ret; |
| 2243 | } else { |
| 2244 | if (!out->playback_started) { |
| 2245 | compress_start(handle->compr); |
| 2246 | } |
| 2247 | /*TODO:Do we need this |
| 2248 | if(renderType == ROUTE_UNCOMPRESSED || |
| 2249 | (renderType == ROUTE_UNCOMPRESSED_MCH && !out->open_dec_route)) { |
| 2250 | mFrameCount++; |
| 2251 | renderedPcmBytes += out->output_meta_data.bufferLength; |
| 2252 | }*/ |
| 2253 | renderedPcmBytes += ret; |
| 2254 | #if USE_SWDECODE |
| 2255 | /*iTODO: enable for MS11 |
| 2256 | audio_bitstream_copy_residue_output_start(out->bitstrm, renderType, |
| 2257 | bytes_to_write); |
| 2258 | TODO:what if ret<bytes_to_write*/ |
| 2259 | #endif |
| 2260 | } |
| 2261 | #if USE_SWDECODE |
| 2262 | } |
| 2263 | #endif |
| 2264 | } |
| 2265 | out->playback_started = 1; |
| 2266 | out->offload_state = OFFLOAD_STATE_PLAYING; |
| 2267 | out->send_new_metadata = 0; |
| 2268 | return renderedPcmBytes; |
| 2269 | } |
| 2270 | |
| 2271 | size_t render_pcm_data(struct stream_out *out, const void *buffer, size_t bytes) |
| 2272 | { |
| 2273 | ALOGV("%s", __func__); |
| 2274 | size_t ret = 0; |
| 2275 | struct listnode *node; |
| 2276 | struct alsa_handle *handle; |
| 2277 | list_for_each(node, &out->session_list) { |
| 2278 | handle = node_to_item(node, struct alsa_handle, list); |
| 2279 | ALOGV("%s handle is 0x%x", __func__,(uint32_t)handle); |
| 2280 | ret = write_data(out, handle, buffer, bytes); |
| 2281 | } |
| 2282 | return ret; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2283 | } |
| 2284 | |
| 2285 | static ssize_t out_write(struct audio_stream_out *stream, const void *buffer, |
| 2286 | size_t bytes) |
| 2287 | { |
| 2288 | struct stream_out *out = (struct stream_out *)stream; |
| 2289 | struct audio_device *adev = out->dev; |
| 2290 | ssize_t ret = 0; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2291 | struct listnode *node; |
| 2292 | bool continueDecode; |
| 2293 | struct alsa_handle *handle; |
| 2294 | size_t bytes_consumed; |
| 2295 | size_t total_bytes_consumed = 0; |
| 2296 | |
| 2297 | ALOGV("%s bytes =%d", __func__, bytes); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2298 | |
| 2299 | pthread_mutex_lock(&out->lock); |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2300 | |
| 2301 | //TODO: handle a2dp |
| 2302 | /* if (mRouteAudioToA2dp && |
| 2303 | mA2dpUseCase == AudioHardwareALSA::USECASE_NONE) { |
| 2304 | a2dpRenderingControl(A2DP_RENDER_SETUP); |
| 2305 | } |
| 2306 | */ |
| 2307 | /* TODO: meta data comes in set_parameter it will be passed in compre_open |
| 2308 | for all format exxce ms11 format |
| 2309 | and for ms11 it will be set sdecode fucntion while opneing ms11 instance |
| 2310 | hence below piece of code is no required*/ |
| 2311 | /* |
| 2312 | if(!out->dec_conf_set && is_decoder_config_required(out)) { |
| 2313 | if (setDecodeConfig(out, (char *)buffer, bytes)) |
| 2314 | ALOGD("decoder configuration set"); |
| 2315 | } |
| 2316 | */ |
| 2317 | |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2318 | if (out->standby) { |
| 2319 | out->standby = false; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2320 | list_for_each(node, &out->session_list) { |
| 2321 | handle = node_to_item(node, struct alsa_handle, list); |
| 2322 | pthread_mutex_lock(&adev->lock); |
| 2323 | ret = start_output_stream(out, handle); |
| 2324 | pthread_mutex_unlock(&adev->lock); |
| 2325 | /* ToDo: If use case is compress offload should return 0 */ |
| 2326 | if (ret != 0) { |
| 2327 | out->standby = true; |
| 2328 | goto exit; |
| 2329 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2330 | } |
| 2331 | } |
| 2332 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2333 | if (out->uc_strm_type == OFFLOAD_PLAYBACK_STREAM) { |
| 2334 | #if USE_SWDECODE |
| 2335 | //TODO: Enable for MS11 |
| 2336 | copy_bitstream_internal_buffer(out->bitstrm, (char *)buffer, bytes); |
| 2337 | //DO check if timestamp mode handle partial buffer |
| 2338 | do { |
| 2339 | |
| 2340 | bytes_consumed = 0; |
| 2341 | ret = decode(out, (char *)buffer, bytes, &bytes_consumed, &continueDecode); |
| 2342 | if(ret < 0) |
| 2343 | goto exit; |
| 2344 | /*TODO: check for return size from write when ms11 is removed*/ |
| 2345 | render_offload_data(out, continueDecode); |
| 2346 | total_bytes_consumed += bytes_consumed; |
| 2347 | |
| 2348 | } while(continueDecode == true); |
| 2349 | #endif |
| 2350 | #if 0 |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2351 | ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes); |
| 2352 | if (out->send_new_metadata) { |
| 2353 | ALOGVV("send new gapless metadata"); |
| 2354 | compress_set_gapless_metadata(out->compr, &out->gapless_mdata); |
| 2355 | out->send_new_metadata = 0; |
| 2356 | } |
| 2357 | |
| 2358 | ret = compress_write(out->compr, buffer, bytes); |
| 2359 | ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret); |
| 2360 | if (ret >= 0 && ret < (ssize_t)bytes) { |
| 2361 | send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); |
| 2362 | } |
| 2363 | if (!out->playback_started) { |
| 2364 | compress_start(out->compr); |
| 2365 | out->playback_started = 1; |
| 2366 | out->offload_state = OFFLOAD_STATE_PLAYING; |
| 2367 | } |
| 2368 | pthread_mutex_unlock(&out->lock); |
| 2369 | return ret; |
| 2370 | } else { |
| 2371 | if (out->pcm) { |
| 2372 | if (out->muted) |
| 2373 | memset((void *)buffer, 0, bytes); |
| 2374 | ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes); |
| 2375 | ret = pcm_write(out->pcm, (void *)buffer, bytes); |
| 2376 | if (ret == 0) |
| 2377 | out->written += bytes / (out->config.channels * sizeof(short)); |
| 2378 | } |
| 2379 | } |
| 2380 | |
| 2381 | exit: |
| 2382 | pthread_mutex_unlock(&out->lock); |
| 2383 | |
| 2384 | if (ret != 0) { |
| 2385 | if (out->pcm) |
| 2386 | ALOGE("%s: error %d - %s", __func__, ret, pcm_get_error(out->pcm)); |
| 2387 | out_standby(&out->stream.common); |
| 2388 | usleep(bytes * 1000000 / audio_stream_frame_size(&out->stream.common) / |
| 2389 | out_get_sample_rate(&out->stream.common)); |
| 2390 | } |
| 2391 | return bytes; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2392 | #endif |
| 2393 | ret = render_offload_data(out, buffer, bytes); |
| 2394 | total_bytes_consumed = ret; |
| 2395 | } else { |
| 2396 | ret = render_pcm_data(out, buffer, bytes); |
| 2397 | total_bytes_consumed = ret; |
| 2398 | } |
| 2399 | |
| 2400 | exit: |
| 2401 | pthread_mutex_unlock(&out->lock); |
| 2402 | ALOGV("total_bytes_consumed %d",total_bytes_consumed); |
| 2403 | return total_bytes_consumed; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2404 | } |
| 2405 | |
| 2406 | static int out_get_render_position(const struct audio_stream_out *stream, |
| 2407 | uint32_t *dsp_frames) |
| 2408 | { |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2409 | struct listnode *node; |
| 2410 | struct alsa_handle *handle; |
| 2411 | struct stream_out *out = (struct stream_out *)stream; |
| 2412 | struct audio_device *adev = out->dev; |
| 2413 | *dsp_frames = 0; |
| 2414 | ALOGV("%s", __func__); |
| 2415 | pthread_mutex_lock(&out->lock); |
| 2416 | if ((out->uc_strm_type == OFFLOAD_PLAYBACK_STREAM) && (dsp_frames != NULL)) { |
| 2417 | list_for_each(node, &out->session_list) { |
| 2418 | handle = node_to_item(node, struct alsa_handle, list); |
| 2419 | if ((handle && handle->compr && |
| 2420 | handle->route_format != ROUTE_DSP_TRANSCODED_COMPRESSED)){ |
| 2421 | compress_get_tstamp(handle->compr, (unsigned long *)dsp_frames, |
| 2422 | &out->sample_rate); |
| 2423 | ALOGV("%s rendered frames %d sample_rate %d", |
| 2424 | __func__, *dsp_frames, out->sample_rate); |
| 2425 | } |
| 2426 | pthread_mutex_unlock(&out->lock); |
| 2427 | return 0; |
| 2428 | } |
| 2429 | } |
| 2430 | else { |
| 2431 | pthread_mutex_unlock(&out->lock); |
| 2432 | return -EINVAL; |
| 2433 | } |
| 2434 | return 0; |
| 2435 | #if 0 |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2436 | struct stream_out *out = (struct stream_out *)stream; |
| 2437 | *dsp_frames = 0; |
| 2438 | if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) { |
| 2439 | pthread_mutex_lock(&out->lock); |
| 2440 | if (out->compr != NULL) { |
| 2441 | compress_get_tstamp(out->compr, (unsigned long *)dsp_frames, |
| 2442 | &out->sample_rate); |
| 2443 | ALOGVV("%s rendered frames %d sample_rate %d", |
| 2444 | __func__, *dsp_frames, out->sample_rate); |
| 2445 | } |
| 2446 | pthread_mutex_unlock(&out->lock); |
| 2447 | return 0; |
| 2448 | } else |
| 2449 | return -EINVAL; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2450 | #endif |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2451 | } |
| 2452 | |
| 2453 | static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) |
| 2454 | { |
| 2455 | return 0; |
| 2456 | } |
| 2457 | |
| 2458 | static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) |
| 2459 | { |
| 2460 | return 0; |
| 2461 | } |
| 2462 | |
| 2463 | static int out_get_next_write_timestamp(const struct audio_stream_out *stream, |
| 2464 | int64_t *timestamp) |
| 2465 | { |
| 2466 | return -EINVAL; |
| 2467 | } |
| 2468 | |
| 2469 | static int out_get_presentation_position(const struct audio_stream_out *stream, |
| 2470 | uint64_t *frames, struct timespec *timestamp) |
| 2471 | { |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2472 | struct listnode *node; |
| 2473 | struct alsa_handle *handle; |
| 2474 | struct stream_out *out = (struct stream_out *)stream; |
| 2475 | struct audio_device *adev = out->dev; |
| 2476 | *frames = 0; |
| 2477 | ALOGV("%s", __func__); |
| 2478 | pthread_mutex_lock(&out->lock); |
| 2479 | if ((frames != NULL)) { |
| 2480 | list_for_each(node, &out->session_list) { |
| 2481 | handle = node_to_item(node, struct alsa_handle, list); |
| 2482 | if ((handle && handle->compr && |
| 2483 | handle->route_format != ROUTE_DSP_TRANSCODED_COMPRESSED)){ |
| 2484 | compress_get_tstamp(handle->compr, (unsigned long *)frames, |
| 2485 | &out->sample_rate); |
| 2486 | clock_gettime(CLOCK_MONOTONIC, timestamp); |
| 2487 | ALOGV("%s rendered frames %d sample_rate %d", |
| 2488 | __func__, *frames, out->sample_rate); |
| 2489 | } |
| 2490 | else if (handle->pcm) { |
| 2491 | size_t avail; |
| 2492 | if (pcm_get_htimestamp(handle->pcm, &avail, timestamp) == 0) { |
| 2493 | size_t kernel_buffer_size = handle->config.period_size * handle->config.period_count; |
| 2494 | int64_t signed_frames = out->written - kernel_buffer_size + avail; |
| 2495 | // This adjustment accounts for buffering after app processor. |
| 2496 | // It is based on estimated DSP latency per use case, rather than exact. |
| 2497 | signed_frames -= |
| 2498 | (platform_render_latency(handle->usecase) * out->sample_rate / 1000000LL); |
| 2499 | |
| 2500 | // It would be unusual for this value to be negative, but check just in case ... |
| 2501 | if (signed_frames >= 0) { |
| 2502 | *frames = signed_frames; |
| 2503 | } |
| 2504 | } |
| 2505 | } |
| 2506 | |
| 2507 | } |
| 2508 | } |
| 2509 | pthread_mutex_unlock(&out->lock); |
| 2510 | return -EINVAL; |
| 2511 | #if 0 |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2512 | struct stream_out *out = (struct stream_out *)stream; |
| 2513 | int ret = -1; |
| 2514 | unsigned long dsp_frames; |
| 2515 | |
| 2516 | pthread_mutex_lock(&out->lock); |
| 2517 | |
| 2518 | if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { |
| 2519 | if (out->compr != NULL) { |
| 2520 | compress_get_tstamp(out->compr, &dsp_frames, |
| 2521 | &out->sample_rate); |
| 2522 | ALOGVV("%s rendered frames %ld sample_rate %d", |
| 2523 | __func__, dsp_frames, out->sample_rate); |
| 2524 | *frames = dsp_frames; |
| 2525 | ret = 0; |
| 2526 | /* this is the best we can do */ |
| 2527 | clock_gettime(CLOCK_MONOTONIC, timestamp); |
| 2528 | } |
| 2529 | } else { |
| 2530 | if (out->pcm) { |
| 2531 | size_t avail; |
| 2532 | if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) { |
| 2533 | size_t kernel_buffer_size = out->config.period_size * out->config.period_count; |
| 2534 | int64_t signed_frames = out->written - kernel_buffer_size + avail; |
| 2535 | // This adjustment accounts for buffering after app processor. |
| 2536 | // It is based on estimated DSP latency per use case, rather than exact. |
| 2537 | signed_frames -= |
| 2538 | (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL); |
| 2539 | |
| 2540 | // It would be unusual for this value to be negative, but check just in case ... |
| 2541 | if (signed_frames >= 0) { |
| 2542 | *frames = signed_frames; |
| 2543 | ret = 0; |
| 2544 | } |
| 2545 | } |
| 2546 | } |
| 2547 | } |
| 2548 | |
| 2549 | pthread_mutex_unlock(&out->lock); |
| 2550 | |
| 2551 | return ret; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2552 | #endif |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2553 | } |
| 2554 | |
| 2555 | static int out_set_callback(struct audio_stream_out *stream, |
| 2556 | stream_callback_t callback, void *cookie) |
| 2557 | { |
| 2558 | struct stream_out *out = (struct stream_out *)stream; |
| 2559 | |
| 2560 | ALOGV("%s", __func__); |
| 2561 | pthread_mutex_lock(&out->lock); |
| 2562 | out->offload_callback = callback; |
| 2563 | out->offload_cookie = cookie; |
| 2564 | pthread_mutex_unlock(&out->lock); |
| 2565 | return 0; |
| 2566 | } |
| 2567 | |
| 2568 | static int out_pause(struct audio_stream_out* stream) |
| 2569 | { |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2570 | struct listnode *node; |
| 2571 | struct alsa_handle *handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2572 | struct stream_out *out = (struct stream_out *)stream; |
| 2573 | int status = -ENOSYS; |
| 2574 | ALOGV("%s", __func__); |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2575 | pthread_mutex_lock(&out->lock); |
| 2576 | list_for_each(node, &out->session_list) { |
| 2577 | handle = node_to_item(node, struct alsa_handle, list); |
| 2578 | if (handle->compr != NULL && out->offload_state == |
| 2579 | OFFLOAD_STATE_PLAYING) { |
| 2580 | status = compress_pause(handle->compr); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2581 | out->offload_state = OFFLOAD_STATE_PAUSED; |
| 2582 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2583 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2584 | pthread_mutex_unlock(&out->lock); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2585 | return status; |
| 2586 | } |
| 2587 | |
| 2588 | static int out_resume(struct audio_stream_out* stream) |
| 2589 | { |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2590 | struct listnode *node; |
| 2591 | struct alsa_handle *handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2592 | struct stream_out *out = (struct stream_out *)stream; |
| 2593 | int status = -ENOSYS; |
| 2594 | ALOGV("%s", __func__); |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2595 | pthread_mutex_lock(&out->lock); |
| 2596 | list_for_each(node, &out->session_list) { |
| 2597 | handle = node_to_item(node, struct alsa_handle, list); |
| 2598 | status = 0; |
| 2599 | if (handle->compr != NULL && out->offload_state == |
| 2600 | OFFLOAD_STATE_PAUSED) { |
| 2601 | status = compress_resume(handle->compr); |
| 2602 | out->offload_state = OFFLOAD_STATE_PLAYING; |
| 2603 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2604 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2605 | pthread_mutex_unlock(&out->lock); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2606 | return status; |
| 2607 | } |
| 2608 | |
| 2609 | static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type ) |
| 2610 | { |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2611 | struct listnode *node; |
| 2612 | struct alsa_handle *handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2613 | struct stream_out *out = (struct stream_out *)stream; |
| 2614 | int status = -ENOSYS; |
| 2615 | ALOGV("%s", __func__); |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2616 | pthread_mutex_lock(&out->lock); |
| 2617 | list_for_each(node, &out->session_list) { |
| 2618 | handle = node_to_item(node, struct alsa_handle, list); |
| 2619 | status = 0; |
| 2620 | if (handle->compr != NULL) { |
| 2621 | if (type == AUDIO_DRAIN_EARLY_NOTIFY) |
| 2622 | status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN); |
| 2623 | else |
| 2624 | status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN); |
| 2625 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2626 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2627 | pthread_mutex_unlock(&out->lock); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2628 | return status; |
| 2629 | } |
| 2630 | |
| 2631 | static int out_flush(struct audio_stream_out* stream) |
| 2632 | { |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2633 | struct listnode *node; |
| 2634 | struct alsa_handle *handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2635 | struct stream_out *out = (struct stream_out *)stream; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2636 | int status = -ENOSYS; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2637 | ALOGV("%s", __func__); |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2638 | pthread_mutex_lock(&out->lock); |
| 2639 | list_for_each(node, &out->session_list) { |
| 2640 | handle = node_to_item(node, struct alsa_handle, list); |
| 2641 | status = 0; |
| 2642 | if (handle->compr != NULL) { |
| 2643 | stop_compressed_output_l(out); |
| 2644 | } |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2645 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2646 | pthread_mutex_unlock(&out->lock); |
| 2647 | return status; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2648 | } |
| 2649 | |
| 2650 | int adev_open_output_stream(struct audio_hw_device *dev, |
| 2651 | audio_io_handle_t handle, |
| 2652 | audio_devices_t devices, |
| 2653 | audio_output_flags_t flags, |
| 2654 | struct audio_config *config, |
| 2655 | struct audio_stream_out **stream_out) |
| 2656 | { |
| 2657 | struct audio_device *adev = (struct audio_device *)dev; |
| 2658 | struct stream_out *out; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2659 | struct alsa_handle *device_handle = NULL; |
| 2660 | int i, ret, channels; |
| 2661 | struct listnode *item; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2662 | |
| 2663 | ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)", |
| 2664 | __func__, config->sample_rate, config->channel_mask, devices, flags); |
| 2665 | *stream_out = NULL; |
| 2666 | out = (struct stream_out *)calloc(1, sizeof(struct stream_out)); |
| 2667 | |
| 2668 | if (devices == AUDIO_DEVICE_NONE) |
| 2669 | devices = AUDIO_DEVICE_OUT_SPEAKER; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2670 | list_init(&out->session_list); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2671 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2672 | reset_out_parameters(out); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2673 | out->flags = flags; |
| 2674 | out->devices = devices; |
| 2675 | out->dev = adev; |
| 2676 | out->format = config->format; |
| 2677 | out->sample_rate = config->sample_rate; |
| 2678 | out->channel_mask = AUDIO_CHANNEL_OUT_STEREO; |
| 2679 | out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2680 | out->config = config; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2681 | out->handle = handle; |
| 2682 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2683 | //*TODO: get hdmi/spdif format/channels from routing manager and intialize out->spdif_format & out->hdmi_format*/ |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2684 | /* Init use case and pcm_config */ |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2685 | out->hdmi_format = UNCOMPRESSED; |
| 2686 | out->spdif_format = UNCOMPRESSED; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2687 | out->stream.common.get_sample_rate = out_get_sample_rate; |
| 2688 | out->stream.common.set_sample_rate = out_set_sample_rate; |
| 2689 | out->stream.common.get_buffer_size = out_get_buffer_size; |
| 2690 | out->stream.common.get_channels = out_get_channels; |
| 2691 | out->stream.common.get_format = out_get_format; |
| 2692 | out->stream.common.set_format = out_set_format; |
| 2693 | out->stream.common.standby = out_standby; |
| 2694 | out->stream.common.dump = out_dump; |
| 2695 | out->stream.common.set_parameters = out_set_parameters; |
| 2696 | out->stream.common.get_parameters = out_get_parameters; |
| 2697 | out->stream.common.add_audio_effect = out_add_audio_effect; |
| 2698 | out->stream.common.remove_audio_effect = out_remove_audio_effect; |
| 2699 | out->stream.get_latency = out_get_latency; |
| 2700 | out->stream.set_volume = out_set_volume; |
| 2701 | out->stream.write = out_write; |
| 2702 | out->stream.get_render_position = out_get_render_position; |
| 2703 | out->stream.get_next_write_timestamp = out_get_next_write_timestamp; |
| 2704 | out->stream.get_presentation_position = out_get_presentation_position; |
| 2705 | |
| 2706 | out->standby = 1; |
| 2707 | /* out->muted = false; by calloc() */ |
| 2708 | /* out->written = 0; by calloc() */ |
| 2709 | |
| 2710 | pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL); |
| 2711 | pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL); |
| 2712 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2713 | if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { |
| 2714 | ALOGE("%s: Usecase is OFFLOAD", __func__); |
| 2715 | if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version || |
| 2716 | config->offload_info.size != AUDIO_INFO_INITIALIZER.size) { |
| 2717 | ALOGE("%s: Unsupported Offload information", __func__); |
| 2718 | ret = -EINVAL; |
| 2719 | goto error_open; |
| 2720 | } |
| 2721 | |
| 2722 | if (!is_supported_format(config->offload_info.format)) { |
| 2723 | ALOGE("%s: Unsupported audio format", __func__); |
| 2724 | ret = -EINVAL; |
| 2725 | goto error_open; |
| 2726 | } |
| 2727 | out->compr_config.codec = (struct snd_codec *) |
| 2728 | calloc(1, sizeof(struct snd_codec)); |
| 2729 | //Session/clip config. |
| 2730 | out->format = config->offload_info.format; |
| 2731 | out->sample_rate = config->offload_info.sample_rate; |
| 2732 | out->compr_config.codec->id = |
| 2733 | get_snd_codec_id(config->offload_info.format); |
| 2734 | out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; |
| 2735 | out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS; |
| 2736 | out->compr_config.codec->sample_rate = |
| 2737 | compress_get_alsa_rate(config->offload_info.sample_rate); |
| 2738 | out->compr_config.codec->bit_rate = |
| 2739 | config->offload_info.bit_rate; |
| 2740 | out->compr_config.codec->ch_in = |
| 2741 | popcount(config->channel_mask); |
| 2742 | out->compr_config.codec->ch_out = out->compr_config.codec->ch_in; |
| 2743 | |
| 2744 | if (config->offload_info.channel_mask) |
| 2745 | out->channel_mask = config->offload_info.channel_mask; |
| 2746 | else if (config->channel_mask) |
| 2747 | out->channel_mask = config->channel_mask; |
| 2748 | out->uc_strm_type = OFFLOAD_PLAYBACK_STREAM; |
| 2749 | |
| 2750 | //Initialize the handles |
| 2751 | /* ------------------------------------------------------------------------ |
| 2752 | Update use decoder type and routing flags and corresponding states |
| 2753 | decoderType will cache the decode types such as decode/passthrough/transcode |
| 2754 | and in s/w or dsp. Besides, the states to open decode/passthrough/transcode |
| 2755 | handles with the corresponding devices and device formats are updated |
| 2756 | -------------------------------------------------------------------------*/ |
| 2757 | update_decode_type_and_routing_states(out); |
| 2758 | |
| 2759 | /* ------------------------------------------------------------------------ |
| 2760 | Update rxHandle states |
| 2761 | Based on the states, we open the driver and store the handle at appropriate |
| 2762 | index |
| 2763 | -------------------------------------------------------------------------*/ |
| 2764 | update_alsa_handle_state(out); |
| 2765 | |
| 2766 | /* ------------------------------------------------------------------------ |
| 2767 | setup routing |
| 2768 | -------------------------------------------------------------------------*/ |
| 2769 | ret = allocate_internal_buffers(out); |
| 2770 | if(ret < 0) { |
| 2771 | ALOGE("%s:Error %d",__func__, ret); |
| 2772 | goto error_handle; |
| 2773 | } |
| 2774 | |
| 2775 | //Callbacks |
| 2776 | out->stream.set_callback = out_set_callback; |
| 2777 | out->stream.pause = out_pause; |
| 2778 | out->stream.resume = out_resume; |
| 2779 | out->stream.drain = out_drain; |
| 2780 | out->stream.flush = out_flush; |
| 2781 | |
| 2782 | if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) |
| 2783 | out->non_blocking = 1; |
| 2784 | |
| 2785 | out->send_new_metadata = 1; |
| 2786 | create_offload_callback_thread(out); |
| 2787 | ALOGV("%s: offloaded output offload_info version %04x bit rate %d", |
| 2788 | __func__, config->offload_info.version, |
| 2789 | config->offload_info.bit_rate); |
| 2790 | } else { //if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) { |
| 2791 | ALOGE("%s: Usecase is DEEP_BUFFER", __func__); |
| 2792 | if((device_handle = get_alsa_handle())== NULL) |
| 2793 | goto error_handle; |
| 2794 | list_add_tail(&out->session_list, &device_handle->list); |
| 2795 | device_handle->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER; |
| 2796 | device_handle->config = pcm_config_deep_buffer; |
| 2797 | device_handle->out = out; |
| 2798 | device_handle->cmd_pending = false; |
| 2799 | out->sample_rate = device_handle->config.rate; |
| 2800 | out->uc_strm_type = DEEP_BUFFER_PLAYBACK_STREAM; |
| 2801 | out->buffer_size = device_handle->config.period_size * |
| 2802 | audio_stream_frame_size(&out->stream.common); |
| 2803 | }/* else { |
| 2804 | if((device_handle = get_alsa_handle())== NULL) |
| 2805 | goto error_handle; |
| 2806 | list_add_tail(&out->session_list, &device_handle->list); |
| 2807 | device_handle->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY; |
| 2808 | device_handle->config = pcm_config_low_latency; |
| 2809 | device_handle->sample_rate = device_handle->config.rate; |
| 2810 | device_handle->out = out; |
| 2811 | device_handle->cmd_pending = false; |
| 2812 | out->uc_strm_type = LOW_LATENCY_PLAYBACK_STREAM; |
| 2813 | out->buffer_size = device_handle->config.period_size * |
| 2814 | audio_stream_frame_size(&out->stream.common); |
| 2815 | }*/ |
| 2816 | |
| 2817 | if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) { |
| 2818 | ALOGE("%s: Usecase is primary ", __func__); |
| 2819 | if(adev->primary_output == NULL) |
| 2820 | adev->primary_output = out; |
| 2821 | else { |
| 2822 | ALOGE("%s: Primary output is already opened", __func__); |
| 2823 | ret = -EEXIST; |
| 2824 | goto error_open; |
| 2825 | } |
| 2826 | } |
| 2827 | |
| 2828 | /* Check if this usecase is already existing */ |
| 2829 | pthread_mutex_lock(&adev->lock); |
| 2830 | if (out->uc_strm_type != OFFLOAD_PLAYBACK_STREAM) { |
| 2831 | if (get_usecase_from_list(adev, device_handle->usecase) != NULL) { |
| 2832 | ALOGE("%s: Usecase (%d) is already present", __func__, |
| 2833 | device_handle->usecase); |
| 2834 | pthread_mutex_unlock(&adev->lock); |
| 2835 | ret = -EEXIST; |
| 2836 | goto error_open; |
| 2837 | } |
| 2838 | } |
| 2839 | pthread_mutex_unlock(&adev->lock); |
| 2840 | |
| 2841 | |
| 2842 | /* out->muted = false; by calloc() */ |
| 2843 | |
| 2844 | |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2845 | config->format = out->stream.common.get_format(&out->stream.common); |
| 2846 | config->channel_mask = out->stream.common.get_channels(&out->stream.common); |
| 2847 | config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common); |
| 2848 | |
| 2849 | *stream_out = &out->stream; |
| 2850 | ALOGV("%s: exit", __func__); |
| 2851 | return 0; |
| 2852 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2853 | error_handle: |
| 2854 | ret = -EINVAL; |
| 2855 | ALOGE("%s: exit: error handle %d", __func__, ret); |
| 2856 | while (!list_empty(&out->session_list)) { |
| 2857 | item = list_head(&out->session_list); |
| 2858 | list_remove(item); |
| 2859 | device_handle = node_to_item(item, struct alsa_handle, list); |
| 2860 | platform_free_usecase(device_handle->usecase); |
| 2861 | free_alsa_handle(device_handle); |
| 2862 | } |
| 2863 | |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2864 | error_open: |
| 2865 | free(out); |
| 2866 | *stream_out = NULL; |
| 2867 | ALOGD("%s: exit: ret %d", __func__, ret); |
| 2868 | return ret; |
| 2869 | } |
| 2870 | |
| 2871 | void adev_close_output_stream(struct audio_hw_device *dev, |
| 2872 | struct audio_stream_out *stream) |
| 2873 | { |
| 2874 | struct stream_out *out = (struct stream_out *)stream; |
| 2875 | struct audio_device *adev = out->dev; |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2876 | struct listnode *item; |
| 2877 | struct alsa_handle *handle; |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2878 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2879 | ALOGV("%s", __func__); |
| 2880 | |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2881 | out_standby(&stream->common); |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2882 | if (out->uc_strm_type == OFFLOAD_PLAYBACK_STREAM) { |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2883 | destroy_offload_callback_thread(out); |
| 2884 | |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2885 | while (!list_empty(&out->session_list)) { |
| 2886 | item = list_head(&out->session_list); |
| 2887 | list_remove(item); |
| 2888 | handle = node_to_item(item, struct alsa_handle, list); |
| 2889 | if(handle->compr_config.codec != NULL) |
| 2890 | free(handle->compr_config.codec); |
| 2891 | platform_free_usecase(handle->usecase); |
| 2892 | free_alsa_handle(handle); |
| 2893 | } |
| 2894 | free(out->compr_config.codec); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2895 | } |
Dhananjay Kumar | 09cbbf0 | 2013-11-25 15:49:57 +0530 | [diff] [blame] | 2896 | |
| 2897 | free_internal_buffers(out); |
Dhananjay Kumar | 01e921a | 2013-11-26 23:33:22 +0530 | [diff] [blame] | 2898 | pthread_cond_destroy(&out->cond); |
| 2899 | pthread_mutex_destroy(&out->lock); |
| 2900 | free(stream); |
| 2901 | ALOGV("%s: exit", __func__); |
| 2902 | } |