Remove webrtc/sound/ subdir.

webrtc/sound/alsasoundsystem.cc

Index: webrtc/sound/alsasoundsystem.cc
diff --git a/webrtc/sound/alsasoundsystem.cc b/webrtc/sound/alsasoundsystem.cc
deleted file mode 100644
index 696ff1e45059aa00857a2a35705f2f7665916545..0000000000000000000000000000000000000000
--- a/webrtc/sound/alsasoundsystem.cc
+++ /dev/null
@@ -1,741 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/sound/alsasoundsystem.h"
-
-#include <algorithm>
-#include <string>
-
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/worker.h"
-#include "webrtc/sound/sounddevicelocator.h"
-#include "webrtc/sound/soundinputstreaminterface.h"
-#include "webrtc/sound/soundoutputstreaminterface.h"
-
-namespace rtc {
-
-// Lookup table from the rtc format enum in soundsysteminterface.h to
-// ALSA's enums.
-static const snd_pcm_format_t kCricketFormatToAlsaFormatTable[] = {
-  // The order here must match the order in soundsysteminterface.h
-  SND_PCM_FORMAT_S16_LE,
-};
-
-// Lookup table for the size of a single sample of a given format.
-static const size_t kCricketFormatToSampleSizeTable[] = {
-  // The order here must match the order in soundsysteminterface.h
-  sizeof(int16_t),  // 2
-};
-
-// Minimum latency we allow, in microseconds. This is more or less arbitrary,
-// but it has to be at least large enough to be able to buffer data during a
-// missed context switch, and the typical Linux scheduling quantum is 10ms.
-static const int kMinimumLatencyUsecs = 20 * 1000;
-
-// The latency we'll use for kNoLatencyRequirements (chosen arbitrarily).
-static const int kDefaultLatencyUsecs = kMinimumLatencyUsecs * 2;
-
-// We translate newlines in ALSA device descriptions to hyphens.
-static const char kAlsaDescriptionSearch[] = "\n";
-static const char kAlsaDescriptionReplace[] = " - ";
-
-class AlsaDeviceLocator : public SoundDeviceLocator {
- public:
-  AlsaDeviceLocator(const std::string &name,
-                    const std::string &device_name)
-      : SoundDeviceLocator(name, device_name) {
-    // The ALSA descriptions have newlines in them, which won't show up in
-    // a drop-down box. Replace them with hyphens.
-    rtc::replace_substrs(kAlsaDescriptionSearch,
-                               sizeof(kAlsaDescriptionSearch) - 1,
-                               kAlsaDescriptionReplace,
-                               sizeof(kAlsaDescriptionReplace) - 1,
-                               &name_);
-  }
-
-  SoundDeviceLocator *Copy() const override {
-    return new AlsaDeviceLocator(*this);
-  }
-};
-
-// Functionality that is common to both AlsaInputStream and AlsaOutputStream.
-class AlsaStream {
- public:
-  AlsaStream(AlsaSoundSystem *alsa,
-             snd_pcm_t *handle,
-             size_t frame_size,
-             int wait_timeout_ms,
-             int flags,
-             int freq)
-      : alsa_(alsa),
-        handle_(handle),
-        frame_size_(frame_size),
-        wait_timeout_ms_(wait_timeout_ms),
-        flags_(flags),
-        freq_(freq) {
-  }
-
-  ~AlsaStream() {
-    Close();
-  }
-
-  // Waits for the stream to be ready to accept/return more data, and returns
-  // how much can be written/read, or 0 if we need to Wait() again.
-  snd_pcm_uframes_t Wait() {
-    snd_pcm_sframes_t frames;
-    // Ideally we would not use snd_pcm_wait() and instead hook snd_pcm_poll_*
-    // into PhysicalSocketServer, but PhysicalSocketServer is nasty enough
-    // already and the current clients of SoundSystemInterface do not run
-    // anything else on their worker threads, so snd_pcm_wait() is good enough.
-    frames = symbol_table()->snd_pcm_avail_update()(handle_);
-    if (frames < 0) {
-      LOG(LS_ERROR) << "snd_pcm_avail_update(): " << GetError(frames);
-      Recover(frames);
-      return 0;
-    } else if (frames > 0) {
-      // Already ready, so no need to wait.
-      return frames;
-    }
-    // Else no space/data available, so must wait.
-    int ready = symbol_table()->snd_pcm_wait()(handle_, wait_timeout_ms_);
-    if (ready < 0) {
-      LOG(LS_ERROR) << "snd_pcm_wait(): " << GetError(ready);
-      Recover(ready);
-      return 0;
-    } else if (ready == 0) {
-      // Timeout, so nothing can be written/read right now.
-      // We set the timeout to twice the requested latency, so continuous
-      // timeouts are indicative of a problem, so log as a warning.
-      LOG(LS_WARNING) << "Timeout while waiting on stream";
-      return 0;
-    }
-    // Else ready > 0 (i.e., 1), so it's ready. Get count.
-    frames = symbol_table()->snd_pcm_avail_update()(handle_);
-    if (frames < 0) {
-      LOG(LS_ERROR) << "snd_pcm_avail_update(): " << GetError(frames);
-      Recover(frames);
-      return 0;
-    } else if (frames == 0) {
-      // wait() said we were ready, so this ought to have been positive. Has
-      // been observed to happen in practice though.
-      LOG(LS_WARNING) << "Spurious wake-up";
-    }
-    return frames;
-  }
-
-  int CurrentDelayUsecs() {
-    if (!(flags_ & SoundSystemInterface::FLAG_REPORT_LATENCY)) {
-      return 0;
-    }
-
-    snd_pcm_sframes_t delay;
-    int err = symbol_table()->snd_pcm_delay()(handle_, &delay);
-    if (err != 0) {
-      LOG(LS_ERROR) << "snd_pcm_delay(): " << GetError(err);
-      Recover(err);
-      // We'd rather continue playout/capture with an incorrect delay than stop
-      // it altogether, so return a valid value.
-      return 0;
-    }
-    // The delay is in frames. Convert to microseconds.
-    return delay * rtc::kNumMicrosecsPerSec / freq_;
-  }
-
-  // Used to recover from certain recoverable errors, principally buffer overrun
-  // or underrun (identified as EPIPE). Without calling this the stream stays
-  // in the error state forever.
-  bool Recover(int error) {
-    int err;
-    err = symbol_table()->snd_pcm_recover()(
-        handle_,
-        error,
-        // Silent; i.e., no logging on stderr.
-        1);
-    if (err != 0) {
-      // Docs say snd_pcm_recover returns the original error if it is not one
-      // of the recoverable ones, so this log message will probably contain the
-      // same error twice.
-      LOG(LS_ERROR) << "Unable to recover from \"" << GetError(error) << "\": "
-                    << GetError(err);
-      return false;
-    }
-    if (error == -EPIPE &&  // Buffer underrun/overrun.
-        symbol_table()->snd_pcm_stream()(handle_) == SND_PCM_STREAM_CAPTURE) {
-      // For capture streams we also have to repeat the explicit start() to get
-      // data flowing again.
-      err = symbol_table()->snd_pcm_start()(handle_);
-      if (err != 0) {
-        LOG(LS_ERROR) << "snd_pcm_start(): " << GetError(err);
-        return false;
-      }
-    }
-    return true;
-  }
-
-  bool Close() {
-    if (handle_) {
-      int err;
-      err = symbol_table()->snd_pcm_drop()(handle_);
-      if (err != 0) {
-        LOG(LS_ERROR) << "snd_pcm_drop(): " << GetError(err);
-        // Continue anyways.
-      }
-      err = symbol_table()->snd_pcm_close()(handle_);
-      if (err != 0) {
-        LOG(LS_ERROR) << "snd_pcm_close(): " << GetError(err);
-        // Continue anyways.
-      }
-      handle_ = NULL;
-    }
-    return true;
-  }
-
-  AlsaSymbolTable *symbol_table() {
-    return &alsa_->symbol_table_;
-  }
-
-  snd_pcm_t *handle() {
-    return handle_;
-  }
-
-  const char *GetError(int err) {
-    return alsa_->GetError(err);
-  }
-
-  size_t frame_size() {
-    return frame_size_;
-  }
-
- private:
-  AlsaSoundSystem *alsa_;
-  snd_pcm_t *handle_;
-  size_t frame_size_;
-  int wait_timeout_ms_;
-  int flags_;
-  int freq_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AlsaStream);
-};
-
-// Implementation of an input stream. See soundinputstreaminterface.h regarding
-// thread-safety.
-class AlsaInputStream :
-    public SoundInputStreamInterface,
-    private rtc::Worker {
- public:
-  AlsaInputStream(AlsaSoundSystem *alsa,
-                  snd_pcm_t *handle,
-                  size_t frame_size,
-                  int wait_timeout_ms,
-                  int flags,
-                  int freq)
-      : stream_(alsa, handle, frame_size, wait_timeout_ms, flags, freq),
-        buffer_size_(0) {
-  }
-
-  ~AlsaInputStream() override {
-    bool success = StopReading();
-    // We need that to live.
-    VERIFY(success);
-  }
-
-  bool StartReading() override {
-    return StartWork();
-  }
-
-  bool StopReading() override {
-    return StopWork();
-  }
-
-  bool GetVolume(int *volume) override {
-    // TODO(henrika): Implement this.
-    return false;
-  }
-
-  bool SetVolume(int volume) override {
-    // TODO(henrika): Implement this.
-    return false;
-  }
-
-  bool Close() override {
-    return StopReading() && stream_.Close();
-  }
-
-  int LatencyUsecs() override {
-    return stream_.CurrentDelayUsecs();
-  }
-
- private:
-  // Inherited from Worker.
-  void OnStart() override {
-    HaveWork();
-  }
-
-  // Inherited from Worker.
-  void OnHaveWork() override {
-    // Block waiting for data.
-    snd_pcm_uframes_t avail = stream_.Wait();
-    if (avail > 0) {
-      // Data is available.
-      size_t size = avail * stream_.frame_size();
-      if (size > buffer_size_) {
-        // Must increase buffer size.
-        buffer_.reset(new char[size]);
-        buffer_size_ = size;
-      }
-      // Read all the data.
-      snd_pcm_sframes_t read = stream_.symbol_table()->snd_pcm_readi()(
-          stream_.handle(),
-          buffer_.get(),
-          avail);
-      if (read < 0) {
-        LOG(LS_ERROR) << "snd_pcm_readi(): " << GetError(read);
-        stream_.Recover(read);
-      } else if (read == 0) {
-        // Docs say this shouldn't happen.
-        ASSERT(false);
-        LOG(LS_ERROR) << "No data?";
-      } else {
-        // Got data. Pass it off to the app.
-        SignalSamplesRead(buffer_.get(),
-                          read * stream_.frame_size(),
-                          this);
-      }
-    }
-    // Check for more data with no delay, after any pending messages are
-    // dispatched.
-    HaveWork();
-  }
-
-  // Inherited from Worker.
-  void OnStop() override {
-    // Nothing to do.
-  }
-
-  const char *GetError(int err) {
-    return stream_.GetError(err);
-  }
-
-  AlsaStream stream_;
-  rtc::scoped_ptr<char[]> buffer_;
-  size_t buffer_size_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AlsaInputStream);
-};
-
-// Implementation of an output stream. See soundoutputstreaminterface.h
-// regarding thread-safety.
-class AlsaOutputStream : public SoundOutputStreamInterface,
-                         private rtc::Worker {
- public:
-  AlsaOutputStream(AlsaSoundSystem *alsa,
-                   snd_pcm_t *handle,
-                   size_t frame_size,
-                   int wait_timeout_ms,
-                   int flags,
-                   int freq)
-      : stream_(alsa, handle, frame_size, wait_timeout_ms, flags, freq) {
-  }
-
-  ~AlsaOutputStream() override {
-    bool success = DisableBufferMonitoring();
-    // We need that to live.
-    VERIFY(success);
-  }
-
-  bool EnableBufferMonitoring() override {
-    return StartWork();
-  }
-
-  bool DisableBufferMonitoring() override {
-    return StopWork();
-  }
-
-  bool WriteSamples(const void *sample_data, size_t size) override {
-    if (size % stream_.frame_size() != 0) {
-      // No client of SoundSystemInterface does this, so let's not support it.
-      // (If we wanted to support it, we'd basically just buffer the fractional
-      // frame until we get more data.)
-      ASSERT(false);
-      LOG(LS_ERROR) << "Writes with fractional frames are not supported";
-      return false;
-    }
-    snd_pcm_uframes_t frames = size / stream_.frame_size();
-    snd_pcm_sframes_t written = stream_.symbol_table()->snd_pcm_writei()(
-        stream_.handle(),
-        sample_data,
-        frames);
-    if (written < 0) {
-      LOG(LS_ERROR) << "snd_pcm_writei(): " << GetError(written);
-      stream_.Recover(written);
-      return false;
-    } else if (static_cast<snd_pcm_uframes_t>(written) < frames) {
-      // Shouldn't happen. Drop the rest of the data.
-      LOG(LS_ERROR) << "Stream wrote only " << written << " of " << frames
-                    << " frames!";
-      return false;
-    }
-    return true;
-  }
-
-  bool GetVolume(int *volume) override {
-    // TODO(henrika): Implement this.
-    return false;
-  }
-
-  bool SetVolume(int volume) override {
-    // TODO(henrika): Implement this.
-    return false;
-  }
-
-  bool Close() override {
-    return DisableBufferMonitoring() && stream_.Close();
-  }
-
-  int LatencyUsecs() override {
-    return stream_.CurrentDelayUsecs();
-  }
-
- private:
-  // Inherited from Worker.
-  void OnStart() override {
-    HaveWork();
-  }
-
-  // Inherited from Worker.
-  void OnHaveWork() override {
-    snd_pcm_uframes_t avail = stream_.Wait();
-    if (avail > 0) {
-      size_t space = avail * stream_.frame_size();
-      SignalBufferSpace(space, this);
-    }
-    HaveWork();
-  }
-
-  // Inherited from Worker.
-  void OnStop() override {
-    // Nothing to do.
-  }
-
-  const char *GetError(int err) {
-    return stream_.GetError(err);
-  }
-
-  AlsaStream stream_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AlsaOutputStream);
-};
-
-AlsaSoundSystem::AlsaSoundSystem() : initialized_(false) {}
-
-AlsaSoundSystem::~AlsaSoundSystem() {
-  // Not really necessary, because Terminate() doesn't really do anything.
-  Terminate();
-}
-
-bool AlsaSoundSystem::Init() {
-  if (IsInitialized()) {
-    return true;
-  }
-
-  // Load libasound.
-  if (!symbol_table_.Load()) {
-    // Very odd for a Linux machine to not have a working libasound ...
-    LOG(LS_ERROR) << "Failed to load symbol table";
-    return false;
-  }
-
-  initialized_ = true;
-
-  return true;
-}
-
-void AlsaSoundSystem::Terminate() {
-  if (!IsInitialized()) {
-    return;
-  }
-
-  initialized_ = false;
-
-  // We do not unload the symbol table because we may need it again soon if
-  // Init() is called again.
-}
-
-bool AlsaSoundSystem::EnumeratePlaybackDevices(
-    SoundDeviceLocatorList *devices) {
-  return EnumerateDevices(devices, false);
-}
-
-bool AlsaSoundSystem::EnumerateCaptureDevices(
-    SoundDeviceLocatorList *devices) {
-  return EnumerateDevices(devices, true);
-}
-
-bool AlsaSoundSystem::GetDefaultPlaybackDevice(SoundDeviceLocator **device) {
-  return GetDefaultDevice(device);
-}
-
-bool AlsaSoundSystem::GetDefaultCaptureDevice(SoundDeviceLocator **device) {
-  return GetDefaultDevice(device);
-}
-
-SoundOutputStreamInterface *AlsaSoundSystem::OpenPlaybackDevice(
-    const SoundDeviceLocator *device,
-    const OpenParams &params) {
-  return OpenDevice<SoundOutputStreamInterface>(
-      device,
-      params,
-      SND_PCM_STREAM_PLAYBACK,
-      &AlsaSoundSystem::StartOutputStream);
-}
-
-SoundInputStreamInterface *AlsaSoundSystem::OpenCaptureDevice(
-    const SoundDeviceLocator *device,
-    const OpenParams &params) {
-  return OpenDevice<SoundInputStreamInterface>(
-      device,
-      params,
-      SND_PCM_STREAM_CAPTURE,
-      &AlsaSoundSystem::StartInputStream);
-}
-
-const char *AlsaSoundSystem::GetName() const {
-  return "ALSA";
-}
-
-bool AlsaSoundSystem::EnumerateDevices(
-    SoundDeviceLocatorList *devices,
-    bool capture_not_playback) {
-  ClearSoundDeviceLocatorList(devices);
-
-  if (!IsInitialized()) {
-    return false;
-  }
-
-  const char *type = capture_not_playback ? "Input" : "Output";
-  // dmix and dsnoop are only for playback and capture, respectively, but ALSA
-  // stupidly includes them in both lists.
-  const char *ignore_prefix = capture_not_playback ? "dmix:" : "dsnoop:";
-  // (ALSA lists many more "devices" of questionable interest, but we show them
-  // just in case the weird devices may actually be desirable for some
-  // users/systems.)
-  const char *ignore_default = "default";
-  const char *ignore_null = "null";
-  const char *ignore_pulse = "pulse";
-  // The 'pulse' entry has a habit of mysteriously disappearing when you query
-  // a second time. Remove it from our list. (GIPS lib did the same thing.)
-  int err;
-
-  void **hints;
-  err = symbol_table_.snd_device_name_hint()(-1,     // All cards
-                                             "pcm",  // Only PCM devices
-                                             &hints);
-  if (err != 0) {
-    LOG(LS_ERROR) << "snd_device_name_hint(): " << GetError(err);
-    return false;
-  }
-
-  for (void **list = hints; *list != NULL; ++list) {
-    char *actual_type = symbol_table_.snd_device_name_get_hint()(*list, "IOID");
-    if (actual_type) {  // NULL means it's both.
-      bool wrong_type = (strcmp(actual_type, type) != 0);
-      free(actual_type);
-      if (wrong_type) {
-        // Wrong type of device (i.e., input vs. output).
-        continue;
-      }
-    }
-
-    char *name = symbol_table_.snd_device_name_get_hint()(*list, "NAME");
-    if (!name) {
-      LOG(LS_ERROR) << "Device has no name???";
-      // Skip it.
-      continue;
-    }
-
-    // Now check if we actually want to show this device.
-    if (strcmp(name, ignore_default) != 0 &&
-        strcmp(name, ignore_null) != 0 &&
-        strcmp(name, ignore_pulse) != 0 &&
-        !rtc::starts_with(name, ignore_prefix)) {
-      // Yes, we do.
-      char *desc = symbol_table_.snd_device_name_get_hint()(*list, "DESC");
-      if (!desc) {
-        // Virtual devices don't necessarily have descriptions. Use their names
-        // instead (not pretty!).
-        desc = name;
-      }
-
-      AlsaDeviceLocator *device = new AlsaDeviceLocator(desc, name);
-
-      devices->push_back(device);
-
-      if (desc != name) {
-        free(desc);
-      }
-    }
-
-    free(name);
-  }
-
-  err = symbol_table_.snd_device_name_free_hint()(hints);
-  if (err != 0) {
-    LOG(LS_ERROR) << "snd_device_name_free_hint(): " << GetError(err);
-    // Continue and return true anyways, since we did get the whole list.
-  }
-
-  return true;
-}
-
-bool AlsaSoundSystem::GetDefaultDevice(SoundDeviceLocator **device) {
-  if (!IsInitialized()) {
-    return false;
-  }
-  *device = new AlsaDeviceLocator("Default device", "default");
-  return true;
-}
-
-inline size_t AlsaSoundSystem::FrameSize(const OpenParams &params) {
-  return kCricketFormatToSampleSizeTable[params.format] * params.channels;
-}
-
-template <typename StreamInterface>
-StreamInterface *AlsaSoundSystem::OpenDevice(
-    const SoundDeviceLocator *device,
-    const OpenParams &params,
-    snd_pcm_stream_t type,
-    StreamInterface *(AlsaSoundSystem::*start_fn)(
-        snd_pcm_t *handle,
-        size_t frame_size,
-        int wait_timeout_ms,
-        int flags,
-        int freq)) {
-  if (!IsInitialized()) {
-    return NULL;
-  }
-
-  StreamInterface *stream;
-  int err;
-
-  const char *dev = static_cast<const AlsaDeviceLocator *>(device)->
-      device_name().c_str();
-
-  snd_pcm_t *handle = NULL;
-  err = symbol_table_.snd_pcm_open()(
-      &handle,
-      dev,
-      type,
-      // No flags.
-      0);
-  if (err != 0) {
-    LOG(LS_ERROR) << "snd_pcm_open(" << dev << "): " << GetError(err);
-    return NULL;
-  }
-  LOG(LS_VERBOSE) << "Opening " << dev;
-  ASSERT(handle);  // If open succeeded, handle ought to be valid
-
-  // Compute requested latency in microseconds.
-  int latency;
-  if (params.latency == kNoLatencyRequirements) {
-    latency = kDefaultLatencyUsecs;
-  } else {
-    // kLowLatency is 0, so we treat it the same as a request for zero latency.
-    // Compute what the user asked for.
-    latency = rtc::kNumMicrosecsPerSec *
-        params.latency /
-        params.freq /
-        FrameSize(params);
-    // And this is what we'll actually use.
-    latency = std::max(latency, kMinimumLatencyUsecs);
-  }
-
-  ASSERT(params.format < arraysize(kCricketFormatToAlsaFormatTable));
-
-  err = symbol_table_.snd_pcm_set_params()(
-      handle,
-      kCricketFormatToAlsaFormatTable[params.format],
-      // SoundSystemInterface only supports interleaved audio.
-      SND_PCM_ACCESS_RW_INTERLEAVED,
-      params.channels,
-      params.freq,
-      1,  // Allow ALSA to resample.
-      latency);
-  if (err != 0) {
-    LOG(LS_ERROR) << "snd_pcm_set_params(): " << GetError(err);
-    goto fail;
-  }
-
-  err = symbol_table_.snd_pcm_prepare()(handle);
-  if (err != 0) {
-    LOG(LS_ERROR) << "snd_pcm_prepare(): " << GetError(err);
-    goto fail;
-  }
-
-  stream = (this->*start_fn)(
-      handle,
-      FrameSize(params),
-      // We set the wait time to twice the requested latency, so that wait
-      // timeouts should be rare.
-      2 * latency / rtc::kNumMicrosecsPerMillisec,
-      params.flags,
-      params.freq);
-  if (stream) {
-    return stream;
-  }
-  // Else fall through.
-
- fail:
-  err = symbol_table_.snd_pcm_close()(handle);
-  if (err != 0) {
-    LOG(LS_ERROR) << "snd_pcm_close(): " << GetError(err);
-  }
-  return NULL;
-}
-
-SoundOutputStreamInterface *AlsaSoundSystem::StartOutputStream(
-    snd_pcm_t *handle,
-    size_t frame_size,
-    int wait_timeout_ms,
-    int flags,
-    int freq) {
-  // Nothing to do here but instantiate the stream.
-  return new AlsaOutputStream(
-      this, handle, frame_size, wait_timeout_ms, flags, freq);
-}
-
-SoundInputStreamInterface *AlsaSoundSystem::StartInputStream(
-    snd_pcm_t *handle,
-    size_t frame_size,
-    int wait_timeout_ms,
-    int flags,
-    int freq) {
-  // Output streams start automatically once enough data has been written, but
-  // input streams must be started manually or else snd_pcm_wait() will never
-  // return true.
-  int err;
-  err = symbol_table_.snd_pcm_start()(handle);
-  if (err != 0) {
-    LOG(LS_ERROR) << "snd_pcm_start(): " << GetError(err);
-    return NULL;
-  }
-  return new AlsaInputStream(
-      this, handle, frame_size, wait_timeout_ms, flags, freq);
-}
-
-inline const char *AlsaSoundSystem::GetError(int err) {
-  return symbol_table_.snd_strerror()(err);
-}
-
-}  // namespace rtc