1/*
2 *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
12#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
13
14#include "webrtc/typedefs.h"
15
16namespace webrtc {
17
18static const int kAdmMaxDeviceNameSize = 128;
19static const int kAdmMaxFileNameSize = 512;
20static const int kAdmMaxGuidSize = 128;
21
22static const int kAdmMinPlayoutBufferSizeMs = 10;
23static const int kAdmMaxPlayoutBufferSizeMs = 250;
24
25// ----------------------------------------------------------------------------
26//  AudioDeviceObserver
27// ----------------------------------------------------------------------------
28
29class AudioDeviceObserver
30{
31public:
32    enum ErrorCode
33    {
34        kRecordingError = 0,
35        kPlayoutError = 1
36    };
37    enum WarningCode
38    {
39        kRecordingWarning = 0,
40        kPlayoutWarning = 1
41    };
42
43    virtual void OnErrorIsReported(const ErrorCode error) = 0;
44    virtual void OnWarningIsReported(const WarningCode warning) = 0;
45
46protected:
47    virtual ~AudioDeviceObserver() {}
48};
49
50// ----------------------------------------------------------------------------
51//  AudioTransport
52// ----------------------------------------------------------------------------
53
54class AudioTransport
55{
56public:
57    virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
58                                            const uint32_t nSamples,
59                                            const uint8_t nBytesPerSample,
60                                            const uint8_t nChannels,
61                                            const uint32_t samplesPerSec,
62                                            const uint32_t totalDelayMS,
63                                            const int32_t clockDrift,
64                                            const uint32_t currentMicLevel,
65                                            const bool keyPressed,
66                                            uint32_t& newMicLevel) = 0;
67
68    virtual int32_t NeedMorePlayData(const uint32_t nSamples,
69                                     const uint8_t nBytesPerSample,
70                                     const uint8_t nChannels,
71                                     const uint32_t samplesPerSec,
72                                     void* audioSamples,
73                                     uint32_t& nSamplesOut,
74                                     int64_t* elapsed_time_ms,
75                                     int64_t* ntp_time_ms) = 0;
76
77    // Method to pass captured data directly and unmixed to network channels.
78    // |channel_ids| contains a list of VoE channels which are the
79    // sinks to the capture data. |audio_delay_milliseconds| is the sum of
80    // recording delay and playout delay of the hardware. |current_volume| is
81    // in the range of [0, 255], representing the current microphone analog
82    // volume. |key_pressed| is used by the typing detection.
83    // |need_audio_processing| specify if the data needs to be processed by APM.
84    // Currently WebRtc supports only one APM, and Chrome will make sure only
85    // one stream goes through APM. When |need_audio_processing| is false, the
86    // values of |audio_delay_milliseconds|, |current_volume| and |key_pressed|
87    // will be ignored.
88    // The return value is the new microphone volume, in the range of |0, 255].
89    // When the volume does not need to be updated, it returns 0.
90    // TODO(xians): Remove this interface after Chrome and Libjingle switches
91    // to OnData().
92    virtual int OnDataAvailable(const int voe_channels[],
93                                int number_of_voe_channels,
94                                const int16_t* audio_data,
95                                int sample_rate,
96                                int number_of_channels,
97                                int number_of_frames,
98                                int audio_delay_milliseconds,
99                                int current_volume,
100                                bool key_pressed,
101                                bool need_audio_processing) { return 0; }
102
103    // Method to pass the captured audio data to the specific VoE channel.
104    // |voe_channel| is the id of the VoE channel which is the sink to the
105    // capture data.
106    // TODO(xians): Remove this interface after Libjingle switches to
107    // PushCaptureData().
108    virtual void OnData(int voe_channel, const void* audio_data,
109                        int bits_per_sample, int sample_rate,
110                        int number_of_channels,
111                        int number_of_frames) {}
112
113    // Method to push the captured audio data to the specific VoE channel.
114    // The data will not undergo audio processing.
115    // |voe_channel| is the id of the VoE channel which is the sink to the
116    // capture data.
117    // TODO(xians): Make the interface pure virtual after Libjingle
118    // has its implementation.
119    virtual void PushCaptureData(int voe_channel, const void* audio_data,
120                                 int bits_per_sample, int sample_rate,
121                                 int number_of_channels,
122                                 int number_of_frames) {}
123
124    // Method to pull mixed render audio data from all active VoE channels.
125    // The data will not be passed as reference for audio processing internally.
126    // TODO(xians): Support getting the unmixed render data from specific VoE
127    // channel.
128    virtual void PullRenderData(int bits_per_sample, int sample_rate,
129                                int number_of_channels, int number_of_frames,
130                                void* audio_data,
131                                int64_t* elapsed_time_ms,
132                                int64_t* ntp_time_ms) {}
133
134protected:
135    virtual ~AudioTransport() {}
136};
137
138}  // namespace webrtc
139
140#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
141