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
|
/*
* Copyright (C) 2013-2014 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 QCOM_AUDIO_PLATFORM_H
#define QCOM_AUDIO_PLATFORM_H
/*
* Below are the devices for which is back end is same, SLIMBUS_0_RX.
* All these devices are handled by the internal HW codec. We can
* enable any one of these devices at any time
*/
#define AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND \
(AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \
AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE)
/* Sound devices specific to the platform
* The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound
* devices to enable corresponding mixer paths
*/
enum {
SND_DEVICE_NONE = 0,
/* Playback devices */
SND_DEVICE_MIN,
SND_DEVICE_OUT_BEGIN = SND_DEVICE_MIN,
SND_DEVICE_OUT_HANDSET = SND_DEVICE_OUT_BEGIN,
SND_DEVICE_OUT_SPEAKER,
SND_DEVICE_OUT_SPEAKER_REVERSE,
SND_DEVICE_OUT_SPEAKER_SAFE,
SND_DEVICE_OUT_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
SND_DEVICE_OUT_VOICE_SPEAKER,
SND_DEVICE_OUT_VOICE_HEADPHONES,
SND_DEVICE_OUT_VOICE_HEADSET,
SND_DEVICE_OUT_HDMI,
SND_DEVICE_OUT_SPEAKER_AND_HDMI,
SND_DEVICE_OUT_BT_SCO,
SND_DEVICE_OUT_BT_SCO_WB,
SND_DEVICE_OUT_BT_A2DP,
SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP,
SND_DEVICE_OUT_VOICE_HANDSET_TMUS,
SND_DEVICE_OUT_VOICE_HANDSET,
SND_DEVICE_OUT_VOICE_HAC_HANDSET,
SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
SND_DEVICE_OUT_USB_HEADSET,
SND_DEVICE_OUT_USB_HEADPHONES,
SND_DEVICE_OUT_VOICE_USB_HEADSET,
SND_DEVICE_OUT_VOICE_USB_HEADPHONES,
SND_DEVICE_OUT_END,
/*
* Note: IN_BEGIN should be same as OUT_END because total number of devices
* SND_DEVICES_MAX should not exceed MAX_RX + MAX_TX devices.
*/
/* Capture devices */
SND_DEVICE_IN_BEGIN = SND_DEVICE_OUT_END,
SND_DEVICE_IN_HANDSET_MIC = SND_DEVICE_IN_BEGIN,
SND_DEVICE_IN_SPEAKER_MIC,
SND_DEVICE_IN_HEADSET_MIC,
SND_DEVICE_IN_HANDSET_MIC_AEC,
SND_DEVICE_IN_SPEAKER_MIC_AEC,
SND_DEVICE_IN_HEADSET_MIC_AEC,
SND_DEVICE_IN_VOICE_SPEAKER_MIC,
SND_DEVICE_IN_VOICE_HEADSET_MIC,
SND_DEVICE_IN_HDMI_MIC,
SND_DEVICE_IN_BT_SCO_MIC,
SND_DEVICE_IN_BT_SCO_MIC_NREC,
SND_DEVICE_IN_BT_SCO_MIC_WB,
SND_DEVICE_IN_BT_SCO_MIC_WB_NREC,
SND_DEVICE_IN_CAMCORDER_MIC,
SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
SND_DEVICE_IN_VOICE_DMIC_EF,
SND_DEVICE_IN_VOICE_DMIC_BS,
SND_DEVICE_IN_VOICE_DMIC_EF_TMUS,
SND_DEVICE_IN_VOICE_SPEAKER_DMIC_EF,
SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BS,
SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC,
SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC,
SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC,
SND_DEVICE_IN_VOICE_REC_MIC,
SND_DEVICE_IN_VOICE_REC_DMIC_EF,
SND_DEVICE_IN_VOICE_REC_DMIC_BS,
SND_DEVICE_IN_VOICE_REC_DMIC_EF_FLUENCE,
SND_DEVICE_IN_VOICE_REC_DMIC_BS_FLUENCE,
SND_DEVICE_IN_END,
SND_DEVICE_MAX = SND_DEVICE_IN_END,
};
#define MIXER_CARD 0
#define SOUND_CARD 0
#define MIXER_PATH_MAX_LENGTH 100
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
#define DEFAULT_INPUT_SAMPLING_RATE 48000
/*
* tinyAlsa library interprets period size as number of frames
* one frame = channel_count * sizeof (pcm sample)
* so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
* DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
* We should take care of returning proper size when AudioFlinger queries for
* the buffer size of an input/output stream
*/
#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 960
#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 8
#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
#define HDMI_MULTI_PERIOD_SIZE 336
#define HDMI_MULTI_PERIOD_COUNT 8
#define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
#define HDMI_MULTI_PERIOD_BYTES (HDMI_MULTI_PERIOD_SIZE * HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)
#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
#define AUDIO_CAPTURE_PERIOD_COUNT 2
#define VOIP_CAPTURE_PERIOD_DURATION_MSEC 20
#define VOIP_CAPTURE_PERIOD_COUNT 2
#define VOIP_PLAYBACK_PERIOD_DURATION_MSEC 20
#define VOIP_PLAYBACK_PERIOD_COUNT 2
#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
#define LOW_LATENCY_CAPTURE_USE_CASE 0
#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
#define AFE_PROXY_RECORD_PCM_DEVICE 8
#define HFP_ASM_RX_TX 18
#define PLATFORM_INFO_XML_PATH "audio_platform_info.xml"
#define PLATFORM_INFO_XML_BASE_STRING "audio_platform_info"
#define DEVICE_NAME_MAX_SIZE 128
#define AUDIO_MAKE_STRING_FROM_ENUM(X) { #X, X }
#endif // QCOM_AUDIO_PLATFORM_H
|
