Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/audio_device/linux/audio_device_pulse_linux.h
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356
modules/audio_device/linux/audio_device_pulse_linux.h
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
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#include <memory>
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#include "webrtc/modules/audio_device/audio_device_generic.h"
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#include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/platform_thread.h"
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#include "webrtc/rtc_base/thread_checker.h"
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#include <X11/Xlib.h>
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#include <pulse/pulseaudio.h>
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// We define this flag if it's missing from our headers, because we want to be
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// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
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// if run against a recent version of the library.
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#ifndef PA_STREAM_ADJUST_LATENCY
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#define PA_STREAM_ADJUST_LATENCY 0x2000U
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#endif
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#ifndef PA_STREAM_START_MUTED
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#define PA_STREAM_START_MUTED 0x1000U
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#endif
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// Set this constant to 0 to disable latency reading
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const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;
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// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]
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// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
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const uint32_t WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;
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// Some timing constants for optimal operation. See
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// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
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// for a good explanation of some of the factors that go into this.
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// Playback.
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// For playback, there is a round-trip delay to fill the server-side playback
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// buffer, so setting too low of a latency is a buffer underflow risk. We will
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// automatically increase the latency if a buffer underflow does occur, but we
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// also enforce a sane minimum at start-up time. Anything lower would be
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// virtually guaranteed to underflow at least once, so there's no point in
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// allowing lower latencies.
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const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;
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// Every time a playback stream underflows, we will reconfigure it with target
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// latency that is greater by this amount.
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const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;
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// We also need to configure a suitable request size. Too small and we'd burn
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// CPU from the overhead of transfering small amounts of data at once. Too large
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// and the amount of data remaining in the buffer right before refilling it
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// would be a buffer underflow risk. We set it to half of the buffer size.
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const uint32_t WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;
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// Capture.
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// For capture, low latency is not a buffer overflow risk, but it makes us burn
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// CPU from the overhead of transfering small amounts of data at once, so we set
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// a recommended value that we use for the kLowLatency constant (but if the user
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// explicitly requests something lower then we will honour it).
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// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
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const uint32_t WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;
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// There is a round-trip delay to ack the data to the server, so the
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// server-side buffer needs extra space to prevent buffer overflow. 20ms is
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// sufficient, but there is no penalty to making it bigger, so we make it huge.
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// (750ms is libpulse's default value for the _total_ buffer size in the
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// kNoLatencyRequirements case.)
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const uint32_t WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
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const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;
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// Init _configuredLatencyRec/Play to this value to disable latency requirements
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const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
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// Set this const to 1 to account for peeked and used data in latency calculation
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const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
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namespace webrtc
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{
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class EventWrapper;
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class AudioDeviceLinuxPulse: public AudioDeviceGeneric
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{
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public:
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AudioDeviceLinuxPulse();
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virtual ~AudioDeviceLinuxPulse();
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// Retrieve the currently utilized audio layer
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int32_t ActiveAudioLayer(
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AudioDeviceModule::AudioLayer& audioLayer) const override;
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// Main initializaton and termination
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InitStatus Init() override;
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int32_t Terminate() override;
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bool Initialized() const override;
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// Device enumeration
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int16_t PlayoutDevices() override;
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int16_t RecordingDevices() override;
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int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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int32_t RecordingDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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// Device selection
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int32_t SetPlayoutDevice(uint16_t index) override;
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int32_t SetPlayoutDevice(
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AudioDeviceModule::WindowsDeviceType device) override;
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int32_t SetRecordingDevice(uint16_t index) override;
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int32_t SetRecordingDevice(
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AudioDeviceModule::WindowsDeviceType device) override;
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// Audio transport initialization
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int32_t PlayoutIsAvailable(bool& available) override;
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int32_t InitPlayout() override;
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bool PlayoutIsInitialized() const override;
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int32_t RecordingIsAvailable(bool& available) override;
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int32_t InitRecording() override;
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bool RecordingIsInitialized() const override;
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// Audio transport control
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int32_t StartPlayout() override;
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int32_t StopPlayout() override;
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bool Playing() const override;
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int32_t StartRecording() override;
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int32_t StopRecording() override;
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bool Recording() const override;
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// Microphone Automatic Gain Control (AGC)
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int32_t SetAGC(bool enable) override;
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bool AGC() const override;
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// Audio mixer initialization
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int32_t InitSpeaker() override;
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bool SpeakerIsInitialized() const override;
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int32_t InitMicrophone() override;
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bool MicrophoneIsInitialized() const override;
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// Speaker volume controls
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int32_t SpeakerVolumeIsAvailable(bool& available) override;
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int32_t SetSpeakerVolume(uint32_t volume) override;
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int32_t SpeakerVolume(uint32_t& volume) const override;
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int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
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int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
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// Microphone volume controls
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int32_t MicrophoneVolumeIsAvailable(bool& available) override;
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int32_t SetMicrophoneVolume(uint32_t volume) override;
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int32_t MicrophoneVolume(uint32_t& volume) const override;
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int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
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int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
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// Speaker mute control
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int32_t SpeakerMuteIsAvailable(bool& available) override;
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int32_t SetSpeakerMute(bool enable) override;
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int32_t SpeakerMute(bool& enabled) const override;
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// Microphone mute control
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int32_t MicrophoneMuteIsAvailable(bool& available) override;
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int32_t SetMicrophoneMute(bool enable) override;
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int32_t MicrophoneMute(bool& enabled) const override;
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// Stereo support
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int32_t StereoPlayoutIsAvailable(bool& available) override;
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int32_t SetStereoPlayout(bool enable) override;
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int32_t StereoPlayout(bool& enabled) const override;
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int32_t StereoRecordingIsAvailable(bool& available) override;
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int32_t SetStereoRecording(bool enable) override;
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int32_t StereoRecording(bool& enabled) const override;
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// Delay information and control
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int32_t PlayoutDelay(uint16_t& delayMS) const override;
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int32_t RecordingDelay(uint16_t& delayMS) const override;
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bool PlayoutWarning() const override;
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bool PlayoutError() const override;
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bool RecordingWarning() const override;
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bool RecordingError() const override;
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void ClearPlayoutWarning() override;
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void ClearPlayoutError() override;
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void ClearRecordingWarning() override;
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void ClearRecordingError() override;
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void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
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private:
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void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); }
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void UnLock() RTC_UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); }
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void WaitForOperationCompletion(pa_operation* paOperation) const;
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void WaitForSuccess(pa_operation* paOperation) const;
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bool KeyPressed() const;
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static void PaContextStateCallback(pa_context* c, void* pThis);
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static void PaSinkInfoCallback(pa_context* c,
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const pa_sink_info* i,
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int eol,
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void* pThis);
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static void PaSourceInfoCallback(pa_context* c,
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const pa_source_info* i,
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int eol,
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void* pThis);
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static void PaServerInfoCallback(pa_context* c,
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const pa_server_info* i,
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void* pThis);
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static void PaStreamStateCallback(pa_stream* p, void* pThis);
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void PaContextStateCallbackHandler(pa_context* c);
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void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
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void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
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void PaServerInfoCallbackHandler(const pa_server_info* i);
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void PaStreamStateCallbackHandler(pa_stream* p);
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void EnableWriteCallback();
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void DisableWriteCallback();
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static void PaStreamWriteCallback(pa_stream* unused,
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size_t buffer_space,
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void* pThis);
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void PaStreamWriteCallbackHandler(size_t buffer_space);
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static void PaStreamUnderflowCallback(pa_stream* unused, void* pThis);
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void PaStreamUnderflowCallbackHandler();
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void EnableReadCallback();
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void DisableReadCallback();
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static void PaStreamReadCallback(pa_stream* unused1,
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size_t unused2,
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void* pThis);
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void PaStreamReadCallbackHandler();
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static void PaStreamOverflowCallback(pa_stream* unused, void* pThis);
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void PaStreamOverflowCallbackHandler();
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int32_t LatencyUsecs(pa_stream* stream);
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int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
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int32_t ProcessRecordedData(int8_t* bufferData,
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uint32_t bufferSizeInSamples,
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uint32_t recDelay);
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int32_t CheckPulseAudioVersion();
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int32_t InitSamplingFrequency();
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int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
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int32_t InitPulseAudio();
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int32_t TerminatePulseAudio();
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void PaLock();
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void PaUnLock();
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static bool RecThreadFunc(void*);
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static bool PlayThreadFunc(void*);
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bool RecThreadProcess();
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bool PlayThreadProcess();
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AudioDeviceBuffer* _ptrAudioBuffer;
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rtc::CriticalSection _critSect;
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EventWrapper& _timeEventRec;
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EventWrapper& _timeEventPlay;
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EventWrapper& _recStartEvent;
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EventWrapper& _playStartEvent;
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// TODO(pbos): Remove unique_ptr and use directly without resetting.
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std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
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std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
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AudioMixerManagerLinuxPulse _mixerManager;
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uint16_t _inputDeviceIndex;
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uint16_t _outputDeviceIndex;
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bool _inputDeviceIsSpecified;
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bool _outputDeviceIsSpecified;
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int sample_rate_hz_;
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uint8_t _recChannels;
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uint8_t _playChannels;
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// Stores thread ID in constructor.
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// We can then use ThreadChecker::CalledOnValidThread() to ensure that
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// other methods are called from the same thread.
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// Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()).
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rtc::ThreadChecker thread_checker_;
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bool _initialized;
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bool _recording;
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bool _playing;
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bool _recIsInitialized;
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bool _playIsInitialized;
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bool _startRec;
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bool _stopRec;
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bool _startPlay;
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bool _stopPlay;
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bool _AGC;
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bool update_speaker_volume_at_startup_;
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uint32_t _sndCardPlayDelay;
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uint32_t _sndCardRecDelay;
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int32_t _writeErrors;
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uint16_t _playWarning;
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uint16_t _playError;
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uint16_t _recWarning;
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uint16_t _recError;
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uint16_t _deviceIndex;
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int16_t _numPlayDevices;
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int16_t _numRecDevices;
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char* _playDeviceName;
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char* _recDeviceName;
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char* _playDisplayDeviceName;
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char* _recDisplayDeviceName;
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char _paServerVersion[32];
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int8_t* _playBuffer;
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size_t _playbackBufferSize;
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size_t _playbackBufferUnused;
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size_t _tempBufferSpace;
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int8_t* _recBuffer;
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size_t _recordBufferSize;
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size_t _recordBufferUsed;
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const void* _tempSampleData;
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size_t _tempSampleDataSize;
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int32_t _configuredLatencyPlay;
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int32_t _configuredLatencyRec;
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// PulseAudio
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uint16_t _paDeviceIndex;
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bool _paStateChanged;
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pa_threaded_mainloop* _paMainloop;
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pa_mainloop_api* _paMainloopApi;
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pa_context* _paContext;
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pa_stream* _recStream;
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pa_stream* _playStream;
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uint32_t _recStreamFlags;
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uint32_t _playStreamFlags;
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pa_buffer_attr _playBufferAttr;
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pa_buffer_attr _recBufferAttr;
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char _oldKeyState[32];
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Display* _XDisplay;
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};
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}
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#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
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