The CL broke the builds is :
http://code.google.com/p/webrtc/source/detail?r=2804 Review URL: https://webrtc-codereview.appspot.com/873012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2911 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
xians@webrtc.org
parent
58849fd1ec
commit
597b4472eb
@ -8,8 +8,8 @@
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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 MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
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#define MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
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#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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#include "modules/audio_device/include/audio_device_defines.h"
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#include "modules/interface/module.h"
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@ -204,4 +204,4 @@ AudioDeviceModule* CreateAudioDeviceModule(
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
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#endif // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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@ -11,197 +11,6 @@
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#ifndef MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
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#define MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
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#include "modules/audio_device/include/audio_device_defines.h"
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#include "modules/interface/module.h"
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namespace webrtc {
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class AudioDeviceModule : public RefCountedModule {
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public:
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enum ErrorCode {
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kAdmErrNone = 0,
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kAdmErrArgument = 1
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};
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enum AudioLayer {
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kPlatformDefaultAudio = 0,
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kWindowsWaveAudio = 1,
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kWindowsCoreAudio = 2,
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kLinuxAlsaAudio = 3,
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kLinuxPulseAudio = 4,
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kDummyAudio = 5
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};
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enum WindowsDeviceType {
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kDefaultCommunicationDevice = -1,
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kDefaultDevice = -2
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};
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enum BufferType {
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kFixedBufferSize = 0,
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kAdaptiveBufferSize = 1
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};
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enum ChannelType {
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kChannelLeft = 0,
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kChannelRight = 1,
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kChannelBoth = 2
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};
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public:
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// Retrieve the currently utilized audio layer
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virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
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// Error handling
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virtual ErrorCode LastError() const = 0;
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virtual int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) = 0;
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// Full-duplex transportation of PCM audio
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virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
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// Main initialization and termination
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virtual int32_t Init() = 0;
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virtual int32_t Terminate() = 0;
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virtual bool Initialized() const = 0;
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// Device enumeration
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virtual int16_t PlayoutDevices() = 0;
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virtual int16_t RecordingDevices() = 0;
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virtual int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) = 0;
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virtual int32_t RecordingDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) = 0;
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// Device selection
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virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
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virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
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virtual int32_t SetRecordingDevice(uint16_t index) = 0;
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virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
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// Audio transport initialization
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virtual int32_t PlayoutIsAvailable(bool* available) = 0;
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virtual int32_t InitPlayout() = 0;
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virtual bool PlayoutIsInitialized() const = 0;
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virtual int32_t RecordingIsAvailable(bool* available) = 0;
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virtual int32_t InitRecording() = 0;
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virtual bool RecordingIsInitialized() const = 0;
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// Audio transport control
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virtual int32_t StartPlayout() = 0;
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virtual int32_t StopPlayout() = 0;
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virtual bool Playing() const = 0;
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virtual int32_t StartRecording() = 0;
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virtual int32_t StopRecording() = 0;
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virtual bool Recording() const = 0;
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// Microphone Automatic Gain Control (AGC)
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virtual int32_t SetAGC(bool enable) = 0;
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virtual bool AGC() const = 0;
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// Volume control based on the Windows Wave API (Windows only)
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virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
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uint16_t volumeRight) = 0;
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virtual int32_t WaveOutVolume(uint16_t* volumeLeft,
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uint16_t* volumeRight) const = 0;
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// Audio mixer initialization
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virtual int32_t SpeakerIsAvailable(bool* available) = 0;
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virtual int32_t InitSpeaker() = 0;
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virtual bool SpeakerIsInitialized() const = 0;
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virtual int32_t MicrophoneIsAvailable(bool* available) = 0;
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virtual int32_t InitMicrophone() = 0;
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virtual bool MicrophoneIsInitialized() const = 0;
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// Speaker volume controls
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virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
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virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
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virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
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virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
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virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
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virtual int32_t SpeakerVolumeStepSize(uint16_t* stepSize) const = 0;
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// Microphone volume controls
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virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
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virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
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virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
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virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
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virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
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virtual int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) const = 0;
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// Speaker mute control
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virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
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virtual int32_t SetSpeakerMute(bool enable) = 0;
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virtual int32_t SpeakerMute(bool* enabled) const = 0;
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// Microphone mute control
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virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
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virtual int32_t SetMicrophoneMute(bool enable) = 0;
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virtual int32_t MicrophoneMute(bool* enabled) const = 0;
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// Microphone boost control
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virtual int32_t MicrophoneBoostIsAvailable(bool* available) = 0;
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virtual int32_t SetMicrophoneBoost(bool enable) = 0;
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virtual int32_t MicrophoneBoost(bool* enabled) const = 0;
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// Stereo support
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virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
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virtual int32_t SetStereoPlayout(bool enable) = 0;
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virtual int32_t StereoPlayout(bool* enabled) const = 0;
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virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
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virtual int32_t SetStereoRecording(bool enable) = 0;
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virtual int32_t StereoRecording(bool* enabled) const = 0;
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virtual int32_t SetRecordingChannel(const ChannelType channel) = 0;
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virtual int32_t RecordingChannel(ChannelType* channel) const = 0;
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// Delay information and control
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virtual int32_t SetPlayoutBuffer(const BufferType type,
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uint16_t sizeMS = 0) = 0;
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virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const = 0;
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virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
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virtual int32_t RecordingDelay(uint16_t* delayMS) const = 0;
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// CPU load
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virtual int32_t CPULoad(uint16_t* load) const = 0;
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// Recording of raw PCM data
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virtual int32_t StartRawOutputFileRecording(
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const char pcmFileNameUTF8[kAdmMaxFileNameSize]) = 0;
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virtual int32_t StopRawOutputFileRecording() = 0;
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virtual int32_t StartRawInputFileRecording(
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const char pcmFileNameUTF8[kAdmMaxFileNameSize]) = 0;
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virtual int32_t StopRawInputFileRecording() = 0;
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// Native sample rate controls (samples/sec)
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virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) = 0;
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virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const = 0;
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virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) = 0;
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virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const = 0;
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// Mobile device specific functions
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virtual int32_t ResetAudioDevice() = 0;
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virtual int32_t SetLoudspeakerStatus(bool enable) = 0;
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virtual int32_t GetLoudspeakerStatus(bool* enabled) const = 0;
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// *Experimental - not recommended for use.*
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// Enables the Windows Core Audio built-in AEC. Fails on other platforms.
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//
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// Must be called before InitRecording(). When enabled:
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// 1. StartPlayout() must be called before StartRecording().
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// 2. StopRecording() should be called before StopPlayout().
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// The reverse order may cause garbage audio to be rendered or the
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// capture side to halt until StopRecording() is called.
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virtual int32_t EnableBuiltInAEC(bool enable) { return -1; }
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virtual bool BuiltInAECIsEnabled() const { return false; }
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protected:
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virtual ~AudioDeviceModule() {};
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};
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AudioDeviceModule* CreateAudioDeviceModule(
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WebRtc_Word32 id, AudioDeviceModule::AudioLayer audioLayer);
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} // namespace webrtc
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#include "../../include/audio_device.h"
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#endif // MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
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@ -1,80 +0,0 @@
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/*
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* Copyright (c) 2011 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_DEFINES_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
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#include "typedefs.h"
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namespace webrtc {
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static const int kAdmMaxDeviceNameSize = 128;
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static const int kAdmMaxFileNameSize = 512;
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static const int kAdmMaxGuidSize = 128;
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static const int kAdmMinPlayoutBufferSizeMs = 10;
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static const int kAdmMaxPlayoutBufferSizeMs = 250;
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// ----------------------------------------------------------------------------
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// AudioDeviceObserver
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// ----------------------------------------------------------------------------
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class AudioDeviceObserver
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{
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public:
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enum ErrorCode
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{
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kRecordingError = 0,
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kPlayoutError = 1
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};
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enum WarningCode
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{
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kRecordingWarning = 0,
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kPlayoutWarning = 1
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};
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virtual void OnErrorIsReported(const ErrorCode error) = 0;
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virtual void OnWarningIsReported(const WarningCode warning) = 0;
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protected:
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virtual ~AudioDeviceObserver() {}
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};
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// ----------------------------------------------------------------------------
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// AudioTransport
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// ----------------------------------------------------------------------------
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class AudioTransport
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{
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public:
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virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
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const uint32_t nSamples,
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const uint8_t nBytesPerSample,
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const uint8_t nChannels,
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const uint32_t samplesPerSec,
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const uint32_t totalDelayMS,
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const int32_t clockDrift,
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const uint32_t currentMicLevel,
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uint32_t& newMicLevel) = 0;
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virtual int32_t NeedMorePlayData(const uint32_t nSamples,
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const uint8_t nBytesPerSample,
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const uint8_t nChannels,
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const uint32_t samplesPerSec,
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void* audioSamples,
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uint32_t& nSamplesOut) = 0;
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protected:
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virtual ~AudioTransport() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
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14
src/modules/audio_device/main/source/audio_device.gypi
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14
src/modules/audio_device/main/source/audio_device.gypi
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@ -0,0 +1,14 @@
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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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'includes': [
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'../../audio_device.gypi',
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],
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}
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