| /* |
| ** |
| ** 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 { |
| |
| // ---------------------------------------------------------------------------- |
| |
| class AudioMixer |
| { |
| public: |
| AudioMixer(size_t frameCount, uint32_t sampleRate, |
| uint32_t maxNumTracks = MAX_NUM_TRACKS); |
| |
| /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed |
| |
| 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 names (MAX_NUM_TRACKS units) |
| TRACK0 = 0x1000, |
| |
| // 0x2000 is unused |
| |
| // 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 RAMP_VOLUME and VOLUME (8 channels max) |
| VOLUME0 = 0x4200, |
| VOLUME1 = 0x4201, |
| AUXLEVEL = 0x4210, |
| }; |
| |
| |
| // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS |
| |
| // Allocate a track name. Returns new track name if successful, -1 on failure. |
| int getTrackName(); |
| |
| // Free an allocated track by name |
| void deleteTrackName(int name); |
| |
| // Enable or disable an allocated track by name |
| void enable(int name); |
| void disable(int name); |
| |
| void setParameter(int name, int target, int param, void *value); |
| |
| void setBufferProvider(int name, AudioBufferProvider* bufferProvider); |
| void process(int64_t pts); |
| |
| uint32_t trackNames() const { return mTrackNames; } |
| |
| size_t getUnreleasedFrames(int name) const; |
| |
| 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, |
| }; |
| |
| struct state_t; |
| struct track_t; |
| |
| 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[MAX_NUM_CHANNELS]; // [0]3.12 fixed point |
| int32_t volumeRL; |
| }; |
| |
| int32_t prevVolume[MAX_NUM_CHANNELS]; |
| |
| // 16-byte boundary |
| |
| int32_t volumeInc[MAX_NUM_CHANNELS]; |
| int32_t auxInc; |
| int32_t prevAuxLevel; |
| |
| // 16-byte boundary |
| |
| int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance |
| uint16_t frameCount; |
| |
| uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) |
| uint8_t format; // always 16 |
| uint16_t enabled; // actually bool |
| uint32_t channelMask; // currently under-used |
| |
| AudioBufferProvider* bufferProvider; |
| |
| // 16-byte boundary |
| |
| mutable AudioBufferProvider::Buffer buffer; // 8 bytes |
| |
| hook_t hook; |
| const void* in; // current location in buffer |
| |
| // 16-byte boundary |
| |
| AudioResampler* resampler; |
| uint32_t sampleRate; |
| int32_t* mainBuffer; |
| int32_t* auxBuffer; |
| |
| // 16-byte boundary |
| |
| uint64_t localTimeFreq; |
| |
| int64_t padding; |
| |
| // 16-byte boundary |
| |
| bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); |
| bool doesResample() const { return resampler != NULL; } |
| void resetResampler() { if (resampler != NULL) resampler->reset(); } |
| void adjustVolumeRamp(bool aux); |
| size_t getUnreleasedFrames() const { return resampler != NULL ? |
| resampler->getUnreleasedFrames() : 0; }; |
| }; |
| |
| // pad to 32-bytes to fill cache line |
| struct state_t { |
| uint32_t enabledTracks; |
| uint32_t needsChanged; |
| size_t frameCount; |
| void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL |
| int32_t *outputTemp; |
| int32_t *resampleTemp; |
| int32_t reserved[2]; |
| // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS |
| track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32))); |
| }; |
| |
| // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc. |
| uint32_t mTrackNames; |
| |
| // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS, |
| // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS |
| const uint32_t mConfiguredNames; |
| |
| 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, int64_t pts); |
| static void process__nop(state_t* state, int64_t pts); |
| static void process__genericNoResampling(state_t* state, int64_t pts); |
| static void process__genericResampling(state_t* state, int64_t pts); |
| static void process__OneTrack16BitsStereoNoResampling(state_t* state, |
| int64_t pts); |
| #if 0 |
| static void process__TwoTracks16BitsStereoNoResampling(state_t* state, |
| int64_t pts); |
| #endif |
| |
| static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, |
| int outputFrameIndex); |
| }; |
| |
| // ---------------------------------------------------------------------------- |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIO_MIXER_H |