Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

14 files changed, 7096 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.cc
new file mode 100644
index 0000000000..5dfb91d6f4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.cc
@@ -0,0 +1,40 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "modules/audio_device/linux/alsasymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_alsa {
+
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(AlsaSymbolTable, "libasound.so.2")
+#define X(sym) LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(AlsaSymbolTable, sym)
+ALSA_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DEFINE_END(AlsaSymbolTable)
+
+}  // namespace adm_linux_alsa
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.h b/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.h
new file mode 100644
index 0000000000..c9970b02bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.h
@@ -0,0 +1,148 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_alsa {
+
+// The ALSA symbols we need, as an X-Macro list.
+// This list must contain precisely every libasound function that is used in
+// alsasoundsystem.cc.
+#define ALSA_SYMBOLS_LIST                      \
+  X(snd_device_name_free_hint)                 \
+  X(snd_device_name_get_hint)                  \
+  X(snd_device_name_hint)                      \
+  X(snd_pcm_avail_update)                      \
+  X(snd_pcm_close)                             \
+  X(snd_pcm_delay)                             \
+  X(snd_pcm_drop)                              \
+  X(snd_pcm_open)                              \
+  X(snd_pcm_prepare)                           \
+  X(snd_pcm_readi)                             \
+  X(snd_pcm_recover)                           \
+  X(snd_pcm_resume)                            \
+  X(snd_pcm_reset)                             \
+  X(snd_pcm_state)                             \
+  X(snd_pcm_set_params)                        \
+  X(snd_pcm_get_params)                        \
+  X(snd_pcm_start)                             \
+  X(snd_pcm_stream)                            \
+  X(snd_pcm_frames_to_bytes)                   \
+  X(snd_pcm_bytes_to_frames)                   \
+  X(snd_pcm_wait)                              \
+  X(snd_pcm_writei)                            \
+  X(snd_pcm_info_get_class)                    \
+  X(snd_pcm_info_get_subdevices_avail)         \
+  X(snd_pcm_info_get_subdevice_name)           \
+  X(snd_pcm_info_set_subdevice)                \
+  X(snd_pcm_info_get_id)                       \
+  X(snd_pcm_info_set_device)                   \
+  X(snd_pcm_info_set_stream)                   \
+  X(snd_pcm_info_get_name)                     \
+  X(snd_pcm_info_get_subdevices_count)         \
+  X(snd_pcm_info_sizeof)                       \
+  X(snd_pcm_hw_params)                         \
+  X(snd_pcm_hw_params_malloc)                  \
+  X(snd_pcm_hw_params_free)                    \
+  X(snd_pcm_hw_params_any)                     \
+  X(snd_pcm_hw_params_set_access)              \
+  X(snd_pcm_hw_params_set_format)              \
+  X(snd_pcm_hw_params_set_channels)            \
+  X(snd_pcm_hw_params_set_rate_near)           \
+  X(snd_pcm_hw_params_set_buffer_size_near)    \
+  X(snd_card_next)                             \
+  X(snd_card_get_name)                         \
+  X(snd_config_update)                         \
+  X(snd_config_copy)                           \
+  X(snd_config_get_id)                         \
+  X(snd_ctl_open)                              \
+  X(snd_ctl_close)                             \
+  X(snd_ctl_card_info)                         \
+  X(snd_ctl_card_info_sizeof)                  \
+  X(snd_ctl_card_info_get_id)                  \
+  X(snd_ctl_card_info_get_name)                \
+  X(snd_ctl_pcm_next_device)                   \
+  X(snd_ctl_pcm_info)                          \
+  X(snd_mixer_load)                            \
+  X(snd_mixer_free)                            \
+  X(snd_mixer_detach)                          \
+  X(snd_mixer_close)                           \
+  X(snd_mixer_open)                            \
+  X(snd_mixer_attach)                          \
+  X(snd_mixer_first_elem)                      \
+  X(snd_mixer_elem_next)                       \
+  X(snd_mixer_selem_get_name)                  \
+  X(snd_mixer_selem_is_active)                 \
+  X(snd_mixer_selem_register)                  \
+  X(snd_mixer_selem_set_playback_volume_all)   \
+  X(snd_mixer_selem_get_playback_volume)       \
+  X(snd_mixer_selem_has_playback_volume)       \
+  X(snd_mixer_selem_get_playback_volume_range) \
+  X(snd_mixer_selem_has_playback_switch)       \
+  X(snd_mixer_selem_get_playback_switch)       \
+  X(snd_mixer_selem_set_playback_switch_all)   \
+  X(snd_mixer_selem_has_capture_switch)        \
+  X(snd_mixer_selem_get_capture_switch)        \
+  X(snd_mixer_selem_set_capture_switch_all)    \
+  X(snd_mixer_selem_has_capture_volume)        \
+  X(snd_mixer_selem_set_capture_volume_all)    \
+  X(snd_mixer_selem_get_capture_volume)        \
+  X(snd_mixer_selem_get_capture_volume_range)  \
+  X(snd_dlopen)                                \
+  X(snd_dlclose)                               \
+  X(snd_config)                                \
+  X(snd_config_search)                         \
+  X(snd_config_get_string)                     \
+  X(snd_config_search_definition)              \
+  X(snd_config_get_type)                       \
+  X(snd_config_delete)                         \
+  X(snd_config_iterator_entry)                 \
+  X(snd_config_iterator_first)                 \
+  X(snd_config_iterator_next)                  \
+  X(snd_config_iterator_end)                   \
+  X(snd_config_delete_compound_members)        \
+  X(snd_config_get_integer)                    \
+  X(snd_config_get_bool)                       \
+  X(snd_dlsym)                                 \
+  X(snd_strerror)                              \
+  X(snd_lib_error)                             \
+  X(snd_lib_error_set_handler)
+
+LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(AlsaSymbolTable)
+#define X(sym) LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(AlsaSymbolTable, sym)
+ALSA_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DECLARE_END(AlsaSymbolTable)
+
+}  // namespace adm_linux_alsa
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
new file mode 100644
index 0000000000..1e0ac8be28
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -0,0 +1,1636 @@
+/*
+ *  Copyright (c) 2012 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 "modules/audio_device/linux/audio_device_alsa_linux.h"
+
+#include "modules/audio_device/audio_device_config.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/sleep.h"
+
+WebRTCAlsaSymbolTable* GetAlsaSymbolTable() {
+  static WebRTCAlsaSymbolTable* alsa_symbol_table = new WebRTCAlsaSymbolTable();
+  return alsa_symbol_table;
+}
+
+// Accesses ALSA functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libasound, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                                            \
+  LATESYM_GET(webrtc::adm_linux_alsa::AlsaSymbolTable, GetAlsaSymbolTable(), \
+              sym)
+
+// Redefine these here to be able to do late-binding
+#undef snd_ctl_card_info_alloca
+#define snd_ctl_card_info_alloca(ptr)                  \
+  do {                                                 \
+    *ptr = (snd_ctl_card_info_t*)__builtin_alloca(     \
+        LATE(snd_ctl_card_info_sizeof)());             \
+    memset(*ptr, 0, LATE(snd_ctl_card_info_sizeof)()); \
+  } while (0)
+
+#undef snd_pcm_info_alloca
+#define snd_pcm_info_alloca(pInfo)                                           \
+  do {                                                                       \
+    *pInfo = (snd_pcm_info_t*)__builtin_alloca(LATE(snd_pcm_info_sizeof)()); \
+    memset(*pInfo, 0, LATE(snd_pcm_info_sizeof)());                          \
+  } while (0)
+
+// snd_lib_error_handler_t
+void WebrtcAlsaErrorHandler(const char* file,
+                            int line,
+                            const char* function,
+                            int err,
+                            const char* fmt,
+                            ...) {}
+
+namespace webrtc {
+static const unsigned int ALSA_PLAYOUT_FREQ = 48000;
+static const unsigned int ALSA_PLAYOUT_CH = 2;
+static const unsigned int ALSA_PLAYOUT_LATENCY = 40 * 1000;  // in us
+static const unsigned int ALSA_CAPTURE_FREQ = 48000;
+static const unsigned int ALSA_CAPTURE_CH = 2;
+static const unsigned int ALSA_CAPTURE_LATENCY = 40 * 1000;  // in us
+static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5;     // in ms
+
+#define FUNC_GET_NUM_OF_DEVICE 0
+#define FUNC_GET_DEVICE_NAME 1
+#define FUNC_GET_DEVICE_NAME_FOR_AN_ENUM 2
+
+AudioDeviceLinuxALSA::AudioDeviceLinuxALSA()
+    : _ptrAudioBuffer(NULL),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0),
+      _inputDeviceIsSpecified(false),
+      _outputDeviceIsSpecified(false),
+      _handleRecord(NULL),
+      _handlePlayout(NULL),
+      _recordingBuffersizeInFrame(0),
+      _recordingPeriodSizeInFrame(0),
+      _playoutBufferSizeInFrame(0),
+      _playoutPeriodSizeInFrame(0),
+      _recordingBufferSizeIn10MS(0),
+      _playoutBufferSizeIn10MS(0),
+      _recordingFramesIn10MS(0),
+      _playoutFramesIn10MS(0),
+      _recordingFreq(ALSA_CAPTURE_FREQ),
+      _playoutFreq(ALSA_PLAYOUT_FREQ),
+      _recChannels(ALSA_CAPTURE_CH),
+      _playChannels(ALSA_PLAYOUT_CH),
+      _recordingBuffer(NULL),
+      _playoutBuffer(NULL),
+      _recordingFramesLeft(0),
+      _playoutFramesLeft(0),
+      _initialized(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _recordingDelay(0),
+      _playoutDelay(0) {
+  memset(_oldKeyState, 0, sizeof(_oldKeyState));
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+}
+
+// ----------------------------------------------------------------------------
+//  AudioDeviceLinuxALSA - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+
+  Terminate();
+
+  // Clean up the recording buffer and playout buffer.
+  if (_recordingBuffer) {
+    delete[] _recordingBuffer;
+    _recordingBuffer = NULL;
+  }
+  if (_playoutBuffer) {
+    delete[] _playoutBuffer;
+    _playoutBuffer = NULL;
+  }
+}
+
+void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  MutexLock lock(&mutex_);
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // Inform the AudioBuffer about default settings for this implementation.
+  // Set all values to zero here since the actual settings will be done by
+  // InitPlayout and InitRecording later.
+  _ptrAudioBuffer->SetRecordingSampleRate(0);
+  _ptrAudioBuffer->SetPlayoutSampleRate(0);
+  _ptrAudioBuffer->SetRecordingChannels(0);
+  _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+int32_t AudioDeviceLinuxALSA::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kLinuxAlsaAudio;
+  return 0;
+}
+
+AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() {
+  MutexLock lock(&mutex_);
+
+  // Load libasound
+  if (!GetAlsaSymbolTable()->Load()) {
+    // Alsa is not installed on this system
+    RTC_LOG(LS_ERROR) << "failed to load symbol table";
+    return InitStatus::OTHER_ERROR;
+  }
+
+  if (_initialized) {
+    return InitStatus::OK;
+  }
+#if defined(WEBRTC_USE_X11)
+  // Get X display handle for typing detection
+  _XDisplay = XOpenDisplay(NULL);
+  if (!_XDisplay) {
+    RTC_LOG(LS_WARNING)
+        << "failed to open X display, typing detection will not work";
+  }
+#endif
+
+  _initialized = true;
+
+  return InitStatus::OK;
+}
+
+int32_t AudioDeviceLinuxALSA::Terminate() {
+  if (!_initialized) {
+    return 0;
+  }
+
+  MutexLock lock(&mutex_);
+
+  _mixerManager.Close();
+
+  // RECORDING
+  mutex_.Unlock();
+  _ptrThreadRec.Finalize();
+
+  // PLAYOUT
+  _ptrThreadPlay.Finalize();
+  mutex_.Lock();
+
+#if defined(WEBRTC_USE_X11)
+  if (_XDisplay) {
+    XCloseDisplay(_XDisplay);
+    _XDisplay = NULL;
+  }
+#endif
+  _initialized = false;
+  _outputDeviceIsSpecified = false;
+  _inputDeviceIsSpecified = false;
+
+  return 0;
+}
+
+bool AudioDeviceLinuxALSA::Initialized() const {
+  return (_initialized);
+}
+
+int32_t AudioDeviceLinuxALSA::InitSpeaker() {
+  MutexLock lock(&mutex_);
+  return InitSpeakerLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::InitSpeakerLocked() {
+  if (_playing) {
+    return -1;
+  }
+
+  char devName[kAdmMaxDeviceNameSize] = {0};
+  GetDevicesInfo(2, true, _outputDeviceIndex, devName, kAdmMaxDeviceNameSize);
+  return _mixerManager.OpenSpeaker(devName);
+}
+
+int32_t AudioDeviceLinuxALSA::InitMicrophone() {
+  MutexLock lock(&mutex_);
+  return InitMicrophoneLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::InitMicrophoneLocked() {
+  if (_recording) {
+    return -1;
+  }
+
+  char devName[kAdmMaxDeviceNameSize] = {0};
+  GetDevicesInfo(2, false, _inputDeviceIndex, devName, kAdmMaxDeviceNameSize);
+  return _mixerManager.OpenMicrophone(devName);
+}
+
+bool AudioDeviceLinuxALSA::SpeakerIsInitialized() const {
+  return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceLinuxALSA::MicrophoneIsInitialized() const {
+  return (_mixerManager.MicrophoneIsInitialized());
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know that a volume control
+  // exists
+  available = true;
+
+  // Close the initialized output mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetSpeakerVolume(uint32_t volume) {
+  return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.SpeakerVolume(level) == -1) {
+    return -1;
+  }
+
+  volume = level;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MinSpeakerVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected speaker has a mute control
+  _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+  available = isAvailable;
+
+  // Close the initialized output mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetSpeakerMute(bool enable) {
+  return (_mixerManager.SetSpeakerMute(enable));
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerMute(bool& enabled) const {
+  bool muted(0);
+
+  if (_mixerManager.SpeakerMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MicrophoneMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone has a mute control
+  //
+  _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetMicrophoneMute(bool enable) {
+  return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+// ----------------------------------------------------------------------------
+//  MicrophoneMute
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceLinuxALSA::MicrophoneMute(bool& enabled) const {
+  bool muted(0);
+
+  if (_mixerManager.MicrophoneMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+
+  // If we already have initialized in stereo it's obviously available
+  if (_recIsInitialized && (2 == _recChannels)) {
+    available = true;
+    return 0;
+  }
+
+  // Save rec states and the number of rec channels
+  bool recIsInitialized = _recIsInitialized;
+  bool recording = _recording;
+  int recChannels = _recChannels;
+
+  available = false;
+
+  // Stop/uninitialize recording if initialized (and possibly started)
+  if (_recIsInitialized) {
+    StopRecordingLocked();
+  }
+
+  // Try init in stereo;
+  _recChannels = 2;
+  if (InitRecordingLocked() == 0) {
+    available = true;
+  }
+
+  // Stop/uninitialize recording
+  StopRecordingLocked();
+
+  // Recover previous states
+  _recChannels = recChannels;
+  if (recIsInitialized) {
+    InitRecordingLocked();
+  }
+  if (recording) {
+    StartRecording();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetStereoRecording(bool enable) {
+  if (enable)
+    _recChannels = 2;
+  else
+    _recChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const {
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+
+  // If we already have initialized in stereo it's obviously available
+  if (_playIsInitialized && (2 == _playChannels)) {
+    available = true;
+    return 0;
+  }
+
+  // Save rec states and the number of rec channels
+  bool playIsInitialized = _playIsInitialized;
+  bool playing = _playing;
+  int playChannels = _playChannels;
+
+  available = false;
+
+  // Stop/uninitialize recording if initialized (and possibly started)
+  if (_playIsInitialized) {
+    StopPlayoutLocked();
+  }
+
+  // Try init in stereo;
+  _playChannels = 2;
+  if (InitPlayoutLocked() == 0) {
+    available = true;
+  }
+
+  // Stop/uninitialize recording
+  StopPlayoutLocked();
+
+  // Recover previous states
+  _playChannels = playChannels;
+  if (playIsInitialized) {
+    InitPlayoutLocked();
+  }
+  if (playing) {
+    StartPlayout();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetStereoPlayout(bool enable) {
+  if (enable)
+    _playChannels = 2;
+  else
+    _playChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const {
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a volume control
+  // exists
+  available = true;
+
+  // Close the initialized input mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetMicrophoneVolume(uint32_t volume) {
+  return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+int32_t AudioDeviceLinuxALSA::MicrophoneVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.MicrophoneVolume(level) == -1) {
+    RTC_LOG(LS_WARNING) << "failed to retrive current microphone level";
+    return -1;
+  }
+
+  volume = level;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MinMicrophoneVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+
+  return 0;
+}
+
+int16_t AudioDeviceLinuxALSA::PlayoutDevices() {
+  return (int16_t)GetDevicesInfo(0, true);
+}
+
+int32_t AudioDeviceLinuxALSA::SetPlayoutDevice(uint16_t index) {
+  if (_playIsInitialized) {
+    return -1;
+  }
+
+  int32_t nDevices = GetDevicesInfo(0, true);
+  RTC_LOG(LS_VERBOSE) << "number of available audio output devices is "
+                      << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _outputDeviceIndex = index;
+  _outputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceLinuxALSA::PlayoutDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(PlayoutDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDevicesInfo(1, true, index, name, kAdmMaxDeviceNameSize);
+}
+
+int32_t AudioDeviceLinuxALSA::RecordingDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(RecordingDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize);
+}
+
+int16_t AudioDeviceLinuxALSA::RecordingDevices() {
+  return (int16_t)GetDevicesInfo(0, false);
+}
+
+int32_t AudioDeviceLinuxALSA::SetRecordingDevice(uint16_t index) {
+  if (_recIsInitialized) {
+    return -1;
+  }
+
+  int32_t nDevices = GetDevicesInfo(0, false);
+  RTC_LOG(LS_VERBOSE) << "number of availiable audio input devices is "
+                      << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _inputDeviceIndex = index;
+  _inputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceLinuxALSA::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceLinuxALSA::PlayoutIsAvailable(bool& available) {
+  available = false;
+
+  // Try to initialize the playout side with mono
+  // Assumes that user set num channels after calling this function
+  _playChannels = 1;
+  int32_t res = InitPlayout();
+
+  // Cancel effect of initialization
+  StopPlayout();
+
+  if (res != -1) {
+    available = true;
+  } else {
+    // It may be possible to play out in stereo
+    res = StereoPlayoutIsAvailable(available);
+    if (available) {
+      // Then set channels to 2 so InitPlayout doesn't fail
+      _playChannels = 2;
+    }
+  }
+
+  return res;
+}
+
+int32_t AudioDeviceLinuxALSA::RecordingIsAvailable(bool& available) {
+  available = false;
+
+  // Try to initialize the recording side with mono
+  // Assumes that user set num channels after calling this function
+  _recChannels = 1;
+  int32_t res = InitRecording();
+
+  // Cancel effect of initialization
+  StopRecording();
+
+  if (res != -1) {
+    available = true;
+  } else {
+    // It may be possible to record in stereo
+    res = StereoRecordingIsAvailable(available);
+    if (available) {
+      // Then set channels to 2 so InitPlayout doesn't fail
+      _recChannels = 2;
+    }
+  }
+
+  return res;
+}
+
+int32_t AudioDeviceLinuxALSA::InitPlayout() {
+  MutexLock lock(&mutex_);
+  return InitPlayoutLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::InitPlayoutLocked() {
+  int errVal = 0;
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_playIsInitialized) {
+    return 0;
+  }
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeakerLocked() == -1) {
+    RTC_LOG(LS_WARNING) << "InitSpeaker() failed";
+  }
+
+  // Start by closing any existing wave-output devices
+  //
+  if (_handlePlayout != NULL) {
+    LATE(snd_pcm_close)(_handlePlayout);
+    _handlePlayout = NULL;
+    _playIsInitialized = false;
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error closing current playout sound device, error: "
+                        << LATE(snd_strerror)(errVal);
+    }
+  }
+
+  // Open PCM device for playout
+  char deviceName[kAdmMaxDeviceNameSize] = {0};
+  GetDevicesInfo(2, true, _outputDeviceIndex, deviceName,
+                 kAdmMaxDeviceNameSize);
+
+  RTC_LOG(LS_VERBOSE) << "InitPlayout open (" << deviceName << ")";
+
+  errVal = LATE(snd_pcm_open)(&_handlePlayout, deviceName,
+                              SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+
+  if (errVal == -EBUSY)  // Device busy - try some more!
+  {
+    for (int i = 0; i < 5; i++) {
+      SleepMs(1000);
+      errVal = LATE(snd_pcm_open)(&_handlePlayout, deviceName,
+                                  SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+      if (errVal == 0) {
+        break;
+      }
+    }
+  }
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "unable to open playback device: "
+                      << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+    _handlePlayout = NULL;
+    return -1;
+  }
+
+  _playoutFramesIn10MS = _playoutFreq / 100;
+  if ((errVal = LATE(snd_pcm_set_params)(
+           _handlePlayout,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+           SND_PCM_FORMAT_S16_BE,
+#else
+           SND_PCM_FORMAT_S16_LE,                             // format
+#endif
+           SND_PCM_ACCESS_RW_INTERLEAVED,  // access
+           _playChannels,                  // channels
+           _playoutFreq,                   // rate
+           1,                              // soft_resample
+           ALSA_PLAYOUT_LATENCY  // 40*1000 //latency required overall latency
+                                 // in us
+           )) < 0) {             /* 0.5sec */
+    _playoutFramesIn10MS = 0;
+    RTC_LOG(LS_ERROR) << "unable to set playback device: "
+                      << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+    ErrorRecovery(errVal, _handlePlayout);
+    errVal = LATE(snd_pcm_close)(_handlePlayout);
+    _handlePlayout = NULL;
+    return -1;
+  }
+
+  errVal = LATE(snd_pcm_get_params)(_handlePlayout, &_playoutBufferSizeInFrame,
+                                    &_playoutPeriodSizeInFrame);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_pcm_get_params: " << LATE(snd_strerror)(errVal)
+                      << " (" << errVal << ")";
+    _playoutBufferSizeInFrame = 0;
+    _playoutPeriodSizeInFrame = 0;
+  } else {
+    RTC_LOG(LS_VERBOSE) << "playout snd_pcm_get_params buffer_size:"
+                        << _playoutBufferSizeInFrame
+                        << " period_size :" << _playoutPeriodSizeInFrame;
+  }
+
+  if (_ptrAudioBuffer) {
+    // Update webrtc audio buffer with the selected parameters
+    _ptrAudioBuffer->SetPlayoutSampleRate(_playoutFreq);
+    _ptrAudioBuffer->SetPlayoutChannels(_playChannels);
+  }
+
+  // Set play buffer size
+  _playoutBufferSizeIn10MS =
+      LATE(snd_pcm_frames_to_bytes)(_handlePlayout, _playoutFramesIn10MS);
+
+  // Init varaibles used for play
+
+  if (_handlePlayout != NULL) {
+    _playIsInitialized = true;
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+int32_t AudioDeviceLinuxALSA::InitRecording() {
+  MutexLock lock(&mutex_);
+  return InitRecordingLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::InitRecordingLocked() {
+  int errVal = 0;
+
+  if (_recording) {
+    return -1;
+  }
+
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_recIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophoneLocked() == -1) {
+    RTC_LOG(LS_WARNING) << "InitMicrophone() failed";
+  }
+
+  // Start by closing any existing pcm-input devices
+  //
+  if (_handleRecord != NULL) {
+    int errVal = LATE(snd_pcm_close)(_handleRecord);
+    _handleRecord = NULL;
+    _recIsInitialized = false;
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR)
+          << "Error closing current recording sound device, error: "
+          << LATE(snd_strerror)(errVal);
+    }
+  }
+
+  // Open PCM device for recording
+  // The corresponding settings for playout are made after the record settings
+  char deviceName[kAdmMaxDeviceNameSize] = {0};
+  GetDevicesInfo(2, false, _inputDeviceIndex, deviceName,
+                 kAdmMaxDeviceNameSize);
+
+  RTC_LOG(LS_VERBOSE) << "InitRecording open (" << deviceName << ")";
+  errVal = LATE(snd_pcm_open)(&_handleRecord, deviceName,
+                              SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
+
+  // Available modes: 0 = blocking, SND_PCM_NONBLOCK, SND_PCM_ASYNC
+  if (errVal == -EBUSY)  // Device busy - try some more!
+  {
+    for (int i = 0; i < 5; i++) {
+      SleepMs(1000);
+      errVal = LATE(snd_pcm_open)(&_handleRecord, deviceName,
+                                  SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
+      if (errVal == 0) {
+        break;
+      }
+    }
+  }
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "unable to open record device: "
+                      << LATE(snd_strerror)(errVal);
+    _handleRecord = NULL;
+    return -1;
+  }
+
+  _recordingFramesIn10MS = _recordingFreq / 100;
+  if ((errVal =
+           LATE(snd_pcm_set_params)(_handleRecord,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+                                    SND_PCM_FORMAT_S16_BE,  // format
+#else
+                                    SND_PCM_FORMAT_S16_LE,    // format
+#endif
+                                    SND_PCM_ACCESS_RW_INTERLEAVED,  // access
+                                    _recChannels,                   // channels
+                                    _recordingFreq,                 // rate
+                                    1,                    // soft_resample
+                                    ALSA_CAPTURE_LATENCY  // latency in us
+                                    )) < 0) {
+    // Fall back to another mode then.
+    if (_recChannels == 1)
+      _recChannels = 2;
+    else
+      _recChannels = 1;
+
+    if ((errVal =
+             LATE(snd_pcm_set_params)(_handleRecord,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+                                      SND_PCM_FORMAT_S16_BE,  // format
+#else
+                                      SND_PCM_FORMAT_S16_LE,  // format
+#endif
+                                      SND_PCM_ACCESS_RW_INTERLEAVED,  // access
+                                      _recChannels,         // channels
+                                      _recordingFreq,       // rate
+                                      1,                    // soft_resample
+                                      ALSA_CAPTURE_LATENCY  // latency in us
+                                      )) < 0) {
+      _recordingFramesIn10MS = 0;
+      RTC_LOG(LS_ERROR) << "unable to set record settings: "
+                        << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+      ErrorRecovery(errVal, _handleRecord);
+      errVal = LATE(snd_pcm_close)(_handleRecord);
+      _handleRecord = NULL;
+      return -1;
+    }
+  }
+
+  errVal = LATE(snd_pcm_get_params)(_handleRecord, &_recordingBuffersizeInFrame,
+                                    &_recordingPeriodSizeInFrame);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_pcm_get_params " << LATE(snd_strerror)(errVal)
+                      << " (" << errVal << ")";
+    _recordingBuffersizeInFrame = 0;
+    _recordingPeriodSizeInFrame = 0;
+  } else {
+    RTC_LOG(LS_VERBOSE) << "capture snd_pcm_get_params, buffer_size:"
+                        << _recordingBuffersizeInFrame
+                        << ", period_size:" << _recordingPeriodSizeInFrame;
+  }
+
+  if (_ptrAudioBuffer) {
+    // Update webrtc audio buffer with the selected parameters
+    _ptrAudioBuffer->SetRecordingSampleRate(_recordingFreq);
+    _ptrAudioBuffer->SetRecordingChannels(_recChannels);
+  }
+
+  // Set rec buffer size and create buffer
+  _recordingBufferSizeIn10MS =
+      LATE(snd_pcm_frames_to_bytes)(_handleRecord, _recordingFramesIn10MS);
+
+  if (_handleRecord != NULL) {
+    // Mark recording side as initialized
+    _recIsInitialized = true;
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+int32_t AudioDeviceLinuxALSA::StartRecording() {
+  if (!_recIsInitialized) {
+    return -1;
+  }
+
+  if (_recording) {
+    return 0;
+  }
+
+  _recording = true;
+
+  int errVal = 0;
+  _recordingFramesLeft = _recordingFramesIn10MS;
+
+  // Make sure we only create the buffer once.
+  if (!_recordingBuffer)
+    _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
+  if (!_recordingBuffer) {
+    RTC_LOG(LS_ERROR) << "failed to alloc recording buffer";
+    _recording = false;
+    return -1;
+  }
+  // RECORDING
+  _ptrThreadRec = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (RecThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_capture_thread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  errVal = LATE(snd_pcm_prepare)(_handleRecord);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "capture snd_pcm_prepare failed ("
+                      << LATE(snd_strerror)(errVal) << ")\n";
+    // just log error
+    // if snd_pcm_open fails will return -1
+  }
+
+  errVal = LATE(snd_pcm_start)(_handleRecord);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "capture snd_pcm_start err: "
+                      << LATE(snd_strerror)(errVal);
+    errVal = LATE(snd_pcm_start)(_handleRecord);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "capture snd_pcm_start 2nd try err: "
+                        << LATE(snd_strerror)(errVal);
+      StopRecording();
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StopRecording() {
+  MutexLock lock(&mutex_);
+  return StopRecordingLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::StopRecordingLocked() {
+  if (!_recIsInitialized) {
+    return 0;
+  }
+
+  if (_handleRecord == NULL) {
+    return -1;
+  }
+
+  // Make sure we don't start recording (it's asynchronous).
+  _recIsInitialized = false;
+  _recording = false;
+
+  _ptrThreadRec.Finalize();
+
+  _recordingFramesLeft = 0;
+  if (_recordingBuffer) {
+    delete[] _recordingBuffer;
+    _recordingBuffer = NULL;
+  }
+
+  // Stop and close pcm recording device.
+  int errVal = LATE(snd_pcm_drop)(_handleRecord);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error stop recording: " << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  errVal = LATE(snd_pcm_close)(_handleRecord);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error closing record sound device, error: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  // Check if we have muted and unmute if so.
+  bool muteEnabled = false;
+  MicrophoneMute(muteEnabled);
+  if (muteEnabled) {
+    SetMicrophoneMute(false);
+  }
+
+  // set the pcm input handle to NULL
+  _handleRecord = NULL;
+  return 0;
+}
+
+bool AudioDeviceLinuxALSA::RecordingIsInitialized() const {
+  return (_recIsInitialized);
+}
+
+bool AudioDeviceLinuxALSA::Recording() const {
+  return (_recording);
+}
+
+bool AudioDeviceLinuxALSA::PlayoutIsInitialized() const {
+  return (_playIsInitialized);
+}
+
+int32_t AudioDeviceLinuxALSA::StartPlayout() {
+  if (!_playIsInitialized) {
+    return -1;
+  }
+
+  if (_playing) {
+    return 0;
+  }
+
+  _playing = true;
+
+  _playoutFramesLeft = 0;
+  if (!_playoutBuffer)
+    _playoutBuffer = new int8_t[_playoutBufferSizeIn10MS];
+  if (!_playoutBuffer) {
+    RTC_LOG(LS_ERROR) << "failed to alloc playout buf";
+    _playing = false;
+    return -1;
+  }
+
+  // PLAYOUT
+  _ptrThreadPlay = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (PlayThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_play_thread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "playout snd_pcm_prepare failed ("
+                      << LATE(snd_strerror)(errVal) << ")\n";
+    // just log error
+    // if snd_pcm_open fails will return -1
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StopPlayout() {
+  MutexLock lock(&mutex_);
+  return StopPlayoutLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::StopPlayoutLocked() {
+  if (!_playIsInitialized) {
+    return 0;
+  }
+
+  if (_handlePlayout == NULL) {
+    return -1;
+  }
+
+  _playing = false;
+
+  // stop playout thread first
+  _ptrThreadPlay.Finalize();
+
+  _playoutFramesLeft = 0;
+  delete[] _playoutBuffer;
+  _playoutBuffer = NULL;
+
+  // stop and close pcm playout device
+  int errVal = LATE(snd_pcm_drop)(_handlePlayout);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error stop playing: " << LATE(snd_strerror)(errVal);
+  }
+
+  errVal = LATE(snd_pcm_close)(_handlePlayout);
+  if (errVal < 0)
+    RTC_LOG(LS_ERROR) << "Error closing playout sound device, error: "
+                      << LATE(snd_strerror)(errVal);
+
+  // set the pcm input handle to NULL
+  _playIsInitialized = false;
+  _handlePlayout = NULL;
+  RTC_LOG(LS_VERBOSE) << "handle_playout is now set to NULL";
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::PlayoutDelay(uint16_t& delayMS) const {
+  delayMS = (uint16_t)_playoutDelay * 1000 / _playoutFreq;
+  return 0;
+}
+
+bool AudioDeviceLinuxALSA::Playing() const {
+  return (_playing);
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+int32_t AudioDeviceLinuxALSA::GetDevicesInfo(const int32_t function,
+                                             const bool playback,
+                                             const int32_t enumDeviceNo,
+                                             char* enumDeviceName,
+                                             const int32_t ednLen) const {
+  // Device enumeration based on libjingle implementation
+  // by Tristan Schmelcher at Google Inc.
+
+  const char* type = playback ? "Output" : "Input";
+  // dmix and dsnoop are only for playback and capture, respectively, but ALSA
+  // stupidly includes them in both lists.
+  const char* ignorePrefix = playback ? "dsnoop:" : "dmix:";
+  // (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.)
+
+  int err;
+  int enumCount(0);
+  bool keepSearching(true);
+
+  // From Chromium issue 95797
+  // Loop through the sound cards to get Alsa device hints.
+  // Don't use snd_device_name_hint(-1,..) since there is a access violation
+  // inside this ALSA API with libasound.so.2.0.0.
+  int card = -1;
+  while (!(LATE(snd_card_next)(&card)) && (card >= 0) && keepSearching) {
+    void** hints;
+    err = LATE(snd_device_name_hint)(card, "pcm", &hints);
+    if (err != 0) {
+      RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name hint error: "
+                        << LATE(snd_strerror)(err);
+      return -1;
+    }
+
+    enumCount++;  // default is 0
+    if ((function == FUNC_GET_DEVICE_NAME ||
+         function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM) &&
+        enumDeviceNo == 0) {
+      strcpy(enumDeviceName, "default");
+
+      err = LATE(snd_device_name_free_hint)(hints);
+      if (err != 0) {
+        RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: "
+                          << LATE(snd_strerror)(err);
+      }
+
+      return 0;
+    }
+
+    for (void** list = hints; *list != NULL; ++list) {
+      char* actualType = LATE(snd_device_name_get_hint)(*list, "IOID");
+      if (actualType) {  // NULL means it's both.
+        bool wrongType = (strcmp(actualType, type) != 0);
+        free(actualType);
+        if (wrongType) {
+          // Wrong type of device (i.e., input vs. output).
+          continue;
+        }
+      }
+
+      char* name = LATE(snd_device_name_get_hint)(*list, "NAME");
+      if (!name) {
+        RTC_LOG(LS_ERROR) << "Device has no name";
+        // Skip it.
+        continue;
+      }
+
+      // Now check if we actually want to show this device.
+      if (strcmp(name, "default") != 0 && strcmp(name, "null") != 0 &&
+          strcmp(name, "pulse") != 0 &&
+          strncmp(name, ignorePrefix, strlen(ignorePrefix)) != 0) {
+        // Yes, we do.
+        char* desc = LATE(snd_device_name_get_hint)(*list, "DESC");
+        if (!desc) {
+          // Virtual devices don't necessarily have descriptions.
+          // Use their names instead.
+          desc = name;
+        }
+
+        if (FUNC_GET_NUM_OF_DEVICE == function) {
+          RTC_LOG(LS_VERBOSE) << "Enum device " << enumCount << " - " << name;
+        }
+        if ((FUNC_GET_DEVICE_NAME == function) && (enumDeviceNo == enumCount)) {
+          // We have found the enum device, copy the name to buffer.
+          strncpy(enumDeviceName, desc, ednLen);
+          enumDeviceName[ednLen - 1] = '\0';
+          keepSearching = false;
+          // Replace '\n' with '-'.
+          char* pret = strchr(enumDeviceName, '\n' /*0xa*/);  // LF
+          if (pret)
+            *pret = '-';
+        }
+        if ((FUNC_GET_DEVICE_NAME_FOR_AN_ENUM == function) &&
+            (enumDeviceNo == enumCount)) {
+          // We have found the enum device, copy the name to buffer.
+          strncpy(enumDeviceName, name, ednLen);
+          enumDeviceName[ednLen - 1] = '\0';
+          keepSearching = false;
+        }
+
+        if (keepSearching)
+          ++enumCount;
+
+        if (desc != name)
+          free(desc);
+      }
+
+      free(name);
+
+      if (!keepSearching)
+        break;
+    }
+
+    err = LATE(snd_device_name_free_hint)(hints);
+    if (err != 0) {
+      RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: "
+                        << LATE(snd_strerror)(err);
+      // Continue and return true anyway, since we did get the whole list.
+    }
+  }
+
+  if (FUNC_GET_NUM_OF_DEVICE == function) {
+    if (enumCount == 1)  // only default?
+      enumCount = 0;
+    return enumCount;  // Normal return point for function 0
+  }
+
+  if (keepSearching) {
+    // If we get here for function 1 and 2, we didn't find the specified
+    // enum device.
+    RTC_LOG(LS_ERROR)
+        << "GetDevicesInfo - Could not find device name or numbers";
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const {
+  if (_handleRecord == NULL) {
+    RTC_LOG(LS_ERROR) << "input state has been modified during unlocked period";
+    return -1;
+  }
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const {
+  if (_handlePlayout == NULL) {
+    RTC_LOG(LS_ERROR)
+        << "output state has been modified during unlocked period";
+    return -1;
+  }
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::ErrorRecovery(int32_t error,
+                                            snd_pcm_t* deviceHandle) {
+  int st = LATE(snd_pcm_state)(deviceHandle);
+  RTC_LOG(LS_VERBOSE) << "Trying to recover from "
+                      << ((LATE(snd_pcm_stream)(deviceHandle) ==
+                           SND_PCM_STREAM_CAPTURE)
+                              ? "capture"
+                              : "playout")
+                      << " error: " << LATE(snd_strerror)(error) << " ("
+                      << error << ") (state " << st << ")";
+
+  // It is recommended to use snd_pcm_recover for all errors. If that function
+  // cannot handle the error, the input error code will be returned, otherwise
+  // 0 is returned. From snd_pcm_recover API doc: "This functions handles
+  // -EINTR (4) (interrupted system call), -EPIPE (32) (playout overrun or
+  // capture underrun) and -ESTRPIPE (86) (stream is suspended) error codes
+  // trying to prepare given stream for next I/O."
+
+  /** Open */
+  //    SND_PCM_STATE_OPEN = 0,
+  /** Setup installed */
+  //    SND_PCM_STATE_SETUP,
+  /** Ready to start */
+  //    SND_PCM_STATE_PREPARED,
+  /** Running */
+  //    SND_PCM_STATE_RUNNING,
+  /** Stopped: underrun (playback) or overrun (capture) detected */
+  //    SND_PCM_STATE_XRUN,= 4
+  /** Draining: running (playback) or stopped (capture) */
+  //    SND_PCM_STATE_DRAINING,
+  /** Paused */
+  //    SND_PCM_STATE_PAUSED,
+  /** Hardware is suspended */
+  //    SND_PCM_STATE_SUSPENDED,
+  //  ** Hardware is disconnected */
+  //    SND_PCM_STATE_DISCONNECTED,
+  //    SND_PCM_STATE_LAST = SND_PCM_STATE_DISCONNECTED
+
+  // snd_pcm_recover isn't available in older alsa, e.g. on the FC4 machine
+  // in Sthlm lab.
+
+  int res = LATE(snd_pcm_recover)(deviceHandle, error, 1);
+  if (0 == res) {
+    RTC_LOG(LS_VERBOSE) << "Recovery - snd_pcm_recover OK";
+
+    if ((error == -EPIPE || error == -ESTRPIPE) &&  // Buf underrun/overrun.
+        _recording &&
+        LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE) {
+      // For capture streams we also have to repeat the explicit start()
+      // to get data flowing again.
+      int err = LATE(snd_pcm_start)(deviceHandle);
+      if (err != 0) {
+        RTC_LOG(LS_ERROR) << "Recovery - snd_pcm_start error: " << err;
+        return -1;
+      }
+    }
+
+    if ((error == -EPIPE || error == -ESTRPIPE) &&  // Buf underrun/overrun.
+        _playing &&
+        LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_PLAYBACK) {
+      // For capture streams we also have to repeat the explicit start() to get
+      // data flowing again.
+      int err = LATE(snd_pcm_start)(deviceHandle);
+      if (err != 0) {
+        RTC_LOG(LS_ERROR) << "Recovery - snd_pcm_start error: "
+                          << LATE(snd_strerror)(err);
+        return -1;
+      }
+    }
+
+    return -EPIPE == error ? 1 : 0;
+  } else {
+    RTC_LOG(LS_ERROR) << "Unrecoverable alsa stream error: " << res;
+  }
+
+  return res;
+}
+
+// ============================================================================
+//                                  Thread Methods
+// ============================================================================
+
+bool AudioDeviceLinuxALSA::PlayThreadProcess() {
+  if (!_playing)
+    return false;
+
+  int err;
+  snd_pcm_sframes_t frames;
+  snd_pcm_sframes_t avail_frames;
+
+  Lock();
+  // return a positive number of frames ready otherwise a negative error code
+  avail_frames = LATE(snd_pcm_avail_update)(_handlePlayout);
+  if (avail_frames < 0) {
+    RTC_LOG(LS_ERROR) << "playout snd_pcm_avail_update error: "
+                      << LATE(snd_strerror)(avail_frames);
+    ErrorRecovery(avail_frames, _handlePlayout);
+    UnLock();
+    return true;
+  } else if (avail_frames == 0) {
+    UnLock();
+
+    // maximum tixe in milliseconds to wait, a negative value means infinity
+    err = LATE(snd_pcm_wait)(_handlePlayout, 2);
+    if (err == 0) {  // timeout occured
+      RTC_LOG(LS_VERBOSE) << "playout snd_pcm_wait timeout";
+    }
+
+    return true;
+  }
+
+  if (_playoutFramesLeft <= 0) {
+    UnLock();
+    _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
+    Lock();
+
+    _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+    RTC_DCHECK_EQ(_playoutFramesLeft, _playoutFramesIn10MS);
+  }
+
+  if (static_cast<uint32_t>(avail_frames) > _playoutFramesLeft)
+    avail_frames = _playoutFramesLeft;
+
+  int size = LATE(snd_pcm_frames_to_bytes)(_handlePlayout, _playoutFramesLeft);
+  frames = LATE(snd_pcm_writei)(
+      _handlePlayout, &_playoutBuffer[_playoutBufferSizeIn10MS - size],
+      avail_frames);
+
+  if (frames < 0) {
+    RTC_LOG(LS_VERBOSE) << "playout snd_pcm_writei error: "
+                        << LATE(snd_strerror)(frames);
+    _playoutFramesLeft = 0;
+    ErrorRecovery(frames, _handlePlayout);
+    UnLock();
+    return true;
+  } else {
+    RTC_DCHECK_EQ(frames, avail_frames);
+    _playoutFramesLeft -= frames;
+  }
+
+  UnLock();
+  return true;
+}
+
+bool AudioDeviceLinuxALSA::RecThreadProcess() {
+  if (!_recording)
+    return false;
+
+  int err;
+  snd_pcm_sframes_t frames;
+  snd_pcm_sframes_t avail_frames;
+  int8_t buffer[_recordingBufferSizeIn10MS];
+
+  Lock();
+
+  // return a positive number of frames ready otherwise a negative error code
+  avail_frames = LATE(snd_pcm_avail_update)(_handleRecord);
+  if (avail_frames < 0) {
+    RTC_LOG(LS_ERROR) << "capture snd_pcm_avail_update error: "
+                      << LATE(snd_strerror)(avail_frames);
+    ErrorRecovery(avail_frames, _handleRecord);
+    UnLock();
+    return true;
+  } else if (avail_frames == 0) {  // no frame is available now
+    UnLock();
+
+    // maximum time in milliseconds to wait, a negative value means infinity
+    err = LATE(snd_pcm_wait)(_handleRecord, ALSA_CAPTURE_WAIT_TIMEOUT);
+    if (err == 0)  // timeout occured
+      RTC_LOG(LS_VERBOSE) << "capture snd_pcm_wait timeout";
+
+    return true;
+  }
+
+  if (static_cast<uint32_t>(avail_frames) > _recordingFramesLeft)
+    avail_frames = _recordingFramesLeft;
+
+  frames = LATE(snd_pcm_readi)(_handleRecord, buffer,
+                               avail_frames);  // frames to be written
+  if (frames < 0) {
+    RTC_LOG(LS_ERROR) << "capture snd_pcm_readi error: "
+                      << LATE(snd_strerror)(frames);
+    ErrorRecovery(frames, _handleRecord);
+    UnLock();
+    return true;
+  } else if (frames > 0) {
+    RTC_DCHECK_EQ(frames, avail_frames);
+
+    int left_size =
+        LATE(snd_pcm_frames_to_bytes)(_handleRecord, _recordingFramesLeft);
+    int size = LATE(snd_pcm_frames_to_bytes)(_handleRecord, frames);
+
+    memcpy(&_recordingBuffer[_recordingBufferSizeIn10MS - left_size], buffer,
+           size);
+    _recordingFramesLeft -= frames;
+
+    if (!_recordingFramesLeft) {  // buf is full
+      _recordingFramesLeft = _recordingFramesIn10MS;
+
+      // store the recorded buffer (no action will be taken if the
+      // #recorded samples is not a full buffer)
+      _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+                                         _recordingFramesIn10MS);
+
+      // calculate delay
+      _playoutDelay = 0;
+      _recordingDelay = 0;
+      if (_handlePlayout) {
+        err = LATE(snd_pcm_delay)(_handlePlayout,
+                                  &_playoutDelay);  // returned delay in frames
+        if (err < 0) {
+          // TODO(xians): Shall we call ErrorRecovery() here?
+          _playoutDelay = 0;
+          RTC_LOG(LS_ERROR)
+              << "playout snd_pcm_delay: " << LATE(snd_strerror)(err);
+        }
+      }
+
+      err = LATE(snd_pcm_delay)(_handleRecord,
+                                &_recordingDelay);  // returned delay in frames
+      if (err < 0) {
+        // TODO(xians): Shall we call ErrorRecovery() here?
+        _recordingDelay = 0;
+        RTC_LOG(LS_ERROR) << "capture snd_pcm_delay: "
+                          << LATE(snd_strerror)(err);
+      }
+
+      // TODO(xians): Shall we add 10ms buffer delay to the record delay?
+      _ptrAudioBuffer->SetVQEData(_playoutDelay * 1000 / _playoutFreq,
+                                  _recordingDelay * 1000 / _recordingFreq);
+
+      _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+      // Deliver recorded samples at specified sample rate, mic level etc.
+      // to the observer using callback.
+      UnLock();
+      _ptrAudioBuffer->DeliverRecordedData();
+      Lock();
+    }
+  }
+
+  UnLock();
+  return true;
+}
+
+bool AudioDeviceLinuxALSA::KeyPressed() const {
+#if defined(WEBRTC_USE_X11)
+  char szKey[32];
+  unsigned int i = 0;
+  char state = 0;
+
+  if (!_XDisplay)
+    return false;
+
+  // Check key map status
+  XQueryKeymap(_XDisplay, szKey);
+
+  // A bit change in keymap means a key is pressed
+  for (i = 0; i < sizeof(szKey); i++)
+    state |= (szKey[i] ^ _oldKeyState[i]) & szKey[i];
+
+  // Save old state
+  memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState));
+  return (state != 0);
+#else
+  return false;
+#endif
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.h
new file mode 100644
index 0000000000..23e21d3ce9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.h
@@ -0,0 +1,208 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
+
+#include <memory>
+
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+
+#if defined(WEBRTC_USE_X11)
+#include <X11/Xlib.h>
+#endif
+#include <alsa/asoundlib.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+
+typedef webrtc::adm_linux_alsa::AlsaSymbolTable WebRTCAlsaSymbolTable;
+WebRTCAlsaSymbolTable* GetAlsaSymbolTable();
+
+namespace webrtc {
+
+class AudioDeviceLinuxALSA : public AudioDeviceGeneric {
+ public:
+  AudioDeviceLinuxALSA();
+  virtual ~AudioDeviceLinuxALSA();
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+
+ private:
+  int32_t InitRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t StopRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t StopPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t GetDevicesInfo(int32_t function,
+                         bool playback,
+                         int32_t enumDeviceNo = 0,
+                         char* enumDeviceName = NULL,
+                         int32_t ednLen = 0) const;
+  int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
+
+  bool KeyPressed() const;
+
+  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
+  void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
+
+  inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
+  inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
+
+  static void RecThreadFunc(void*);
+  static void PlayThreadFunc(void*);
+  bool RecThreadProcess();
+  bool PlayThreadProcess();
+
+  AudioDeviceBuffer* _ptrAudioBuffer;
+
+  Mutex mutex_;
+
+  rtc::PlatformThread _ptrThreadRec;
+  rtc::PlatformThread _ptrThreadPlay;
+
+  AudioMixerManagerLinuxALSA _mixerManager;
+
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
+
+  snd_pcm_t* _handleRecord;
+  snd_pcm_t* _handlePlayout;
+
+  snd_pcm_uframes_t _recordingBuffersizeInFrame;
+  snd_pcm_uframes_t _recordingPeriodSizeInFrame;
+  snd_pcm_uframes_t _playoutBufferSizeInFrame;
+  snd_pcm_uframes_t _playoutPeriodSizeInFrame;
+
+  ssize_t _recordingBufferSizeIn10MS;
+  ssize_t _playoutBufferSizeIn10MS;
+  uint32_t _recordingFramesIn10MS;
+  uint32_t _playoutFramesIn10MS;
+
+  uint32_t _recordingFreq;
+  uint32_t _playoutFreq;
+  uint8_t _recChannels;
+  uint8_t _playChannels;
+
+  int8_t* _recordingBuffer;  // in byte
+  int8_t* _playoutBuffer;    // in byte
+  uint32_t _recordingFramesLeft;
+  uint32_t _playoutFramesLeft;
+
+  bool _initialized;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+
+  snd_pcm_sframes_t _recordingDelay;
+  snd_pcm_sframes_t _playoutDelay;
+
+  char _oldKeyState[32];
+#if defined(WEBRTC_USE_X11)
+  Display* _XDisplay;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_ALSA_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
new file mode 100644
index 0000000000..90cd58c497
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
@@ -0,0 +1,2286 @@
+/*
+ *  Copyright (c) 2012 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 "modules/audio_device/linux/audio_device_pulse_linux.h"
+
+#include <string.h>
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+
+WebRTCPulseSymbolTable* GetPulseSymbolTable() {
+  static WebRTCPulseSymbolTable* pulse_symbol_table =
+      new WebRTCPulseSymbolTable();
+  return pulse_symbol_table;
+}
+
+// Accesses Pulse functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libpulse, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                             \
+  LATESYM_GET(webrtc::adm_linux_pulse::PulseAudioSymbolTable, \
+              GetPulseSymbolTable(), sym)
+
+namespace webrtc {
+
+AudioDeviceLinuxPulse::AudioDeviceLinuxPulse()
+    : _ptrAudioBuffer(NULL),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0),
+      _inputDeviceIsSpecified(false),
+      _outputDeviceIsSpecified(false),
+      sample_rate_hz_(0),
+      _recChannels(1),
+      _playChannels(1),
+      _initialized(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _startRec(false),
+      _startPlay(false),
+      update_speaker_volume_at_startup_(false),
+      quit_(false),
+      _sndCardPlayDelay(0),
+      _writeErrors(0),
+      _deviceIndex(-1),
+      _numPlayDevices(0),
+      _numRecDevices(0),
+      _playDeviceName(NULL),
+      _recDeviceName(NULL),
+      _playDisplayDeviceName(NULL),
+      _recDisplayDeviceName(NULL),
+      _playBuffer(NULL),
+      _playbackBufferSize(0),
+      _playbackBufferUnused(0),
+      _tempBufferSpace(0),
+      _recBuffer(NULL),
+      _recordBufferSize(0),
+      _recordBufferUsed(0),
+      _tempSampleData(NULL),
+      _tempSampleDataSize(0),
+      _configuredLatencyPlay(0),
+      _configuredLatencyRec(0),
+      _paDeviceIndex(-1),
+      _paStateChanged(false),
+      _paMainloop(NULL),
+      _paMainloopApi(NULL),
+      _paContext(NULL),
+      _recStream(NULL),
+      _playStream(NULL),
+      _recStreamFlags(0),
+      _playStreamFlags(0) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+
+  memset(_paServerVersion, 0, sizeof(_paServerVersion));
+  memset(&_playBufferAttr, 0, sizeof(_playBufferAttr));
+  memset(&_recBufferAttr, 0, sizeof(_recBufferAttr));
+  memset(_oldKeyState, 0, sizeof(_oldKeyState));
+}
+
+AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+
+  if (_recBuffer) {
+    delete[] _recBuffer;
+    _recBuffer = NULL;
+  }
+  if (_playBuffer) {
+    delete[] _playBuffer;
+    _playBuffer = NULL;
+  }
+  if (_playDeviceName) {
+    delete[] _playDeviceName;
+    _playDeviceName = NULL;
+  }
+  if (_recDeviceName) {
+    delete[] _recDeviceName;
+    _recDeviceName = NULL;
+  }
+}
+
+void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // Inform the AudioBuffer about default settings for this implementation.
+  // Set all values to zero here since the actual settings will be done by
+  // InitPlayout and InitRecording later.
+  _ptrAudioBuffer->SetRecordingSampleRate(0);
+  _ptrAudioBuffer->SetPlayoutSampleRate(0);
+  _ptrAudioBuffer->SetRecordingChannels(0);
+  _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+// ----------------------------------------------------------------------------
+//  ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceLinuxPulse::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kLinuxPulseAudio;
+  return 0;
+}
+
+AudioDeviceGeneric::InitStatus AudioDeviceLinuxPulse::Init() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_initialized) {
+    return InitStatus::OK;
+  }
+
+  // Initialize PulseAudio
+  if (InitPulseAudio() < 0) {
+    RTC_LOG(LS_ERROR) << "failed to initialize PulseAudio";
+    if (TerminatePulseAudio() < 0) {
+      RTC_LOG(LS_ERROR) << "failed to terminate PulseAudio";
+    }
+    return InitStatus::OTHER_ERROR;
+  }
+
+#if defined(WEBRTC_USE_X11)
+  // Get X display handle for typing detection
+  _XDisplay = XOpenDisplay(NULL);
+  if (!_XDisplay) {
+    RTC_LOG(LS_WARNING)
+        << "failed to open X display, typing detection will not work";
+  }
+#endif
+
+  // RECORDING
+  const auto attributes =
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
+  _ptrThreadRec = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (RecThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_rec_thread", attributes);
+
+  // PLAYOUT
+  _ptrThreadPlay = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (PlayThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_play_thread", attributes);
+  _initialized = true;
+
+  return InitStatus::OK;
+}
+
+int32_t AudioDeviceLinuxPulse::Terminate() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!_initialized) {
+    return 0;
+  }
+  {
+    MutexLock lock(&mutex_);
+    quit_ = true;
+  }
+  _mixerManager.Close();
+
+  // RECORDING
+  _timeEventRec.Set();
+  _ptrThreadRec.Finalize();
+
+  // PLAYOUT
+  _timeEventPlay.Set();
+  _ptrThreadPlay.Finalize();
+
+  // Terminate PulseAudio
+  if (TerminatePulseAudio() < 0) {
+    RTC_LOG(LS_ERROR) << "failed to terminate PulseAudio";
+    return -1;
+  }
+
+#if defined(WEBRTC_USE_X11)
+  if (_XDisplay) {
+    XCloseDisplay(_XDisplay);
+    _XDisplay = NULL;
+  }
+#endif
+
+  _initialized = false;
+  _outputDeviceIsSpecified = false;
+  _inputDeviceIsSpecified = false;
+
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::Initialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_initialized);
+}
+
+int32_t AudioDeviceLinuxPulse::InitSpeaker() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  // check if default device
+  if (_outputDeviceIndex == 0) {
+    uint16_t deviceIndex = 0;
+    GetDefaultDeviceInfo(false, NULL, deviceIndex);
+    _paDeviceIndex = deviceIndex;
+  } else {
+    // get the PA device index from
+    // the callback
+    _deviceIndex = _outputDeviceIndex;
+
+    // get playout devices
+    PlayoutDevices();
+  }
+
+  // the callback has now set the _paDeviceIndex to
+  // the PulseAudio index of the device
+  if (_mixerManager.OpenSpeaker(_paDeviceIndex) == -1) {
+    return -1;
+  }
+
+  // clear _deviceIndex
+  _deviceIndex = -1;
+  _paDeviceIndex = -1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::InitMicrophone() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_recording) {
+    return -1;
+  }
+
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  // Check if default device
+  if (_inputDeviceIndex == 0) {
+    uint16_t deviceIndex = 0;
+    GetDefaultDeviceInfo(true, NULL, deviceIndex);
+    _paDeviceIndex = deviceIndex;
+  } else {
+    // Get the PA device index from
+    // the callback
+    _deviceIndex = _inputDeviceIndex;
+
+    // get recording devices
+    RecordingDevices();
+  }
+
+  // The callback has now set the _paDeviceIndex to
+  // the PulseAudio index of the device
+  if (_mixerManager.OpenMicrophone(_paDeviceIndex) == -1) {
+    return -1;
+  }
+
+  // Clear _deviceIndex
+  _deviceIndex = -1;
+  _paDeviceIndex = -1;
+
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::SpeakerIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceLinuxPulse::MicrophoneIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_mixerManager.MicrophoneIsInitialized());
+}
+
+int32_t AudioDeviceLinuxPulse::SpeakerVolumeIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know volume control exists.
+  available = true;
+
+  // Close the initialized output mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetSpeakerVolume(uint32_t volume) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!_playing) {
+    // Only update the volume if it's been set while we weren't playing.
+    update_speaker_volume_at_startup_ = true;
+  }
+  return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+int32_t AudioDeviceLinuxPulse::SpeakerVolume(uint32_t& volume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  uint32_t level(0);
+
+  if (_mixerManager.SpeakerVolume(level) == -1) {
+    return -1;
+  }
+
+  volume = level;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MinSpeakerVolume(uint32_t& minVolume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SpeakerMuteIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected speaker has a mute control
+  _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+  available = isAvailable;
+
+  // Close the initialized output mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetSpeakerMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_mixerManager.SetSpeakerMute(enable));
+}
+
+int32_t AudioDeviceLinuxPulse::SpeakerMute(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool muted(0);
+  if (_mixerManager.SpeakerMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MicrophoneMuteIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no
+    // volume control, hence it is safe to state that there is no
+    // boost control already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone has a mute control
+  //
+  _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetMicrophoneMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+int32_t AudioDeviceLinuxPulse::MicrophoneMute(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool muted(0);
+  if (_mixerManager.MicrophoneMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StereoRecordingIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_recChannels == 2 && _recording) {
+    available = true;
+    return 0;
+  }
+
+  available = false;
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+  int error = 0;
+
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // Cannot open the specified device
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone can record stereo.
+  bool isAvailable(false);
+  error = _mixerManager.StereoRecordingIsAvailable(isAvailable);
+  if (!error)
+    available = isAvailable;
+
+  // Close the initialized input mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return error;
+}
+
+int32_t AudioDeviceLinuxPulse::SetStereoRecording(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (enable)
+    _recChannels = 2;
+  else
+    _recChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StereoRecording(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StereoPlayoutIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_playChannels == 2 && _playing) {
+    available = true;
+    return 0;
+  }
+
+  available = false;
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+  int error = 0;
+
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // Cannot open the specified device.
+    return -1;
+  }
+
+  // Check if the selected speaker can play stereo.
+  bool isAvailable(false);
+  error = _mixerManager.StereoPlayoutIsAvailable(isAvailable);
+  if (!error)
+    available = isAvailable;
+
+  // Close the initialized input mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return error;
+}
+
+int32_t AudioDeviceLinuxPulse::SetStereoPlayout(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (enable)
+    _playChannels = 2;
+  else
+    _playChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MicrophoneVolumeIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no
+    // volume control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a volume control
+  // exists.
+  available = true;
+
+  // Close the initialized input mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetMicrophoneVolume(uint32_t volume) {
+  return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+int32_t AudioDeviceLinuxPulse::MicrophoneVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.MicrophoneVolume(level) == -1) {
+    RTC_LOG(LS_WARNING) << "failed to retrieve current microphone level";
+    return -1;
+  }
+
+  volume = level;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MinMicrophoneVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+
+  return 0;
+}
+
+int16_t AudioDeviceLinuxPulse::PlayoutDevices() {
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+  _numPlayDevices = 1;  // init to 1 to account for "default"
+
+  // get the whole list of devices and update _numPlayDevices
+  paOperation =
+      LATE(pa_context_get_sink_info_list)(_paContext, PaSinkInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  return _numPlayDevices;
+}
+
+int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(uint16_t index) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_playIsInitialized) {
+    return -1;
+  }
+
+  const uint16_t nDevices = PlayoutDevices();
+
+  RTC_LOG(LS_VERBOSE) << "number of availiable output devices is " << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _outputDeviceIndex = index;
+  _outputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceLinuxPulse::PlayoutDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  const uint16_t nDevices = PlayoutDevices();
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  // Check if default device
+  if (index == 0) {
+    uint16_t deviceIndex = 0;
+    return GetDefaultDeviceInfo(false, name, deviceIndex);
+  }
+
+  // Tell the callback that we want
+  // The name for this device
+  _playDisplayDeviceName = name;
+  _deviceIndex = index;
+
+  // get playout devices
+  PlayoutDevices();
+
+  // clear device name and index
+  _playDisplayDeviceName = NULL;
+  _deviceIndex = -1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::RecordingDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  const uint16_t nDevices(RecordingDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  // Check if default device
+  if (index == 0) {
+    uint16_t deviceIndex = 0;
+    return GetDefaultDeviceInfo(true, name, deviceIndex);
+  }
+
+  // Tell the callback that we want
+  // the name for this device
+  _recDisplayDeviceName = name;
+  _deviceIndex = index;
+
+  // Get recording devices
+  RecordingDevices();
+
+  // Clear device name and index
+  _recDisplayDeviceName = NULL;
+  _deviceIndex = -1;
+
+  return 0;
+}
+
+int16_t AudioDeviceLinuxPulse::RecordingDevices() {
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+  _numRecDevices = 1;  // Init to 1 to account for "default"
+
+  // Get the whole list of devices and update _numRecDevices
+  paOperation = LATE(pa_context_get_source_info_list)(
+      _paContext, PaSourceInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  return _numRecDevices;
+}
+
+int32_t AudioDeviceLinuxPulse::SetRecordingDevice(uint16_t index) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_recIsInitialized) {
+    return -1;
+  }
+
+  const uint16_t nDevices(RecordingDevices());
+
+  RTC_LOG(LS_VERBOSE) << "number of availiable input devices is " << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _inputDeviceIndex = index;
+  _inputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceLinuxPulse::PlayoutIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  available = false;
+
+  // Try to initialize the playout side
+  int32_t res = InitPlayout();
+
+  // Cancel effect of initialization
+  StopPlayout();
+
+  if (res != -1) {
+    available = true;
+  }
+
+  return res;
+}
+
+int32_t AudioDeviceLinuxPulse::RecordingIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  available = false;
+
+  // Try to initialize the playout side
+  int32_t res = InitRecording();
+
+  // Cancel effect of initialization
+  StopRecording();
+
+  if (res != -1) {
+    available = true;
+  }
+
+  return res;
+}
+
+int32_t AudioDeviceLinuxPulse::InitPlayout() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_playIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeaker() == -1) {
+    RTC_LOG(LS_WARNING) << "InitSpeaker() failed";
+  }
+
+  // Set the play sample specification
+  pa_sample_spec playSampleSpec;
+  playSampleSpec.channels = _playChannels;
+  playSampleSpec.format = PA_SAMPLE_S16LE;
+  playSampleSpec.rate = sample_rate_hz_;
+
+  // Create a new play stream
+  {
+    MutexLock lock(&mutex_);
+    _playStream =
+        LATE(pa_stream_new)(_paContext, "playStream", &playSampleSpec, NULL);
+  }
+
+  if (!_playStream) {
+    RTC_LOG(LS_ERROR) << "failed to create play stream, err="
+                      << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  // Provide the playStream to the mixer
+  _mixerManager.SetPlayStream(_playStream);
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters
+    _ptrAudioBuffer->SetPlayoutSampleRate(sample_rate_hz_);
+    _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
+  }
+
+  RTC_LOG(LS_VERBOSE) << "stream state "
+                      << LATE(pa_stream_get_state)(_playStream);
+
+  // Set stream flags
+  _playStreamFlags = (pa_stream_flags_t)(PA_STREAM_AUTO_TIMING_UPDATE |
+                                         PA_STREAM_INTERPOLATE_TIMING);
+
+  if (_configuredLatencyPlay != WEBRTC_PA_NO_LATENCY_REQUIREMENTS) {
+    // If configuring a specific latency then we want to specify
+    // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
+    // automatically to reach that target latency. However, that flag
+    // doesn't exist in Ubuntu 8.04 and many people still use that,
+    // so we have to check the protocol version of libpulse.
+    if (LATE(pa_context_get_protocol_version)(_paContext) >=
+        WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) {
+      _playStreamFlags |= PA_STREAM_ADJUST_LATENCY;
+    }
+
+    const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream);
+    if (!spec) {
+      RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec()";
+      return -1;
+    }
+
+    size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+    uint32_t latency = bytesPerSec * WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS /
+                       WEBRTC_PA_MSECS_PER_SEC;
+
+    // Set the play buffer attributes
+    _playBufferAttr.maxlength = latency;  // num bytes stored in the buffer
+    _playBufferAttr.tlength = latency;    // target fill level of play buffer
+    // minimum free num bytes before server request more data
+    _playBufferAttr.minreq = latency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
+    // prebuffer tlength before starting playout
+    _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq;
+
+    _configuredLatencyPlay = latency;
+  }
+
+  // num samples in bytes * num channels
+  _playbackBufferSize = sample_rate_hz_ / 100 * 2 * _playChannels;
+  _playbackBufferUnused = _playbackBufferSize;
+  _playBuffer = new int8_t[_playbackBufferSize];
+
+  // Enable underflow callback
+  LATE(pa_stream_set_underflow_callback)
+  (_playStream, PaStreamUnderflowCallback, this);
+
+  // Set the state callback function for the stream
+  LATE(pa_stream_set_state_callback)(_playStream, PaStreamStateCallback, this);
+
+  // Mark playout side as initialized
+  {
+    MutexLock lock(&mutex_);
+    _playIsInitialized = true;
+    _sndCardPlayDelay = 0;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::InitRecording() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (_recording) {
+    return -1;
+  }
+
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_recIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophone() == -1) {
+    RTC_LOG(LS_WARNING) << "InitMicrophone() failed";
+  }
+
+  // Set the rec sample specification
+  pa_sample_spec recSampleSpec;
+  recSampleSpec.channels = _recChannels;
+  recSampleSpec.format = PA_SAMPLE_S16LE;
+  recSampleSpec.rate = sample_rate_hz_;
+
+  // Create a new rec stream
+  _recStream =
+      LATE(pa_stream_new)(_paContext, "recStream", &recSampleSpec, NULL);
+  if (!_recStream) {
+    RTC_LOG(LS_ERROR) << "failed to create rec stream, err="
+                      << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  // Provide the recStream to the mixer
+  _mixerManager.SetRecStream(_recStream);
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters
+    _ptrAudioBuffer->SetRecordingSampleRate(sample_rate_hz_);
+    _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+  }
+
+  if (_configuredLatencyRec != WEBRTC_PA_NO_LATENCY_REQUIREMENTS) {
+    _recStreamFlags = (pa_stream_flags_t)(PA_STREAM_AUTO_TIMING_UPDATE |
+                                          PA_STREAM_INTERPOLATE_TIMING);
+
+    // If configuring a specific latency then we want to specify
+    // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
+    // automatically to reach that target latency. However, that flag
+    // doesn't exist in Ubuntu 8.04 and many people still use that,
+    //  so we have to check the protocol version of libpulse.
+    if (LATE(pa_context_get_protocol_version)(_paContext) >=
+        WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) {
+      _recStreamFlags |= PA_STREAM_ADJUST_LATENCY;
+    }
+
+    const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_recStream);
+    if (!spec) {
+      RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec(rec)";
+      return -1;
+    }
+
+    size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+    uint32_t latency = bytesPerSec * WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS /
+                       WEBRTC_PA_MSECS_PER_SEC;
+
+    // Set the rec buffer attributes
+    // Note: fragsize specifies a maximum transfer size, not a minimum, so
+    // it is not possible to force a high latency setting, only a low one.
+    _recBufferAttr.fragsize = latency;  // size of fragment
+    _recBufferAttr.maxlength =
+        latency + bytesPerSec * WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS /
+                      WEBRTC_PA_MSECS_PER_SEC;
+
+    _configuredLatencyRec = latency;
+  }
+
+  _recordBufferSize = sample_rate_hz_ / 100 * 2 * _recChannels;
+  _recordBufferUsed = 0;
+  _recBuffer = new int8_t[_recordBufferSize];
+
+  // Enable overflow callback
+  LATE(pa_stream_set_overflow_callback)
+  (_recStream, PaStreamOverflowCallback, this);
+
+  // Set the state callback function for the stream
+  LATE(pa_stream_set_state_callback)(_recStream, PaStreamStateCallback, this);
+
+  // Mark recording side as initialized
+  _recIsInitialized = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StartRecording() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!_recIsInitialized) {
+    return -1;
+  }
+
+  if (_recording) {
+    return 0;
+  }
+
+  // Set state to ensure that the recording starts from the audio thread.
+  _startRec = true;
+
+  // The audio thread will signal when recording has started.
+  _timeEventRec.Set();
+  if (!_recStartEvent.Wait(TimeDelta::Seconds(10))) {
+    {
+      MutexLock lock(&mutex_);
+      _startRec = false;
+    }
+    StopRecording();
+    RTC_LOG(LS_ERROR) << "failed to activate recording";
+    return -1;
+  }
+
+  {
+    MutexLock lock(&mutex_);
+    if (_recording) {
+      // The recording state is set by the audio thread after recording
+      // has started.
+    } else {
+      RTC_LOG(LS_ERROR) << "failed to activate recording";
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StopRecording() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  MutexLock lock(&mutex_);
+
+  if (!_recIsInitialized) {
+    return 0;
+  }
+
+  if (_recStream == NULL) {
+    return -1;
+  }
+
+  _recIsInitialized = false;
+  _recording = false;
+
+  RTC_LOG(LS_VERBOSE) << "stopping recording";
+
+  // Stop Recording
+  PaLock();
+
+  DisableReadCallback();
+  LATE(pa_stream_set_overflow_callback)(_recStream, NULL, NULL);
+
+  // Unset this here so that we don't get a TERMINATED callback
+  LATE(pa_stream_set_state_callback)(_recStream, NULL, NULL);
+
+  if (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_UNCONNECTED) {
+    // Disconnect the stream
+    if (LATE(pa_stream_disconnect)(_recStream) != PA_OK) {
+      RTC_LOG(LS_ERROR) << "failed to disconnect rec stream, err="
+                        << LATE(pa_context_errno)(_paContext);
+      PaUnLock();
+      return -1;
+    }
+
+    RTC_LOG(LS_VERBOSE) << "disconnected recording";
+  }
+
+  LATE(pa_stream_unref)(_recStream);
+  _recStream = NULL;
+
+  PaUnLock();
+
+  // Provide the recStream to the mixer
+  _mixerManager.SetRecStream(_recStream);
+
+  if (_recBuffer) {
+    delete[] _recBuffer;
+    _recBuffer = NULL;
+  }
+
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::RecordingIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_recIsInitialized);
+}
+
+bool AudioDeviceLinuxPulse::Recording() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_recording);
+}
+
+bool AudioDeviceLinuxPulse::PlayoutIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_playIsInitialized);
+}
+
+int32_t AudioDeviceLinuxPulse::StartPlayout() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (!_playIsInitialized) {
+    return -1;
+  }
+
+  if (_playing) {
+    return 0;
+  }
+
+  // Set state to ensure that playout starts from the audio thread.
+  {
+    MutexLock lock(&mutex_);
+    _startPlay = true;
+  }
+
+  // Both `_startPlay` and `_playing` needs protction since they are also
+  // accessed on the playout thread.
+
+  // The audio thread will signal when playout has started.
+  _timeEventPlay.Set();
+  if (!_playStartEvent.Wait(TimeDelta::Seconds(10))) {
+    {
+      MutexLock lock(&mutex_);
+      _startPlay = false;
+    }
+    StopPlayout();
+    RTC_LOG(LS_ERROR) << "failed to activate playout";
+    return -1;
+  }
+
+  {
+    MutexLock lock(&mutex_);
+    if (_playing) {
+      // The playing state is set by the audio thread after playout
+      // has started.
+    } else {
+      RTC_LOG(LS_ERROR) << "failed to activate playing";
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StopPlayout() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  MutexLock lock(&mutex_);
+
+  if (!_playIsInitialized) {
+    return 0;
+  }
+
+  if (_playStream == NULL) {
+    return -1;
+  }
+
+  _playIsInitialized = false;
+  _playing = false;
+  _sndCardPlayDelay = 0;
+
+  RTC_LOG(LS_VERBOSE) << "stopping playback";
+
+  // Stop Playout
+  PaLock();
+
+  DisableWriteCallback();
+  LATE(pa_stream_set_underflow_callback)(_playStream, NULL, NULL);
+
+  // Unset this here so that we don't get a TERMINATED callback
+  LATE(pa_stream_set_state_callback)(_playStream, NULL, NULL);
+
+  if (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_UNCONNECTED) {
+    // Disconnect the stream
+    if (LATE(pa_stream_disconnect)(_playStream) != PA_OK) {
+      RTC_LOG(LS_ERROR) << "failed to disconnect play stream, err="
+                        << LATE(pa_context_errno)(_paContext);
+      PaUnLock();
+      return -1;
+    }
+
+    RTC_LOG(LS_VERBOSE) << "disconnected playback";
+  }
+
+  LATE(pa_stream_unref)(_playStream);
+  _playStream = NULL;
+
+  PaUnLock();
+
+  // Provide the playStream to the mixer
+  _mixerManager.SetPlayStream(_playStream);
+
+  if (_playBuffer) {
+    delete[] _playBuffer;
+    _playBuffer = NULL;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::PlayoutDelay(uint16_t& delayMS) const {
+  MutexLock lock(&mutex_);
+  delayMS = (uint16_t)_sndCardPlayDelay;
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::Playing() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_playing);
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+void AudioDeviceLinuxPulse::PaContextStateCallback(pa_context* c, void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaContextStateCallbackHandler(c);
+}
+
+// ----------------------------------------------------------------------------
+//  PaSinkInfoCallback
+// ----------------------------------------------------------------------------
+
+void AudioDeviceLinuxPulse::PaSinkInfoCallback(pa_context* /*c*/,
+                                               const pa_sink_info* i,
+                                               int eol,
+                                               void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaSinkInfoCallbackHandler(i, eol);
+}
+
+void AudioDeviceLinuxPulse::PaSourceInfoCallback(pa_context* /*c*/,
+                                                 const pa_source_info* i,
+                                                 int eol,
+                                                 void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaSourceInfoCallbackHandler(i,
+                                                                          eol);
+}
+
+void AudioDeviceLinuxPulse::PaServerInfoCallback(pa_context* /*c*/,
+                                                 const pa_server_info* i,
+                                                 void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaServerInfoCallbackHandler(i);
+}
+
+void AudioDeviceLinuxPulse::PaStreamStateCallback(pa_stream* p, void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamStateCallbackHandler(p);
+}
+
+void AudioDeviceLinuxPulse::PaContextStateCallbackHandler(pa_context* c) {
+  RTC_LOG(LS_VERBOSE) << "context state cb";
+
+  pa_context_state_t state = LATE(pa_context_get_state)(c);
+  switch (state) {
+    case PA_CONTEXT_UNCONNECTED:
+      RTC_LOG(LS_VERBOSE) << "unconnected";
+      break;
+    case PA_CONTEXT_CONNECTING:
+    case PA_CONTEXT_AUTHORIZING:
+    case PA_CONTEXT_SETTING_NAME:
+      RTC_LOG(LS_VERBOSE) << "no state";
+      break;
+    case PA_CONTEXT_FAILED:
+    case PA_CONTEXT_TERMINATED:
+      RTC_LOG(LS_VERBOSE) << "failed";
+      _paStateChanged = true;
+      LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+      break;
+    case PA_CONTEXT_READY:
+      RTC_LOG(LS_VERBOSE) << "ready";
+      _paStateChanged = true;
+      LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+      break;
+  }
+}
+
+void AudioDeviceLinuxPulse::PaSinkInfoCallbackHandler(const pa_sink_info* i,
+                                                      int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  if (_numPlayDevices == _deviceIndex) {
+    // Convert the device index to the one of the sink
+    _paDeviceIndex = i->index;
+
+    if (_playDeviceName) {
+      // Copy the sink name
+      strncpy(_playDeviceName, i->name, kAdmMaxDeviceNameSize);
+      _playDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+    }
+    if (_playDisplayDeviceName) {
+      // Copy the sink display name
+      strncpy(_playDisplayDeviceName, i->description, kAdmMaxDeviceNameSize);
+      _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+    }
+  }
+
+  _numPlayDevices++;
+}
+
+void AudioDeviceLinuxPulse::PaSourceInfoCallbackHandler(const pa_source_info* i,
+                                                        int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  // We don't want to list output devices
+  if (i->monitor_of_sink == PA_INVALID_INDEX) {
+    if (_numRecDevices == _deviceIndex) {
+      // Convert the device index to the one of the source
+      _paDeviceIndex = i->index;
+
+      if (_recDeviceName) {
+        // copy the source name
+        strncpy(_recDeviceName, i->name, kAdmMaxDeviceNameSize);
+        _recDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+      }
+      if (_recDisplayDeviceName) {
+        // Copy the source display name
+        strncpy(_recDisplayDeviceName, i->description, kAdmMaxDeviceNameSize);
+        _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+      }
+    }
+
+    _numRecDevices++;
+  }
+}
+
+void AudioDeviceLinuxPulse::PaServerInfoCallbackHandler(
+    const pa_server_info* i) {
+  // Use PA native sampling rate
+  sample_rate_hz_ = i->sample_spec.rate;
+
+  // Copy the PA server version
+  strncpy(_paServerVersion, i->server_version, 31);
+  _paServerVersion[31] = '\0';
+
+  if (_recDisplayDeviceName) {
+    // Copy the source name
+    strncpy(_recDisplayDeviceName, i->default_source_name,
+            kAdmMaxDeviceNameSize);
+    _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+  }
+
+  if (_playDisplayDeviceName) {
+    // Copy the sink name
+    strncpy(_playDisplayDeviceName, i->default_sink_name,
+            kAdmMaxDeviceNameSize);
+    _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+  }
+
+  LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+}
+
+void AudioDeviceLinuxPulse::PaStreamStateCallbackHandler(pa_stream* p) {
+  RTC_LOG(LS_VERBOSE) << "stream state cb";
+
+  pa_stream_state_t state = LATE(pa_stream_get_state)(p);
+  switch (state) {
+    case PA_STREAM_UNCONNECTED:
+      RTC_LOG(LS_VERBOSE) << "unconnected";
+      break;
+    case PA_STREAM_CREATING:
+      RTC_LOG(LS_VERBOSE) << "creating";
+      break;
+    case PA_STREAM_FAILED:
+    case PA_STREAM_TERMINATED:
+      RTC_LOG(LS_VERBOSE) << "failed";
+      break;
+    case PA_STREAM_READY:
+      RTC_LOG(LS_VERBOSE) << "ready";
+      break;
+  }
+
+  LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+}
+
+int32_t AudioDeviceLinuxPulse::CheckPulseAudioVersion() {
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+
+  // get the server info and update deviceName
+  paOperation =
+      LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  RTC_LOG(LS_VERBOSE) << "checking PulseAudio version: " << _paServerVersion;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::InitSamplingFrequency() {
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+
+  // Get the server info and update sample_rate_hz_
+  paOperation =
+      LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice,
+                                                    char* name,
+                                                    uint16_t& index) {
+  char tmpName[kAdmMaxDeviceNameSize] = {0};
+  // subtract length of "default: "
+  uint16_t nameLen = kAdmMaxDeviceNameSize - 9;
+  char* pName = NULL;
+
+  if (name) {
+    // Add "default: "
+    strcpy(name, "default: ");
+    pName = &name[9];
+  }
+
+  // Tell the callback that we want
+  // the name for this device
+  if (recDevice) {
+    _recDisplayDeviceName = tmpName;
+  } else {
+    _playDisplayDeviceName = tmpName;
+  }
+
+  // Set members
+  _paDeviceIndex = -1;
+  _deviceIndex = 0;
+  _numPlayDevices = 0;
+  _numRecDevices = 0;
+
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+
+  // Get the server info and update deviceName
+  paOperation =
+      LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  // Get the device index
+  if (recDevice) {
+    paOperation = LATE(pa_context_get_source_info_by_name)(
+        _paContext, (char*)tmpName, PaSourceInfoCallback, this);
+  } else {
+    paOperation = LATE(pa_context_get_sink_info_by_name)(
+        _paContext, (char*)tmpName, PaSinkInfoCallback, this);
+  }
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  // Set the index
+  index = _paDeviceIndex;
+
+  if (name) {
+    // Copy to name string
+    strncpy(pName, tmpName, nameLen);
+  }
+
+  // Clear members
+  _playDisplayDeviceName = NULL;
+  _recDisplayDeviceName = NULL;
+  _paDeviceIndex = -1;
+  _deviceIndex = -1;
+  _numPlayDevices = 0;
+  _numRecDevices = 0;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::InitPulseAudio() {
+  int retVal = 0;
+
+  // Load libpulse
+  if (!GetPulseSymbolTable()->Load()) {
+    // Most likely the Pulse library and sound server are not installed on
+    // this system
+    RTC_LOG(LS_ERROR) << "failed to load symbol table";
+    return -1;
+  }
+
+  // Create a mainloop API and connection to the default server
+  // the mainloop is the internal asynchronous API event loop
+  if (_paMainloop) {
+    RTC_LOG(LS_ERROR) << "PA mainloop has already existed";
+    return -1;
+  }
+  _paMainloop = LATE(pa_threaded_mainloop_new)();
+  if (!_paMainloop) {
+    RTC_LOG(LS_ERROR) << "could not create mainloop";
+    return -1;
+  }
+
+  // Start the threaded main loop
+  retVal = LATE(pa_threaded_mainloop_start)(_paMainloop);
+  if (retVal != PA_OK) {
+    RTC_LOG(LS_ERROR) << "failed to start main loop, error=" << retVal;
+    return -1;
+  }
+
+  RTC_LOG(LS_VERBOSE) << "mainloop running!";
+
+  PaLock();
+
+  _paMainloopApi = LATE(pa_threaded_mainloop_get_api)(_paMainloop);
+  if (!_paMainloopApi) {
+    RTC_LOG(LS_ERROR) << "could not create mainloop API";
+    PaUnLock();
+    return -1;
+  }
+
+  // Create a new PulseAudio context
+  if (_paContext) {
+    RTC_LOG(LS_ERROR) << "PA context has already existed";
+    PaUnLock();
+    return -1;
+  }
+  _paContext = LATE(pa_context_new)(_paMainloopApi, "WEBRTC VoiceEngine");
+
+  if (!_paContext) {
+    RTC_LOG(LS_ERROR) << "could not create context";
+    PaUnLock();
+    return -1;
+  }
+
+  // Set state callback function
+  LATE(pa_context_set_state_callback)(_paContext, PaContextStateCallback, this);
+
+  // Connect the context to a server (default)
+  _paStateChanged = false;
+  retVal =
+      LATE(pa_context_connect)(_paContext, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL);
+
+  if (retVal != PA_OK) {
+    RTC_LOG(LS_ERROR) << "failed to connect context, error=" << retVal;
+    PaUnLock();
+    return -1;
+  }
+
+  // Wait for state change
+  while (!_paStateChanged) {
+    LATE(pa_threaded_mainloop_wait)(_paMainloop);
+  }
+
+  // Now check to see what final state we reached.
+  pa_context_state_t state = LATE(pa_context_get_state)(_paContext);
+
+  if (state != PA_CONTEXT_READY) {
+    if (state == PA_CONTEXT_FAILED) {
+      RTC_LOG(LS_ERROR) << "failed to connect to PulseAudio sound server";
+    } else if (state == PA_CONTEXT_TERMINATED) {
+      RTC_LOG(LS_ERROR) << "PulseAudio connection terminated early";
+    } else {
+      // Shouldn't happen, because we only signal on one of those three
+      // states
+      RTC_LOG(LS_ERROR) << "unknown problem connecting to PulseAudio";
+    }
+    PaUnLock();
+    return -1;
+  }
+
+  PaUnLock();
+
+  // Give the objects to the mixer manager
+  _mixerManager.SetPulseAudioObjects(_paMainloop, _paContext);
+
+  // Check the version
+  if (CheckPulseAudioVersion() < 0) {
+    RTC_LOG(LS_ERROR) << "PulseAudio version " << _paServerVersion
+                      << " not supported";
+    return -1;
+  }
+
+  // Initialize sampling frequency
+  if (InitSamplingFrequency() < 0 || sample_rate_hz_ == 0) {
+    RTC_LOG(LS_ERROR) << "failed to initialize sampling frequency, set to "
+                      << sample_rate_hz_ << " Hz";
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::TerminatePulseAudio() {
+  // Do nothing if the instance doesn't exist
+  // likely GetPulseSymbolTable.Load() fails
+  if (!_paMainloop) {
+    return 0;
+  }
+
+  PaLock();
+
+  // Disconnect the context
+  if (_paContext) {
+    LATE(pa_context_disconnect)(_paContext);
+  }
+
+  // Unreference the context
+  if (_paContext) {
+    LATE(pa_context_unref)(_paContext);
+  }
+
+  PaUnLock();
+  _paContext = NULL;
+
+  // Stop the threaded main loop
+  if (_paMainloop) {
+    LATE(pa_threaded_mainloop_stop)(_paMainloop);
+  }
+
+  // Free the mainloop
+  if (_paMainloop) {
+    LATE(pa_threaded_mainloop_free)(_paMainloop);
+  }
+
+  _paMainloop = NULL;
+
+  RTC_LOG(LS_VERBOSE) << "PulseAudio terminated";
+
+  return 0;
+}
+
+void AudioDeviceLinuxPulse::PaLock() {
+  LATE(pa_threaded_mainloop_lock)(_paMainloop);
+}
+
+void AudioDeviceLinuxPulse::PaUnLock() {
+  LATE(pa_threaded_mainloop_unlock)(_paMainloop);
+}
+
+void AudioDeviceLinuxPulse::WaitForOperationCompletion(
+    pa_operation* paOperation) const {
+  if (!paOperation) {
+    RTC_LOG(LS_ERROR) << "paOperation NULL in WaitForOperationCompletion";
+    return;
+  }
+
+  while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING) {
+    LATE(pa_threaded_mainloop_wait)(_paMainloop);
+  }
+
+  LATE(pa_operation_unref)(paOperation);
+}
+
+// ============================================================================
+//                                  Thread Methods
+// ============================================================================
+
+void AudioDeviceLinuxPulse::EnableWriteCallback() {
+  if (LATE(pa_stream_get_state)(_playStream) == PA_STREAM_READY) {
+    // May already have available space. Must check.
+    _tempBufferSpace = LATE(pa_stream_writable_size)(_playStream);
+    if (_tempBufferSpace > 0) {
+      // Yup, there is already space available, so if we register a
+      // write callback then it will not receive any event. So dispatch
+      // one ourself instead.
+      _timeEventPlay.Set();
+      return;
+    }
+  }
+
+  LATE(pa_stream_set_write_callback)(_playStream, &PaStreamWriteCallback, this);
+}
+
+void AudioDeviceLinuxPulse::DisableWriteCallback() {
+  LATE(pa_stream_set_write_callback)(_playStream, NULL, NULL);
+}
+
+void AudioDeviceLinuxPulse::PaStreamWriteCallback(pa_stream* /*unused*/,
+                                                  size_t buffer_space,
+                                                  void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamWriteCallbackHandler(
+      buffer_space);
+}
+
+void AudioDeviceLinuxPulse::PaStreamWriteCallbackHandler(size_t bufferSpace) {
+  _tempBufferSpace = bufferSpace;
+
+  // Since we write the data asynchronously on a different thread, we have
+  // to temporarily disable the write callback or else Pulse will call it
+  // continuously until we write the data. We re-enable it below.
+  DisableWriteCallback();
+  _timeEventPlay.Set();
+}
+
+void AudioDeviceLinuxPulse::PaStreamUnderflowCallback(pa_stream* /*unused*/,
+                                                      void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)
+      ->PaStreamUnderflowCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler() {
+  RTC_LOG(LS_WARNING) << "Playout underflow";
+
+  if (_configuredLatencyPlay == WEBRTC_PA_NO_LATENCY_REQUIREMENTS) {
+    // We didn't configure a pa_buffer_attr before, so switching to
+    // one now would be questionable.
+    return;
+  }
+
+  // Otherwise reconfigure the stream with a higher target latency.
+
+  const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream);
+  if (!spec) {
+    RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec()";
+    return;
+  }
+
+  size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+  uint32_t newLatency =
+      _configuredLatencyPlay + bytesPerSec *
+                                   WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS /
+                                   WEBRTC_PA_MSECS_PER_SEC;
+
+  // Set the play buffer attributes
+  _playBufferAttr.maxlength = newLatency;
+  _playBufferAttr.tlength = newLatency;
+  _playBufferAttr.minreq = newLatency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
+  _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq;
+
+  pa_operation* op = LATE(pa_stream_set_buffer_attr)(
+      _playStream, &_playBufferAttr, NULL, NULL);
+  if (!op) {
+    RTC_LOG(LS_ERROR) << "pa_stream_set_buffer_attr()";
+    return;
+  }
+
+  // Don't need to wait for this to complete.
+  LATE(pa_operation_unref)(op);
+
+  // Save the new latency in case we underflow again.
+  _configuredLatencyPlay = newLatency;
+}
+
+void AudioDeviceLinuxPulse::EnableReadCallback() {
+  LATE(pa_stream_set_read_callback)(_recStream, &PaStreamReadCallback, this);
+}
+
+void AudioDeviceLinuxPulse::DisableReadCallback() {
+  LATE(pa_stream_set_read_callback)(_recStream, NULL, NULL);
+}
+
+void AudioDeviceLinuxPulse::PaStreamReadCallback(pa_stream* /*unused1*/,
+                                                 size_t /*unused2*/,
+                                                 void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamReadCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler() {
+  // We get the data pointer and size now in order to save one Lock/Unlock
+  // in the worker thread.
+  if (LATE(pa_stream_peek)(_recStream, &_tempSampleData,
+                           &_tempSampleDataSize) != 0) {
+    RTC_LOG(LS_ERROR) << "Can't read data!";
+    return;
+  }
+
+  // Since we consume the data asynchronously on a different thread, we have
+  // to temporarily disable the read callback or else Pulse will call it
+  // continuously until we consume the data. We re-enable it below.
+  DisableReadCallback();
+  _timeEventRec.Set();
+}
+
+void AudioDeviceLinuxPulse::PaStreamOverflowCallback(pa_stream* /*unused*/,
+                                                     void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamOverflowCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamOverflowCallbackHandler() {
+  RTC_LOG(LS_WARNING) << "Recording overflow";
+}
+
+int32_t AudioDeviceLinuxPulse::LatencyUsecs(pa_stream* stream) {
+  if (!WEBRTC_PA_REPORT_LATENCY) {
+    return 0;
+  }
+
+  if (!stream) {
+    return 0;
+  }
+
+  pa_usec_t latency;
+  int negative;
+  if (LATE(pa_stream_get_latency)(stream, &latency, &negative) != 0) {
+    RTC_LOG(LS_ERROR) << "Can't query latency";
+    // We'd rather continue playout/capture with an incorrect delay than
+    // stop it altogether, so return a valid value.
+    return 0;
+  }
+
+  if (negative) {
+    RTC_LOG(LS_VERBOSE)
+        << "warning: pa_stream_get_latency reported negative delay";
+
+    // The delay can be negative for monitoring streams if the captured
+    // samples haven't been played yet. In such a case, "latency"
+    // contains the magnitude, so we must negate it to get the real value.
+    int32_t tmpLatency = (int32_t)-latency;
+    if (tmpLatency < 0) {
+      // Make sure that we don't use a negative delay.
+      tmpLatency = 0;
+    }
+
+    return tmpLatency;
+  } else {
+    return (int32_t)latency;
+  }
+}
+
+int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
+                                                size_t bufferSize)
+    RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_) {
+  size_t size = bufferSize;
+  uint32_t numRecSamples = _recordBufferSize / (2 * _recChannels);
+
+  // Account for the peeked data and the used data.
+  uint32_t recDelay =
+      (uint32_t)((LatencyUsecs(_recStream) / 1000) +
+                 10 * ((size + _recordBufferUsed) / _recordBufferSize));
+
+  if (_playStream) {
+    // Get the playout delay.
+    _sndCardPlayDelay = (uint32_t)(LatencyUsecs(_playStream) / 1000);
+  }
+
+  if (_recordBufferUsed > 0) {
+    // Have to copy to the buffer until it is full.
+    size_t copy = _recordBufferSize - _recordBufferUsed;
+    if (size < copy) {
+      copy = size;
+    }
+
+    memcpy(&_recBuffer[_recordBufferUsed], bufferData, copy);
+    _recordBufferUsed += copy;
+    bufferData = static_cast<const char*>(bufferData) + copy;
+    size -= copy;
+
+    if (_recordBufferUsed != _recordBufferSize) {
+      // Not enough data yet to pass to VoE.
+      return 0;
+    }
+
+    // Provide data to VoiceEngine.
+    if (ProcessRecordedData(_recBuffer, numRecSamples, recDelay) == -1) {
+      // We have stopped recording.
+      return -1;
+    }
+
+    _recordBufferUsed = 0;
+  }
+
+  // Now process full 10ms sample sets directly from the input.
+  while (size >= _recordBufferSize) {
+    // Provide data to VoiceEngine.
+    if (ProcessRecordedData(static_cast<int8_t*>(const_cast<void*>(bufferData)),
+                            numRecSamples, recDelay) == -1) {
+      // We have stopped recording.
+      return -1;
+    }
+
+    bufferData = static_cast<const char*>(bufferData) + _recordBufferSize;
+    size -= _recordBufferSize;
+
+    // We have consumed 10ms of data.
+    recDelay -= 10;
+  }
+
+  // Now save any leftovers for later.
+  if (size > 0) {
+    memcpy(_recBuffer, bufferData, size);
+    _recordBufferUsed = size;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::ProcessRecordedData(int8_t* bufferData,
+                                                   uint32_t bufferSizeInSamples,
+                                                   uint32_t recDelay)
+    RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_) {
+  _ptrAudioBuffer->SetRecordedBuffer(bufferData, bufferSizeInSamples);
+
+  // TODO(andrew): this is a temporary hack, to avoid non-causal far- and
+  // near-end signals at the AEC for PulseAudio. I think the system delay is
+  // being correctly calculated here, but for legacy reasons we add +10 ms
+  // to the value in the AEC. The real fix will be part of a larger
+  // investigation into managing system delay in the AEC.
+  if (recDelay > 10)
+    recDelay -= 10;
+  else
+    recDelay = 0;
+  _ptrAudioBuffer->SetVQEData(_sndCardPlayDelay, recDelay);
+  _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+  // Deliver recorded samples at specified sample rate,
+  // mic level etc. to the observer using callback.
+  UnLock();
+  _ptrAudioBuffer->DeliverRecordedData();
+  Lock();
+
+  // We have been unlocked - check the flag again.
+  if (!_recording) {
+    return -1;
+  }
+
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::PlayThreadProcess() {
+  if (!_timeEventPlay.Wait(TimeDelta::Seconds(1))) {
+    return true;
+  }
+
+  MutexLock lock(&mutex_);
+
+  if (quit_) {
+    return false;
+  }
+
+  if (_startPlay) {
+    RTC_LOG(LS_VERBOSE) << "_startPlay true, performing initial actions";
+
+    _startPlay = false;
+    _playDeviceName = NULL;
+
+    // Set if not default device
+    if (_outputDeviceIndex > 0) {
+      // Get the playout device name
+      _playDeviceName = new char[kAdmMaxDeviceNameSize];
+      _deviceIndex = _outputDeviceIndex;
+      PlayoutDevices();
+    }
+
+    // Start muted only supported on 0.9.11 and up
+    if (LATE(pa_context_get_protocol_version)(_paContext) >=
+        WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) {
+      // Get the currently saved speaker mute status
+      // and set the initial mute status accordingly
+      bool enabled(false);
+      _mixerManager.SpeakerMute(enabled);
+      if (enabled) {
+        _playStreamFlags |= PA_STREAM_START_MUTED;
+      }
+    }
+
+    // Get the currently saved speaker volume
+    uint32_t volume = 0;
+    if (update_speaker_volume_at_startup_)
+      _mixerManager.SpeakerVolume(volume);
+
+    PaLock();
+
+    // NULL gives PA the choice of startup volume.
+    pa_cvolume* ptr_cvolume = NULL;
+    if (update_speaker_volume_at_startup_) {
+      pa_cvolume cVolumes;
+      ptr_cvolume = &cVolumes;
+
+      // Set the same volume for all channels
+      const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream);
+      LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
+      update_speaker_volume_at_startup_ = false;
+    }
+
+    // Connect the stream to a sink
+    if (LATE(pa_stream_connect_playback)(
+            _playStream, _playDeviceName, &_playBufferAttr,
+            (pa_stream_flags_t)_playStreamFlags, ptr_cvolume, NULL) != PA_OK) {
+      RTC_LOG(LS_ERROR) << "failed to connect play stream, err="
+                        << LATE(pa_context_errno)(_paContext);
+    }
+
+    RTC_LOG(LS_VERBOSE) << "play stream connected";
+
+    // Wait for state change
+    while (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_READY) {
+      LATE(pa_threaded_mainloop_wait)(_paMainloop);
+    }
+
+    RTC_LOG(LS_VERBOSE) << "play stream ready";
+
+    // We can now handle write callbacks
+    EnableWriteCallback();
+
+    PaUnLock();
+
+    // Clear device name
+    if (_playDeviceName) {
+      delete[] _playDeviceName;
+      _playDeviceName = NULL;
+    }
+
+    _playing = true;
+    _playStartEvent.Set();
+
+    return true;
+  }
+
+  if (_playing) {
+    if (!_recording) {
+      // Update the playout delay
+      _sndCardPlayDelay = (uint32_t)(LatencyUsecs(_playStream) / 1000);
+    }
+
+    if (_playbackBufferUnused < _playbackBufferSize) {
+      size_t write = _playbackBufferSize - _playbackBufferUnused;
+      if (_tempBufferSpace < write) {
+        write = _tempBufferSpace;
+      }
+
+      PaLock();
+      if (LATE(pa_stream_write)(
+              _playStream, (void*)&_playBuffer[_playbackBufferUnused], write,
+              NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) {
+        _writeErrors++;
+        if (_writeErrors > 10) {
+          RTC_LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors
+                            << ", error=" << LATE(pa_context_errno)(_paContext);
+          _writeErrors = 0;
+        }
+      }
+      PaUnLock();
+
+      _playbackBufferUnused += write;
+      _tempBufferSpace -= write;
+    }
+
+    uint32_t numPlaySamples = _playbackBufferSize / (2 * _playChannels);
+    // Might have been reduced to zero by the above.
+    if (_tempBufferSpace > 0) {
+      // Ask for new PCM data to be played out using the
+      // AudioDeviceBuffer ensure that this callback is executed
+      // without taking the audio-thread lock.
+      UnLock();
+      RTC_LOG(LS_VERBOSE) << "requesting data";
+      uint32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(numPlaySamples);
+      Lock();
+
+      // We have been unlocked - check the flag again.
+      if (!_playing) {
+        return true;
+      }
+
+      nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer);
+      if (nSamples != numPlaySamples) {
+        RTC_LOG(LS_ERROR) << "invalid number of output samples(" << nSamples
+                          << ")";
+      }
+
+      size_t write = _playbackBufferSize;
+      if (_tempBufferSpace < write) {
+        write = _tempBufferSpace;
+      }
+
+      RTC_LOG(LS_VERBOSE) << "will write";
+      PaLock();
+      if (LATE(pa_stream_write)(_playStream, (void*)&_playBuffer[0], write,
+                                NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) {
+        _writeErrors++;
+        if (_writeErrors > 10) {
+          RTC_LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors
+                            << ", error=" << LATE(pa_context_errno)(_paContext);
+          _writeErrors = 0;
+        }
+      }
+      PaUnLock();
+
+      _playbackBufferUnused = write;
+    }
+
+    _tempBufferSpace = 0;
+    PaLock();
+    EnableWriteCallback();
+    PaUnLock();
+
+  }  // _playing
+
+  return true;
+}
+
+bool AudioDeviceLinuxPulse::RecThreadProcess() {
+  if (!_timeEventRec.Wait(TimeDelta::Seconds(1))) {
+    return true;
+  }
+
+  MutexLock lock(&mutex_);
+  if (quit_) {
+    return false;
+  }
+  if (_startRec) {
+    RTC_LOG(LS_VERBOSE) << "_startRec true, performing initial actions";
+
+    _recDeviceName = NULL;
+
+    // Set if not default device
+    if (_inputDeviceIndex > 0) {
+      // Get the recording device name
+      _recDeviceName = new char[kAdmMaxDeviceNameSize];
+      _deviceIndex = _inputDeviceIndex;
+      RecordingDevices();
+    }
+
+    PaLock();
+
+    RTC_LOG(LS_VERBOSE) << "connecting stream";
+
+    // Connect the stream to a source
+    if (LATE(pa_stream_connect_record)(
+            _recStream, _recDeviceName, &_recBufferAttr,
+            (pa_stream_flags_t)_recStreamFlags) != PA_OK) {
+      RTC_LOG(LS_ERROR) << "failed to connect rec stream, err="
+                        << LATE(pa_context_errno)(_paContext);
+    }
+
+    RTC_LOG(LS_VERBOSE) << "connected";
+
+    // Wait for state change
+    while (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_READY) {
+      LATE(pa_threaded_mainloop_wait)(_paMainloop);
+    }
+
+    RTC_LOG(LS_VERBOSE) << "done";
+
+    // We can now handle read callbacks
+    EnableReadCallback();
+
+    PaUnLock();
+
+    // Clear device name
+    if (_recDeviceName) {
+      delete[] _recDeviceName;
+      _recDeviceName = NULL;
+    }
+
+    _startRec = false;
+    _recording = true;
+    _recStartEvent.Set();
+
+    return true;
+  }
+
+  if (_recording) {
+    // Read data and provide it to VoiceEngine
+    if (ReadRecordedData(_tempSampleData, _tempSampleDataSize) == -1) {
+      return true;
+    }
+
+    _tempSampleData = NULL;
+    _tempSampleDataSize = 0;
+
+    PaLock();
+    while (true) {
+      // Ack the last thing we read
+      if (LATE(pa_stream_drop)(_recStream) != 0) {
+        RTC_LOG(LS_WARNING)
+            << "failed to drop, err=" << LATE(pa_context_errno)(_paContext);
+      }
+
+      if (LATE(pa_stream_readable_size)(_recStream) <= 0) {
+        // Then that was all the data
+        break;
+      }
+
+      // Else more data.
+      const void* sampleData;
+      size_t sampleDataSize;
+
+      if (LATE(pa_stream_peek)(_recStream, &sampleData, &sampleDataSize) != 0) {
+        RTC_LOG(LS_ERROR) << "RECORD_ERROR, error = "
+                          << LATE(pa_context_errno)(_paContext);
+        break;
+      }
+
+      // Drop lock for sigslot dispatch, which could take a while.
+      PaUnLock();
+      // Read data and provide it to VoiceEngine
+      if (ReadRecordedData(sampleData, sampleDataSize) == -1) {
+        return true;
+      }
+      PaLock();
+
+      // Return to top of loop for the ack and the check for more data.
+    }
+
+    EnableReadCallback();
+    PaUnLock();
+
+  }  // _recording
+
+  return true;
+}
+
+bool AudioDeviceLinuxPulse::KeyPressed() const {
+#if defined(WEBRTC_USE_X11)
+  char szKey[32];
+  unsigned int i = 0;
+  char state = 0;
+
+  if (!_XDisplay)
+    return false;
+
+  // Check key map status
+  XQueryKeymap(_XDisplay, szKey);
+
+  // A bit change in keymap means a key is pressed
+  for (i = 0; i < sizeof(szKey); i++)
+    state |= (szKey[i] ^ _oldKeyState[i]) & szKey[i];
+
+  // Save old state
+  memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState));
+  return (state != 0);
+#else
+  return false;
+#endif
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.h
new file mode 100644
index 0000000000..0cf89ef011
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.h
@@ -0,0 +1,349 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+#include "rtc_base/event.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+#if defined(WEBRTC_USE_X11)
+#include <X11/Xlib.h>
+#endif
+
+#include <pulse/pulseaudio.h>
+#include <stddef.h>
+#include <stdint.h>
+
+// We define this flag if it's missing from our headers, because we want to be
+// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
+// if run against a recent version of the library.
+#ifndef PA_STREAM_ADJUST_LATENCY
+#define PA_STREAM_ADJUST_LATENCY 0x2000U
+#endif
+#ifndef PA_STREAM_START_MUTED
+#define PA_STREAM_START_MUTED 0x1000U
+#endif
+
+// Set this constant to 0 to disable latency reading
+const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;
+
+// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]
+
+// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
+const uint32_t WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;
+
+// Some timing constants for optimal operation. See
+// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
+// for a good explanation of some of the factors that go into this.
+
+// Playback.
+
+// For playback, there is a round-trip delay to fill the server-side playback
+// buffer, so setting too low of a latency is a buffer underflow risk. We will
+// automatically increase the latency if a buffer underflow does occur, but we
+// also enforce a sane minimum at start-up time. Anything lower would be
+// virtually guaranteed to underflow at least once, so there's no point in
+// allowing lower latencies.
+const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;
+
+// Every time a playback stream underflows, we will reconfigure it with target
+// latency that is greater by this amount.
+const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;
+
+// We also need to configure a suitable request size. Too small and we'd burn
+// CPU from the overhead of transfering small amounts of data at once. Too large
+// and the amount of data remaining in the buffer right before refilling it
+// would be a buffer underflow risk. We set it to half of the buffer size.
+const uint32_t WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;
+
+// Capture.
+
+// For capture, low latency is not a buffer overflow risk, but it makes us burn
+// CPU from the overhead of transfering small amounts of data at once, so we set
+// a recommended value that we use for the kLowLatency constant (but if the user
+// explicitly requests something lower then we will honour it).
+// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
+const uint32_t WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;
+
+// There is a round-trip delay to ack the data to the server, so the
+// server-side buffer needs extra space to prevent buffer overflow. 20ms is
+// sufficient, but there is no penalty to making it bigger, so we make it huge.
+// (750ms is libpulse's default value for the _total_ buffer size in the
+// kNoLatencyRequirements case.)
+const uint32_t WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
+
+const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;
+
+// Init _configuredLatencyRec/Play to this value to disable latency requirements
+const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
+
+// Set this const to 1 to account for peeked and used data in latency
+// calculation
+const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
+
+typedef webrtc::adm_linux_pulse::PulseAudioSymbolTable WebRTCPulseSymbolTable;
+WebRTCPulseSymbolTable* GetPulseSymbolTable();
+
+namespace webrtc {
+
+class AudioDeviceLinuxPulse : public AudioDeviceGeneric {
+ public:
+  AudioDeviceLinuxPulse();
+  virtual ~AudioDeviceLinuxPulse();
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() override;
+  int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Playing() const override;
+  int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+
+ private:
+  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
+  void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
+  void WaitForOperationCompletion(pa_operation* paOperation) const;
+  void WaitForSuccess(pa_operation* paOperation) const;
+
+  bool KeyPressed() const;
+
+  static void PaContextStateCallback(pa_context* c, void* pThis);
+  static void PaSinkInfoCallback(pa_context* c,
+                                 const pa_sink_info* i,
+                                 int eol,
+                                 void* pThis);
+  static void PaSourceInfoCallback(pa_context* c,
+                                   const pa_source_info* i,
+                                   int eol,
+                                   void* pThis);
+  static void PaServerInfoCallback(pa_context* c,
+                                   const pa_server_info* i,
+                                   void* pThis);
+  static void PaStreamStateCallback(pa_stream* p, void* pThis);
+  void PaContextStateCallbackHandler(pa_context* c);
+  void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
+  void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
+  void PaServerInfoCallbackHandler(const pa_server_info* i);
+  void PaStreamStateCallbackHandler(pa_stream* p);
+
+  void EnableWriteCallback();
+  void DisableWriteCallback();
+  static void PaStreamWriteCallback(pa_stream* unused,
+                                    size_t buffer_space,
+                                    void* pThis);
+  void PaStreamWriteCallbackHandler(size_t buffer_space);
+  static void PaStreamUnderflowCallback(pa_stream* unused, void* pThis);
+  void PaStreamUnderflowCallbackHandler();
+  void EnableReadCallback();
+  void DisableReadCallback();
+  static void PaStreamReadCallback(pa_stream* unused1,
+                                   size_t unused2,
+                                   void* pThis);
+  void PaStreamReadCallbackHandler();
+  static void PaStreamOverflowCallback(pa_stream* unused, void* pThis);
+  void PaStreamOverflowCallbackHandler();
+  int32_t LatencyUsecs(pa_stream* stream);
+  int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
+  int32_t ProcessRecordedData(int8_t* bufferData,
+                              uint32_t bufferSizeInSamples,
+                              uint32_t recDelay);
+
+  int32_t CheckPulseAudioVersion();
+  int32_t InitSamplingFrequency();
+  int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
+  int32_t InitPulseAudio();
+  int32_t TerminatePulseAudio();
+
+  void PaLock();
+  void PaUnLock();
+
+  static void RecThreadFunc(void*);
+  static void PlayThreadFunc(void*);
+  bool RecThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);
+  bool PlayThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);
+
+  AudioDeviceBuffer* _ptrAudioBuffer;
+
+  mutable Mutex mutex_;
+  rtc::Event _timeEventRec;
+  rtc::Event _timeEventPlay;
+  rtc::Event _recStartEvent;
+  rtc::Event _playStartEvent;
+
+  rtc::PlatformThread _ptrThreadPlay;
+  rtc::PlatformThread _ptrThreadRec;
+
+  AudioMixerManagerLinuxPulse _mixerManager;
+
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
+
+  int sample_rate_hz_;
+  uint8_t _recChannels;
+  uint8_t _playChannels;
+
+  // Stores thread ID in constructor.
+  // We can then use RTC_DCHECK_RUN_ON(&worker_thread_checker_) to ensure that
+  // other methods are called from the same thread.
+  // Currently only does RTC_DCHECK(thread_checker_.IsCurrent()).
+  SequenceChecker thread_checker_;
+
+  bool _initialized;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+  bool _startRec;
+  bool _startPlay;
+  bool update_speaker_volume_at_startup_;
+  bool quit_ RTC_GUARDED_BY(&mutex_);
+
+  uint32_t _sndCardPlayDelay RTC_GUARDED_BY(&mutex_);
+
+  int32_t _writeErrors;
+
+  uint16_t _deviceIndex;
+  int16_t _numPlayDevices;
+  int16_t _numRecDevices;
+  char* _playDeviceName;
+  char* _recDeviceName;
+  char* _playDisplayDeviceName;
+  char* _recDisplayDeviceName;
+  char _paServerVersion[32];
+
+  int8_t* _playBuffer;
+  size_t _playbackBufferSize;
+  size_t _playbackBufferUnused;
+  size_t _tempBufferSpace;
+  int8_t* _recBuffer;
+  size_t _recordBufferSize;
+  size_t _recordBufferUsed;
+  const void* _tempSampleData;
+  size_t _tempSampleDataSize;
+  int32_t _configuredLatencyPlay;
+  int32_t _configuredLatencyRec;
+
+  // PulseAudio
+  uint16_t _paDeviceIndex;
+  bool _paStateChanged;
+
+  pa_threaded_mainloop* _paMainloop;
+  pa_mainloop_api* _paMainloopApi;
+  pa_context* _paContext;
+
+  pa_stream* _recStream;
+  pa_stream* _playStream;
+  uint32_t _recStreamFlags;
+  uint32_t _playStreamFlags;
+  pa_buffer_attr _playBufferAttr;
+  pa_buffer_attr _recBufferAttr;
+
+  char _oldKeyState[32];
+#if defined(WEBRTC_USE_X11)
+  Display* _XDisplay;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
new file mode 100644
index 0000000000..e7e7033173
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
@@ -0,0 +1,979 @@
+/*
+ *  Copyright (c) 2012 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 "modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
+
+#include "modules/audio_device/linux/audio_device_alsa_linux.h"
+#include "rtc_base/logging.h"
+
+// Accesses ALSA functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libasound, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                                            \
+  LATESYM_GET(webrtc::adm_linux_alsa::AlsaSymbolTable, GetAlsaSymbolTable(), \
+              sym)
+
+namespace webrtc {
+
+AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA()
+    : _outputMixerHandle(NULL),
+      _inputMixerHandle(NULL),
+      _outputMixerElement(NULL),
+      _inputMixerElement(NULL) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+
+  memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+  memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+}
+
+AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+  Close();
+}
+
+// ============================================================================
+//                                    PUBLIC METHODS
+// ============================================================================
+
+int32_t AudioMixerManagerLinuxALSA::Close() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  MutexLock lock(&mutex_);
+
+  CloseSpeakerLocked();
+  CloseMicrophoneLocked();
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseSpeaker() {
+  MutexLock lock(&mutex_);
+  return CloseSpeakerLocked();
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseSpeakerLocked() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  int errVal = 0;
+
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing playout mixer";
+    LATE(snd_mixer_free)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+    _outputMixerHandle = NULL;
+    _outputMixerElement = NULL;
+  }
+  memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseMicrophone() {
+  MutexLock lock(&mutex_);
+  return CloseMicrophoneLocked();
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseMicrophoneLocked() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  int errVal = 0;
+
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    LATE(snd_mixer_free)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 2";
+
+    errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 3";
+
+    errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 4";
+    _inputMixerHandle = NULL;
+    _inputMixerElement = NULL;
+  }
+  memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenSpeaker(name="
+                      << deviceName << ")";
+
+  MutexLock lock(&mutex_);
+
+  int errVal = 0;
+
+  // Close any existing output mixer handle
+  //
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing playout mixer";
+
+    LATE(snd_mixer_free)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+  }
+  _outputMixerHandle = NULL;
+  _outputMixerElement = NULL;
+
+  errVal = LATE(snd_mixer_open)(&_outputMixerHandle, 0);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_open(&_outputMixerHandle, 0) - error";
+    return -1;
+  }
+
+  char controlName[kAdmMaxDeviceNameSize] = {0};
+  GetControlName(controlName, deviceName);
+
+  RTC_LOG(LS_VERBOSE) << "snd_mixer_attach(_outputMixerHandle, " << controlName
+                      << ")";
+
+  errVal = LATE(snd_mixer_attach)(_outputMixerHandle, controlName);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_attach(_outputMixerHandle, " << controlName
+                      << ") error: " << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+  strcpy(_outputMixerStr, controlName);
+
+  errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, NULL, NULL);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR)
+        << "snd_mixer_selem_register(_outputMixerHandle, NULL, NULL), "
+           "error: "
+        << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+
+  // Load and find the proper mixer element
+  if (LoadSpeakerMixerElement() < 0) {
+    return -1;
+  }
+
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "the output mixer device is now open ("
+                        << _outputMixerHandle << ")";
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenMicrophone(name="
+                      << deviceName << ")";
+
+  MutexLock lock(&mutex_);
+
+  int errVal = 0;
+
+  // Close any existing input mixer handle
+  //
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    LATE(snd_mixer_free)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+  }
+  _inputMixerHandle = NULL;
+  _inputMixerElement = NULL;
+
+  errVal = LATE(snd_mixer_open)(&_inputMixerHandle, 0);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_open(&_inputMixerHandle, 0) - error";
+    return -1;
+  }
+
+  char controlName[kAdmMaxDeviceNameSize] = {0};
+  GetControlName(controlName, deviceName);
+
+  RTC_LOG(LS_VERBOSE) << "snd_mixer_attach(_inputMixerHandle, " << controlName
+                      << ")";
+
+  errVal = LATE(snd_mixer_attach)(_inputMixerHandle, controlName);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_attach(_inputMixerHandle, " << controlName
+                      << ") error: " << LATE(snd_strerror)(errVal);
+
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+  strcpy(_inputMixerStr, controlName);
+
+  errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, NULL, NULL);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR)
+        << "snd_mixer_selem_register(_inputMixerHandle, NULL, NULL), "
+           "error: "
+        << LATE(snd_strerror)(errVal);
+
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+  // Load and find the proper mixer element
+  if (LoadMicMixerElement() < 0) {
+    return -1;
+  }
+
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "the input mixer device is now open ("
+                        << _inputMixerHandle << ")";
+  }
+
+  return 0;
+}
+
+bool AudioMixerManagerLinuxALSA::SpeakerIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_outputMixerHandle != NULL);
+}
+
+bool AudioMixerManagerLinuxALSA::MicrophoneIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_inputMixerHandle != NULL);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume="
+                      << volume << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  int errVal = LATE(snd_mixer_selem_set_playback_volume_all)(
+      _outputMixerElement, volume);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error changing master volume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerVolume(uint32_t& volume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  long int vol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume)(
+      _outputMixerElement, (snd_mixer_selem_channel_id_t)0, &vol);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting outputvolume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SpeakerVolume() => vol="
+                      << vol;
+
+  volume = static_cast<uint32_t>(vol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MaxSpeakerVolume(
+    uint32_t& maxVolume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avilable output mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume_range)(
+      _outputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  maxVolume = static_cast<uint32_t>(maxVol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MinSpeakerVolume(
+    uint32_t& minVolume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume_range)(
+      _outputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  minVolume = static_cast<uint32_t>(minVol);
+
+  return 0;
+}
+
+// TL: Have done testnig with these but they don't seem reliable and
+// they were therefore not added
+/*
+ // ----------------------------------------------------------------------------
+ //    SetMaxSpeakerVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMaxSpeakerVolume(
+     uint32_t maxVolume)
+ {
+
+ if (_outputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_playback_volume_range(
+ _outputMixerElement, &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting playback volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ maxVol = maxVolume;
+ errVal = snd_mixer_selem_set_playback_volume_range(
+ _outputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting playback volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+
+ // ----------------------------------------------------------------------------
+ //    SetMinSpeakerVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMinSpeakerVolume(
+     uint32_t minVolume)
+ {
+
+ if (_outputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_playback_volume_range(
+ _outputMixerElement, &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting playback volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ minVol = minVolume;
+ errVal = snd_mixer_selem_set_playback_volume_range(
+ _outputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting playback volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+ */
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(bool& available) {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_playback_volume)(_outputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerMuteIsAvailable(bool& available) {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerMute(enable="
+                      << enable << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  // Ensure that the selected speaker destination has a valid mute control.
+  bool available(false);
+  SpeakerMuteIsAvailable(available);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the speaker";
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  int errVal = LATE(snd_mixer_selem_set_playback_switch_all)(
+      _outputMixerElement, !enable);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error setting playback switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer exists";
+    return -1;
+  }
+
+  // Ensure that the selected speaker destination has a valid mute control.
+  bool available =
+      LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the speaker";
+    return -1;
+  }
+
+  int value(false);
+
+  // Retrieve one boolean control value for a specified mute-control
+  //
+  int errVal = LATE(snd_mixer_selem_get_playback_switch)(
+      _outputMixerElement, (snd_mixer_selem_channel_id_t)0, &value);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting playback switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  enabled = (bool)!value;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneMuteIsAvailable(bool& available) {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable="
+                      << enable << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  // Ensure that the selected microphone destination has a valid mute control.
+  bool available(false);
+  MicrophoneMuteIsAvailable(available);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the microphone";
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  int errVal =
+      LATE(snd_mixer_selem_set_capture_switch_all)(_inputMixerElement, !enable);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error setting capture switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer exists";
+    return -1;
+  }
+
+  // Ensure that the selected microphone destination has a valid mute control.
+  bool available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the microphone";
+    return -1;
+  }
+
+  int value(false);
+
+  // Retrieve one boolean control value for a specified mute-control
+  //
+  int errVal = LATE(snd_mixer_selem_get_capture_switch)(
+      _inputMixerElement, (snd_mixer_selem_channel_id_t)0, &value);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting capture switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  enabled = (bool)!value;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneVolumeIsAvailable(
+    bool& available) {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume=" << volume
+      << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  int errVal =
+      LATE(snd_mixer_selem_set_capture_volume_all)(_inputMixerElement, volume);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error changing microphone volume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+// TL: Have done testnig with these but they don't seem reliable and
+// they were therefore not added
+/*
+ // ----------------------------------------------------------------------------
+ //    SetMaxMicrophoneVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMaxMicrophoneVolume(
+     uint32_t maxVolume)
+ {
+
+ if (_inputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_capture_volume_range(_inputMixerElement,
+  &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting capture volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ maxVol = (long int)maxVolume;
+ printf("min %d max %d", minVol, maxVol);
+ errVal = snd_mixer_selem_set_capture_volume_range(_inputMixerElement, minVol,
+ maxVol); RTC_LOG(LS_VERBOSE) << "Capture hardware volume range, min: " <<
+ minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting capture volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+
+ // ----------------------------------------------------------------------------
+ //    SetMinMicrophoneVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMinMicrophoneVolume(
+ uint32_t minVolume)
+ {
+
+ if (_inputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_capture_volume_range(
+ _inputMixerElement, &minVol, &maxVol);
+ if (maxVol <= minVol)
+ {
+ //maxVol = 255;
+ RTC_LOG(LS_WARNING) << "Error getting capture volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ printf("min %d max %d", minVol, maxVol);
+ minVol = (long int)minVolume;
+ errVal = snd_mixer_selem_set_capture_volume_range(
+ _inputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Capture hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting capture volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+ */
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneVolume(uint32_t& volume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int vol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume)(
+      _inputMixerElement, (snd_mixer_selem_channel_id_t)0, &vol);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting inputvolume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxALSA::MicrophoneVolume() => vol=" << vol;
+
+  volume = static_cast<uint32_t>(vol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MaxMicrophoneVolume(
+    uint32_t& maxVolume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  // check if we have mic volume at all
+  if (!LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement)) {
+    RTC_LOG(LS_ERROR) << "No microphone volume available";
+    return -1;
+  }
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume_range)(
+      _inputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting microphone volume range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  maxVolume = static_cast<uint32_t>(maxVol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MinMicrophoneVolume(
+    uint32_t& minVolume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume_range)(
+      _inputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting microphone volume range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  minVolume = static_cast<uint32_t>(minVol);
+
+  return 0;
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+int32_t AudioMixerManagerLinuxALSA::LoadMicMixerElement() const {
+  int errVal = LATE(snd_mixer_load)(_inputMixerHandle);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_load(_inputMixerHandle), error: "
+                      << LATE(snd_strerror)(errVal);
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+
+  snd_mixer_elem_t* elem = NULL;
+  snd_mixer_elem_t* micElem = NULL;
+  unsigned mixerIdx = 0;
+  const char* selemName = NULL;
+
+  // Find and store handles to the right mixer elements
+  for (elem = LATE(snd_mixer_first_elem)(_inputMixerHandle); elem;
+       elem = LATE(snd_mixer_elem_next)(elem), mixerIdx++) {
+    if (LATE(snd_mixer_selem_is_active)(elem)) {
+      selemName = LATE(snd_mixer_selem_get_name)(elem);
+      if (strcmp(selemName, "Capture") == 0)  // "Capture", "Mic"
+      {
+        _inputMixerElement = elem;
+        RTC_LOG(LS_VERBOSE) << "Capture element set";
+      } else if (strcmp(selemName, "Mic") == 0) {
+        micElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Mic element found";
+      }
+    }
+
+    if (_inputMixerElement) {
+      // Use the first Capture element that is found
+      // The second one may not work
+      break;
+    }
+  }
+
+  if (_inputMixerElement == NULL) {
+    // We didn't find a Capture handle, use Mic.
+    if (micElem != NULL) {
+      _inputMixerElement = micElem;
+      RTC_LOG(LS_VERBOSE) << "Using Mic as capture volume.";
+    } else {
+      _inputMixerElement = NULL;
+      RTC_LOG(LS_ERROR) << "Could not find capture volume on the mixer.";
+
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const {
+  int errVal = LATE(snd_mixer_load)(_outputMixerHandle);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_load(_outputMixerHandle), error: "
+                      << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+
+  snd_mixer_elem_t* elem = NULL;
+  snd_mixer_elem_t* masterElem = NULL;
+  snd_mixer_elem_t* speakerElem = NULL;
+  unsigned mixerIdx = 0;
+  const char* selemName = NULL;
+
+  // Find and store handles to the right mixer elements
+  for (elem = LATE(snd_mixer_first_elem)(_outputMixerHandle); elem;
+       elem = LATE(snd_mixer_elem_next)(elem), mixerIdx++) {
+    if (LATE(snd_mixer_selem_is_active)(elem)) {
+      selemName = LATE(snd_mixer_selem_get_name)(elem);
+      RTC_LOG(LS_VERBOSE) << "snd_mixer_selem_get_name " << mixerIdx << ": "
+                          << selemName << " =" << elem;
+
+      // "Master", "PCM", "Wave", "Master Mono", "PC Speaker", "PCM", "Wave"
+      if (strcmp(selemName, "PCM") == 0) {
+        _outputMixerElement = elem;
+        RTC_LOG(LS_VERBOSE) << "PCM element set";
+      } else if (strcmp(selemName, "Master") == 0) {
+        masterElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Master element found";
+      } else if (strcmp(selemName, "Speaker") == 0) {
+        speakerElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Speaker element found";
+      }
+    }
+
+    if (_outputMixerElement) {
+      // We have found the element we want
+      break;
+    }
+  }
+
+  // If we didn't find a PCM Handle, use Master or Speaker
+  if (_outputMixerElement == NULL) {
+    if (masterElem != NULL) {
+      _outputMixerElement = masterElem;
+      RTC_LOG(LS_VERBOSE) << "Using Master as output volume.";
+    } else if (speakerElem != NULL) {
+      _outputMixerElement = speakerElem;
+      RTC_LOG(LS_VERBOSE) << "Using Speaker as output volume.";
+    } else {
+      _outputMixerElement = NULL;
+      RTC_LOG(LS_ERROR) << "Could not find output volume in the mixer.";
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+void AudioMixerManagerLinuxALSA::GetControlName(char* controlName,
+                                                char* deviceName) const {
+  // Example
+  // deviceName: "front:CARD=Intel,DEV=0"
+  // controlName: "hw:CARD=Intel"
+  char* pos1 = strchr(deviceName, ':');
+  char* pos2 = strchr(deviceName, ',');
+  if (!pos2) {
+    // Can also be default:CARD=Intel
+    pos2 = &deviceName[strlen(deviceName)];
+  }
+  if (pos1 && pos2) {
+    strcpy(controlName, "hw");
+    int nChar = (int)(pos2 - pos1);
+    strncpy(&controlName[2], pos1, nChar);
+    controlName[2 + nChar] = '\0';
+  } else {
+    strcpy(controlName, deviceName);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
new file mode 100644
index 0000000000..d98287822d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
+
+#include <alsa/asoundlib.h>
+
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/linux/alsasymboltable_linux.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class AudioMixerManagerLinuxALSA {
+ public:
+  int32_t OpenSpeaker(char* deviceName) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t OpenMicrophone(char* deviceName) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SetSpeakerVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t Close() RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t CloseSpeaker() RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t CloseMicrophone() RTC_LOCKS_EXCLUDED(mutex_);
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
+
+ public:
+  AudioMixerManagerLinuxALSA();
+  ~AudioMixerManagerLinuxALSA();
+
+ private:
+  int32_t CloseSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t CloseMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t LoadMicMixerElement() const;
+  int32_t LoadSpeakerMixerElement() const;
+  void GetControlName(char* controlName, char* deviceName) const;
+
+ private:
+  Mutex mutex_;
+  mutable snd_mixer_t* _outputMixerHandle;
+  char _outputMixerStr[kAdmMaxDeviceNameSize];
+  mutable snd_mixer_t* _inputMixerHandle;
+  char _inputMixerStr[kAdmMaxDeviceNameSize];
+  mutable snd_mixer_elem_t* _outputMixerElement;
+  mutable snd_mixer_elem_t* _inputMixerElement;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
new file mode 100644
index 0000000000..91beee3c87
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
@@ -0,0 +1,844 @@
+/*
+ *  Copyright (c) 2012 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 "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
+
+#include <stddef.h>
+
+#include "modules/audio_device/linux/audio_device_pulse_linux.h"
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// Accesses Pulse functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libpulse, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                             \
+  LATESYM_GET(webrtc::adm_linux_pulse::PulseAudioSymbolTable, \
+              GetPulseSymbolTable(), sym)
+
+namespace webrtc {
+
+class AutoPulseLock {
+ public:
+  explicit AutoPulseLock(pa_threaded_mainloop* pa_mainloop)
+      : pa_mainloop_(pa_mainloop) {
+    LATE(pa_threaded_mainloop_lock)(pa_mainloop_);
+  }
+
+  ~AutoPulseLock() { LATE(pa_threaded_mainloop_unlock)(pa_mainloop_); }
+
+ private:
+  pa_threaded_mainloop* const pa_mainloop_;
+};
+
+AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse()
+    : _paOutputDeviceIndex(-1),
+      _paInputDeviceIndex(-1),
+      _paPlayStream(NULL),
+      _paRecStream(NULL),
+      _paMainloop(NULL),
+      _paContext(NULL),
+      _paVolume(0),
+      _paMute(0),
+      _paVolSteps(0),
+      _paSpeakerMute(false),
+      _paSpeakerVolume(PA_VOLUME_NORM),
+      _paChannels(0),
+      _paObjectsSet(false) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+}
+
+AudioMixerManagerLinuxPulse::~AudioMixerManagerLinuxPulse() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+
+  Close();
+}
+
+// ===========================================================================
+//                                    PUBLIC METHODS
+// ===========================================================================
+
+int32_t AudioMixerManagerLinuxPulse::SetPulseAudioObjects(
+    pa_threaded_mainloop* mainloop,
+    pa_context* context) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  if (!mainloop || !context) {
+    RTC_LOG(LS_ERROR) << "could not set PulseAudio objects for mixer";
+    return -1;
+  }
+
+  _paMainloop = mainloop;
+  _paContext = context;
+  _paObjectsSet = true;
+
+  RTC_LOG(LS_VERBOSE) << "the PulseAudio objects for the mixer has been set";
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::Close() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  CloseSpeaker();
+  CloseMicrophone();
+
+  _paMainloop = NULL;
+  _paContext = NULL;
+  _paObjectsSet = false;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::CloseSpeaker() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  // Reset the index to -1
+  _paOutputDeviceIndex = -1;
+  _paPlayStream = NULL;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::CloseMicrophone() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  // Reset the index to -1
+  _paInputDeviceIndex = -1;
+  _paRecStream = NULL;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetPlayStream(pa_stream* playStream) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetPlayStream(playStream)";
+
+  _paPlayStream = playStream;
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetRecStream(recStream)";
+
+  _paRecStream = recStream;
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::OpenSpeaker(uint16_t deviceIndex) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex="
+                      << deviceIndex << ")";
+
+  // No point in opening the speaker
+  // if PA objects have not been set
+  if (!_paObjectsSet) {
+    RTC_LOG(LS_ERROR) << "PulseAudio objects has not been set";
+    return -1;
+  }
+
+  // Set the index for the PulseAudio
+  // output device to control
+  _paOutputDeviceIndex = deviceIndex;
+
+  RTC_LOG(LS_VERBOSE) << "the output mixer device is now open";
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::OpenMicrophone(uint16_t deviceIndex) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex="
+      << deviceIndex << ")";
+
+  // No point in opening the microphone
+  // if PA objects have not been set
+  if (!_paObjectsSet) {
+    RTC_LOG(LS_ERROR) << "PulseAudio objects have not been set";
+    return -1;
+  }
+
+  // Set the index for the PulseAudio
+  // input device to control
+  _paInputDeviceIndex = deviceIndex;
+
+  RTC_LOG(LS_VERBOSE) << "the input mixer device is now open";
+
+  return 0;
+}
+
+bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_paOutputDeviceIndex != -1);
+}
+
+bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_paInputDeviceIndex != -1);
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(uint32_t volume) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerVolume(volume="
+                      << volume << ")";
+
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only really set the volume if we have a connected stream
+    AutoPulseLock auto_lock(_paMainloop);
+
+    // Get the number of channels from the sample specification
+    const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_paPlayStream);
+    if (!spec) {
+      RTC_LOG(LS_ERROR) << "could not get sample specification";
+      return -1;
+    }
+
+    // Set the same volume for all channels
+    pa_cvolume cVolumes;
+    LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
+
+    pa_operation* paOperation = NULL;
+    paOperation = LATE(pa_context_set_sink_input_volume)(
+        _paContext, LATE(pa_stream_get_index)(_paPlayStream), &cVolumes,
+        PaSetVolumeCallback, NULL);
+    if (!paOperation) {
+      setFailed = true;
+    }
+
+    // Don't need to wait for the completion
+    LATE(pa_operation_unref)(paOperation);
+  } else {
+    // We have not created a stream or it's not connected to the sink
+    // Save the volume to be set at connection
+    _paSpeakerVolume = volume;
+  }
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not set speaker volume, error="
+                        << LATE(pa_context_errno)(_paContext);
+
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerVolume(uint32_t& volume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only get the volume if we have a connected stream
+    if (!GetSinkInputInfo())
+      return -1;
+
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = static_cast<uint32_t>(_paVolume);
+  } else {
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = _paSpeakerVolume;
+  }
+
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SpeakerVolume() => vol="
+                      << volume;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MaxSpeakerVolume(
+    uint32_t& maxVolume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // PA_VOLUME_NORM corresponds to 100% (0db)
+  // but PA allows up to 150 db amplification
+  maxVolume = static_cast<uint32_t>(PA_VOLUME_NORM);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MinSpeakerVolume(
+    uint32_t& minVolume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  minVolume = static_cast<uint32_t>(PA_VOLUME_MUTED);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerMute(enable="
+                      << enable << ")";
+
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only really mute if we have a connected stream
+    AutoPulseLock auto_lock(_paMainloop);
+
+    pa_operation* paOperation = NULL;
+    paOperation = LATE(pa_context_set_sink_input_mute)(
+        _paContext, LATE(pa_stream_get_index)(_paPlayStream), (int)enable,
+        PaSetVolumeCallback, NULL);
+    if (!paOperation) {
+      setFailed = true;
+    }
+
+    // Don't need to wait for the completion
+    LATE(pa_operation_unref)(paOperation);
+  } else {
+    // We have not created a stream or it's not connected to the sink
+    // Save the mute status to be set at connection
+    _paSpeakerMute = enable;
+  }
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not mute speaker, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only get the mute status if we have a connected stream
+    if (!GetSinkInputInfo())
+      return -1;
+
+    enabled = static_cast<bool>(_paMute);
+  } else {
+    enabled = _paSpeakerMute;
+  }
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SpeakerMute() => enabled=" << enabled;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paOutputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+
+    // Get the actual stream device index if we have a connected stream
+    // The device used by the stream can be changed
+    // during the call
+    if (_paPlayStream &&
+        (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
+    }
+  }
+
+  if (!GetSinkInfoByIndex(deviceIndex))
+    return -1;
+
+  available = static_cast<bool>(_paChannels == 2);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  AutoPulseLock auto_lock(_paMainloop);
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  pa_operation* paOperation = NULL;
+
+  // Get info for this source
+  // We want to know if the actual device can record in stereo
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, deviceIndex, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+
+  available = static_cast<bool>(_paChannels == 2);
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
+         " => available="
+      << available;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneMuteIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetMicrophoneMute(enable=" << enable
+      << ")";
+
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+  pa_operation* paOperation = NULL;
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  AutoPulseLock auto_lock(_paMainloop);
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  // Set mute switch for the source
+  paOperation = LATE(pa_context_set_source_mute_by_index)(
+      _paContext, deviceIndex, enable, PaSetVolumeCallback, NULL);
+
+  if (!paOperation) {
+    setFailed = true;
+  }
+
+  // Don't need to wait for this to complete.
+  LATE(pa_operation_unref)(paOperation);
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not mute microphone, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    // Get the actual stream device index if we have a connected stream
+    // The device used by the stream can be changed
+    // during the call
+    if (_paRecStream &&
+        (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+    }
+  }
+
+  if (!GetSourceInfoByIndex(deviceIndex))
+    return -1;
+
+  enabled = static_cast<bool>(_paMute);
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::MicrophoneMute() => enabled=" << enabled;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetMicrophoneVolume(volume=" << volume
+      << ")";
+
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Unlike output streams, input streams have no concept of a stream
+  // volume, only a device volume. So we have to change the volume of the
+  // device itself.
+
+  // The device may have a different number of channels than the stream and
+  // their mapping may be different, so we don't want to use the channel
+  // count from our sample spec. We could use PA_CHANNELS_MAX to cover our
+  // bases, and the server allows that even if the device's channel count
+  // is lower, but some buggy PA clients don't like that (the pavucontrol
+  // on Hardy dies in an assert if the channel count is different). So
+  // instead we look up the actual number of channels that the device has.
+  AutoPulseLock auto_lock(_paMainloop);
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  bool setFailed(false);
+  pa_operation* paOperation = NULL;
+
+  // Get the number of channels for this source
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, deviceIndex, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+
+  uint8_t channels = _paChannels;
+  pa_cvolume cVolumes;
+  LATE(pa_cvolume_set)(&cVolumes, channels, volume);
+
+  // Set the volume for the source
+  paOperation = LATE(pa_context_set_source_volume_by_index)(
+      _paContext, deviceIndex, &cVolumes, PaSetVolumeCallback, NULL);
+
+  if (!paOperation) {
+    setFailed = true;
+  }
+
+  // Don't need to wait for this to complete.
+  LATE(pa_operation_unref)(paOperation);
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not set microphone volume, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneVolume(uint32_t& volume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    // Get the actual stream device index if we have a connected stream.
+    // The device used by the stream can be changed during the call.
+    if (_paRecStream &&
+        (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+    }
+  }
+
+  if (!GetSourceInfoByIndex(deviceIndex))
+    return -1;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = static_cast<uint32_t>(_paVolume);
+  }
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::MicrophoneVolume() => vol=" << volume;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MaxMicrophoneVolume(
+    uint32_t& maxVolume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // PA_VOLUME_NORM corresponds to 100% (0db)
+  // PA allows up to 150 db amplification (PA_VOLUME_MAX)
+  // but that doesn't work well for all sound cards
+  maxVolume = static_cast<uint32_t>(PA_VOLUME_NORM);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MinMicrophoneVolume(
+    uint32_t& minVolume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  minVolume = static_cast<uint32_t>(PA_VOLUME_MUTED);
+
+  return 0;
+}
+
+// ===========================================================================
+//                                 Private Methods
+// ===========================================================================
+
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context* /*c*/,
+                                                     const pa_sink_info* i,
+                                                     int eol,
+                                                     void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)->PaSinkInfoCallbackHandler(
+      i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
+    pa_context* /*c*/,
+    const pa_sink_input_info* i,
+    int eol,
+    void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)
+      ->PaSinkInputInfoCallbackHandler(i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context* /*c*/,
+                                                       const pa_source_info* i,
+                                                       int eol,
+                                                       void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)->PaSourceInfoCallbackHandler(
+      i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context* c,
+                                                      int success,
+                                                      void* /*pThis*/) {
+  if (!success) {
+    RTC_LOG(LS_ERROR) << "failed to set volume";
+  }
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallbackHandler(
+    const pa_sink_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // get the max volume for any channel
+  _paMute = i->mute;     // get mute status
+
+  // supported since PA 0.9.15
+  //_paVolSteps = i->n_volume_steps; // get the number of volume steps
+  // default value is PA_VOLUME_NORM+1
+  _paVolSteps = PA_VOLUME_NORM + 1;
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallbackHandler(
+    const pa_sink_input_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // Get the max volume for any channel
+  _paMute = i->mute;     // Get mute status
+}
+
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallbackHandler(
+    const pa_source_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // Get the max volume for any channel
+  _paMute = i->mute;     // Get mute status
+
+  // supported since PA 0.9.15
+  //_paVolSteps = i->n_volume_steps; // Get the number of volume steps
+  // default value is PA_VOLUME_NORM+1
+  _paVolSteps = PA_VOLUME_NORM + 1;
+}
+
+void AudioMixerManagerLinuxPulse::WaitForOperationCompletion(
+    pa_operation* paOperation) const {
+  while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING) {
+    LATE(pa_threaded_mainloop_wait)(_paMainloop);
+  }
+
+  LATE(pa_operation_unref)(paOperation);
+}
+
+bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  // Get info for this stream (sink input).
+  paOperation = LATE(pa_context_get_sink_input_info)(
+      _paContext, LATE(pa_stream_get_index)(_paPlayStream),
+      PaSinkInputInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(int device_index) const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  paOperation = LATE(pa_context_get_sink_info_by_index)(
+      _paContext, device_index, PaSinkInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(int device_index) const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, device_index, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
new file mode 100644
index 0000000000..546440c4a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
+
+#include <pulse/pulseaudio.h>
+#include <stdint.h>
+
+#include "api/sequence_checker.h"
+
+#ifndef UINT32_MAX
+#define UINT32_MAX ((uint32_t)-1)
+#endif
+
+namespace webrtc {
+
+class AudioMixerManagerLinuxPulse {
+ public:
+  int32_t SetPlayStream(pa_stream* playStream);
+  int32_t SetRecStream(pa_stream* recStream);
+  int32_t OpenSpeaker(uint16_t deviceIndex);
+  int32_t OpenMicrophone(uint16_t deviceIndex);
+  int32_t SetSpeakerVolume(uint32_t volume);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable);
+  int32_t StereoPlayoutIsAvailable(bool& available);
+  int32_t StereoRecordingIsAvailable(bool& available);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t SetPulseAudioObjects(pa_threaded_mainloop* mainloop,
+                               pa_context* context);
+  int32_t Close();
+  int32_t CloseSpeaker();
+  int32_t CloseMicrophone();
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
+
+ public:
+  AudioMixerManagerLinuxPulse();
+  ~AudioMixerManagerLinuxPulse();
+
+ private:
+  static void PaSinkInfoCallback(pa_context* c,
+                                 const pa_sink_info* i,
+                                 int eol,
+                                 void* pThis);
+  static void PaSinkInputInfoCallback(pa_context* c,
+                                      const pa_sink_input_info* i,
+                                      int eol,
+                                      void* pThis);
+  static void PaSourceInfoCallback(pa_context* c,
+                                   const pa_source_info* i,
+                                   int eol,
+                                   void* pThis);
+  static void PaSetVolumeCallback(pa_context* /*c*/,
+                                  int success,
+                                  void* /*pThis*/);
+  void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
+  void PaSinkInputInfoCallbackHandler(const pa_sink_input_info* i, int eol);
+  void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
+
+  void WaitForOperationCompletion(pa_operation* paOperation) const;
+
+  bool GetSinkInputInfo() const;
+  bool GetSinkInfoByIndex(int device_index) const;
+  bool GetSourceInfoByIndex(int device_index) const;
+
+ private:
+  int16_t _paOutputDeviceIndex;
+  int16_t _paInputDeviceIndex;
+
+  pa_stream* _paPlayStream;
+  pa_stream* _paRecStream;
+
+  pa_threaded_mainloop* _paMainloop;
+  pa_context* _paContext;
+
+  mutable uint32_t _paVolume;
+  mutable uint32_t _paMute;
+  mutable uint32_t _paVolSteps;
+  bool _paSpeakerMute;
+  mutable uint32_t _paSpeakerVolume;
+  mutable uint8_t _paChannels;
+  bool _paObjectsSet;
+
+  // Stores thread ID in constructor.
+  // We can then use RTC_DCHECK_RUN_ON(&worker_thread_checker_) to ensure that
+  // other methods are called from the same thread.
+  // Currently only does RTC_DCHECK(thread_checker_.IsCurrent()).
+  SequenceChecker thread_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc
new file mode 100644
index 0000000000..751edafd8b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2010 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 "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/logging.h"
+
+#ifdef WEBRTC_LINUX
+#include <dlfcn.h>
+#endif
+
+namespace webrtc {
+namespace adm_linux {
+
+inline static const char* GetDllError() {
+#ifdef WEBRTC_LINUX
+  char* err = dlerror();
+  if (err) {
+    return err;
+  } else {
+    return "No error";
+  }
+#else
+#error Not implemented
+#endif
+}
+
+DllHandle InternalLoadDll(absl::string_view dll_name) {
+#ifdef WEBRTC_LINUX
+  DllHandle handle = dlopen(std::string(dll_name).c_str(), RTLD_NOW);
+#else
+#error Not implemented
+#endif
+  if (handle == kInvalidDllHandle) {
+    RTC_LOG(LS_WARNING) << "Can't load " << dll_name << " : " << GetDllError();
+  }
+  return handle;
+}
+
+void InternalUnloadDll(DllHandle handle) {
+#ifdef WEBRTC_LINUX
+// TODO(pbos): Remove this dlclose() exclusion when leaks and suppressions from
+// here are gone (or AddressSanitizer can display them properly).
+//
+// Skip dlclose() on AddressSanitizer as leaks including this module in the
+// stack trace gets displayed as <unknown module> instead of the actual library
+// -> it can not be suppressed.
+// https://code.google.com/p/address-sanitizer/issues/detail?id=89
+#if !defined(ADDRESS_SANITIZER)
+  if (dlclose(handle) != 0) {
+    RTC_LOG(LS_ERROR) << GetDllError();
+  }
+#endif  // !defined(ADDRESS_SANITIZER)
+#else
+#error Not implemented
+#endif
+}
+
+static bool LoadSymbol(DllHandle handle,
+                       absl::string_view symbol_name,
+                       void** symbol) {
+#ifdef WEBRTC_LINUX
+  *symbol = dlsym(handle, std::string(symbol_name).c_str());
+  char* err = dlerror();
+  if (err) {
+    RTC_LOG(LS_ERROR) << "Error loading symbol " << symbol_name << " : " << err;
+    return false;
+  } else if (!*symbol) {
+    RTC_LOG(LS_ERROR) << "Symbol " << symbol_name << " is NULL";
+    return false;
+  }
+  return true;
+#else
+#error Not implemented
+#endif
+}
+
+// This routine MUST assign SOME value for every symbol, even if that value is
+// NULL, or else some symbols may be left with uninitialized data that the
+// caller may later interpret as a valid address.
+bool InternalLoadSymbols(DllHandle handle,
+                         int num_symbols,
+                         const char* const symbol_names[],
+                         void* symbols[]) {
+#ifdef WEBRTC_LINUX
+  // Clear any old errors.
+  dlerror();
+#endif
+  for (int i = 0; i < num_symbols; ++i) {
+    if (!LoadSymbol(handle, symbol_names[i], &symbols[i])) {
+      return false;
+    }
+  }
+  return true;
+}
+
+}  // namespace adm_linux
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.h b/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
new file mode 100644
index 0000000000..00f3c5a449
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2010 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.
+ */
+
+#ifndef AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H_
+
+#include <stddef.h>  // for NULL
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+// This file provides macros for creating "symbol table" classes to simplify the
+// dynamic loading of symbols from DLLs. Currently the implementation only
+// supports Linux and pure C symbols.
+// See talk/sound/pulseaudiosymboltable.(h|cc) for an example.
+
+namespace webrtc {
+namespace adm_linux {
+
+#ifdef WEBRTC_LINUX
+typedef void* DllHandle;
+
+const DllHandle kInvalidDllHandle = NULL;
+#else
+#error Not implemented
+#endif
+
+// These are helpers for use only by the class below.
+DllHandle InternalLoadDll(absl::string_view);
+
+void InternalUnloadDll(DllHandle handle);
+
+bool InternalLoadSymbols(DllHandle handle,
+                         int num_symbols,
+                         const char* const symbol_names[],
+                         void* symbols[]);
+
+template <int SYMBOL_TABLE_SIZE,
+          const char kDllName[],
+          const char* const kSymbolNames[]>
+class LateBindingSymbolTable {
+ public:
+  LateBindingSymbolTable()
+      : handle_(kInvalidDllHandle), undefined_symbols_(false) {
+    memset(symbols_, 0, sizeof(symbols_));
+  }
+
+  ~LateBindingSymbolTable() { Unload(); }
+
+  LateBindingSymbolTable(const LateBindingSymbolTable&) = delete;
+  LateBindingSymbolTable& operator=(LateBindingSymbolTable&) = delete;
+
+  static int NumSymbols() { return SYMBOL_TABLE_SIZE; }
+
+  // We do not use this, but we offer it for theoretical convenience.
+  static const char* GetSymbolName(int index) {
+    RTC_DCHECK_LT(index, NumSymbols());
+    return kSymbolNames[index];
+  }
+
+  bool IsLoaded() const { return handle_ != kInvalidDllHandle; }
+
+  // Loads the DLL and the symbol table. Returns true iff the DLL and symbol
+  // table loaded successfully.
+  bool Load() {
+    if (IsLoaded()) {
+      return true;
+    }
+    if (undefined_symbols_) {
+      // We do not attempt to load again because repeated attempts are not
+      // likely to succeed and DLL loading is costly.
+      return false;
+    }
+    handle_ = InternalLoadDll(kDllName);
+    if (!IsLoaded()) {
+      return false;
+    }
+    if (!InternalLoadSymbols(handle_, NumSymbols(), kSymbolNames, symbols_)) {
+      undefined_symbols_ = true;
+      Unload();
+      return false;
+    }
+    return true;
+  }
+
+  void Unload() {
+    if (!IsLoaded()) {
+      return;
+    }
+    InternalUnloadDll(handle_);
+    handle_ = kInvalidDllHandle;
+    memset(symbols_, 0, sizeof(symbols_));
+  }
+
+  // Retrieves the given symbol. NOTE: Recommended to use LATESYM_GET below
+  // instead of this.
+  void* GetSymbol(int index) const {
+    RTC_DCHECK(IsLoaded());
+    RTC_DCHECK_LT(index, NumSymbols());
+    return symbols_[index];
+  }
+
+ private:
+  DllHandle handle_;
+  bool undefined_symbols_;
+  void* symbols_[SYMBOL_TABLE_SIZE];
+};
+
+// This macro must be invoked in a header to declare a symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(ClassName) enum {
+// This macro must be invoked in the header declaration once for each symbol
+// (recommended to use an X-Macro to avoid duplication).
+// This macro defines an enum with names built from the symbols, which
+// essentially creates a hash table in the compiler from symbol names to their
+// indices in the symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(ClassName, sym) \
+  ClassName##_SYMBOL_TABLE_INDEX_##sym,
+
+// This macro completes the header declaration.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_END(ClassName)       \
+  ClassName##_SYMBOL_TABLE_SIZE                                \
+  }                                                            \
+  ;                                                            \
+                                                               \
+  extern const char ClassName##_kDllName[];                    \
+  extern const char* const                                     \
+      ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE]; \
+                                                               \
+  typedef ::webrtc::adm_linux::LateBindingSymbolTable<         \
+      ClassName##_SYMBOL_TABLE_SIZE, ClassName##_kDllName,     \
+      ClassName##_kSymbolNames>                                \
+      ClassName;
+
+// This macro must be invoked in a .cc file to define a previously-declared
+// symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(ClassName, dllName) \
+  const char ClassName##_kDllName[] = dllName;                     \
+  const char* const ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE] = {
+// This macro must be invoked in the .cc definition once for each symbol
+// (recommended to use an X-Macro to avoid duplication).
+// This would have to use the mangled name if we were to ever support C++
+// symbols.
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(ClassName, sym) #sym,
+
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_END(ClassName) \
+  }                                                     \
+  ;
+
+// Index of a given symbol in the given symbol table class.
+#define LATESYM_INDEXOF(ClassName, sym) (ClassName##_SYMBOL_TABLE_INDEX_##sym)
+
+// Returns a reference to the given late-binded symbol, with the correct type.
+#define LATESYM_GET(ClassName, inst, sym) \
+  (*reinterpret_cast<__typeof__(&sym)>(   \
+      (inst)->GetSymbol(LATESYM_INDEXOF(ClassName, sym))))
+
+}  // namespace adm_linux
+}  // namespace webrtc
+
+#endif  // ADM_LATEBINDINGSYMBOLTABLE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc
new file mode 100644
index 0000000000..e0759e6ca3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc
@@ -0,0 +1,41 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_pulse {
+
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")
+#define X(sym) \
+  LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
+PULSE_AUDIO_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DEFINE_END(PulseAudioSymbolTable)
+
+}  // namespace adm_linux_pulse
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h b/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h
new file mode 100644
index 0000000000..2f6a9510d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h
@@ -0,0 +1,106 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_pulse {
+
+// The PulseAudio symbols we need, as an X-Macro list.
+// This list must contain precisely every libpulse function that is used in
+// the ADM LINUX PULSE Device and Mixer classes
+#define PULSE_AUDIO_SYMBOLS_LIST           \
+  X(pa_bytes_per_second)                   \
+  X(pa_context_connect)                    \
+  X(pa_context_disconnect)                 \
+  X(pa_context_errno)                      \
+  X(pa_context_get_protocol_version)       \
+  X(pa_context_get_server_info)            \
+  X(pa_context_get_sink_info_list)         \
+  X(pa_context_get_sink_info_by_index)     \
+  X(pa_context_get_sink_info_by_name)      \
+  X(pa_context_get_sink_input_info)        \
+  X(pa_context_get_source_info_by_index)   \
+  X(pa_context_get_source_info_by_name)    \
+  X(pa_context_get_source_info_list)       \
+  X(pa_context_get_state)                  \
+  X(pa_context_new)                        \
+  X(pa_context_set_sink_input_volume)      \
+  X(pa_context_set_sink_input_mute)        \
+  X(pa_context_set_source_volume_by_index) \
+  X(pa_context_set_source_mute_by_index)   \
+  X(pa_context_set_state_callback)         \
+  X(pa_context_unref)                      \
+  X(pa_cvolume_set)                        \
+  X(pa_operation_get_state)                \
+  X(pa_operation_unref)                    \
+  X(pa_stream_connect_playback)            \
+  X(pa_stream_connect_record)              \
+  X(pa_stream_disconnect)                  \
+  X(pa_stream_drop)                        \
+  X(pa_stream_get_device_index)            \
+  X(pa_stream_get_index)                   \
+  X(pa_stream_get_latency)                 \
+  X(pa_stream_get_sample_spec)             \
+  X(pa_stream_get_state)                   \
+  X(pa_stream_new)                         \
+  X(pa_stream_peek)                        \
+  X(pa_stream_readable_size)               \
+  X(pa_stream_set_buffer_attr)             \
+  X(pa_stream_set_overflow_callback)       \
+  X(pa_stream_set_read_callback)           \
+  X(pa_stream_set_state_callback)          \
+  X(pa_stream_set_underflow_callback)      \
+  X(pa_stream_set_write_callback)          \
+  X(pa_stream_unref)                       \
+  X(pa_stream_writable_size)               \
+  X(pa_stream_write)                       \
+  X(pa_strerror)                           \
+  X(pa_threaded_mainloop_free)             \
+  X(pa_threaded_mainloop_get_api)          \
+  X(pa_threaded_mainloop_lock)             \
+  X(pa_threaded_mainloop_new)              \
+  X(pa_threaded_mainloop_signal)           \
+  X(pa_threaded_mainloop_start)            \
+  X(pa_threaded_mainloop_stop)             \
+  X(pa_threaded_mainloop_unlock)           \
+  X(pa_threaded_mainloop_wait)
+
+LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(PulseAudioSymbolTable)
+#define X(sym) \
+  LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(PulseAudioSymbolTable, sym)
+PULSE_AUDIO_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DECLARE_END(PulseAudioSymbolTable)
+
+}  // namespace adm_linux_pulse
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_