AudioMixer.h revision e8a1ced4da17dc6c07803dc2af8060f62a8389c1
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 33namespace android { 34 35// ---------------------------------------------------------------------------- 36 37class AudioMixer 38{ 39public: 40 AudioMixer(size_t frameCount, uint32_t sampleRate, 41 uint32_t maxNumTracks = MAX_NUM_TRACKS); 42 43 /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed 44 45 46 // This mixer has a hard-coded upper limit of 32 active track inputs. 47 // Adding support for > 32 tracks would require more than simply changing this value. 48 static const uint32_t MAX_NUM_TRACKS = 32; 49 // maximum number of channels supported by the mixer 50 51 // This mixer has a hard-coded upper limit of 2 channels for output. 52 // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 53 // Adding support for > 2 channel output would require more than simply changing this value. 54 static const uint32_t MAX_NUM_CHANNELS = 2; 55 // maximum number of channels supported for the content 56 static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8; 57 58 static const uint16_t UNITY_GAIN = 0x1000; 59 60 enum { // names 61 62 // track names (MAX_NUM_TRACKS units) 63 TRACK0 = 0x1000, 64 65 // 0x2000 is unused 66 67 // setParameter targets 68 TRACK = 0x3000, 69 RESAMPLE = 0x3001, 70 RAMP_VOLUME = 0x3002, // ramp to new volume 71 VOLUME = 0x3003, // don't ramp 72 73 // set Parameter names 74 // for target TRACK 75 CHANNEL_MASK = 0x4000, 76 FORMAT = 0x4001, 77 MAIN_BUFFER = 0x4002, 78 AUX_BUFFER = 0x4003, 79 DOWNMIX_TYPE = 0X4004, 80 MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 81 // for target RESAMPLE 82 SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name; 83 // parameter 'value' is the new sample rate in Hz. 84 // Only creates a sample rate converter the first time that 85 // the track sample rate is different from the mix sample rate. 86 // If the new sample rate is the same as the mix sample rate, 87 // and a sample rate converter already exists, 88 // then the sample rate converter remains present but is a no-op. 89 RESET = 0x4101, // Reset sample rate converter without changing sample rate. 90 // This clears out the resampler's input buffer. 91 REMOVE = 0x4102, // Remove the sample rate converter on this track name; 92 // the track is restored to the mix sample rate. 93 // for target RAMP_VOLUME and VOLUME (8 channels max) 94 VOLUME0 = 0x4200, 95 VOLUME1 = 0x4201, 96 AUXLEVEL = 0x4210, 97 }; 98 99 100 // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS 101 102 // Allocate a track name. Returns new track name if successful, -1 on failure. 103 // The failure could be because of an invalid channelMask or format, or that 104 // the track capacity of the mixer is exceeded. 105 int getTrackName(audio_channel_mask_t channelMask, 106 audio_format_t format, int sessionId); 107 108 // Free an allocated track by name 109 void deleteTrackName(int name); 110 111 // Enable or disable an allocated track by name 112 void enable(int name); 113 void disable(int name); 114 115 void setParameter(int name, int target, int param, void *value); 116 117 void setBufferProvider(int name, AudioBufferProvider* bufferProvider); 118 void process(int64_t pts); 119 120 uint32_t trackNames() const { return mTrackNames; } 121 122 size_t getUnreleasedFrames(int name) const; 123 124 static inline bool isValidPcmTrackFormat(audio_format_t format) { 125 return format == AUDIO_FORMAT_PCM_16_BIT; 126 } 127 128private: 129 130 enum { 131 // FIXME this representation permits up to 8 channels 132 NEEDS_CHANNEL_COUNT__MASK = 0x00000007, 133 }; 134 135 enum { 136 NEEDS_CHANNEL_1 = 0x00000000, // mono 137 NEEDS_CHANNEL_2 = 0x00000001, // stereo 138 139 // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT 140 141 NEEDS_MUTE = 0x00000100, 142 NEEDS_RESAMPLE = 0x00001000, 143 NEEDS_AUX = 0x00010000, 144 }; 145 146 struct state_t; 147 struct track_t; 148 class DownmixerBufferProvider; 149 150 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, 151 int32_t* aux); 152 static const int BLOCKSIZE = 16; // 4 cache lines 153 154 struct track_t { 155 uint32_t needs; 156 157 union { 158 int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point 159 int32_t volumeRL; 160 }; 161 162 int32_t prevVolume[MAX_NUM_CHANNELS]; 163 164 // 16-byte boundary 165 166 int32_t volumeInc[MAX_NUM_CHANNELS]; 167 int32_t auxInc; 168 int32_t prevAuxLevel; 169 170 // 16-byte boundary 171 172 int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance 173 uint16_t frameCount; 174 175 uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) 176 uint8_t format; // always 16 177 uint16_t enabled; // actually bool 178 audio_channel_mask_t channelMask; 179 180 // actual buffer provider used by the track hooks, see DownmixerBufferProvider below 181 // for how the Track buffer provider is wrapped by another one when dowmixing is required 182 AudioBufferProvider* bufferProvider; 183 184 // 16-byte boundary 185 186 mutable AudioBufferProvider::Buffer buffer; // 8 bytes 187 188 hook_t hook; 189 const void* in; // current location in buffer 190 191 // 16-byte boundary 192 193 AudioResampler* resampler; 194 uint32_t sampleRate; 195 int32_t* mainBuffer; 196 int32_t* auxBuffer; 197 198 // 16-byte boundary 199 200 DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes 201 202 int32_t sessionId; 203 204 audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT) 205 audio_format_t mFormat; // input track format 206 207 // 16-byte boundary 208 209 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); 210 bool doesResample() const { return resampler != NULL; } 211 void resetResampler() { if (resampler != NULL) resampler->reset(); } 212 void adjustVolumeRamp(bool aux); 213 size_t getUnreleasedFrames() const { return resampler != NULL ? 214 resampler->getUnreleasedFrames() : 0; }; 215 }; 216 217 // pad to 32-bytes to fill cache line 218 struct state_t { 219 uint32_t enabledTracks; 220 uint32_t needsChanged; 221 size_t frameCount; 222 void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL 223 int32_t *outputTemp; 224 int32_t *resampleTemp; 225 NBLog::Writer* mLog; 226 int32_t reserved[1]; 227 // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS 228 track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32))); 229 }; 230 231 // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect 232 class DownmixerBufferProvider : public AudioBufferProvider { 233 public: 234 virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); 235 virtual void releaseBuffer(Buffer* buffer); 236 DownmixerBufferProvider(); 237 virtual ~DownmixerBufferProvider(); 238 239 AudioBufferProvider* mTrackBufferProvider; 240 effect_handle_t mDownmixHandle; 241 effect_config_t mDownmixConfig; 242 }; 243 244 // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. 245 uint32_t mTrackNames; 246 247 // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, 248 // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS 249 const uint32_t mConfiguredNames; 250 251 const uint32_t mSampleRate; 252 253 NBLog::Writer mDummyLog; 254public: 255 void setLog(NBLog::Writer* log); 256private: 257 state_t mState __attribute__((aligned(32))); 258 259 // effect descriptor for the downmixer used by the mixer 260 static effect_descriptor_t sDwnmFxDesc; 261 // indicates whether a downmix effect has been found and is usable by this mixer 262 static bool sIsMultichannelCapable; 263 264 // Call after changing either the enabled status of a track, or parameters of an enabled track. 265 // OK to call more often than that, but unnecessary. 266 void invalidateState(uint32_t mask); 267 268 static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask); 269 static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); 270 static void unprepareTrackForDownmix(track_t* pTrack, int trackName); 271 272 static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 273 int32_t* aux); 274 static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 275 static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 276 int32_t* aux); 277 static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 278 int32_t* aux); 279 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 280 int32_t* aux); 281 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 282 int32_t* aux); 283 284 static void process__validate(state_t* state, int64_t pts); 285 static void process__nop(state_t* state, int64_t pts); 286 static void process__genericNoResampling(state_t* state, int64_t pts); 287 static void process__genericResampling(state_t* state, int64_t pts); 288 static void process__OneTrack16BitsStereoNoResampling(state_t* state, 289 int64_t pts); 290#if 0 291 static void process__TwoTracks16BitsStereoNoResampling(state_t* state, 292 int64_t pts); 293#endif 294 295 static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, 296 int outputFrameIndex); 297 298 static uint64_t sLocalTimeFreq; 299 static pthread_once_t sOnceControl; 300 static void sInitRoutine(); 301}; 302 303// ---------------------------------------------------------------------------- 304}; // namespace android 305 306#endif // ANDROID_AUDIO_MIXER_H 307