| /* //device/include/server/AudioFlinger/AudioMixer.h |
| ** |
| ** Copyright 2007, The Android Open Source Project |
| ** |
| ** Licensed under the Apache License, Version 2.0 (the "License"); |
| ** you may not use this file except in compliance with the License. |
| ** You may obtain a copy of the License at |
| ** |
| ** http://www.apache.org/licenses/LICENSE-2.0 |
| ** |
| ** Unless required by applicable law or agreed to in writing, software |
| ** distributed under the License is distributed on an "AS IS" BASIS, |
| ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| ** See the License for the specific language governing permissions and |
| ** limitations under the License. |
| */ |
| |
| #ifndef ANDROID_AUDIO_MIXER_H |
| #define ANDROID_AUDIO_MIXER_H |
| |
| #include <stdint.h> |
| #include <sys/types.h> |
| |
| #include "AudioBufferProvider.h" |
| #include "AudioResampler.h" |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| |
| #define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) |
| #define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) |
| |
| // ---------------------------------------------------------------------------- |
| |
| class AudioMixer |
| { |
| public: |
| AudioMixer(size_t frameCount, uint32_t sampleRate); |
| |
| ~AudioMixer(); |
| |
| static const uint32_t MAX_NUM_TRACKS = 32; |
| static const uint32_t MAX_NUM_CHANNELS = 2; |
| |
| static const uint16_t UNITY_GAIN = 0x1000; |
| |
| enum { // names |
| |
| // track units (32 units) |
| TRACK0 = 0x1000, |
| |
| // enable/disable |
| MIXING = 0x2000, |
| |
| // setParameter targets |
| TRACK = 0x3000, |
| RESAMPLE = 0x3001, |
| RAMP_VOLUME = 0x3002, // ramp to new volume |
| VOLUME = 0x3003, // don't ramp |
| |
| // set Parameter names |
| // for target TRACK |
| CHANNEL_MASK = 0x4000, |
| FORMAT = 0x4001, |
| MAIN_BUFFER = 0x4002, |
| AUX_BUFFER = 0x4003, |
| // for TARGET RESAMPLE |
| SAMPLE_RATE = 0x4100, |
| RESET = 0x4101, |
| // for TARGET VOLUME (8 channels max) |
| VOLUME0 = 0x4200, |
| VOLUME1 = 0x4201, |
| AUXLEVEL = 0x4210, |
| }; |
| |
| |
| int getTrackName(); |
| void deleteTrackName(int name); |
| |
| status_t enable(int name); |
| status_t disable(int name); |
| |
| status_t setActiveTrack(int track); |
| status_t setParameter(int target, int name, void *value); |
| |
| status_t setBufferProvider(AudioBufferProvider* bufferProvider); |
| void process(); |
| |
| uint32_t trackNames() const { return mTrackNames; } |
| |
| static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c); |
| |
| private: |
| |
| enum { |
| NEEDS_CHANNEL_COUNT__MASK = 0x00000003, |
| NEEDS_FORMAT__MASK = 0x000000F0, |
| NEEDS_MUTE__MASK = 0x00000100, |
| NEEDS_RESAMPLE__MASK = 0x00001000, |
| NEEDS_AUX__MASK = 0x00010000, |
| }; |
| |
| enum { |
| NEEDS_CHANNEL_1 = 0x00000000, |
| NEEDS_CHANNEL_2 = 0x00000001, |
| |
| NEEDS_FORMAT_16 = 0x00000010, |
| |
| NEEDS_MUTE_DISABLED = 0x00000000, |
| NEEDS_MUTE_ENABLED = 0x00000100, |
| |
| NEEDS_RESAMPLE_DISABLED = 0x00000000, |
| NEEDS_RESAMPLE_ENABLED = 0x00001000, |
| |
| NEEDS_AUX_DISABLED = 0x00000000, |
| NEEDS_AUX_ENABLED = 0x00010000, |
| }; |
| |
| static inline int32_t applyVolume(int32_t in, int32_t v) { |
| return in * v; |
| } |
| |
| |
| struct state_t; |
| struct track_t; |
| |
| typedef void (*mix_t)(state_t* state); |
| typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux); |
| static const int BLOCKSIZE = 16; // 4 cache lines |
| |
| struct track_t { |
| uint32_t needs; |
| |
| union { |
| int16_t volume[2]; // [0]3.12 fixed point |
| int32_t volumeRL; |
| }; |
| |
| int32_t prevVolume[2]; |
| |
| int32_t volumeInc[2]; |
| int32_t auxLevel; |
| int32_t auxInc; |
| int32_t prevAuxLevel; |
| |
| uint16_t frameCount; |
| |
| uint8_t channelCount : 4; |
| uint8_t enabled : 1; |
| uint8_t reserved0 : 3; |
| uint8_t format; |
| uint32_t channelMask; |
| |
| AudioBufferProvider* bufferProvider; |
| mutable AudioBufferProvider::Buffer buffer; |
| |
| hook_t hook; |
| void const* in; // current location in buffer |
| |
| AudioResampler* resampler; |
| uint32_t sampleRate; |
| int32_t* mainBuffer; |
| int32_t* auxBuffer; |
| |
| bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); |
| bool doesResample() const; |
| void resetResampler(); |
| void adjustVolumeRamp(bool aux); |
| }; |
| |
| // pad to 32-bytes to fill cache line |
| struct state_t { |
| uint32_t enabledTracks; |
| uint32_t needsChanged; |
| size_t frameCount; |
| mix_t hook; |
| int32_t *outputTemp; |
| int32_t *resampleTemp; |
| int32_t reserved[2]; |
| track_t tracks[32]; __attribute__((aligned(32))); |
| }; |
| |
| int mActiveTrack; |
| uint32_t mTrackNames; |
| const uint32_t mSampleRate; |
| |
| state_t mState __attribute__((aligned(32))); |
| |
| void invalidateState(uint32_t mask); |
| |
| static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); |
| static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); |
| static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); |
| static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); |
| static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); |
| static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); |
| |
| static void process__validate(state_t* state); |
| static void process__nop(state_t* state); |
| static void process__genericNoResampling(state_t* state); |
| static void process__genericResampling(state_t* state); |
| static void process__OneTrack16BitsStereoNoResampling(state_t* state); |
| static void process__TwoTracks16BitsStereoNoResampling(state_t* state); |
| }; |
| |
| // ---------------------------------------------------------------------------- |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIO_MIXER_H |