AudioMixer.h revision 01d3acba9de861cb2b718338e787cff3566fc5ec
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 // for target RESAMPLE 81 SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name; 82 // parameter 'value' is the new sample rate in Hz. 83 // Only creates a sample rate converter the first time that 84 // the track sample rate is different from the mix sample rate. 85 // If the new sample rate is the same as the mix sample rate, 86 // and a sample rate converter already exists, 87 // then the sample rate converter remains present but is a no-op. 88 RESET = 0x4101, // Reset sample rate converter without changing sample rate. 89 // This clears out the resampler's input buffer. 90 REMOVE = 0x4102, // Remove the sample rate converter on this track name; 91 // the track is restored to the mix sample rate. 92 // for target RAMP_VOLUME and VOLUME (8 channels max) 93 VOLUME0 = 0x4200, 94 VOLUME1 = 0x4201, 95 AUXLEVEL = 0x4210, 96 }; 97 98 99 // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS 100 101 // Allocate a track name. Returns new track name if successful, -1 on failure. 102 int getTrackName(audio_channel_mask_t channelMask, int sessionId); 103 104 // Free an allocated track by name 105 void deleteTrackName(int name); 106 107 // Enable or disable an allocated track by name 108 void enable(int name); 109 void disable(int name); 110 111 void setParameter(int name, int target, int param, void *value); 112 113 void setBufferProvider(int name, AudioBufferProvider* bufferProvider); 114 void process(int64_t pts); 115 116 uint32_t trackNames() const { return mTrackNames; } 117 118 size_t getUnreleasedFrames(int name) const; 119 120private: 121 122 enum { 123 // FIXME this representation permits up to 8 channels 124 NEEDS_CHANNEL_COUNT__MASK = 0x00000007, 125 }; 126 127 enum { 128 NEEDS_CHANNEL_1 = 0x00000000, // mono 129 NEEDS_CHANNEL_2 = 0x00000001, // stereo 130 131 // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT 132 133 NEEDS_MUTE = 0x00000100, 134 NEEDS_RESAMPLE = 0x00001000, 135 NEEDS_AUX = 0x00010000, 136 }; 137 138 struct state_t; 139 struct track_t; 140 class DownmixerBufferProvider; 141 142 typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, 143 int32_t* aux); 144 static const int BLOCKSIZE = 16; // 4 cache lines 145 146 struct track_t { 147 uint32_t needs; 148 149 union { 150 int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point 151 int32_t volumeRL; 152 }; 153 154 int32_t prevVolume[MAX_NUM_CHANNELS]; 155 156 // 16-byte boundary 157 158 int32_t volumeInc[MAX_NUM_CHANNELS]; 159 int32_t auxInc; 160 int32_t prevAuxLevel; 161 162 // 16-byte boundary 163 164 int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance 165 uint16_t frameCount; 166 167 uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) 168 uint8_t format; // always 16 169 uint16_t enabled; // actually bool 170 audio_channel_mask_t channelMask; 171 172 // actual buffer provider used by the track hooks, see DownmixerBufferProvider below 173 // for how the Track buffer provider is wrapped by another one when dowmixing is required 174 AudioBufferProvider* bufferProvider; 175 176 // 16-byte boundary 177 178 mutable AudioBufferProvider::Buffer buffer; // 8 bytes 179 180 hook_t hook; 181 const void* in; // current location in buffer 182 183 // 16-byte boundary 184 185 AudioResampler* resampler; 186 uint32_t sampleRate; 187 int32_t* mainBuffer; 188 int32_t* auxBuffer; 189 190 // 16-byte boundary 191 192 DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes 193 194 int32_t sessionId; 195 196 int32_t padding[2]; 197 198 // 16-byte boundary 199 200 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); 201 bool doesResample() const { return resampler != NULL; } 202 void resetResampler() { if (resampler != NULL) resampler->reset(); } 203 void adjustVolumeRamp(bool aux); 204 size_t getUnreleasedFrames() const { return resampler != NULL ? 205 resampler->getUnreleasedFrames() : 0; }; 206 }; 207 208 // pad to 32-bytes to fill cache line 209 struct state_t { 210 uint32_t enabledTracks; 211 uint32_t needsChanged; 212 size_t frameCount; 213 void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL 214 int32_t *outputTemp; 215 int32_t *resampleTemp; 216 NBLog::Writer* mLog; 217 int32_t reserved[1]; 218 // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS 219 track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32))); 220 }; 221 222 // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect 223 class DownmixerBufferProvider : public AudioBufferProvider { 224 public: 225 virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); 226 virtual void releaseBuffer(Buffer* buffer); 227 DownmixerBufferProvider(); 228 virtual ~DownmixerBufferProvider(); 229 230 AudioBufferProvider* mTrackBufferProvider; 231 effect_handle_t mDownmixHandle; 232 effect_config_t mDownmixConfig; 233 }; 234 235 // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. 236 uint32_t mTrackNames; 237 238 // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, 239 // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS 240 const uint32_t mConfiguredNames; 241 242 const uint32_t mSampleRate; 243 244 NBLog::Writer mDummyLog; 245public: 246 void setLog(NBLog::Writer* log); 247private: 248 state_t mState __attribute__((aligned(32))); 249 250 // effect descriptor for the downmixer used by the mixer 251 static effect_descriptor_t sDwnmFxDesc; 252 // indicates whether a downmix effect has been found and is usable by this mixer 253 static bool sIsMultichannelCapable; 254 255 // Call after changing either the enabled status of a track, or parameters of an enabled track. 256 // OK to call more often than that, but unnecessary. 257 void invalidateState(uint32_t mask); 258 259 static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask); 260 static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); 261 static void unprepareTrackForDownmix(track_t* pTrack, int trackName); 262 263 static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 264 int32_t* aux); 265 static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 266 static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 267 int32_t* aux); 268 static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, 269 int32_t* aux); 270 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 271 int32_t* aux); 272 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 273 int32_t* aux); 274 275 static void process__validate(state_t* state, int64_t pts); 276 static void process__nop(state_t* state, int64_t pts); 277 static void process__genericNoResampling(state_t* state, int64_t pts); 278 static void process__genericResampling(state_t* state, int64_t pts); 279 static void process__OneTrack16BitsStereoNoResampling(state_t* state, 280 int64_t pts); 281#if 0 282 static void process__TwoTracks16BitsStereoNoResampling(state_t* state, 283 int64_t pts); 284#endif 285 286 static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, 287 int outputFrameIndex); 288 289 static uint64_t sLocalTimeFreq; 290 static pthread_once_t sOnceControl; 291 static void sInitRoutine(); 292}; 293 294// ---------------------------------------------------------------------------- 295}; // namespace android 296 297#endif // ANDROID_AUDIO_MIXER_H 298