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 <hardware/audio_effect.h> 25 #include <media/AudioBufferProvider.h> 26 #include <media/AudioResamplerPublic.h> 27 #include <media/nbaio/NBLog.h> 28 #include <system/audio.h> 29 #include <utils/Compat.h> 30 #include <utils/threads.h> 31 32 #include "AudioResampler.h" 33 #include "BufferProviders.h" 34 35 // FIXME This is actually unity gain, which might not be max in future, expressed in U.12 36 #define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT 37 38 namespace android { 39 40 // ---------------------------------------------------------------------------- 41 42 class AudioMixer 43 { 44 public: 45 AudioMixer(size_t frameCount, uint32_t sampleRate, 46 uint32_t maxNumTracks = MAX_NUM_TRACKS); 47 48 /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed 49 50 51 // This mixer has a hard-coded upper limit of 32 active track inputs. 52 // Adding support for > 32 tracks would require more than simply changing this value. 53 static const uint32_t MAX_NUM_TRACKS = 32; 54 // maximum number of channels supported by the mixer 55 56 // This mixer has a hard-coded upper limit of 8 channels for output. 57 static const uint32_t MAX_NUM_CHANNELS = 8; 58 static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only 59 // maximum number of channels supported for the content 60 static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX; 61 62 static const uint16_t UNITY_GAIN_INT = 0x1000; 63 static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f; 64 65 enum { // names 66 67 // track names (MAX_NUM_TRACKS units) 68 TRACK0 = 0x1000, 69 70 // 0x2000 is unused 71 72 // setParameter targets 73 TRACK = 0x3000, 74 RESAMPLE = 0x3001, 75 RAMP_VOLUME = 0x3002, // ramp to new volume 76 VOLUME = 0x3003, // don't ramp 77 TIMESTRETCH = 0x3004, 78 79 // set Parameter names 80 // for target TRACK 81 CHANNEL_MASK = 0x4000, 82 FORMAT = 0x4001, 83 MAIN_BUFFER = 0x4002, 84 AUX_BUFFER = 0x4003, 85 DOWNMIX_TYPE = 0X4004, 86 MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 87 MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output 88 // for target RESAMPLE 89 SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name; 90 // parameter 'value' is the new sample rate in Hz. 91 // Only creates a sample rate converter the first time that 92 // the track sample rate is different from the mix sample rate. 93 // If the new sample rate is the same as the mix sample rate, 94 // and a sample rate converter already exists, 95 // then the sample rate converter remains present but is a no-op. 96 RESET = 0x4101, // Reset sample rate converter without changing sample rate. 97 // This clears out the resampler's input buffer. 98 REMOVE = 0x4102, // Remove the sample rate converter on this track name; 99 // the track is restored to the mix sample rate. 100 // for target RAMP_VOLUME and VOLUME (8 channels max) 101 // FIXME use float for these 3 to improve the dynamic range 102 VOLUME0 = 0x4200, 103 VOLUME1 = 0x4201, 104 AUXLEVEL = 0x4210, 105 // for target TIMESTRETCH 106 PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name; 107 // parameter 'value' is a pointer to the new playback rate. 108 }; 109 110 111 // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS 112 113 // Allocate a track name. Returns new track name if successful, -1 on failure. 114 // The failure could be because of an invalid channelMask or format, or that 115 // the track capacity of the mixer is exceeded. 116 int getTrackName(audio_channel_mask_t channelMask, 117 audio_format_t format, int sessionId); 118 119 // Free an allocated track by name 120 void deleteTrackName(int name); 121 122 // Enable or disable an allocated track by name 123 void enable(int name); 124 void disable(int name); 125 126 void setParameter(int name, int target, int param, void *value); 127 128 void setBufferProvider(int name, AudioBufferProvider* bufferProvider); 129 void process(int64_t pts); 130 trackNames()131 uint32_t trackNames() const { return mTrackNames; } 132 133 size_t getUnreleasedFrames(int name) const; 134 isValidPcmTrackFormat(audio_format_t format)135 static inline bool isValidPcmTrackFormat(audio_format_t format) { 136 switch (format) { 137 case AUDIO_FORMAT_PCM_8_BIT: 138 case AUDIO_FORMAT_PCM_16_BIT: 139 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 140 case AUDIO_FORMAT_PCM_32_BIT: 141 case AUDIO_FORMAT_PCM_FLOAT: 142 return true; 143 default: 144 return false; 145 } 146 } 147 148 private: 149 150 enum { 151 // FIXME this representation permits up to 8 channels 152 NEEDS_CHANNEL_COUNT__MASK = 0x00000007, 153 }; 154 155 enum { 156 NEEDS_CHANNEL_1 = 0x00000000, // mono 157 NEEDS_CHANNEL_2 = 0x00000001, // stereo 158 159 // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT 160 161 NEEDS_MUTE = 0x00000100, 162 NEEDS_RESAMPLE = 0x00001000, 163 NEEDS_AUX = 0x00010000, 164 }; 165 166 struct state_t; 167 struct track_t; 168 169 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, 170 int32_t* aux); 171 static const int BLOCKSIZE = 16; // 4 cache lines 172 173 struct track_t { 174 uint32_t needs; 175 176 // TODO: Eventually remove legacy integer volume settings 177 union { 178 int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero) 179 int32_t volumeRL; 180 }; 181 182 int32_t prevVolume[MAX_NUM_VOLUMES]; 183 184 // 16-byte boundary 185 186 int32_t volumeInc[MAX_NUM_VOLUMES]; 187 int32_t auxInc; 188 int32_t prevAuxLevel; 189 190 // 16-byte boundary 191 192 int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance 193 uint16_t frameCount; 194 195 uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) 196 uint8_t unused_padding; // formerly format, was always 16 197 uint16_t enabled; // actually bool 198 audio_channel_mask_t channelMask; 199 200 // actual buffer provider used by the track hooks, see DownmixerBufferProvider below 201 // for how the Track buffer provider is wrapped by another one when dowmixing is required 202 AudioBufferProvider* bufferProvider; 203 204 // 16-byte boundary 205 206 mutable AudioBufferProvider::Buffer buffer; // 8 bytes 207 208 hook_t hook; 209 const void* in; // current location in buffer 210 211 // 16-byte boundary 212 213 AudioResampler* resampler; 214 uint32_t sampleRate; 215 int32_t* mainBuffer; 216 int32_t* auxBuffer; 217 218 // 16-byte boundary 219 220 /* Buffer providers are constructed to translate the track input data as needed. 221 * 222 * TODO: perhaps make a single PlaybackConverterProvider class to move 223 * all pre-mixer track buffer conversions outside the AudioMixer class. 224 * 225 * 1) mInputBufferProvider: The AudioTrack buffer provider. 226 * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to 227 * match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer 228 * requires reformat. For example, it may convert floating point input to 229 * PCM_16_bit if that's required by the downmixer. 230 * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match 231 * the number of channels required by the mixer sink. 232 * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from 233 * the downmixer requirements to the mixer engine input requirements. 234 * 5) mTimestretchBufferProvider: Adds timestretching for playback rate 235 */ 236 AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider. 237 PassthruBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting. 238 PassthruBufferProvider* downmixerBufferProvider; // wrapper for channel conversion. 239 PassthruBufferProvider* mPostDownmixReformatBufferProvider; 240 PassthruBufferProvider* mTimestretchBufferProvider; 241 242 int32_t sessionId; 243 244 audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 245 audio_format_t mFormat; // input track format 246 audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 247 // each track must be converted to this format. 248 audio_format_t mDownmixRequiresFormat; // required downmixer format 249 // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary 250 // AUDIO_FORMAT_INVALID if no required format 251 252 float mVolume[MAX_NUM_VOLUMES]; // floating point set volume 253 float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume 254 float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment 255 256 float mAuxLevel; // floating point set aux level 257 float mPrevAuxLevel; // floating point prev aux level 258 float mAuxInc; // floating point aux increment 259 260 audio_channel_mask_t mMixerChannelMask; 261 uint32_t mMixerChannelCount; 262 263 AudioPlaybackRate mPlaybackRate; 264 needsRamptrack_t265 bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; } 266 bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate); doesResampletrack_t267 bool doesResample() const { return resampler != NULL; } resetResamplertrack_t268 void resetResampler() { if (resampler != NULL) resampler->reset(); } 269 void adjustVolumeRamp(bool aux, bool useFloat = false); getUnreleasedFramestrack_t270 size_t getUnreleasedFrames() const { return resampler != NULL ? 271 resampler->getUnreleasedFrames() : 0; }; 272 273 status_t prepareForDownmix(); 274 void unprepareForDownmix(); 275 status_t prepareForReformat(); 276 void unprepareForReformat(); 277 bool setPlaybackRate(const AudioPlaybackRate &playbackRate); 278 void reconfigureBufferProviders(); 279 }; 280 281 typedef void (*process_hook_t)(state_t* state, int64_t pts); 282 283 // pad to 32-bytes to fill cache line 284 struct state_t { 285 uint32_t enabledTracks; 286 uint32_t needsChanged; 287 size_t frameCount; 288 process_hook_t hook; // one of process__*, never NULL 289 int32_t *outputTemp; 290 int32_t *resampleTemp; 291 NBLog::Writer* mLog; 292 int32_t reserved[1]; 293 // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS 294 track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32))); 295 }; 296 297 // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. 298 uint32_t mTrackNames; 299 300 // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, 301 // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS 302 const uint32_t mConfiguredNames; 303 304 const uint32_t mSampleRate; 305 306 NBLog::Writer mDummyLog; 307 public: 308 void setLog(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, int64_t pts); 332 static void process__nop(state_t* state, int64_t pts); 333 static void process__genericNoResampling(state_t* state, int64_t pts); 334 static void process__genericResampling(state_t* state, int64_t pts); 335 static void process__OneTrack16BitsStereoNoResampling(state_t* state, 336 int64_t pts); 337 338 static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, 339 int outputFrameIndex); 340 341 static uint64_t sLocalTimeFreq; 342 static pthread_once_t sOnceControl; 343 static void sInitRoutine(); 344 345 /* multi-format volume mixing function (calls template functions 346 * in AudioMixerOps.h). The template parameters are as follows: 347 * 348 * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) 349 * USEFLOATVOL (set to true if float volume is used) 350 * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards) 351 * TO: int32_t (Q4.27) or float 352 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float 353 * TA: int32_t (Q4.27) 354 */ 355 template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL, 356 typename TO, typename TI, typename TA> 357 static void volumeMix(TO *out, size_t outFrames, 358 const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t); 359 360 // multi-format process hooks 361 template <int MIXTYPE, typename TO, typename TI, typename TA> 362 static void process_NoResampleOneTrack(state_t* state, int64_t pts); 363 364 // multi-format track hooks 365 template <int MIXTYPE, typename TO, typename TI, typename TA> 366 static void track__Resample(track_t* t, TO* out, size_t frameCount, 367 TO* temp __unused, TA* aux); 368 template <int MIXTYPE, typename TO, typename TI, typename TA> 369 static void track__NoResample(track_t* t, TO* out, size_t frameCount, 370 TO* temp __unused, TA* aux); 371 372 static void convertMixerFormat(void *out, audio_format_t mixerOutFormat, 373 void *in, audio_format_t mixerInFormat, size_t sampleCount); 374 375 // hook types 376 enum { 377 PROCESSTYPE_NORESAMPLEONETRACK, 378 }; 379 enum { 380 TRACKTYPE_NOP, 381 TRACKTYPE_RESAMPLE, 382 TRACKTYPE_NORESAMPLE, 383 TRACKTYPE_NORESAMPLEMONO, 384 }; 385 386 // functions for determining the proper process and track hooks. 387 static process_hook_t getProcessHook(int processType, uint32_t channelCount, 388 audio_format_t mixerInFormat, audio_format_t mixerOutFormat); 389 static hook_t getTrackHook(int trackType, uint32_t channelCount, 390 audio_format_t mixerInFormat, audio_format_t mixerOutFormat); 391 }; 392 393 // ---------------------------------------------------------------------------- 394 } // namespace android 395 396 #endif // ANDROID_AUDIO_MIXER_H 397