AudioMixer.h revision 7d5b26230a179cd7bcc01f6578cd80d8c15a92a5
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 <utils/threads.h> 25 26#include "AudioBufferProvider.h" 27#include "AudioResampler.h" 28 29#include <audio_effects/effect_downmix.h> 30#include <system/audio.h> 31 32namespace android { 33 34// ---------------------------------------------------------------------------- 35 36class AudioMixer 37{ 38public: 39 AudioMixer(size_t frameCount, uint32_t sampleRate, 40 uint32_t maxNumTracks = MAX_NUM_TRACKS); 41 42 /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed 43 44 static const uint32_t MAX_NUM_TRACKS = 32; 45 // maximum number of channels supported by the mixer 46 static const uint32_t MAX_NUM_CHANNELS = 2; 47 // maximum number of channels supported for the content 48 static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8; 49 50 static const uint16_t UNITY_GAIN = 0x1000; 51 52 enum { // names 53 54 // track names (MAX_NUM_TRACKS units) 55 TRACK0 = 0x1000, 56 57 // 0x2000 is unused 58 59 // setParameter targets 60 TRACK = 0x3000, 61 RESAMPLE = 0x3001, 62 RAMP_VOLUME = 0x3002, // ramp to new volume 63 VOLUME = 0x3003, // don't ramp 64 65 // set Parameter names 66 // for target TRACK 67 CHANNEL_MASK = 0x4000, 68 FORMAT = 0x4001, 69 MAIN_BUFFER = 0x4002, 70 AUX_BUFFER = 0x4003, 71 DOWNMIX_TYPE = 0X4004, 72 // for target RESAMPLE 73 SAMPLE_RATE = 0x4100, 74 RESET = 0x4101, 75 // for target RAMP_VOLUME and VOLUME (8 channels max) 76 VOLUME0 = 0x4200, 77 VOLUME1 = 0x4201, 78 AUXLEVEL = 0x4210, 79 }; 80 81 82 // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS 83 84 // Allocate a track name. Returns new track name if successful, -1 on failure. 85 int getTrackName(); 86 87 // Free an allocated track by name 88 void deleteTrackName(int name); 89 90 // Enable or disable an allocated track by name 91 void enable(int name); 92 void disable(int name); 93 94 void setParameter(int name, int target, int param, void *value); 95 96 void setBufferProvider(int name, AudioBufferProvider* bufferProvider); 97 void process(int64_t pts); 98 99 uint32_t trackNames() const { return mTrackNames; } 100 101 size_t getUnreleasedFrames(int name) const; 102 103private: 104 105 enum { 106 NEEDS_CHANNEL_COUNT__MASK = 0x00000007, 107 NEEDS_FORMAT__MASK = 0x000000F0, 108 NEEDS_MUTE__MASK = 0x00000100, 109 NEEDS_RESAMPLE__MASK = 0x00001000, 110 NEEDS_AUX__MASK = 0x00010000, 111 }; 112 113 enum { 114 NEEDS_CHANNEL_1 = 0x00000000, 115 NEEDS_CHANNEL_2 = 0x00000001, 116 117 NEEDS_FORMAT_16 = 0x00000010, 118 119 NEEDS_MUTE_DISABLED = 0x00000000, 120 NEEDS_MUTE_ENABLED = 0x00000100, 121 122 NEEDS_RESAMPLE_DISABLED = 0x00000000, 123 NEEDS_RESAMPLE_ENABLED = 0x00001000, 124 125 NEEDS_AUX_DISABLED = 0x00000000, 126 NEEDS_AUX_ENABLED = 0x00010000, 127 }; 128 129 struct state_t; 130 struct track_t; 131 class DownmixerBufferProvider; 132 133 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux); 134 static const int BLOCKSIZE = 16; // 4 cache lines 135 136 struct track_t { 137 uint32_t needs; 138 139 union { 140 int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point 141 int32_t volumeRL; 142 }; 143 144 int32_t prevVolume[MAX_NUM_CHANNELS]; 145 146 // 16-byte boundary 147 148 int32_t volumeInc[MAX_NUM_CHANNELS]; 149 int32_t auxInc; 150 int32_t prevAuxLevel; 151 152 // 16-byte boundary 153 154 int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance 155 uint16_t frameCount; 156 157 uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) 158 uint8_t format; // always 16 159 uint16_t enabled; // actually bool 160 audio_channel_mask_t channelMask; 161 162 // actual buffer provider used by the track hooks, see DownmixerBufferProvider below 163 // for how the Track buffer provider is wrapped by another one when dowmixing is required 164 AudioBufferProvider* bufferProvider; 165 166 // 16-byte boundary 167 168 mutable AudioBufferProvider::Buffer buffer; // 8 bytes 169 170 hook_t hook; 171 const void* in; // current location in buffer 172 173 // 16-byte boundary 174 175 AudioResampler* resampler; 176 uint32_t sampleRate; 177 int32_t* mainBuffer; 178 int32_t* auxBuffer; 179 180 // 16-byte boundary 181 182 uint64_t localTimeFreq; 183 184 DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes 185 186 int32_t padding; 187 188 // 16-byte boundary 189 190 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); 191 bool doesResample() const { return resampler != NULL; } 192 void resetResampler() { if (resampler != NULL) resampler->reset(); } 193 void adjustVolumeRamp(bool aux); 194 size_t getUnreleasedFrames() const { return resampler != NULL ? 195 resampler->getUnreleasedFrames() : 0; }; 196 }; 197 198 // pad to 32-bytes to fill cache line 199 struct state_t { 200 uint32_t enabledTracks; 201 uint32_t needsChanged; 202 size_t frameCount; 203 void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL 204 int32_t *outputTemp; 205 int32_t *resampleTemp; 206 int32_t reserved[2]; 207 // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS 208 track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32))); 209 }; 210 211 // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect 212 class DownmixerBufferProvider : public AudioBufferProvider { 213 public: 214 virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); 215 virtual void releaseBuffer(Buffer* buffer); 216 DownmixerBufferProvider(); 217 virtual ~DownmixerBufferProvider(); 218 219 AudioBufferProvider* mTrackBufferProvider; 220 effect_handle_t mDownmixHandle; 221 effect_config_t mDownmixConfig; 222 }; 223 224 // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. 225 uint32_t mTrackNames; 226 227 // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, 228 // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS 229 const uint32_t mConfiguredNames; 230 231 const uint32_t mSampleRate; 232 233 state_t mState __attribute__((aligned(32))); 234 235 // effect descriptor for the downmixer used by the mixer 236 static effect_descriptor_t dwnmFxDesc; 237 // indicates whether a downmix effect has been found and is usable by this mixer 238 static bool isMultichannelCapable; 239 240 void invalidateState(uint32_t mask); 241 static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); 242 243 static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 244 static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 245 static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 246 static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 247 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 248 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 249 250 static void process__validate(state_t* state, int64_t pts); 251 static void process__nop(state_t* state, int64_t pts); 252 static void process__genericNoResampling(state_t* state, int64_t pts); 253 static void process__genericResampling(state_t* state, int64_t pts); 254 static void process__OneTrack16BitsStereoNoResampling(state_t* state, 255 int64_t pts); 256#if 0 257 static void process__TwoTracks16BitsStereoNoResampling(state_t* state, 258 int64_t pts); 259#endif 260 261 static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, 262 int outputFrameIndex); 263}; 264 265// ---------------------------------------------------------------------------- 266}; // namespace android 267 268#endif // ANDROID_AUDIO_MIXER_H 269