AudioMixer.h revision 3b81acab52b7140c1b8b20be2d67be3e221637e7 
1/*
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
31class AudioMixer
32{
33public:
34                            AudioMixer(size_t frameCount, uint32_t sampleRate);
35
36    /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
37
38    static const uint32_t MAX_NUM_TRACKS = 32;
39    static const uint32_t MAX_NUM_CHANNELS = 2;
40
41    static const uint16_t UNITY_GAIN = 0x1000;
42
43    enum { // names
44
45        // track names (MAX_NUM_TRACKS units)
46        TRACK0          = 0x1000,
47
48        // 0x2000 is unused
49
50        // setParameter targets
51        TRACK           = 0x3000,
52        RESAMPLE        = 0x3001,
53        RAMP_VOLUME     = 0x3002, // ramp to new volume
54        VOLUME          = 0x3003, // don't ramp
55
56        // set Parameter names
57        // for target TRACK
58        CHANNEL_MASK    = 0x4000,
59        FORMAT          = 0x4001,
60        MAIN_BUFFER     = 0x4002,
61        AUX_BUFFER      = 0x4003,
62        // for target RESAMPLE
63        SAMPLE_RATE     = 0x4100,
64        RESET           = 0x4101,
65        // for target RAMP_VOLUME and VOLUME (8 channels max)
66        VOLUME0         = 0x4200,
67        VOLUME1         = 0x4201,
68        AUXLEVEL        = 0x4210,
69    };
70
71
72    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
73    int         getTrackName();
74    void        deleteTrackName(int name);
75
76    void        enable(int name);
77    void        disable(int name);
78
79    void        setParameter(int name, int target, int param, void *value);
80
81    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
82    void        process(int64_t pts);
83
84    uint32_t    trackNames() const { return mTrackNames; }
85
86    size_t      getUnreleasedFrames(int name) const;
87
88private:
89
90    enum {
91        NEEDS_CHANNEL_COUNT__MASK   = 0x00000003,
92        NEEDS_FORMAT__MASK          = 0x000000F0,
93        NEEDS_MUTE__MASK            = 0x00000100,
94        NEEDS_RESAMPLE__MASK        = 0x00001000,
95        NEEDS_AUX__MASK             = 0x00010000,
96    };
97
98    enum {
99        NEEDS_CHANNEL_1             = 0x00000000,
100        NEEDS_CHANNEL_2             = 0x00000001,
101
102        NEEDS_FORMAT_16             = 0x00000010,
103
104        NEEDS_MUTE_DISABLED         = 0x00000000,
105        NEEDS_MUTE_ENABLED          = 0x00000100,
106
107        NEEDS_RESAMPLE_DISABLED     = 0x00000000,
108        NEEDS_RESAMPLE_ENABLED      = 0x00001000,
109
110        NEEDS_AUX_DISABLED     = 0x00000000,
111        NEEDS_AUX_ENABLED      = 0x00010000,
112    };
113
114    struct state_t;
115    struct track_t;
116
117    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
118    static const int BLOCKSIZE = 16; // 4 cache lines
119
120    struct track_t {
121        uint32_t    needs;
122
123        union {
124        int16_t     volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
125        int32_t     volumeRL;
126        };
127
128        int32_t     prevVolume[MAX_NUM_CHANNELS];
129
130        // 16-byte boundary
131
132        int32_t     volumeInc[MAX_NUM_CHANNELS];
133        int32_t     auxInc;
134        int32_t     prevAuxLevel;
135
136        // 16-byte boundary
137
138        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
139        uint16_t    frameCount;
140
141        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
142        uint8_t     format;         // always 16
143        uint16_t    enabled;        // actually bool
144        uint32_t    channelMask;    // currently under-used
145
146        AudioBufferProvider*                bufferProvider;
147
148        // 16-byte boundary
149
150        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
151
152        hook_t      hook;
153        const void* in;             // current location in buffer
154
155        // 16-byte boundary
156
157        AudioResampler*     resampler;
158        uint32_t            sampleRate;
159        int32_t*           mainBuffer;
160        int32_t*           auxBuffer;
161
162        // 16-byte boundary
163
164        uint64_t    localTimeFreq;
165
166        int64_t     padding;
167
168        // 16-byte boundary
169
170        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
171        bool        doesResample() const { return resampler != NULL; }
172        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
173        void        adjustVolumeRamp(bool aux);
174        size_t      getUnreleasedFrames() const { return resampler != NULL ?
175                                                    resampler->getUnreleasedFrames() : 0; };
176    };
177
178    // pad to 32-bytes to fill cache line
179    struct state_t {
180        uint32_t        enabledTracks;
181        uint32_t        needsChanged;
182        size_t          frameCount;
183        void            (*hook)(state_t* state, int64_t pts);   // one of process__*, never NULL
184        int32_t         *outputTemp;
185        int32_t         *resampleTemp;
186        int32_t         reserved[2];
187        track_t         tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
188    };
189
190    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
191    uint32_t        mTrackNames;
192    const uint32_t  mSampleRate;
193
194    state_t         mState __attribute__((aligned(32)));
195
196    void invalidateState(uint32_t mask);
197
198    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
199    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
200    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
201    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
202    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
203    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
204
205    static void process__validate(state_t* state, int64_t pts);
206    static void process__nop(state_t* state, int64_t pts);
207    static void process__genericNoResampling(state_t* state, int64_t pts);
208    static void process__genericResampling(state_t* state, int64_t pts);
209    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
210                                                          int64_t pts);
211#if 0
212    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
213                                                           int64_t pts);
214#endif
215
216    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
217                                      int outputFrameIndex);
218};
219
220// ----------------------------------------------------------------------------
221}; // namespace android
222
223#endif // ANDROID_AUDIO_MIXER_H
224