/* ** ** 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 #include #include #include #include #include #include #include #include #include "AudioResampler.h" #include "BufferProviders.h" // FIXME This is actually unity gain, which might not be max in future, expressed in U.12 #define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT 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 // This mixer has a hard-coded upper limit of 32 active track inputs. // Adding support for > 32 tracks would require more than simply changing this value. static const uint32_t MAX_NUM_TRACKS = 32; // maximum number of channels supported by the mixer // This mixer has a hard-coded upper limit of 8 channels for output. static const uint32_t MAX_NUM_CHANNELS = 8; static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only // maximum number of channels supported for the content static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX; static const uint16_t UNITY_GAIN_INT = 0x1000; static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f; 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 TIMESTRETCH = 0x3004, // set Parameter names // for target TRACK CHANNEL_MASK = 0x4000, FORMAT = 0x4001, MAIN_BUFFER = 0x4002, AUX_BUFFER = 0x4003, DOWNMIX_TYPE = 0X4004, MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT) MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output // for target RESAMPLE SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name; // parameter 'value' is the new sample rate in Hz. // Only creates a sample rate converter the first time that // the track sample rate is different from the mix sample rate. // If the new sample rate is the same as the mix sample rate, // and a sample rate converter already exists, // then the sample rate converter remains present but is a no-op. RESET = 0x4101, // Reset sample rate converter without changing sample rate. // This clears out the resampler's input buffer. REMOVE = 0x4102, // Remove the sample rate converter on this track name; // the track is restored to the mix sample rate. // for target RAMP_VOLUME and VOLUME (8 channels max) // FIXME use float for these 3 to improve the dynamic range VOLUME0 = 0x4200, VOLUME1 = 0x4201, AUXLEVEL = 0x4210, // for target TIMESTRETCH PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name; // parameter 'value' is a pointer to the new playback rate. }; // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS // Allocate a track name. Returns new track name if successful, -1 on failure. // The failure could be because of an invalid channelMask or format, or that // the track capacity of the mixer is exceeded. int getTrackName(audio_channel_mask_t channelMask, audio_format_t format, int sessionId); // 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; static inline bool isValidPcmTrackFormat(audio_format_t format) { switch (format) { case AUDIO_FORMAT_PCM_8_BIT: case AUDIO_FORMAT_PCM_16_BIT: case AUDIO_FORMAT_PCM_24_BIT_PACKED: case AUDIO_FORMAT_PCM_8_24_BIT: case AUDIO_FORMAT_PCM_32_BIT: case AUDIO_FORMAT_PCM_FLOAT: return true; default: return false; } } private: enum { // FIXME this representation permits up to 8 channels NEEDS_CHANNEL_COUNT__MASK = 0x00000007, }; enum { NEEDS_CHANNEL_1 = 0x00000000, // mono NEEDS_CHANNEL_2 = 0x00000001, // stereo // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT NEEDS_MUTE = 0x00000100, NEEDS_RESAMPLE = 0x00001000, NEEDS_AUX = 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; // TODO: Eventually remove legacy integer volume settings union { int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero) int32_t volumeRL; }; int32_t prevVolume[MAX_NUM_VOLUMES]; // 16-byte boundary int32_t volumeInc[MAX_NUM_VOLUMES]; 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 unused_padding; // formerly format, was always 16 uint16_t enabled; // actually bool audio_channel_mask_t channelMask; // actual buffer provider used by the track hooks, see DownmixerBufferProvider below // for how the Track buffer provider is wrapped by another one when dowmixing is required 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 /* Buffer providers are constructed to translate the track input data as needed. * * TODO: perhaps make a single PlaybackConverterProvider class to move * all pre-mixer track buffer conversions outside the AudioMixer class. * * 1) mInputBufferProvider: The AudioTrack buffer provider. * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to * match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer * requires reformat. For example, it may convert floating point input to * PCM_16_bit if that's required by the downmixer. * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match * the number of channels required by the mixer sink. * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from * the downmixer requirements to the mixer engine input requirements. * 5) mTimestretchBufferProvider: Adds timestretching for playback rate */ AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider. PassthruBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting. PassthruBufferProvider* downmixerBufferProvider; // wrapper for channel conversion. PassthruBufferProvider* mPostDownmixReformatBufferProvider; PassthruBufferProvider* mTimestretchBufferProvider; int32_t sessionId; audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT) audio_format_t mFormat; // input track format audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT) // each track must be converted to this format. audio_format_t mDownmixRequiresFormat; // required downmixer format // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary // AUDIO_FORMAT_INVALID if no required format float mVolume[MAX_NUM_VOLUMES]; // floating point set volume float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment float mAuxLevel; // floating point set aux level float mPrevAuxLevel; // floating point prev aux level float mAuxInc; // floating point aux increment audio_channel_mask_t mMixerChannelMask; uint32_t mMixerChannelCount; AudioPlaybackRate mPlaybackRate; bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; } bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate); bool doesResample() const { return resampler != NULL; } void resetResampler() { if (resampler != NULL) resampler->reset(); } void adjustVolumeRamp(bool aux, bool useFloat = false); size_t getUnreleasedFrames() const { return resampler != NULL ? resampler->getUnreleasedFrames() : 0; }; status_t prepareForDownmix(); void unprepareForDownmix(); status_t prepareForReformat(); void unprepareForReformat(); bool setPlaybackRate(const AudioPlaybackRate &playbackRate); void reconfigureBufferProviders(); }; typedef void (*process_hook_t)(state_t* state, int64_t pts); // pad to 32-bytes to fill cache line struct state_t { uint32_t enabledTracks; uint32_t needsChanged; size_t frameCount; process_hook_t hook; // one of process__*, never NULL int32_t *outputTemp; int32_t *resampleTemp; NBLog::Writer* mLog; int32_t reserved[1]; // 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; NBLog::Writer mDummyLog; public: void setLog(NBLog::Writer* log); private: state_t mState __attribute__((aligned(32))); // Call after changing either the enabled status of a track, or parameters of an enabled track. // OK to call more often than that, but unnecessary. void invalidateState(uint32_t mask); bool setChannelMasks(int name, audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask); 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); static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, int outputFrameIndex); static uint64_t sLocalTimeFreq; static pthread_once_t sOnceControl; static void sInitRoutine(); /* multi-format volume mixing function (calls template functions * in AudioMixerOps.h). The template parameters are as follows: * * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) * USEFLOATVOL (set to true if float volume is used) * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards) * TO: int32_t (Q4.27) or float * TI: int32_t (Q4.27) or int16_t (Q0.15) or float * TA: int32_t (Q4.27) */ template static void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t); // multi-format process hooks template static void process_NoResampleOneTrack(state_t* state, int64_t pts); // multi-format track hooks template static void track__Resample(track_t* t, TO* out, size_t frameCount, TO* temp __unused, TA* aux); template static void track__NoResample(track_t* t, TO* out, size_t frameCount, TO* temp __unused, TA* aux); static void convertMixerFormat(void *out, audio_format_t mixerOutFormat, void *in, audio_format_t mixerInFormat, size_t sampleCount); // hook types enum { PROCESSTYPE_NORESAMPLEONETRACK, }; enum { TRACKTYPE_NOP, TRACKTYPE_RESAMPLE, TRACKTYPE_NORESAMPLE, TRACKTYPE_NORESAMPLEMONO, }; // functions for determining the proper process and track hooks. static process_hook_t getProcessHook(int processType, uint32_t channelCount, audio_format_t mixerInFormat, audio_format_t mixerOutFormat); static hook_t getTrackHook(int trackType, uint32_t channelCount, audio_format_t mixerInFormat, audio_format_t mixerOutFormat); }; // ---------------------------------------------------------------------------- } // namespace android #endif // ANDROID_AUDIO_MIXER_H