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
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
|
/*
**
** Copyright 2007, 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_MIXER_H
#define ANDROID_AUDIO_MIXER_H
#include <stdint.h>
#include <sys/types.h>
#include "AudioBufferProvider.h"
#include "AudioResampler.h"
namespace android {
// ----------------------------------------------------------------------------
class AudioMixer
{
public:
AudioMixer(size_t frameCount, uint32_t sampleRate);
/*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
static const uint32_t MAX_NUM_TRACKS = 32;
static const uint32_t MAX_NUM_CHANNELS = 2;
static const uint16_t UNITY_GAIN = 0x1000;
enum { // names
// track names (MAX_NUM_TRACKS units)
TRACK0 = 0x1000,
// 0x2000 is unused
// setParameter targets
TRACK = 0x3000,
RESAMPLE = 0x3001,
RAMP_VOLUME = 0x3002, // ramp to new volume
VOLUME = 0x3003, // don't ramp
// set Parameter names
// for target TRACK
CHANNEL_MASK = 0x4000,
FORMAT = 0x4001,
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
// for target RESAMPLE
SAMPLE_RATE = 0x4100,
RESET = 0x4101,
// for target RAMP_VOLUME and VOLUME (8 channels max)
VOLUME0 = 0x4200,
VOLUME1 = 0x4201,
AUXLEVEL = 0x4210,
};
// For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
int getTrackName();
void deleteTrackName(int name);
void enable(int name);
void disable(int name);
void setParameter(int name, int target, int param, void *value);
void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
void process(int64_t pts);
uint32_t trackNames() const { return mTrackNames; }
size_t getUnreleasedFrames(int name) const;
private:
enum {
NEEDS_CHANNEL_COUNT__MASK = 0x00000003,
NEEDS_FORMAT__MASK = 0x000000F0,
NEEDS_MUTE__MASK = 0x00000100,
NEEDS_RESAMPLE__MASK = 0x00001000,
NEEDS_AUX__MASK = 0x00010000,
};
enum {
NEEDS_CHANNEL_1 = 0x00000000,
NEEDS_CHANNEL_2 = 0x00000001,
NEEDS_FORMAT_16 = 0x00000010,
NEEDS_MUTE_DISABLED = 0x00000000,
NEEDS_MUTE_ENABLED = 0x00000100,
NEEDS_RESAMPLE_DISABLED = 0x00000000,
NEEDS_RESAMPLE_ENABLED = 0x00001000,
NEEDS_AUX_DISABLED = 0x00000000,
NEEDS_AUX_ENABLED = 0x00010000,
};
struct state_t;
struct track_t;
typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
static const int BLOCKSIZE = 16; // 4 cache lines
struct track_t {
uint32_t needs;
union {
int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
int32_t volumeRL;
};
int32_t prevVolume[MAX_NUM_CHANNELS];
// 16-byte boundary
int32_t volumeInc[MAX_NUM_CHANNELS];
int32_t auxInc;
int32_t prevAuxLevel;
// 16-byte boundary
int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
uint16_t frameCount;
uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
uint8_t format; // always 16
uint16_t enabled; // actually bool
uint32_t channelMask; // currently under-used
AudioBufferProvider* bufferProvider;
// 16-byte boundary
mutable AudioBufferProvider::Buffer buffer; // 8 bytes
hook_t hook;
const void* in; // current location in buffer
// 16-byte boundary
AudioResampler* resampler;
uint32_t sampleRate;
int32_t* mainBuffer;
int32_t* auxBuffer;
// 16-byte boundary
uint64_t localTimeFreq;
int64_t padding;
// 16-byte boundary
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
bool doesResample() const { return resampler != NULL; }
void resetResampler() { if (resampler != NULL) resampler->reset(); }
void adjustVolumeRamp(bool aux);
size_t getUnreleasedFrames() const { return resampler != NULL ?
resampler->getUnreleasedFrames() : 0; };
};
// pad to 32-bytes to fill cache line
struct state_t {
uint32_t enabledTracks;
uint32_t needsChanged;
size_t frameCount;
void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL
int32_t *outputTemp;
int32_t *resampleTemp;
int32_t reserved[2];
track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
};
// bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
uint32_t mTrackNames;
const uint32_t mSampleRate;
state_t mState __attribute__((aligned(32)));
void invalidateState(uint32_t mask);
static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
static void process__validate(state_t* state, int64_t pts);
static void process__nop(state_t* state, int64_t pts);
static void process__genericNoResampling(state_t* state, int64_t pts);
static void process__genericResampling(state_t* state, int64_t pts);
static void process__OneTrack16BitsStereoNoResampling(state_t* state,
int64_t pts);
#if 0
static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
int64_t pts);
#endif
static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
int outputFrameIndex);
};
// ----------------------------------------------------------------------------
}; // namespace android
#endif // ANDROID_AUDIO_MIXER_H
|
