diff options
Diffstat (limited to 'services/audioflinger/AudioMixer.h')
| -rw-r--r-- | services/audioflinger/AudioMixer.h | 235 |
1 files changed, 235 insertions, 0 deletions
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h new file mode 100644 index 0000000..856450c --- /dev/null +++ b/services/audioflinger/AudioMixer.h @@ -0,0 +1,235 @@ +/* +** +** 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 |