Initial import.

1 files changed, 389 insertions, 0 deletions

diff --git a/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
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index 00000000000..c4ca743853a
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+++ b/chromium/third_party/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
@@ -0,0 +1,389 @@
+/*
+ *  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 WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
+
+#include "webrtc/modules/audio_device/audio_device_generic.h"
+#include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+
+#include <X11/Xlib.h>
+#include <pulse/pulseaudio.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;
+
+namespace webrtc
+{
+class EventWrapper;
+class ThreadWrapper;
+
+class AudioDeviceLinuxPulse: public AudioDeviceGeneric
+{
+public:
+    AudioDeviceLinuxPulse(const int32_t id);
+    virtual ~AudioDeviceLinuxPulse();
+
+    static bool PulseAudioIsSupported();
+
+    // Retrieve the currently utilized audio layer
+    virtual int32_t ActiveAudioLayer(
+        AudioDeviceModule::AudioLayer& audioLayer) const OVERRIDE;
+
+    // Main initializaton and termination
+    virtual int32_t Init() OVERRIDE;
+    virtual int32_t Terminate() OVERRIDE;
+    virtual bool Initialized() const OVERRIDE;
+
+    // Device enumeration
+    virtual int16_t PlayoutDevices() OVERRIDE;
+    virtual int16_t RecordingDevices() OVERRIDE;
+    virtual int32_t PlayoutDeviceName(
+        uint16_t index,
+        char name[kAdmMaxDeviceNameSize],
+        char guid[kAdmMaxGuidSize]) OVERRIDE;
+    virtual int32_t RecordingDeviceName(
+        uint16_t index,
+        char name[kAdmMaxDeviceNameSize],
+        char guid[kAdmMaxGuidSize]) OVERRIDE;
+
+    // Device selection
+    virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE;
+    virtual int32_t SetPlayoutDevice(
+        AudioDeviceModule::WindowsDeviceType device) OVERRIDE;
+    virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE;
+    virtual int32_t SetRecordingDevice(
+        AudioDeviceModule::WindowsDeviceType device) OVERRIDE;
+
+    // Audio transport initialization
+    virtual int32_t PlayoutIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t InitPlayout() OVERRIDE;
+    virtual bool PlayoutIsInitialized() const OVERRIDE;
+    virtual int32_t RecordingIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t InitRecording() OVERRIDE;
+    virtual bool RecordingIsInitialized() const OVERRIDE;
+
+    // Audio transport control
+    virtual int32_t StartPlayout() OVERRIDE;
+    virtual int32_t StopPlayout() OVERRIDE;
+    virtual bool Playing() const OVERRIDE;
+    virtual int32_t StartRecording() OVERRIDE;
+    virtual int32_t StopRecording() OVERRIDE;
+    virtual bool Recording() const OVERRIDE;
+
+    // Microphone Automatic Gain Control (AGC)
+    virtual int32_t SetAGC(bool enable) OVERRIDE;
+    virtual bool AGC() const OVERRIDE;
+
+    // Volume control based on the Windows Wave API (Windows only)
+    virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
+                                     uint16_t volumeRight) OVERRIDE;
+    virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
+                                  uint16_t& volumeRight) const OVERRIDE;
+
+    // Audio mixer initialization
+    virtual int32_t SpeakerIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t InitSpeaker() OVERRIDE;
+    virtual bool SpeakerIsInitialized() const OVERRIDE;
+    virtual int32_t MicrophoneIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t InitMicrophone() OVERRIDE;
+    virtual bool MicrophoneIsInitialized() const OVERRIDE;
+
+    // Speaker volume controls
+    virtual int32_t SpeakerVolumeIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE;
+    virtual int32_t SpeakerVolume(uint32_t& volume) const OVERRIDE;
+    virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const OVERRIDE;
+    virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const OVERRIDE;
+    virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const OVERRIDE;
+
+    // Microphone volume controls
+    virtual int32_t MicrophoneVolumeIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE;
+    virtual int32_t MicrophoneVolume(uint32_t& volume) const OVERRIDE;
+    virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const OVERRIDE;
+    virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const OVERRIDE;
+    virtual int32_t MicrophoneVolumeStepSize(
+        uint16_t& stepSize) const OVERRIDE;
+
+    // Speaker mute control
+    virtual int32_t SpeakerMuteIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t SetSpeakerMute(bool enable) OVERRIDE;
+    virtual int32_t SpeakerMute(bool& enabled) const OVERRIDE;
+
+    // Microphone mute control
+    virtual int32_t MicrophoneMuteIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE;
+    virtual int32_t MicrophoneMute(bool& enabled) const OVERRIDE;
+
+    // Microphone boost control
+    virtual int32_t MicrophoneBoostIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE;
+    virtual int32_t MicrophoneBoost(bool& enabled) const OVERRIDE;
+
+    // Stereo support
+    virtual int32_t StereoPlayoutIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t SetStereoPlayout(bool enable) OVERRIDE;
+    virtual int32_t StereoPlayout(bool& enabled) const OVERRIDE;
+    virtual int32_t StereoRecordingIsAvailable(bool& available) OVERRIDE;
+    virtual int32_t SetStereoRecording(bool enable) OVERRIDE;
+    virtual int32_t StereoRecording(bool& enabled) const OVERRIDE;
+
+    // Delay information and control
+    virtual int32_t
+        SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+                         uint16_t sizeMS) OVERRIDE;
+    virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                  uint16_t& sizeMS) const OVERRIDE;
+    virtual int32_t PlayoutDelay(uint16_t& delayMS) const OVERRIDE;
+    virtual int32_t RecordingDelay(uint16_t& delayMS) const OVERRIDE;
+
+    // CPU load
+    virtual int32_t CPULoad(uint16_t& load) const OVERRIDE;
+
+public:
+    virtual bool PlayoutWarning() const OVERRIDE;
+    virtual bool PlayoutError() const OVERRIDE;
+    virtual bool RecordingWarning() const OVERRIDE;
+    virtual bool RecordingError() const OVERRIDE;
+    virtual void ClearPlayoutWarning() OVERRIDE;
+    virtual void ClearPlayoutError() OVERRIDE;
+    virtual void ClearRecordingWarning() OVERRIDE;
+    virtual void ClearRecordingError() OVERRIDE;
+
+public:
+    virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) OVERRIDE;
+
+private:
+    void Lock()
+    {
+        _critSect.Enter();
+    }
+    ;
+    void UnLock()
+    {
+        _critSect.Leave();
+    }
+    ;
+    void WaitForOperationCompletion(pa_operation* paOperation) const;
+    void WaitForSuccess(pa_operation* paOperation) const;
+
+private:
+    bool KeyPressed() const;
+
+private:
+    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 bool RecThreadFunc(void*);
+    static bool PlayThreadFunc(void*);
+    bool RecThreadProcess();
+    bool PlayThreadProcess();
+
+private:
+    AudioDeviceBuffer* _ptrAudioBuffer;
+
+    CriticalSectionWrapper& _critSect;
+    EventWrapper& _timeEventRec;
+    EventWrapper& _timeEventPlay;
+    EventWrapper& _recStartEvent;
+    EventWrapper& _playStartEvent;
+
+    ThreadWrapper* _ptrThreadPlay;
+    ThreadWrapper* _ptrThreadRec;
+    uint32_t _recThreadID;
+    uint32_t _playThreadID;
+    int32_t _id;
+
+    AudioMixerManagerLinuxPulse _mixerManager;
+
+    uint16_t _inputDeviceIndex;
+    uint16_t _outputDeviceIndex;
+    bool _inputDeviceIsSpecified;
+    bool _outputDeviceIsSpecified;
+
+    int sample_rate_hz_;
+    uint8_t _recChannels;
+    uint8_t _playChannels;
+
+    AudioDeviceModule::BufferType _playBufType;
+
+private:
+    bool _initialized;
+    bool _recording;
+    bool _playing;
+    bool _recIsInitialized;
+    bool _playIsInitialized;
+    bool _startRec;
+    bool _stopRec;
+    bool _startPlay;
+    bool _stopPlay;
+    bool _AGC;
+    bool update_speaker_volume_at_startup_;
+
+private:
+    uint16_t _playBufDelayFixed; // fixed playback delay
+
+    uint32_t _sndCardPlayDelay;
+    uint32_t _sndCardRecDelay;
+
+    int32_t _writeErrors;
+    uint16_t _playWarning;
+    uint16_t _playError;
+    uint16_t _recWarning;
+    uint16_t _recError;
+
+    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];
+    Display* _XDisplay;
+};
+
+}
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_