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 <media/AudioBufferProvider.h> 25 #include <media/AudioResampler.h> 26 #include <media/AudioResamplerPublic.h> 27 #include <media/BufferProviders.h> 28 #include <media/nbaio/NBLog.h> 29 #include <system/audio.h> 30 #include <utils/Compat.h> 31 #include <utils/threads.h> 32 33 // FIXME This is actually unity gain, which might not be max in future, expressed in U.12 34 #define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT 35 36 namespace android { 37 38 // ---------------------------------------------------------------------------- 39 40 class AudioMixer 41 { 42 public: 43 AudioMixer(size_t frameCount, uint32_t sampleRate, 44 uint32_t maxNumTracks = MAX_NUM_TRACKS); 45 46 /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed 47 48 49 // This mixer has a hard-coded upper limit of 32 active track inputs. 50 // Adding support for > 32 tracks would require more than simply changing this value. 51 static const uint32_t MAX_NUM_TRACKS = 32; 52 // maximum number of channels supported by the mixer 53 54 // This mixer has a hard-coded upper limit of 8 channels for output. 55 static const uint32_t MAX_NUM_CHANNELS = 8; 56 static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only 57 // maximum number of channels supported for the content 58 static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX; 59 60 static const uint16_t UNITY_GAIN_INT = 0x1000; 61 static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f; 62 63 enum { // names 64 65 // track names (MAX_NUM_TRACKS units) 66 TRACK0 = 0x1000, 67 68 // 0x2000 is unused 69 70 // setParameter targets 71 TRACK = 0x3000, 72 RESAMPLE = 0x3001, 73 RAMP_VOLUME = 0x3002, // ramp to new volume 74 VOLUME = 0x3003, // don't ramp 75 TIMESTRETCH = 0x3004, 76 77 // set Parameter names 78 // for target TRACK 79 CHANNEL_MASK = 0x4000, 80 FORMAT = 0x4001, 81 MAIN_BUFFER = 0x4002, 82 AUX_BUFFER = 0x4003, 83 DOWNMIX_TYPE = 0X4004, 84 MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 85 MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output 86 // for target RESAMPLE 87 SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name; 88 // parameter 'value' is the new sample rate in Hz. 89 // Only creates a sample rate converter the first time that 90 // the track sample rate is different from the mix sample rate. 91 // If the new sample rate is the same as the mix sample rate, 92 // and a sample rate converter already exists, 93 // then the sample rate converter remains present but is a no-op. 94 RESET = 0x4101, // Reset sample rate converter without changing sample rate. 95 // This clears out the resampler's input buffer. 96 REMOVE = 0x4102, // Remove the sample rate converter on this track name; 97 // the track is restored to the mix sample rate. 98 // for target RAMP_VOLUME and VOLUME (8 channels max) 99 // FIXME use float for these 3 to improve the dynamic range 100 VOLUME0 = 0x4200, 101 VOLUME1 = 0x4201, 102 AUXLEVEL = 0x4210, 103 // for target TIMESTRETCH 104 PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name; 105 // parameter 'value' is a pointer to the new playback rate. 106 }; 107 108 109 // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS 110 111 // Allocate a track name. Returns new track name if successful, -1 on failure. 112 // The failure could be because of an invalid channelMask or format, or that 113 // the track capacity of the mixer is exceeded. 114 int getTrackName(audio_channel_mask_t channelMask, 115 audio_format_t format, int sessionId); 116 117 // Free an allocated track by name 118 void deleteTrackName(int name); 119 120 // Enable or disable an allocated track by name 121 void enable(int name); 122 void disable(int name); 123 124 void setParameter(int name, int target, int param, void *value); 125 126 void setBufferProvider(int name, AudioBufferProvider* bufferProvider); 127 void process(); 128 129 uint32_t trackNames() const { return mTrackNames; } 130 131 size_t getUnreleasedFrames(int name) const; 132 133 static inline bool isValidPcmTrackFormat(audio_format_t format) { 134 switch (format) { 135 case AUDIO_FORMAT_PCM_8_BIT: 136 case AUDIO_FORMAT_PCM_16_BIT: 137 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 138 case AUDIO_FORMAT_PCM_32_BIT: 139 case AUDIO_FORMAT_PCM_FLOAT: 140 return true; 141 default: 142 return false; 143 } 144 } 145 146 private: 147 148 enum { 149 // FIXME this representation permits up to 8 channels 150 NEEDS_CHANNEL_COUNT__MASK = 0x00000007, 151 }; 152 153 enum { 154 NEEDS_CHANNEL_1 = 0x00000000, // mono 155 NEEDS_CHANNEL_2 = 0x00000001, // stereo 156 157 // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT 158 159 NEEDS_MUTE = 0x00000100, 160 NEEDS_RESAMPLE = 0x00001000, 161 NEEDS_AUX = 0x00010000, 162 }; 163 164 struct state_t; 165 struct track_t; 166 167 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, 168 int32_t* aux); 169 static const int BLOCKSIZE = 16; // 4 cache lines 170 171 struct track_t { 172 uint32_t needs; 173 174 // TODO: Eventually remove legacy integer volume settings 175 union { 176 int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero) 177 int32_t volumeRL; 178 }; 179 180 int32_t prevVolume[MAX_NUM_VOLUMES]; 181 182 // 16-byte boundary 183 184 int32_t volumeInc[MAX_NUM_VOLUMES]; 185 int32_t auxInc; 186 int32_t prevAuxLevel; 187 188 // 16-byte boundary 189 190 int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance 191 uint16_t frameCount; 192 193 uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) 194 uint8_t unused_padding; // formerly format, was always 16 195 uint16_t enabled; // actually bool 196 audio_channel_mask_t channelMask; 197 198 // actual buffer provider used by the track hooks, see DownmixerBufferProvider below 199 // for how the Track buffer provider is wrapped by another one when dowmixing is required 200 AudioBufferProvider* bufferProvider; 201 202 // 16-byte boundary 203 204 mutable AudioBufferProvider::Buffer buffer; // 8 bytes 205 206 hook_t hook; 207 const void* in; // current location in buffer 208 209 // 16-byte boundary 210 211 AudioResampler* resampler; 212 uint32_t sampleRate; 213 int32_t* mainBuffer; 214 int32_t* auxBuffer; 215 216 // 16-byte boundary 217 218 /* Buffer providers are constructed to translate the track input data as needed. 219 * 220 * TODO: perhaps make a single PlaybackConverterProvider class to move 221 * all pre-mixer track buffer conversions outside the AudioMixer class. 222 * 223 * 1) mInputBufferProvider: The AudioTrack buffer provider. 224 * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to 225 * match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer 226 * requires reformat. For example, it may convert floating point input to 227 * PCM_16_bit if that's required by the downmixer. 228 * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match 229 * the number of channels required by the mixer sink. 230 * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from 231 * the downmixer requirements to the mixer engine input requirements. 232 * 5) mTimestretchBufferProvider: Adds timestretching for playback rate 233 */ 234 AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider. 235 PassthruBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting. 236 PassthruBufferProvider* downmixerBufferProvider; // wrapper for channel conversion. 237 PassthruBufferProvider* mPostDownmixReformatBufferProvider; 238 PassthruBufferProvider* mTimestretchBufferProvider; 239 240 int32_t sessionId; 241 242 audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 243 audio_format_t mFormat; // input track format 244 audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 245 // each track must be converted to this format. 246 audio_format_t mDownmixRequiresFormat; // required downmixer format 247 // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary 248 // AUDIO_FORMAT_INVALID if no required format 249 250 float mVolume[MAX_NUM_VOLUMES]; // floating point set volume 251 float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume 252 float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment 253 254 float mAuxLevel; // floating point set aux level 255 float mPrevAuxLevel; // floating point prev aux level 256 float mAuxInc; // floating point aux increment 257 258 audio_channel_mask_t mMixerChannelMask; 259 uint32_t mMixerChannelCount; 260 261 AudioPlaybackRate mPlaybackRate; 262 263 bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; } 264 bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate); 265 bool doesResample() const { return resampler != NULL; } 266 void resetResampler() { if (resampler != NULL) resampler->reset(); } 267 void adjustVolumeRamp(bool aux, bool useFloat = false); 268 size_t getUnreleasedFrames() const { return resampler != NULL ? 269 resampler->getUnreleasedFrames() : 0; }; 270 271 status_t prepareForDownmix(); 272 void unprepareForDownmix(); 273 status_t prepareForReformat(); 274 void unprepareForReformat(); 275 bool setPlaybackRate(const AudioPlaybackRate &playbackRate); 276 void reconfigureBufferProviders(); 277 }; 278 279 typedef void (*process_hook_t)(state_t* state); 280 281 // pad to 32-bytes to fill cache line 282 struct state_t { 283 uint32_t enabledTracks; 284 uint32_t needsChanged; 285 size_t frameCount; 286 process_hook_t hook; // one of process__*, never NULL 287 int32_t *outputTemp; 288 int32_t *resampleTemp; 289 NBLog::Writer* mNBLogWriter; // associated NBLog::Writer or &mDummyLog 290 int32_t reserved[1]; 291 // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS 292 track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32))); 293 }; 294 295 // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. 296 uint32_t mTrackNames; 297 298 // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, 299 // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS 300 const uint32_t mConfiguredNames; 301 302 const uint32_t mSampleRate; 303 304 NBLog::Writer mDummyLogWriter; 305 public: 306 // Called by FastMixer to inform AudioMixer of it's associated NBLog::Writer. 307 // FIXME It would be safer to use TLS for this, so we don't accidentally use wrong one. 308 void setNBLogWriter(NBLog::Writer* log); 309 private: 310 state_t mState __attribute__((aligned(32))); 311 312 // Call after changing either the enabled status of a track, or parameters of an enabled track. 313 // OK to call more often than that, but unnecessary. 314 void invalidateState(uint32_t mask); 315 316 bool setChannelMasks(int name, 317 audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask); 318 319 static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 320 int32_t* aux); 321 static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 322 static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 323 int32_t* aux); 324 static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 325 int32_t* aux); 326 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 327 int32_t* aux); 328 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 329 int32_t* aux); 330 331 static void process__validate(state_t* state); 332 static void process__nop(state_t* state); 333 static void process__genericNoResampling(state_t* state); 334 static void process__genericResampling(state_t* state); 335 static void process__OneTrack16BitsStereoNoResampling(state_t* state); 336 337 static pthread_once_t sOnceControl; 338 static void sInitRoutine(); 339 340 /* multi-format volume mixing function (calls template functions 341 * in AudioMixerOps.h). The template parameters are as follows: 342 * 343 * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) 344 * USEFLOATVOL (set to true if float volume is used) 345 * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards) 346 * TO: int32_t (Q4.27) or float 347 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float 348 * TA: int32_t (Q4.27) 349 */ 350 template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL, 351 typename TO, typename TI, typename TA> 352 static void volumeMix(TO *out, size_t outFrames, 353 const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t); 354 355 // multi-format process hooks 356 template <int MIXTYPE, typename TO, typename TI, typename TA> 357 static void process_NoResampleOneTrack(state_t* state); 358 359 // multi-format track hooks 360 template <int MIXTYPE, typename TO, typename TI, typename TA> 361 static void track__Resample(track_t* t, TO* out, size_t frameCount, 362 TO* temp __unused, TA* aux); 363 template <int MIXTYPE, typename TO, typename TI, typename TA> 364 static void track__NoResample(track_t* t, TO* out, size_t frameCount, 365 TO* temp __unused, TA* aux); 366 367 static void convertMixerFormat(void *out, audio_format_t mixerOutFormat, 368 void *in, audio_format_t mixerInFormat, size_t sampleCount); 369 370 // hook types 371 enum { 372 PROCESSTYPE_NORESAMPLEONETRACK, 373 }; 374 enum { 375 TRACKTYPE_NOP, 376 TRACKTYPE_RESAMPLE, 377 TRACKTYPE_NORESAMPLE, 378 TRACKTYPE_NORESAMPLEMONO, 379 }; 380 381 // functions for determining the proper process and track hooks. 382 static process_hook_t getProcessHook(int processType, uint32_t channelCount, 383 audio_format_t mixerInFormat, audio_format_t mixerOutFormat); 384 static hook_t getTrackHook(int trackType, uint32_t channelCount, 385 audio_format_t mixerInFormat, audio_format_t mixerOutFormat); 386 }; 387 388 // ---------------------------------------------------------------------------- 389 } // namespace android 390 391 #endif // ANDROID_AUDIO_MIXER_H 392
