henrike@webrtc.org | 91bac04 | 2014-08-26 22:04:04 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright 2004 The WebRTC Project Authors. All rights reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #include "webrtc/sound/alsasoundsystem.h" |
| 12 | |
| 13 | #include "webrtc/sound/sounddevicelocator.h" |
| 14 | #include "webrtc/sound/soundinputstreaminterface.h" |
| 15 | #include "webrtc/sound/soundoutputstreaminterface.h" |
| 16 | #include "webrtc/base/common.h" |
| 17 | #include "webrtc/base/logging.h" |
| 18 | #include "webrtc/base/scoped_ptr.h" |
| 19 | #include "webrtc/base/stringutils.h" |
| 20 | #include "webrtc/base/timeutils.h" |
| 21 | #include "webrtc/base/worker.h" |
| 22 | |
| 23 | namespace rtc { |
| 24 | |
| 25 | // Lookup table from the rtc format enum in soundsysteminterface.h to |
| 26 | // ALSA's enums. |
| 27 | static const snd_pcm_format_t kCricketFormatToAlsaFormatTable[] = { |
| 28 | // The order here must match the order in soundsysteminterface.h |
| 29 | SND_PCM_FORMAT_S16_LE, |
| 30 | }; |
| 31 | |
| 32 | // Lookup table for the size of a single sample of a given format. |
| 33 | static const size_t kCricketFormatToSampleSizeTable[] = { |
| 34 | // The order here must match the order in soundsysteminterface.h |
| 35 | sizeof(int16_t), // 2 |
| 36 | }; |
| 37 | |
| 38 | // Minimum latency we allow, in microseconds. This is more or less arbitrary, |
| 39 | // but it has to be at least large enough to be able to buffer data during a |
| 40 | // missed context switch, and the typical Linux scheduling quantum is 10ms. |
| 41 | static const int kMinimumLatencyUsecs = 20 * 1000; |
| 42 | |
| 43 | // The latency we'll use for kNoLatencyRequirements (chosen arbitrarily). |
| 44 | static const int kDefaultLatencyUsecs = kMinimumLatencyUsecs * 2; |
| 45 | |
| 46 | // We translate newlines in ALSA device descriptions to hyphens. |
| 47 | static const char kAlsaDescriptionSearch[] = "\n"; |
| 48 | static const char kAlsaDescriptionReplace[] = " - "; |
| 49 | |
| 50 | class AlsaDeviceLocator : public SoundDeviceLocator { |
| 51 | public: |
| 52 | AlsaDeviceLocator(const std::string &name, |
| 53 | const std::string &device_name) |
| 54 | : SoundDeviceLocator(name, device_name) { |
| 55 | // The ALSA descriptions have newlines in them, which won't show up in |
| 56 | // a drop-down box. Replace them with hyphens. |
| 57 | rtc::replace_substrs(kAlsaDescriptionSearch, |
| 58 | sizeof(kAlsaDescriptionSearch) - 1, |
| 59 | kAlsaDescriptionReplace, |
| 60 | sizeof(kAlsaDescriptionReplace) - 1, |
| 61 | &name_); |
| 62 | } |
| 63 | |
| 64 | virtual SoundDeviceLocator *Copy() const { |
| 65 | return new AlsaDeviceLocator(*this); |
| 66 | } |
| 67 | }; |
| 68 | |
| 69 | // Functionality that is common to both AlsaInputStream and AlsaOutputStream. |
| 70 | class AlsaStream { |
| 71 | public: |
| 72 | AlsaStream(AlsaSoundSystem *alsa, |
| 73 | snd_pcm_t *handle, |
| 74 | size_t frame_size, |
| 75 | int wait_timeout_ms, |
| 76 | int flags, |
| 77 | int freq) |
| 78 | : alsa_(alsa), |
| 79 | handle_(handle), |
| 80 | frame_size_(frame_size), |
| 81 | wait_timeout_ms_(wait_timeout_ms), |
| 82 | flags_(flags), |
| 83 | freq_(freq) { |
| 84 | } |
| 85 | |
| 86 | ~AlsaStream() { |
| 87 | Close(); |
| 88 | } |
| 89 | |
| 90 | // Waits for the stream to be ready to accept/return more data, and returns |
| 91 | // how much can be written/read, or 0 if we need to Wait() again. |
| 92 | snd_pcm_uframes_t Wait() { |
| 93 | snd_pcm_sframes_t frames; |
| 94 | // Ideally we would not use snd_pcm_wait() and instead hook snd_pcm_poll_* |
| 95 | // into PhysicalSocketServer, but PhysicalSocketServer is nasty enough |
| 96 | // already and the current clients of SoundSystemInterface do not run |
| 97 | // anything else on their worker threads, so snd_pcm_wait() is good enough. |
| 98 | frames = symbol_table()->snd_pcm_avail_update()(handle_); |
| 99 | if (frames < 0) { |
| 100 | LOG(LS_ERROR) << "snd_pcm_avail_update(): " << GetError(frames); |
| 101 | Recover(frames); |
| 102 | return 0; |
| 103 | } else if (frames > 0) { |
| 104 | // Already ready, so no need to wait. |
| 105 | return frames; |
| 106 | } |
| 107 | // Else no space/data available, so must wait. |
| 108 | int ready = symbol_table()->snd_pcm_wait()(handle_, wait_timeout_ms_); |
| 109 | if (ready < 0) { |
| 110 | LOG(LS_ERROR) << "snd_pcm_wait(): " << GetError(ready); |
| 111 | Recover(ready); |
| 112 | return 0; |
| 113 | } else if (ready == 0) { |
| 114 | // Timeout, so nothing can be written/read right now. |
| 115 | // We set the timeout to twice the requested latency, so continuous |
| 116 | // timeouts are indicative of a problem, so log as a warning. |
| 117 | LOG(LS_WARNING) << "Timeout while waiting on stream"; |
| 118 | return 0; |
| 119 | } |
| 120 | // Else ready > 0 (i.e., 1), so it's ready. Get count. |
| 121 | frames = symbol_table()->snd_pcm_avail_update()(handle_); |
| 122 | if (frames < 0) { |
| 123 | LOG(LS_ERROR) << "snd_pcm_avail_update(): " << GetError(frames); |
| 124 | Recover(frames); |
| 125 | return 0; |
| 126 | } else if (frames == 0) { |
| 127 | // wait() said we were ready, so this ought to have been positive. Has |
| 128 | // been observed to happen in practice though. |
| 129 | LOG(LS_WARNING) << "Spurious wake-up"; |
| 130 | } |
| 131 | return frames; |
| 132 | } |
| 133 | |
| 134 | int CurrentDelayUsecs() { |
| 135 | if (!(flags_ & SoundSystemInterface::FLAG_REPORT_LATENCY)) { |
| 136 | return 0; |
| 137 | } |
| 138 | |
| 139 | snd_pcm_sframes_t delay; |
| 140 | int err = symbol_table()->snd_pcm_delay()(handle_, &delay); |
| 141 | if (err != 0) { |
| 142 | LOG(LS_ERROR) << "snd_pcm_delay(): " << GetError(err); |
| 143 | Recover(err); |
| 144 | // We'd rather continue playout/capture with an incorrect delay than stop |
| 145 | // it altogether, so return a valid value. |
| 146 | return 0; |
| 147 | } |
| 148 | // The delay is in frames. Convert to microseconds. |
| 149 | return delay * rtc::kNumMicrosecsPerSec / freq_; |
| 150 | } |
| 151 | |
| 152 | // Used to recover from certain recoverable errors, principally buffer overrun |
| 153 | // or underrun (identified as EPIPE). Without calling this the stream stays |
| 154 | // in the error state forever. |
| 155 | bool Recover(int error) { |
| 156 | int err; |
| 157 | err = symbol_table()->snd_pcm_recover()( |
| 158 | handle_, |
| 159 | error, |
| 160 | // Silent; i.e., no logging on stderr. |
| 161 | 1); |
| 162 | if (err != 0) { |
| 163 | // Docs say snd_pcm_recover returns the original error if it is not one |
| 164 | // of the recoverable ones, so this log message will probably contain the |
| 165 | // same error twice. |
| 166 | LOG(LS_ERROR) << "Unable to recover from \"" << GetError(error) << "\": " |
| 167 | << GetError(err); |
| 168 | return false; |
| 169 | } |
| 170 | if (error == -EPIPE && // Buffer underrun/overrun. |
| 171 | symbol_table()->snd_pcm_stream()(handle_) == SND_PCM_STREAM_CAPTURE) { |
| 172 | // For capture streams we also have to repeat the explicit start() to get |
| 173 | // data flowing again. |
| 174 | err = symbol_table()->snd_pcm_start()(handle_); |
| 175 | if (err != 0) { |
| 176 | LOG(LS_ERROR) << "snd_pcm_start(): " << GetError(err); |
| 177 | return false; |
| 178 | } |
| 179 | } |
| 180 | return true; |
| 181 | } |
| 182 | |
| 183 | bool Close() { |
| 184 | if (handle_) { |
| 185 | int err; |
| 186 | err = symbol_table()->snd_pcm_drop()(handle_); |
| 187 | if (err != 0) { |
| 188 | LOG(LS_ERROR) << "snd_pcm_drop(): " << GetError(err); |
| 189 | // Continue anyways. |
| 190 | } |
| 191 | err = symbol_table()->snd_pcm_close()(handle_); |
| 192 | if (err != 0) { |
| 193 | LOG(LS_ERROR) << "snd_pcm_close(): " << GetError(err); |
| 194 | // Continue anyways. |
| 195 | } |
| 196 | handle_ = NULL; |
| 197 | } |
| 198 | return true; |
| 199 | } |
| 200 | |
| 201 | AlsaSymbolTable *symbol_table() { |
| 202 | return &alsa_->symbol_table_; |
| 203 | } |
| 204 | |
| 205 | snd_pcm_t *handle() { |
| 206 | return handle_; |
| 207 | } |
| 208 | |
| 209 | const char *GetError(int err) { |
| 210 | return alsa_->GetError(err); |
| 211 | } |
| 212 | |
| 213 | size_t frame_size() { |
| 214 | return frame_size_; |
| 215 | } |
| 216 | |
| 217 | private: |
| 218 | AlsaSoundSystem *alsa_; |
| 219 | snd_pcm_t *handle_; |
| 220 | size_t frame_size_; |
| 221 | int wait_timeout_ms_; |
| 222 | int flags_; |
| 223 | int freq_; |
| 224 | |
| 225 | DISALLOW_COPY_AND_ASSIGN(AlsaStream); |
| 226 | }; |
| 227 | |
| 228 | // Implementation of an input stream. See soundinputstreaminterface.h regarding |
| 229 | // thread-safety. |
| 230 | class AlsaInputStream : |
| 231 | public SoundInputStreamInterface, |
| 232 | private rtc::Worker { |
| 233 | public: |
| 234 | AlsaInputStream(AlsaSoundSystem *alsa, |
| 235 | snd_pcm_t *handle, |
| 236 | size_t frame_size, |
| 237 | int wait_timeout_ms, |
| 238 | int flags, |
| 239 | int freq) |
| 240 | : stream_(alsa, handle, frame_size, wait_timeout_ms, flags, freq), |
| 241 | buffer_size_(0) { |
| 242 | } |
| 243 | |
| 244 | virtual ~AlsaInputStream() { |
| 245 | bool success = StopReading(); |
| 246 | // We need that to live. |
| 247 | VERIFY(success); |
| 248 | } |
| 249 | |
| 250 | virtual bool StartReading() { |
| 251 | return StartWork(); |
| 252 | } |
| 253 | |
| 254 | virtual bool StopReading() { |
| 255 | return StopWork(); |
| 256 | } |
| 257 | |
| 258 | virtual bool GetVolume(int *volume) { |
| 259 | // TODO: Implement this. |
| 260 | return false; |
| 261 | } |
| 262 | |
| 263 | virtual bool SetVolume(int volume) { |
| 264 | // TODO: Implement this. |
| 265 | return false; |
| 266 | } |
| 267 | |
| 268 | virtual bool Close() { |
| 269 | return StopReading() && stream_.Close(); |
| 270 | } |
| 271 | |
| 272 | virtual int LatencyUsecs() { |
| 273 | return stream_.CurrentDelayUsecs(); |
| 274 | } |
| 275 | |
| 276 | private: |
| 277 | // Inherited from Worker. |
| 278 | virtual void OnStart() { |
| 279 | HaveWork(); |
| 280 | } |
| 281 | |
| 282 | // Inherited from Worker. |
| 283 | virtual void OnHaveWork() { |
| 284 | // Block waiting for data. |
| 285 | snd_pcm_uframes_t avail = stream_.Wait(); |
| 286 | if (avail > 0) { |
| 287 | // Data is available. |
| 288 | size_t size = avail * stream_.frame_size(); |
| 289 | if (size > buffer_size_) { |
| 290 | // Must increase buffer size. |
| 291 | buffer_.reset(new char[size]); |
| 292 | buffer_size_ = size; |
| 293 | } |
| 294 | // Read all the data. |
| 295 | snd_pcm_sframes_t read = stream_.symbol_table()->snd_pcm_readi()( |
| 296 | stream_.handle(), |
| 297 | buffer_.get(), |
| 298 | avail); |
| 299 | if (read < 0) { |
| 300 | LOG(LS_ERROR) << "snd_pcm_readi(): " << GetError(read); |
| 301 | stream_.Recover(read); |
| 302 | } else if (read == 0) { |
| 303 | // Docs say this shouldn't happen. |
| 304 | ASSERT(false); |
| 305 | LOG(LS_ERROR) << "No data?"; |
| 306 | } else { |
| 307 | // Got data. Pass it off to the app. |
| 308 | SignalSamplesRead(buffer_.get(), |
| 309 | read * stream_.frame_size(), |
| 310 | this); |
| 311 | } |
| 312 | } |
| 313 | // Check for more data with no delay, after any pending messages are |
| 314 | // dispatched. |
| 315 | HaveWork(); |
| 316 | } |
| 317 | |
| 318 | // Inherited from Worker. |
| 319 | virtual void OnStop() { |
| 320 | // Nothing to do. |
| 321 | } |
| 322 | |
| 323 | const char *GetError(int err) { |
| 324 | return stream_.GetError(err); |
| 325 | } |
| 326 | |
| 327 | AlsaStream stream_; |
| 328 | rtc::scoped_ptr<char[]> buffer_; |
| 329 | size_t buffer_size_; |
| 330 | |
| 331 | DISALLOW_COPY_AND_ASSIGN(AlsaInputStream); |
| 332 | }; |
| 333 | |
| 334 | // Implementation of an output stream. See soundoutputstreaminterface.h |
| 335 | // regarding thread-safety. |
| 336 | class AlsaOutputStream : |
| 337 | public SoundOutputStreamInterface, |
| 338 | private rtc::Worker { |
| 339 | public: |
| 340 | AlsaOutputStream(AlsaSoundSystem *alsa, |
| 341 | snd_pcm_t *handle, |
| 342 | size_t frame_size, |
| 343 | int wait_timeout_ms, |
| 344 | int flags, |
| 345 | int freq) |
| 346 | : stream_(alsa, handle, frame_size, wait_timeout_ms, flags, freq) { |
| 347 | } |
| 348 | |
| 349 | virtual ~AlsaOutputStream() { |
| 350 | bool success = DisableBufferMonitoring(); |
| 351 | // We need that to live. |
| 352 | VERIFY(success); |
| 353 | } |
| 354 | |
| 355 | virtual bool EnableBufferMonitoring() { |
| 356 | return StartWork(); |
| 357 | } |
| 358 | |
| 359 | virtual bool DisableBufferMonitoring() { |
| 360 | return StopWork(); |
| 361 | } |
| 362 | |
| 363 | virtual bool WriteSamples(const void *sample_data, |
| 364 | size_t size) { |
| 365 | if (size % stream_.frame_size() != 0) { |
| 366 | // No client of SoundSystemInterface does this, so let's not support it. |
| 367 | // (If we wanted to support it, we'd basically just buffer the fractional |
| 368 | // frame until we get more data.) |
| 369 | ASSERT(false); |
| 370 | LOG(LS_ERROR) << "Writes with fractional frames are not supported"; |
| 371 | return false; |
| 372 | } |
| 373 | snd_pcm_uframes_t frames = size / stream_.frame_size(); |
| 374 | snd_pcm_sframes_t written = stream_.symbol_table()->snd_pcm_writei()( |
| 375 | stream_.handle(), |
| 376 | sample_data, |
| 377 | frames); |
| 378 | if (written < 0) { |
| 379 | LOG(LS_ERROR) << "snd_pcm_writei(): " << GetError(written); |
| 380 | stream_.Recover(written); |
| 381 | return false; |
| 382 | } else if (static_cast<snd_pcm_uframes_t>(written) < frames) { |
| 383 | // Shouldn't happen. Drop the rest of the data. |
| 384 | LOG(LS_ERROR) << "Stream wrote only " << written << " of " << frames |
| 385 | << " frames!"; |
| 386 | return false; |
| 387 | } |
| 388 | return true; |
| 389 | } |
| 390 | |
| 391 | virtual bool GetVolume(int *volume) { |
| 392 | // TODO: Implement this. |
| 393 | return false; |
| 394 | } |
| 395 | |
| 396 | virtual bool SetVolume(int volume) { |
| 397 | // TODO: Implement this. |
| 398 | return false; |
| 399 | } |
| 400 | |
| 401 | virtual bool Close() { |
| 402 | return DisableBufferMonitoring() && stream_.Close(); |
| 403 | } |
| 404 | |
| 405 | virtual int LatencyUsecs() { |
| 406 | return stream_.CurrentDelayUsecs(); |
| 407 | } |
| 408 | |
| 409 | private: |
| 410 | // Inherited from Worker. |
| 411 | virtual void OnStart() { |
| 412 | HaveWork(); |
| 413 | } |
| 414 | |
| 415 | // Inherited from Worker. |
| 416 | virtual void OnHaveWork() { |
| 417 | snd_pcm_uframes_t avail = stream_.Wait(); |
| 418 | if (avail > 0) { |
| 419 | size_t space = avail * stream_.frame_size(); |
| 420 | SignalBufferSpace(space, this); |
| 421 | } |
| 422 | HaveWork(); |
| 423 | } |
| 424 | |
| 425 | // Inherited from Worker. |
| 426 | virtual void OnStop() { |
| 427 | // Nothing to do. |
| 428 | } |
| 429 | |
| 430 | const char *GetError(int err) { |
| 431 | return stream_.GetError(err); |
| 432 | } |
| 433 | |
| 434 | AlsaStream stream_; |
| 435 | |
| 436 | DISALLOW_COPY_AND_ASSIGN(AlsaOutputStream); |
| 437 | }; |
| 438 | |
| 439 | AlsaSoundSystem::AlsaSoundSystem() : initialized_(false) {} |
| 440 | |
| 441 | AlsaSoundSystem::~AlsaSoundSystem() { |
| 442 | // Not really necessary, because Terminate() doesn't really do anything. |
| 443 | Terminate(); |
| 444 | } |
| 445 | |
| 446 | bool AlsaSoundSystem::Init() { |
| 447 | if (IsInitialized()) { |
| 448 | return true; |
| 449 | } |
| 450 | |
| 451 | // Load libasound. |
| 452 | if (!symbol_table_.Load()) { |
| 453 | // Very odd for a Linux machine to not have a working libasound ... |
| 454 | LOG(LS_ERROR) << "Failed to load symbol table"; |
| 455 | return false; |
| 456 | } |
| 457 | |
| 458 | initialized_ = true; |
| 459 | |
| 460 | return true; |
| 461 | } |
| 462 | |
| 463 | void AlsaSoundSystem::Terminate() { |
| 464 | if (!IsInitialized()) { |
| 465 | return; |
| 466 | } |
| 467 | |
| 468 | initialized_ = false; |
| 469 | |
| 470 | // We do not unload the symbol table because we may need it again soon if |
| 471 | // Init() is called again. |
| 472 | } |
| 473 | |
| 474 | bool AlsaSoundSystem::EnumeratePlaybackDevices( |
| 475 | SoundDeviceLocatorList *devices) { |
| 476 | return EnumerateDevices(devices, false); |
| 477 | } |
| 478 | |
| 479 | bool AlsaSoundSystem::EnumerateCaptureDevices( |
| 480 | SoundDeviceLocatorList *devices) { |
| 481 | return EnumerateDevices(devices, true); |
| 482 | } |
| 483 | |
| 484 | bool AlsaSoundSystem::GetDefaultPlaybackDevice(SoundDeviceLocator **device) { |
| 485 | return GetDefaultDevice(device); |
| 486 | } |
| 487 | |
| 488 | bool AlsaSoundSystem::GetDefaultCaptureDevice(SoundDeviceLocator **device) { |
| 489 | return GetDefaultDevice(device); |
| 490 | } |
| 491 | |
| 492 | SoundOutputStreamInterface *AlsaSoundSystem::OpenPlaybackDevice( |
| 493 | const SoundDeviceLocator *device, |
| 494 | const OpenParams ¶ms) { |
| 495 | return OpenDevice<SoundOutputStreamInterface>( |
| 496 | device, |
| 497 | params, |
| 498 | SND_PCM_STREAM_PLAYBACK, |
| 499 | &AlsaSoundSystem::StartOutputStream); |
| 500 | } |
| 501 | |
| 502 | SoundInputStreamInterface *AlsaSoundSystem::OpenCaptureDevice( |
| 503 | const SoundDeviceLocator *device, |
| 504 | const OpenParams ¶ms) { |
| 505 | return OpenDevice<SoundInputStreamInterface>( |
| 506 | device, |
| 507 | params, |
| 508 | SND_PCM_STREAM_CAPTURE, |
| 509 | &AlsaSoundSystem::StartInputStream); |
| 510 | } |
| 511 | |
| 512 | const char *AlsaSoundSystem::GetName() const { |
| 513 | return "ALSA"; |
| 514 | } |
| 515 | |
| 516 | bool AlsaSoundSystem::EnumerateDevices( |
| 517 | SoundDeviceLocatorList *devices, |
| 518 | bool capture_not_playback) { |
| 519 | ClearSoundDeviceLocatorList(devices); |
| 520 | |
| 521 | if (!IsInitialized()) { |
| 522 | return false; |
| 523 | } |
| 524 | |
| 525 | const char *type = capture_not_playback ? "Input" : "Output"; |
| 526 | // dmix and dsnoop are only for playback and capture, respectively, but ALSA |
| 527 | // stupidly includes them in both lists. |
| 528 | const char *ignore_prefix = capture_not_playback ? "dmix:" : "dsnoop:"; |
| 529 | // (ALSA lists many more "devices" of questionable interest, but we show them |
| 530 | // just in case the weird devices may actually be desirable for some |
| 531 | // users/systems.) |
| 532 | const char *ignore_default = "default"; |
| 533 | const char *ignore_null = "null"; |
| 534 | const char *ignore_pulse = "pulse"; |
| 535 | // The 'pulse' entry has a habit of mysteriously disappearing when you query |
| 536 | // a second time. Remove it from our list. (GIPS lib did the same thing.) |
| 537 | int err; |
| 538 | |
| 539 | void **hints; |
| 540 | err = symbol_table_.snd_device_name_hint()(-1, // All cards |
| 541 | "pcm", // Only PCM devices |
| 542 | &hints); |
| 543 | if (err != 0) { |
| 544 | LOG(LS_ERROR) << "snd_device_name_hint(): " << GetError(err); |
| 545 | return false; |
| 546 | } |
| 547 | |
| 548 | for (void **list = hints; *list != NULL; ++list) { |
| 549 | char *actual_type = symbol_table_.snd_device_name_get_hint()(*list, "IOID"); |
| 550 | if (actual_type) { // NULL means it's both. |
| 551 | bool wrong_type = (strcmp(actual_type, type) != 0); |
| 552 | free(actual_type); |
| 553 | if (wrong_type) { |
| 554 | // Wrong type of device (i.e., input vs. output). |
| 555 | continue; |
| 556 | } |
| 557 | } |
| 558 | |
| 559 | char *name = symbol_table_.snd_device_name_get_hint()(*list, "NAME"); |
| 560 | if (!name) { |
| 561 | LOG(LS_ERROR) << "Device has no name???"; |
| 562 | // Skip it. |
| 563 | continue; |
| 564 | } |
| 565 | |
| 566 | // Now check if we actually want to show this device. |
| 567 | if (strcmp(name, ignore_default) != 0 && |
| 568 | strcmp(name, ignore_null) != 0 && |
| 569 | strcmp(name, ignore_pulse) != 0 && |
| 570 | !rtc::starts_with(name, ignore_prefix)) { |
| 571 | |
| 572 | // Yes, we do. |
| 573 | char *desc = symbol_table_.snd_device_name_get_hint()(*list, "DESC"); |
| 574 | if (!desc) { |
| 575 | // Virtual devices don't necessarily have descriptions. Use their names |
| 576 | // instead (not pretty!). |
| 577 | desc = name; |
| 578 | } |
| 579 | |
| 580 | AlsaDeviceLocator *device = new AlsaDeviceLocator(desc, name); |
| 581 | |
| 582 | devices->push_back(device); |
| 583 | |
| 584 | if (desc != name) { |
| 585 | free(desc); |
| 586 | } |
| 587 | } |
| 588 | |
| 589 | free(name); |
| 590 | } |
| 591 | |
| 592 | err = symbol_table_.snd_device_name_free_hint()(hints); |
| 593 | if (err != 0) { |
| 594 | LOG(LS_ERROR) << "snd_device_name_free_hint(): " << GetError(err); |
| 595 | // Continue and return true anyways, since we did get the whole list. |
| 596 | } |
| 597 | |
| 598 | return true; |
| 599 | } |
| 600 | |
| 601 | bool AlsaSoundSystem::GetDefaultDevice(SoundDeviceLocator **device) { |
| 602 | if (!IsInitialized()) { |
| 603 | return false; |
| 604 | } |
| 605 | *device = new AlsaDeviceLocator("Default device", "default"); |
| 606 | return true; |
| 607 | } |
| 608 | |
| 609 | inline size_t AlsaSoundSystem::FrameSize(const OpenParams ¶ms) { |
| 610 | ASSERT(static_cast<int>(params.format) < |
| 611 | ARRAY_SIZE(kCricketFormatToSampleSizeTable)); |
| 612 | return kCricketFormatToSampleSizeTable[params.format] * params.channels; |
| 613 | } |
| 614 | |
| 615 | template <typename StreamInterface> |
| 616 | StreamInterface *AlsaSoundSystem::OpenDevice( |
| 617 | const SoundDeviceLocator *device, |
| 618 | const OpenParams ¶ms, |
| 619 | snd_pcm_stream_t type, |
| 620 | StreamInterface *(AlsaSoundSystem::*start_fn)( |
| 621 | snd_pcm_t *handle, |
| 622 | size_t frame_size, |
| 623 | int wait_timeout_ms, |
| 624 | int flags, |
| 625 | int freq)) { |
| 626 | |
| 627 | if (!IsInitialized()) { |
| 628 | return NULL; |
| 629 | } |
| 630 | |
| 631 | StreamInterface *stream; |
| 632 | int err; |
| 633 | |
| 634 | const char *dev = static_cast<const AlsaDeviceLocator *>(device)-> |
| 635 | device_name().c_str(); |
| 636 | |
| 637 | snd_pcm_t *handle = NULL; |
| 638 | err = symbol_table_.snd_pcm_open()( |
| 639 | &handle, |
| 640 | dev, |
| 641 | type, |
| 642 | // No flags. |
| 643 | 0); |
| 644 | if (err != 0) { |
| 645 | LOG(LS_ERROR) << "snd_pcm_open(" << dev << "): " << GetError(err); |
| 646 | return NULL; |
| 647 | } |
| 648 | LOG(LS_VERBOSE) << "Opening " << dev; |
| 649 | ASSERT(handle); // If open succeeded, handle ought to be valid |
| 650 | |
| 651 | // Compute requested latency in microseconds. |
| 652 | int latency; |
| 653 | if (params.latency == kNoLatencyRequirements) { |
| 654 | latency = kDefaultLatencyUsecs; |
| 655 | } else { |
| 656 | // kLowLatency is 0, so we treat it the same as a request for zero latency. |
| 657 | // Compute what the user asked for. |
| 658 | latency = rtc::kNumMicrosecsPerSec * |
| 659 | params.latency / |
| 660 | params.freq / |
| 661 | FrameSize(params); |
| 662 | // And this is what we'll actually use. |
| 663 | latency = rtc::_max(latency, kMinimumLatencyUsecs); |
| 664 | } |
| 665 | |
| 666 | ASSERT(static_cast<int>(params.format) < |
| 667 | ARRAY_SIZE(kCricketFormatToAlsaFormatTable)); |
| 668 | |
| 669 | err = symbol_table_.snd_pcm_set_params()( |
| 670 | handle, |
| 671 | kCricketFormatToAlsaFormatTable[params.format], |
| 672 | // SoundSystemInterface only supports interleaved audio. |
| 673 | SND_PCM_ACCESS_RW_INTERLEAVED, |
| 674 | params.channels, |
| 675 | params.freq, |
| 676 | 1, // Allow ALSA to resample. |
| 677 | latency); |
| 678 | if (err != 0) { |
| 679 | LOG(LS_ERROR) << "snd_pcm_set_params(): " << GetError(err); |
| 680 | goto fail; |
| 681 | } |
| 682 | |
| 683 | err = symbol_table_.snd_pcm_prepare()(handle); |
| 684 | if (err != 0) { |
| 685 | LOG(LS_ERROR) << "snd_pcm_prepare(): " << GetError(err); |
| 686 | goto fail; |
| 687 | } |
| 688 | |
| 689 | stream = (this->*start_fn)( |
| 690 | handle, |
| 691 | FrameSize(params), |
| 692 | // We set the wait time to twice the requested latency, so that wait |
| 693 | // timeouts should be rare. |
| 694 | 2 * latency / rtc::kNumMicrosecsPerMillisec, |
| 695 | params.flags, |
| 696 | params.freq); |
| 697 | if (stream) { |
| 698 | return stream; |
| 699 | } |
| 700 | // Else fall through. |
| 701 | |
| 702 | fail: |
| 703 | err = symbol_table_.snd_pcm_close()(handle); |
| 704 | if (err != 0) { |
| 705 | LOG(LS_ERROR) << "snd_pcm_close(): " << GetError(err); |
| 706 | } |
| 707 | return NULL; |
| 708 | } |
| 709 | |
| 710 | SoundOutputStreamInterface *AlsaSoundSystem::StartOutputStream( |
| 711 | snd_pcm_t *handle, |
| 712 | size_t frame_size, |
| 713 | int wait_timeout_ms, |
| 714 | int flags, |
| 715 | int freq) { |
| 716 | // Nothing to do here but instantiate the stream. |
| 717 | return new AlsaOutputStream( |
| 718 | this, handle, frame_size, wait_timeout_ms, flags, freq); |
| 719 | } |
| 720 | |
| 721 | SoundInputStreamInterface *AlsaSoundSystem::StartInputStream( |
| 722 | snd_pcm_t *handle, |
| 723 | size_t frame_size, |
| 724 | int wait_timeout_ms, |
| 725 | int flags, |
| 726 | int freq) { |
| 727 | // Output streams start automatically once enough data has been written, but |
| 728 | // input streams must be started manually or else snd_pcm_wait() will never |
| 729 | // return true. |
| 730 | int err; |
| 731 | err = symbol_table_.snd_pcm_start()(handle); |
| 732 | if (err != 0) { |
| 733 | LOG(LS_ERROR) << "snd_pcm_start(): " << GetError(err); |
| 734 | return NULL; |
| 735 | } |
| 736 | return new AlsaInputStream( |
| 737 | this, handle, frame_size, wait_timeout_ms, flags, freq); |
| 738 | } |
| 739 | |
| 740 | inline const char *AlsaSoundSystem::GetError(int err) { |
| 741 | return symbol_table_.snd_strerror()(err); |
| 742 | } |
| 743 | |
| 744 | } // namespace rtc |