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 "AudioBufferProvider.h" 27 #include "AudioResampler.h" 28 29 #include <audio_effects/effect_downmix.h> 30 #include <system/audio.h> 31 32 namespace android { 33 34 // ---------------------------------------------------------------------------- 35 36 class AudioMixer 37 { 38 public: 39 AudioMixer(size_t frameCount, uint32_t sampleRate, 40 uint32_t maxNumTracks = MAX_NUM_TRACKS); 41 42 /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed 43 44 static const uint32_t MAX_NUM_TRACKS = 32; 45 // maximum number of channels supported by the mixer 46 static const uint32_t MAX_NUM_CHANNELS = 2; 47 // maximum number of channels supported for the content 48 static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8; 49 50 static const uint16_t UNITY_GAIN = 0x1000; 51 52 enum { // names 53 54 // track names (MAX_NUM_TRACKS units) 55 TRACK0 = 0x1000, 56 57 // 0x2000 is unused 58 59 // setParameter targets 60 TRACK = 0x3000, 61 RESAMPLE = 0x3001, 62 RAMP_VOLUME = 0x3002, // ramp to new volume 63 VOLUME = 0x3003, // don't ramp 64 65 // set Parameter names 66 // for target TRACK 67 CHANNEL_MASK = 0x4000, 68 FORMAT = 0x4001, 69 MAIN_BUFFER = 0x4002, 70 AUX_BUFFER = 0x4003, 71 DOWNMIX_TYPE = 0X4004, 72 // for target RESAMPLE 73 SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name; 74 // parameter 'value' is the new sample rate in Hz. 75 // Only creates a sample rate converter the first time that 76 // the track sample rate is different from the mix sample rate. 77 // If the new sample rate is the same as the mix sample rate, 78 // and a sample rate converter already exists, 79 // then the sample rate converter remains present but is a no-op. 80 RESET = 0x4101, // Reset sample rate converter without changing sample rate. 81 // This clears out the resampler's input buffer. 82 REMOVE = 0x4102, // Remove the sample rate converter on this track name; 83 // the track is restored to the mix sample rate. 84 // for target RAMP_VOLUME and VOLUME (8 channels max) 85 VOLUME0 = 0x4200, 86 VOLUME1 = 0x4201, 87 AUXLEVEL = 0x4210, 88 }; 89 90 91 // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS 92 93 // Allocate a track name. Returns new track name if successful, -1 on failure. 94 int getTrackName(audio_channel_mask_t channelMask); 95 96 // Free an allocated track by name 97 void deleteTrackName(int name); 98 99 // Enable or disable an allocated track by name 100 void enable(int name); 101 void disable(int name); 102 103 void setParameter(int name, int target, int param, void *value); 104 105 void setBufferProvider(int name, AudioBufferProvider* bufferProvider); 106 void process(int64_t pts); 107 108 uint32_t trackNames() const { return mTrackNames; } 109 110 size_t getUnreleasedFrames(int name) const; 111 112 private: 113 114 enum { 115 NEEDS_CHANNEL_COUNT__MASK = 0x00000007, 116 NEEDS_FORMAT__MASK = 0x000000F0, 117 NEEDS_MUTE__MASK = 0x00000100, 118 NEEDS_RESAMPLE__MASK = 0x00001000, 119 NEEDS_AUX__MASK = 0x00010000, 120 }; 121 122 enum { 123 NEEDS_CHANNEL_1 = 0x00000000, 124 NEEDS_CHANNEL_2 = 0x00000001, 125 126 NEEDS_FORMAT_16 = 0x00000010, 127 128 NEEDS_MUTE_DISABLED = 0x00000000, 129 NEEDS_MUTE_ENABLED = 0x00000100, 130 131 NEEDS_RESAMPLE_DISABLED = 0x00000000, 132 NEEDS_RESAMPLE_ENABLED = 0x00001000, 133 134 NEEDS_AUX_DISABLED = 0x00000000, 135 NEEDS_AUX_ENABLED = 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, int32_t* aux); 143 static const int BLOCKSIZE = 16; // 4 cache lines 144 145 struct track_t { 146 uint32_t needs; 147 148 union { 149 int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point 150 int32_t volumeRL; 151 }; 152 153 int32_t prevVolume[MAX_NUM_CHANNELS]; 154 155 // 16-byte boundary 156 157 int32_t volumeInc[MAX_NUM_CHANNELS]; 158 int32_t auxInc; 159 int32_t prevAuxLevel; 160 161 // 16-byte boundary 162 163 int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance 164 uint16_t frameCount; 165 166 uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) 167 uint8_t format; // always 16 168 uint16_t enabled; // actually bool 169 audio_channel_mask_t channelMask; 170 171 // actual buffer provider used by the track hooks, see DownmixerBufferProvider below 172 // for how the Track buffer provider is wrapped by another one when dowmixing is required 173 AudioBufferProvider* bufferProvider; 174 175 // 16-byte boundary 176 177 mutable AudioBufferProvider::Buffer buffer; // 8 bytes 178 179 hook_t hook; 180 const void* in; // current location in buffer 181 182 // 16-byte boundary 183 184 AudioResampler* resampler; 185 uint32_t sampleRate; 186 int32_t* mainBuffer; 187 int32_t* auxBuffer; 188 189 // 16-byte boundary 190 191 uint64_t localTimeFreq; 192 193 DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes 194 195 int32_t padding; 196 197 // 16-byte boundary 198 199 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); 200 bool doesResample() const { return resampler != NULL; } 201 void resetResampler() { if (resampler != NULL) resampler->reset(); } 202 void adjustVolumeRamp(bool aux); 203 size_t getUnreleasedFrames() const { return resampler != NULL ? 204 resampler->getUnreleasedFrames() : 0; }; 205 }; 206 207 // pad to 32-bytes to fill cache line 208 struct state_t { 209 uint32_t enabledTracks; 210 uint32_t needsChanged; 211 size_t frameCount; 212 void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL 213 int32_t *outputTemp; 214 int32_t *resampleTemp; 215 int32_t reserved[2]; 216 // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS 217 track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32))); 218 }; 219 220 // AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect 221 class DownmixerBufferProvider : public AudioBufferProvider { 222 public: 223 virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); 224 virtual void releaseBuffer(Buffer* buffer); 225 DownmixerBufferProvider(); 226 virtual ~DownmixerBufferProvider(); 227 228 AudioBufferProvider* mTrackBufferProvider; 229 effect_handle_t mDownmixHandle; 230 effect_config_t mDownmixConfig; 231 }; 232 233 // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. 234 uint32_t mTrackNames; 235 236 // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, 237 // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS 238 const uint32_t mConfiguredNames; 239 240 const uint32_t mSampleRate; 241 242 state_t mState __attribute__((aligned(32))); 243 244 // effect descriptor for the downmixer used by the mixer 245 static effect_descriptor_t dwnmFxDesc; 246 // indicates whether a downmix effect has been found and is usable by this mixer 247 static bool isMultichannelCapable; 248 249 // Call after changing either the enabled status of a track, or parameters of an enabled track. 250 // OK to call more often than that, but unnecessary. 251 void invalidateState(uint32_t mask); 252 253 static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask); 254 static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); 255 static void unprepareTrackForDownmix(track_t* pTrack, int trackName); 256 257 static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 258 static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 259 static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 260 static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); 261 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 262 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 263 264 static void process__validate(state_t* state, int64_t pts); 265 static void process__nop(state_t* state, int64_t pts); 266 static void process__genericNoResampling(state_t* state, int64_t pts); 267 static void process__genericResampling(state_t* state, int64_t pts); 268 static void process__OneTrack16BitsStereoNoResampling(state_t* state, 269 int64_t pts); 270 #if 0 271 static void process__TwoTracks16BitsStereoNoResampling(state_t* state, 272 int64_t pts); 273 #endif 274 275 static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, 276 int outputFrameIndex); 277 }; 278 279 // ---------------------------------------------------------------------------- 280 }; // namespace android 281 282 #endif // ANDROID_AUDIO_MIXER_H 283
