1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
|
/*
** Copyright 2008, The Android Open-Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#ifndef ANDROID_AUDIO_HARDWARE_H
#define ANDROID_AUDIO_HARDWARE_H
#include <stdint.h>
#include <sys/types.h>
#include <utils/threads.h>
#include <utils/SortedVector.h>
#include <hardware_legacy/AudioHardwareBase.h>
extern "C" {
struct pcm;
struct mixer;
struct mixer_ctl;
};
namespace android {
#define CODEC_TYPE_PCM 0
#define PCM_FILL_BUFFER_COUNT 1
// Number of buffers in audio driver for output
#define AUDIO_HW_NUM_OUT_BUF 2
// TODO: determine actual audio DSP and hardware latency
// Additionnal latency introduced by audio DSP and hardware in ms
#define AUDIO_HW_OUT_LATENCY_MS 0
// Default audio output sample rate
#define AUDIO_HW_OUT_SAMPLERATE 44100
// Default audio output channel mask
#define AUDIO_HW_OUT_CHANNELS (AudioSystem::CHANNEL_OUT_STEREO)
// Default audio output sample format
#define AUDIO_HW_OUT_FORMAT (AudioSystem::PCM_16_BIT)
// Default audio output buffer size
#define AUDIO_HW_OUT_BUFSZ 4096
#if 0
// Default audio input sample rate
#define AUDIO_HW_IN_SAMPLERATE 8000
// Default audio input channel mask
#define AUDIO_HW_IN_CHANNELS (AudioSystem::CHANNEL_IN_MONO)
// Default audio input sample format
#define AUDIO_HW_IN_FORMAT (AudioSystem::PCM_16_BIT)
// Default audio input buffer size
#define AUDIO_HW_IN_BUFSZ 256
// Maximum voice volume
#define VOICE_VOLUME_MAX 5
#endif
class AudioHardware : public AudioHardwareBase
{
class AudioStreamOutALSA;
public:
AudioHardware();
virtual ~AudioHardware();
virtual status_t initCheck();
virtual status_t setVoiceVolume(float volume);
virtual status_t setMasterVolume(float volume);
virtual status_t setMode(int mode);
virtual status_t setMicMute(bool state);
virtual status_t getMicMute(bool* state);
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys);
virtual AudioStreamOut* openOutputStream(
uint32_t devices, int *format=0, uint32_t *channels=0,
uint32_t *sampleRate=0, status_t *status=0);
virtual AudioStreamIn* openInputStream(
uint32_t devices, int *format, uint32_t *channels,
uint32_t *sampleRate, status_t *status,
AudioSystem::audio_in_acoustics acoustics);
virtual void closeOutputStream(AudioStreamOut* out);
virtual void closeInputStream(AudioStreamIn* in);
virtual size_t getInputBufferSize(
uint32_t sampleRate, int format, int channelCount);
void clearCurDevice() { }
protected:
virtual status_t dump(int fd, const Vector<String16>& args);
private:
bool mInit;
bool mMicMute;
AudioStreamOutALSA *mOutput;
Mutex mLock;
struct mixer *mMixer;
class AudioStreamOutALSA : public AudioStreamOut
{
public:
AudioStreamOutALSA();
virtual ~AudioStreamOutALSA();
status_t set(AudioHardware* mHardware,
uint32_t devices,
int *pFormat,
uint32_t *pChannels,
uint32_t *pRate);
virtual uint32_t sampleRate()
const { return mSampleRate; }
virtual size_t bufferSize()
const { return mBufferSize; }
virtual uint32_t channels()
const { return mChannels; }
virtual int format()
const { return AUDIO_HW_OUT_FORMAT; }
virtual uint32_t latency()
const { return (1000 * AUDIO_HW_NUM_OUT_BUF *
(bufferSize()/frameSize()))/sampleRate() +
AUDIO_HW_OUT_LATENCY_MS; }
virtual status_t setVolume(float left, float right)
{ return INVALID_OPERATION; }
virtual ssize_t write(const void* buffer, size_t bytes);
virtual status_t standby();
virtual status_t dump(int fd, const Vector<String16>& args);
bool checkStandby();
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys);
uint32_t devices()
{ return mDevices; }
virtual status_t getRenderPosition(uint32_t *dspFrames);
private:
AudioHardware* mHardware;
struct pcm *mPcm;
struct mixer *mMixer;
struct mixer_ctl *mRouteCtl;
const char *next_route;
int mStartCount;
int mRetryCount;
bool mStandby;
uint32_t mDevices;
uint32_t mChannels;
uint32_t mSampleRate;
size_t mBufferSize;
};
};
}; // namespace android
#endif
|