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 16 namespace webrtc { 17 18 static const int kAdmMaxDeviceNameSize = 128; 19 static const int kAdmMaxFileNameSize = 512; 20 static const int kAdmMaxGuidSize = 128; 21 22 static const int kAdmMinPlayoutBufferSizeMs = 10; 23 static const int kAdmMaxPlayoutBufferSizeMs = 250; 24 25 // ---------------------------------------------------------------------------- 26 // AudioDeviceObserver 27 // ---------------------------------------------------------------------------- 28 29 class AudioDeviceObserver 30 { 31 public: 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 46 protected: 47 virtual ~AudioDeviceObserver() {} 48 }; 49 50 // ---------------------------------------------------------------------------- 51 // AudioTransport 52 // ---------------------------------------------------------------------------- 53 54 class AudioTransport 55 { 56 public: 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) {} 133 134 protected: 135 virtual ~AudioTransport() {} 136 }; 137 138 } // namespace webrtc 139 140 #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H 141
