Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_device/include/audio_device_defines.h

@@ -0,0 +1,80 @@
+/*
+ *  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 WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+static const int kAdmMaxDeviceNameSize = 128;
+static const int kAdmMaxFileNameSize = 512;
+static const int kAdmMaxGuidSize = 128;
+
+static const int kAdmMinPlayoutBufferSizeMs = 10;
+static const int kAdmMaxPlayoutBufferSizeMs = 250;
+
+// ----------------------------------------------------------------------------
+//  AudioDeviceObserver
+// ----------------------------------------------------------------------------
+
+class AudioDeviceObserver
+{
+public:
+    enum ErrorCode
+    {
+        kRecordingError = 0,
+        kPlayoutError = 1
+    };
+    enum WarningCode
+    {
+        kRecordingWarning = 0,
+        kPlayoutWarning = 1
+    };
+
+    virtual void OnErrorIsReported(const ErrorCode error) = 0;
+    virtual void OnWarningIsReported(const WarningCode warning) = 0;
+
+protected:
+    virtual ~AudioDeviceObserver() {}
+};
+
+// ----------------------------------------------------------------------------
+//  AudioTransport
+// ----------------------------------------------------------------------------
+
+class AudioTransport
+{
+public:
+    virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                            const uint32_t nSamples,
+                                            const uint8_t nBytesPerSample,
+                                            const uint8_t nChannels,
+                                            const uint32_t samplesPerSec,
+                                            const uint32_t totalDelayMS,
+                                            const int32_t clockDrift,
+                                            const uint32_t currentMicLevel,
+                                            uint32_t& newMicLevel) = 0;   
+
+    virtual int32_t NeedMorePlayData(const uint32_t nSamples,
+                                     const uint8_t nBytesPerSample,
+                                     const uint8_t nChannels,
+                                     const uint32_t samplesPerSec,
+                                     void* audioSamples,
+                                     uint32_t& nSamplesOut) = 0;
+
+protected:
+    virtual ~AudioTransport() {}
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H