blob: aee3e17af3ce8ba85ff062c971460a05ee7116d5 [file] [log] [blame]
Mathias Agopian65ab4712010-07-14 17:59:35 -07001/* //device/include/server/AudioFlinger/AudioMixer.h
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9** http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18#ifndef ANDROID_AUDIO_MIXER_H
19#define ANDROID_AUDIO_MIXER_H
20
21#include <stdint.h>
22#include <sys/types.h>
23
24#include "AudioBufferProvider.h"
25#include "AudioResampler.h"
26
27namespace android {
28
29// ----------------------------------------------------------------------------
30
31#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
32#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
33
34// ----------------------------------------------------------------------------
35
36class AudioMixer
37{
38public:
39 AudioMixer(size_t frameCount, uint32_t sampleRate);
40
41 ~AudioMixer();
42
43 static const uint32_t MAX_NUM_TRACKS = 32;
44 static const uint32_t MAX_NUM_CHANNELS = 2;
45
46 static const uint16_t UNITY_GAIN = 0x1000;
47
48 enum { // names
49
50 // track units (32 units)
51 TRACK0 = 0x1000,
52
53 // enable/disable
54 MIXING = 0x2000,
55
56 // setParameter targets
57 TRACK = 0x3000,
58 RESAMPLE = 0x3001,
59 RAMP_VOLUME = 0x3002, // ramp to new volume
60 VOLUME = 0x3003, // don't ramp
61
62 // set Parameter names
63 // for target TRACK
64 CHANNEL_COUNT = 0x4000,
65 FORMAT = 0x4001,
66 MAIN_BUFFER = 0x4002,
67 AUX_BUFFER = 0x4003,
68 // for TARGET RESAMPLE
69 SAMPLE_RATE = 0x4100,
70 // for TARGET VOLUME (8 channels max)
71 VOLUME0 = 0x4200,
72 VOLUME1 = 0x4201,
73 AUXLEVEL = 0x4210,
74 };
75
76
77 int getTrackName();
78 void deleteTrackName(int name);
79
80 status_t enable(int name);
81 status_t disable(int name);
82
83 status_t setActiveTrack(int track);
84 status_t setParameter(int target, int name, void *value);
85
86 status_t setBufferProvider(AudioBufferProvider* bufferProvider);
87 void process();
88
89 uint32_t trackNames() const { return mTrackNames; }
90
91 static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
92
93private:
94
95 enum {
96 NEEDS_CHANNEL_COUNT__MASK = 0x00000003,
97 NEEDS_FORMAT__MASK = 0x000000F0,
98 NEEDS_MUTE__MASK = 0x00000100,
99 NEEDS_RESAMPLE__MASK = 0x00001000,
100 NEEDS_AUX__MASK = 0x00010000,
101 };
102
103 enum {
104 NEEDS_CHANNEL_1 = 0x00000000,
105 NEEDS_CHANNEL_2 = 0x00000001,
106
107 NEEDS_FORMAT_16 = 0x00000010,
108
109 NEEDS_MUTE_DISABLED = 0x00000000,
110 NEEDS_MUTE_ENABLED = 0x00000100,
111
112 NEEDS_RESAMPLE_DISABLED = 0x00000000,
113 NEEDS_RESAMPLE_ENABLED = 0x00001000,
114
115 NEEDS_AUX_DISABLED = 0x00000000,
116 NEEDS_AUX_ENABLED = 0x00010000,
117 };
118
119 static inline int32_t applyVolume(int32_t in, int32_t v) {
120 return in * v;
121 }
122
123
124 struct state_t;
125 struct track_t;
126
127 typedef void (*mix_t)(state_t* state);
128 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
129 static const int BLOCKSIZE = 16; // 4 cache lines
130
131 struct track_t {
132 uint32_t needs;
133
134 union {
135 int16_t volume[2]; // [0]3.12 fixed point
136 int32_t volumeRL;
137 };
138
139 int32_t prevVolume[2];
140
141 int32_t volumeInc[2];
142 int32_t auxLevel;
143 int32_t auxInc;
144 int32_t prevAuxLevel;
145
146 uint16_t frameCount;
147
148 uint8_t channelCount : 4;
149 uint8_t enabled : 1;
150 uint8_t reserved0 : 3;
151 uint8_t format;
152
153 AudioBufferProvider* bufferProvider;
154 mutable AudioBufferProvider::Buffer buffer;
155
156 hook_t hook;
157 void const* in; // current location in buffer
158
159 AudioResampler* resampler;
160 uint32_t sampleRate;
161 int32_t* mainBuffer;
162 int32_t* auxBuffer;
163
164 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
165 bool doesResample() const;
166 void adjustVolumeRamp(bool aux);
167 };
168
169 // pad to 32-bytes to fill cache line
170 struct state_t {
171 uint32_t enabledTracks;
172 uint32_t needsChanged;
173 size_t frameCount;
174 mix_t hook;
175 int32_t *outputTemp;
176 int32_t *resampleTemp;
177 int32_t reserved[2];
178 track_t tracks[32]; __attribute__((aligned(32)));
179 };
180
181 int mActiveTrack;
182 uint32_t mTrackNames;
183 const uint32_t mSampleRate;
184
185 state_t mState __attribute__((aligned(32)));
186
187 void invalidateState(uint32_t mask);
188
189 static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
190 static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
191 static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
192 static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
193 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
194 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
195
196 static void process__validate(state_t* state);
197 static void process__nop(state_t* state);
198 static void process__genericNoResampling(state_t* state);
199 static void process__genericResampling(state_t* state);
200 static void process__OneTrack16BitsStereoNoResampling(state_t* state);
201 static void process__TwoTracks16BitsStereoNoResampling(state_t* state);
202};
203
204// ----------------------------------------------------------------------------
205}; // namespace android
206
207#endif // ANDROID_AUDIO_MIXER_H