AudioMixer.h revision 5c94b6c7689a335e26a86e8a0d04b56dc627738e
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 uint32_t maxNumTracks = MAX_NUM_TRACKS); 36 37 /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed 38 39 static const uint32_t MAX_NUM_TRACKS = 32; 40 static const uint32_t MAX_NUM_CHANNELS = 2; 41 42 static const uint16_t UNITY_GAIN = 0x1000; 43 44 enum { // names 45 46 // track names (MAX_NUM_TRACKS units) 47 TRACK0 = 0x1000, 48 49 // 0x2000 is unused 50 51 // setParameter targets 52 TRACK = 0x3000, 53 RESAMPLE = 0x3001, 54 RAMP_VOLUME = 0x3002, // ramp to new volume 55 VOLUME = 0x3003, // don't ramp 56 57 // set Parameter names 58 // for target TRACK 59 CHANNEL_MASK = 0x4000, 60 FORMAT = 0x4001, 61 MAIN_BUFFER = 0x4002, 62 AUX_BUFFER = 0x4003, 63 // for target RESAMPLE 64 SAMPLE_RATE = 0x4100, 65 RESET = 0x4101, 66 // for target RAMP_VOLUME and VOLUME (8 channels max) 67 VOLUME0 = 0x4200, 68 VOLUME1 = 0x4201, 69 AUXLEVEL = 0x4210, 70 }; 71 72 73 // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS 74 int getTrackName(); 75 void deleteTrackName(int name); 76 77 void enable(int name); 78 void disable(int name); 79 80 void setParameter(int name, int target, int param, void *value); 81 82 void setBufferProvider(int name, AudioBufferProvider* bufferProvider); 83 void process(int64_t pts); 84 85 uint32_t trackNames() const { return mTrackNames; } 86 87 size_t getUnreleasedFrames(int name) const; 88 89private: 90 91 enum { 92 NEEDS_CHANNEL_COUNT__MASK = 0x00000003, 93 NEEDS_FORMAT__MASK = 0x000000F0, 94 NEEDS_MUTE__MASK = 0x00000100, 95 NEEDS_RESAMPLE__MASK = 0x00001000, 96 NEEDS_AUX__MASK = 0x00010000, 97 }; 98 99 enum { 100 NEEDS_CHANNEL_1 = 0x00000000, 101 NEEDS_CHANNEL_2 = 0x00000001, 102 103 NEEDS_FORMAT_16 = 0x00000010, 104 105 NEEDS_MUTE_DISABLED = 0x00000000, 106 NEEDS_MUTE_ENABLED = 0x00000100, 107 108 NEEDS_RESAMPLE_DISABLED = 0x00000000, 109 NEEDS_RESAMPLE_ENABLED = 0x00001000, 110 111 NEEDS_AUX_DISABLED = 0x00000000, 112 NEEDS_AUX_ENABLED = 0x00010000, 113 }; 114 115 struct state_t; 116 struct track_t; 117 118 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux); 119 static const int BLOCKSIZE = 16; // 4 cache lines 120 121 struct track_t { 122 uint32_t needs; 123 124 union { 125 int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point 126 int32_t volumeRL; 127 }; 128 129 int32_t prevVolume[MAX_NUM_CHANNELS]; 130 131 // 16-byte boundary 132 133 int32_t volumeInc[MAX_NUM_CHANNELS]; 134 int32_t auxInc; 135 int32_t prevAuxLevel; 136 137 // 16-byte boundary 138 139 int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance 140 uint16_t frameCount; 141 142 uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) 143 uint8_t format; // always 16 144 uint16_t enabled; // actually bool 145 uint32_t channelMask; // currently under-used 146 147 AudioBufferProvider* bufferProvider; 148 149 // 16-byte boundary 150 151 mutable AudioBufferProvider::Buffer buffer; // 8 bytes 152 153 hook_t hook; 154 const void* in; // current location in buffer 155 156 // 16-byte boundary 157 158 AudioResampler* resampler; 159 uint32_t sampleRate; 160 int32_t* mainBuffer; 161 int32_t* auxBuffer; 162 163 // 16-byte boundary 164 165 uint64_t localTimeFreq; 166 167 int64_t padding; 168 169 // 16-byte boundary 170 171 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); 172 bool doesResample() const { return resampler != NULL; } 173 void resetResampler() { if (resampler != NULL) resampler->reset(); } 174 void adjustVolumeRamp(bool aux); 175 size_t getUnreleasedFrames() const { return resampler != NULL ? 176 resampler->getUnreleasedFrames() : 0; }; 177 }; 178 179 // pad to 32-bytes to fill cache line 180 struct state_t { 181 uint32_t enabledTracks; 182 uint32_t needsChanged; 183 size_t frameCount; 184 void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL 185 int32_t *outputTemp; 186 int32_t *resampleTemp; 187 int32_t reserved[2]; 188 // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS 189 track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32))); 190 }; 191 192 // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. 193 uint32_t mTrackNames; 194 195 // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, 196 // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS 197 const uint32_t mConfiguredNames; 198 199 const uint32_t mSampleRate; 200 201 state_t mState __attribute__((aligned(32))); 202 203 void invalidateState(uint32_t mask); 204 205 static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 206 static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 207 static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 208 static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 209 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 210 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 211 212 static void process__validate(state_t* state, int64_t pts); 213 static void process__nop(state_t* state, int64_t pts); 214 static void process__genericNoResampling(state_t* state, int64_t pts); 215 static void process__genericResampling(state_t* state, int64_t pts); 216 static void process__OneTrack16BitsStereoNoResampling(state_t* state, 217 int64_t pts); 218#if 0 219 static void process__TwoTracks16BitsStereoNoResampling(state_t* state, 220 int64_t pts); 221#endif 222 223 static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, 224 int outputFrameIndex); 225}; 226 227// ---------------------------------------------------------------------------- 228}; // namespace android 229 230#endif // ANDROID_AUDIO_MIXER_H 231