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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
Jean-Michel Trivi0d255b22011-05-24 15:53:33 -070064 CHANNEL_MASK = 0x4000,
Mathias Agopian65ab4712010-07-14 17:59:35 -070065 FORMAT = 0x4001,
66 MAIN_BUFFER = 0x4002,
67 AUX_BUFFER = 0x4003,
68 // for TARGET RESAMPLE
69 SAMPLE_RATE = 0x4100,
Eric Laurent243f5f92011-02-28 16:52:51 -080070 RESET = 0x4101,
Mathias Agopian65ab4712010-07-14 17:59:35 -070071 // for TARGET VOLUME (8 channels max)
72 VOLUME0 = 0x4200,
73 VOLUME1 = 0x4201,
74 AUXLEVEL = 0x4210,
75 };
76
77
78 int getTrackName();
79 void deleteTrackName(int name);
80
81 status_t enable(int name);
82 status_t disable(int name);
83
84 status_t setActiveTrack(int track);
85 status_t setParameter(int target, int name, void *value);
86
87 status_t setBufferProvider(AudioBufferProvider* bufferProvider);
88 void process();
89
90 uint32_t trackNames() const { return mTrackNames; }
91
92 static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
93
94private:
95
96 enum {
97 NEEDS_CHANNEL_COUNT__MASK = 0x00000003,
98 NEEDS_FORMAT__MASK = 0x000000F0,
99 NEEDS_MUTE__MASK = 0x00000100,
100 NEEDS_RESAMPLE__MASK = 0x00001000,
101 NEEDS_AUX__MASK = 0x00010000,
102 };
103
104 enum {
105 NEEDS_CHANNEL_1 = 0x00000000,
106 NEEDS_CHANNEL_2 = 0x00000001,
107
108 NEEDS_FORMAT_16 = 0x00000010,
109
110 NEEDS_MUTE_DISABLED = 0x00000000,
111 NEEDS_MUTE_ENABLED = 0x00000100,
112
113 NEEDS_RESAMPLE_DISABLED = 0x00000000,
114 NEEDS_RESAMPLE_ENABLED = 0x00001000,
115
116 NEEDS_AUX_DISABLED = 0x00000000,
117 NEEDS_AUX_ENABLED = 0x00010000,
118 };
119
120 static inline int32_t applyVolume(int32_t in, int32_t v) {
121 return in * v;
122 }
123
124
125 struct state_t;
126 struct track_t;
127
128 typedef void (*mix_t)(state_t* state);
129 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
130 static const int BLOCKSIZE = 16; // 4 cache lines
131
132 struct track_t {
133 uint32_t needs;
134
135 union {
136 int16_t volume[2]; // [0]3.12 fixed point
137 int32_t volumeRL;
138 };
139
140 int32_t prevVolume[2];
141
142 int32_t volumeInc[2];
143 int32_t auxLevel;
144 int32_t auxInc;
145 int32_t prevAuxLevel;
146
147 uint16_t frameCount;
148
149 uint8_t channelCount : 4;
150 uint8_t enabled : 1;
151 uint8_t reserved0 : 3;
152 uint8_t format;
Jean-Michel Trivi0d255b22011-05-24 15:53:33 -0700153 uint32_t channelMask;
Mathias Agopian65ab4712010-07-14 17:59:35 -0700154
155 AudioBufferProvider* bufferProvider;
156 mutable AudioBufferProvider::Buffer buffer;
157
158 hook_t hook;
159 void const* in; // current location in buffer
160
161 AudioResampler* resampler;
162 uint32_t sampleRate;
163 int32_t* mainBuffer;
164 int32_t* auxBuffer;
165
166 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
167 bool doesResample() const;
Eric Laurent243f5f92011-02-28 16:52:51 -0800168 void resetResampler();
Mathias Agopian65ab4712010-07-14 17:59:35 -0700169 void adjustVolumeRamp(bool aux);
170 };
171
172 // pad to 32-bytes to fill cache line
173 struct state_t {
174 uint32_t enabledTracks;
175 uint32_t needsChanged;
176 size_t frameCount;
177 mix_t hook;
178 int32_t *outputTemp;
179 int32_t *resampleTemp;
180 int32_t reserved[2];
181 track_t tracks[32]; __attribute__((aligned(32)));
182 };
183
184 int mActiveTrack;
185 uint32_t mTrackNames;
186 const uint32_t mSampleRate;
187
188 state_t mState __attribute__((aligned(32)));
189
190 void invalidateState(uint32_t mask);
191
192 static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
193 static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
194 static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
195 static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
196 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
197 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
198
199 static void process__validate(state_t* state);
200 static void process__nop(state_t* state);
201 static void process__genericNoResampling(state_t* state);
202 static void process__genericResampling(state_t* state);
203 static void process__OneTrack16BitsStereoNoResampling(state_t* state);
204 static void process__TwoTracks16BitsStereoNoResampling(state_t* state);
205};
206
207// ----------------------------------------------------------------------------
208}; // namespace android
209
210#endif // ANDROID_AUDIO_MIXER_H