AudioMixer.h revision 6be494077f8d7970f3a88129c5d139c5a0c88f6d
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 <media/AudioBufferProvider.h> 27#include "AudioResampler.h" 28 29#include <audio_effects/effect_downmix.h> 30#include <system/audio.h> 31#include <media/nbaio/NBLog.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 36namespace android { 37 38// ---------------------------------------------------------------------------- 39 40class AudioMixer 41{ 42public: 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 2 channels for output. 55 // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 56 // Adding support for > 2 channel output would require more than simply changing this value. 57 static const uint32_t MAX_NUM_CHANNELS = 2; 58 // maximum number of channels supported for the content 59 static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8; 60 61 static const uint16_t UNITY_GAIN_INT = 0x1000; 62 static const float UNITY_GAIN_FLOAT = 1.0f; 63 64 enum { // names 65 66 // track names (MAX_NUM_TRACKS units) 67 TRACK0 = 0x1000, 68 69 // 0x2000 is unused 70 71 // setParameter targets 72 TRACK = 0x3000, 73 RESAMPLE = 0x3001, 74 RAMP_VOLUME = 0x3002, // ramp to new volume 75 VOLUME = 0x3003, // don't ramp 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 // for target RESAMPLE 86 SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name; 87 // parameter 'value' is the new sample rate in Hz. 88 // Only creates a sample rate converter the first time that 89 // the track sample rate is different from the mix sample rate. 90 // If the new sample rate is the same as the mix sample rate, 91 // and a sample rate converter already exists, 92 // then the sample rate converter remains present but is a no-op. 93 RESET = 0x4101, // Reset sample rate converter without changing sample rate. 94 // This clears out the resampler's input buffer. 95 REMOVE = 0x4102, // Remove the sample rate converter on this track name; 96 // the track is restored to the mix sample rate. 97 // for target RAMP_VOLUME and VOLUME (8 channels max) 98 // FIXME use float for these 3 to improve the dynamic range 99 VOLUME0 = 0x4200, 100 VOLUME1 = 0x4201, 101 AUXLEVEL = 0x4210, 102 }; 103 104 105 // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS 106 107 // Allocate a track name. Returns new track name if successful, -1 on failure. 108 // The failure could be because of an invalid channelMask or format, or that 109 // the track capacity of the mixer is exceeded. 110 int getTrackName(audio_channel_mask_t channelMask, 111 audio_format_t format, int sessionId); 112 113 // Free an allocated track by name 114 void deleteTrackName(int name); 115 116 // Enable or disable an allocated track by name 117 void enable(int name); 118 void disable(int name); 119 120 void setParameter(int name, int target, int param, void *value); 121 122 void setBufferProvider(int name, AudioBufferProvider* bufferProvider); 123 void process(int64_t pts); 124 125 uint32_t trackNames() const { return mTrackNames; } 126 127 size_t getUnreleasedFrames(int name) const; 128 129 static inline bool isValidPcmTrackFormat(audio_format_t format) { 130 return format == AUDIO_FORMAT_PCM_16_BIT || 131 format == AUDIO_FORMAT_PCM_24_BIT_PACKED || 132 format == AUDIO_FORMAT_PCM_32_BIT || 133 format == AUDIO_FORMAT_PCM_FLOAT; 134 } 135 136private: 137 138 enum { 139 // FIXME this representation permits up to 8 channels 140 NEEDS_CHANNEL_COUNT__MASK = 0x00000007, 141 }; 142 143 enum { 144 NEEDS_CHANNEL_1 = 0x00000000, // mono 145 NEEDS_CHANNEL_2 = 0x00000001, // stereo 146 147 // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT 148 149 NEEDS_MUTE = 0x00000100, 150 NEEDS_RESAMPLE = 0x00001000, 151 NEEDS_AUX = 0x00010000, 152 }; 153 154 struct state_t; 155 struct track_t; 156 class DownmixerBufferProvider; 157 class ReformatBufferProvider; 158 159 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, 160 int32_t* aux); 161 static const int BLOCKSIZE = 16; // 4 cache lines 162 163 struct track_t { 164 uint32_t needs; 165 166 union { 167 int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point 168 int32_t volumeRL; 169 }; 170 171 int32_t prevVolume[MAX_NUM_CHANNELS]; 172 173 // 16-byte boundary 174 175 int32_t volumeInc[MAX_NUM_CHANNELS]; 176 int32_t auxInc; 177 int32_t prevAuxLevel; 178 179 // 16-byte boundary 180 181 int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance 182 uint16_t frameCount; 183 184 uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) 185 uint8_t unused_padding; // formerly format, was always 16 186 uint16_t enabled; // actually bool 187 audio_channel_mask_t channelMask; 188 189 // actual buffer provider used by the track hooks, see DownmixerBufferProvider below 190 // for how the Track buffer provider is wrapped by another one when dowmixing is required 191 AudioBufferProvider* bufferProvider; 192 193 // 16-byte boundary 194 195 mutable AudioBufferProvider::Buffer buffer; // 8 bytes 196 197 hook_t hook; 198 const void* in; // current location in buffer 199 200 // 16-byte boundary 201 202 AudioResampler* resampler; 203 uint32_t sampleRate; 204 int32_t* mainBuffer; 205 int32_t* auxBuffer; 206 207 // 16-byte boundary 208 AudioBufferProvider* mInputBufferProvider; // 4 bytes 209 ReformatBufferProvider* mReformatBufferProvider; // 4 bytes 210 DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes 211 212 int32_t sessionId; 213 214 // 16-byte boundary 215 audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 216 audio_format_t mFormat; // input track format 217 audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 218 // each track must be converted to this format. 219 220 int32_t mUnused[1]; // alignment padding 221 222 // 16-byte boundary 223 224 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); 225 bool doesResample() const { return resampler != NULL; } 226 void resetResampler() { if (resampler != NULL) resampler->reset(); } 227 void adjustVolumeRamp(bool aux); 228 size_t getUnreleasedFrames() const { return resampler != NULL ? 229 resampler->getUnreleasedFrames() : 0; }; 230 }; 231 232 // pad to 32-bytes to fill cache line 233 struct state_t { 234 uint32_t enabledTracks; 235 uint32_t needsChanged; 236 size_t frameCount; 237 void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL 238 int32_t *outputTemp; 239 int32_t *resampleTemp; 240 NBLog::Writer* mLog; 241 int32_t reserved[1]; 242 // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS 243 track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32))); 244 }; 245 246 // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect 247 class DownmixerBufferProvider : public AudioBufferProvider { 248 public: 249 virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); 250 virtual void releaseBuffer(Buffer* buffer); 251 DownmixerBufferProvider(); 252 virtual ~DownmixerBufferProvider(); 253 254 AudioBufferProvider* mTrackBufferProvider; 255 effect_handle_t mDownmixHandle; 256 effect_config_t mDownmixConfig; 257 }; 258 259 // AudioBufferProvider wrapper that reformats track to acceptable mixer input type 260 class ReformatBufferProvider : public AudioBufferProvider { 261 public: 262 ReformatBufferProvider(int32_t channels, 263 audio_format_t inputFormat, audio_format_t outputFormat); 264 virtual ~ReformatBufferProvider(); 265 266 // overrides AudioBufferProvider methods 267 virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); 268 virtual void releaseBuffer(Buffer* buffer); 269 270 void reset(); 271 inline bool requiresInternalBuffers() { 272 return true; //mInputFrameSize < mOutputFrameSize; 273 } 274 275 AudioBufferProvider* mTrackBufferProvider; 276 int32_t mChannels; 277 audio_format_t mInputFormat; 278 audio_format_t mOutputFormat; 279 size_t mInputFrameSize; 280 size_t mOutputFrameSize; 281 // (only) required for reformatting to a larger size. 282 AudioBufferProvider::Buffer mBuffer; 283 void* mOutputData; 284 size_t mOutputCount; 285 size_t mConsumed; 286 }; 287 288 // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. 289 uint32_t mTrackNames; 290 291 // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, 292 // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS 293 const uint32_t mConfiguredNames; 294 295 const uint32_t mSampleRate; 296 297 NBLog::Writer mDummyLog; 298public: 299 void setLog(NBLog::Writer* log); 300private: 301 state_t mState __attribute__((aligned(32))); 302 303 // effect descriptor for the downmixer used by the mixer 304 static effect_descriptor_t sDwnmFxDesc; 305 // indicates whether a downmix effect has been found and is usable by this mixer 306 static bool sIsMultichannelCapable; 307 308 // Call after changing either the enabled status of a track, or parameters of an enabled track. 309 // OK to call more often than that, but unnecessary. 310 void invalidateState(uint32_t mask); 311 312 static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask); 313 static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); 314 static void unprepareTrackForDownmix(track_t* pTrack, int trackName); 315 static status_t prepareTrackForReformat(track_t* pTrack, int trackNum); 316 static void unprepareTrackForReformat(track_t* pTrack, int trackName); 317 static void reconfigureBufferProviders(track_t* pTrack); 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#if 0 338 static void process__TwoTracks16BitsStereoNoResampling(state_t* state, 339 int64_t pts); 340#endif 341 342 static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, 343 int outputFrameIndex); 344 345 static uint64_t sLocalTimeFreq; 346 static pthread_once_t sOnceControl; 347 static void sInitRoutine(); 348}; 349 350// ---------------------------------------------------------------------------- 351}; // namespace android 352 353#endif // ANDROID_AUDIO_MIXER_H 354