| /* |
| ** |
| ** 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 <map> |
| #include <pthread.h> |
| #include <sstream> |
| #include <stdint.h> |
| #include <sys/types.h> |
| #include <unordered_map> |
| #include <vector> |
| |
| #include <android/os/IExternalVibratorService.h> |
| #include <media/AudioBufferProvider.h> |
| #include <media/AudioResampler.h> |
| #include <media/AudioResamplerPublic.h> |
| #include <media/BufferProviders.h> |
| #include <system/audio.h> |
| #include <utils/Compat.h> |
| #include <utils/threads.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 |
| |
| // This must match frameworks/av/services/audioflinger/Configuration.h |
| #define FLOAT_AUX |
| |
| namespace android { |
| |
| namespace NBLog { |
| class Writer; |
| } // namespace NBLog |
| |
| // ---------------------------------------------------------------------------- |
| |
| class AudioMixer |
| { |
| public: |
| // Do not change these unless underlying code changes. |
| // This mixer has a hard-coded upper limit of 8 channels for output. |
| static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8; |
| static constexpr uint32_t MAX_NUM_VOLUMES = FCC_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 |
| // 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 haptic |
| HAPTIC_ENABLED = 0x4007, // Set haptic data from this track should be played or not. |
| HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data. |
| // 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. |
| }; |
| |
| typedef enum { // Haptic intensity, should keep consistent with VibratorService |
| HAPTIC_SCALE_MUTE = os::IExternalVibratorService::SCALE_MUTE, |
| HAPTIC_SCALE_VERY_LOW = os::IExternalVibratorService::SCALE_VERY_LOW, |
| HAPTIC_SCALE_LOW = os::IExternalVibratorService::SCALE_LOW, |
| HAPTIC_SCALE_NONE = os::IExternalVibratorService::SCALE_NONE, |
| HAPTIC_SCALE_HIGH = os::IExternalVibratorService::SCALE_HIGH, |
| HAPTIC_SCALE_VERY_HIGH = os::IExternalVibratorService::SCALE_VERY_HIGH, |
| } haptic_intensity_t; |
| static constexpr float HAPTIC_SCALE_VERY_LOW_RATIO = 2.0f / 3.0f; |
| static constexpr float HAPTIC_SCALE_LOW_RATIO = 3.0f / 4.0f; |
| static const constexpr float HAPTIC_MAX_AMPLITUDE_FLOAT = 1.0f; |
| |
| static inline bool isValidHapticIntensity(haptic_intensity_t hapticIntensity) { |
| switch (hapticIntensity) { |
| case HAPTIC_SCALE_MUTE: |
| case HAPTIC_SCALE_VERY_LOW: |
| case HAPTIC_SCALE_LOW: |
| case HAPTIC_SCALE_NONE: |
| case HAPTIC_SCALE_HIGH: |
| case HAPTIC_SCALE_VERY_HIGH: |
| return true; |
| default: |
| return false; |
| } |
| } |
| |
| AudioMixer(size_t frameCount, uint32_t sampleRate) |
| : mSampleRate(sampleRate) |
| , mFrameCount(frameCount) { |
| pthread_once(&sOnceControl, &sInitRoutine); |
| } |
| |
| // Create a new track in the mixer. |
| // |
| // \param name a unique user-provided integer associated with the track. |
| // If name already exists, the function will abort. |
| // \param channelMask output channel mask. |
| // \param format PCM format |
| // \param sessionId Session id for the track. Tracks with the same |
| // session id will be submixed together. |
| // |
| // \return OK on success. |
| // BAD_VALUE if the format does not satisfy isValidFormat() |
| // or the channelMask does not satisfy isValidChannelMask(). |
| status_t create( |
| int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId); |
| |
| bool exists(int name) const { |
| return mTracks.count(name) > 0; |
| } |
| |
| // Free an allocated track by name. |
| void destroy(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() { |
| for (const auto &pair : mTracks) { |
| // Clear contracted buffer before processing if contracted channels are saved |
| const std::shared_ptr<Track> &t = pair.second; |
| if (t->mKeepContractedChannels) { |
| t->clearContractedBuffer(); |
| } |
| } |
| (this->*mHook)(); |
| processHapticData(); |
| } |
| |
| size_t getUnreleasedFrames(int name) const; |
| |
| std::string trackNames() const { |
| std::stringstream ss; |
| for (const auto &pair : mTracks) { |
| ss << pair.first << " "; |
| } |
| return ss.str(); |
| } |
| |
| void setNBLogWriter(NBLog::Writer *logWriter) { |
| mNBLogWriter = logWriter; |
| } |
| |
| static inline bool isValidFormat(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_32_BIT: |
| case AUDIO_FORMAT_PCM_FLOAT: |
| return true; |
| default: |
| return false; |
| } |
| } |
| |
| static inline bool isValidChannelMask(audio_channel_mask_t channelMask) { |
| return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible. |
| } |
| |
| private: |
| |
| /* For multi-format functions (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) |
| */ |
| |
| 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, |
| }; |
| |
| // hook types |
| enum { |
| PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere |
| }; |
| |
| enum { |
| TRACKTYPE_NOP, |
| TRACKTYPE_RESAMPLE, |
| TRACKTYPE_NORESAMPLE, |
| TRACKTYPE_NORESAMPLEMONO, |
| }; |
| |
| // process hook functionality |
| using process_hook_t = void(AudioMixer::*)(); |
| |
| struct Track; |
| using hook_t = void(Track::*)(int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux); |
| |
| struct Track { |
| Track() |
| : bufferProvider(nullptr) |
| { |
| // TODO: move additional initialization here. |
| } |
| |
| ~Track() |
| { |
| // bufferProvider, mInputBufferProvider need not be deleted. |
| mResampler.reset(nullptr); |
| // Ensure the order of destruction of buffer providers as they |
| // release the upstream provider in the destructor. |
| mTimestretchBufferProvider.reset(nullptr); |
| mPostDownmixReformatBufferProvider.reset(nullptr); |
| mDownmixerBufferProvider.reset(nullptr); |
| mReformatBufferProvider.reset(nullptr); |
| mContractChannelsNonDestructiveBufferProvider.reset(nullptr); |
| mAdjustChannelsBufferProvider.reset(nullptr); |
| } |
| |
| bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; } |
| bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate); |
| bool doesResample() const { return mResampler.get() != nullptr; } |
| void resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); } |
| void adjustVolumeRamp(bool aux, bool useFloat = false); |
| size_t getUnreleasedFrames() const { return mResampler.get() != nullptr ? |
| mResampler->getUnreleasedFrames() : 0; }; |
| |
| status_t prepareForDownmix(); |
| void unprepareForDownmix(); |
| status_t prepareForReformat(); |
| void unprepareForReformat(); |
| status_t prepareForAdjustChannels(); |
| void unprepareForAdjustChannels(); |
| status_t prepareForAdjustChannelsNonDestructive(size_t frames); |
| void unprepareForAdjustChannelsNonDestructive(); |
| void clearContractedBuffer(); |
| bool setPlaybackRate(const AudioPlaybackRate &playbackRate); |
| void reconfigureBufferProviders(); |
| |
| static hook_t getTrackHook(int trackType, uint32_t channelCount, |
| audio_format_t mixerInFormat, audio_format_t mixerOutFormat); |
| |
| void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); |
| |
| template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL, |
| typename TO, typename TI, typename TA> |
| void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp); |
| |
| 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]; |
| int32_t volumeInc[MAX_NUM_VOLUMES]; |
| int32_t auxInc; |
| int32_t prevAuxLevel; |
| 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; |
| |
| mutable AudioBufferProvider::Buffer buffer; // 8 bytes |
| |
| hook_t hook; |
| const void *mIn; // current location in buffer |
| |
| std::unique_ptr<AudioResampler> mResampler; |
| uint32_t sampleRate; |
| int32_t* mainBuffer; |
| int32_t* auxBuffer; |
| |
| /* 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) mAdjustChannelsBufferProvider: Expands or contracts sample data from one interleaved |
| * channel format to another. Expanded channels are filled with zeros and put at the end |
| * of each audio frame. Contracted channels are copied to the end of the buffer. |
| * 3) mContractChannelsNonDestructiveBufferProvider: Non-destructively contract sample data. |
| * This is currently using at audio-haptic coupled playback to separate audio and haptic |
| * data. Contracted channels could be written to given buffer. |
| * 4) 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. |
| * 5) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match |
| * the number of channels required by the mixer sink. |
| * 6) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from |
| * the downmixer requirements to the mixer engine input requirements. |
| * 7) mTimestretchBufferProvider: Adds timestretching for playback rate |
| */ |
| AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider. |
| // TODO: combine mAdjustChannelsBufferProvider and |
| // mContractChannelsNonDestructiveBufferProvider |
| std::unique_ptr<PassthruBufferProvider> mAdjustChannelsBufferProvider; |
| std::unique_ptr<PassthruBufferProvider> mContractChannelsNonDestructiveBufferProvider; |
| std::unique_ptr<PassthruBufferProvider> mReformatBufferProvider; |
| std::unique_ptr<PassthruBufferProvider> mDownmixerBufferProvider; |
| std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider; |
| std::unique_ptr<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; |
| |
| // Haptic |
| bool mHapticPlaybackEnabled; |
| haptic_intensity_t mHapticIntensity; |
| audio_channel_mask_t mHapticChannelMask; |
| uint32_t mHapticChannelCount; |
| audio_channel_mask_t mMixerHapticChannelMask; |
| uint32_t mMixerHapticChannelCount; |
| uint32_t mAdjustInChannelCount; |
| uint32_t mAdjustOutChannelCount; |
| uint32_t mAdjustNonDestructiveInChannelCount; |
| uint32_t mAdjustNonDestructiveOutChannelCount; |
| bool mKeepContractedChannels; |
| |
| float getHapticScaleGamma() const { |
| // Need to keep consistent with the value in VibratorService. |
| switch (mHapticIntensity) { |
| case HAPTIC_SCALE_VERY_LOW: |
| return 2.0f; |
| case HAPTIC_SCALE_LOW: |
| return 1.5f; |
| case HAPTIC_SCALE_HIGH: |
| return 0.5f; |
| case HAPTIC_SCALE_VERY_HIGH: |
| return 0.25f; |
| default: |
| return 1.0f; |
| } |
| } |
| |
| float getHapticMaxAmplitudeRatio() const { |
| // Need to keep consistent with the value in VibratorService. |
| switch (mHapticIntensity) { |
| case HAPTIC_SCALE_VERY_LOW: |
| return HAPTIC_SCALE_VERY_LOW_RATIO; |
| case HAPTIC_SCALE_LOW: |
| return HAPTIC_SCALE_LOW_RATIO; |
| case HAPTIC_SCALE_NONE: |
| case HAPTIC_SCALE_HIGH: |
| case HAPTIC_SCALE_VERY_HIGH: |
| return 1.0f; |
| default: |
| return 0.0f; |
| } |
| } |
| |
| private: |
| // hooks |
| void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); |
| void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); |
| void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); |
| |
| void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); |
| void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); |
| |
| // multi-format track hooks |
| template <int MIXTYPE, typename TO, typename TI, typename TA> |
| void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux); |
| template <int MIXTYPE, typename TO, typename TI, typename TA> |
| void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux); |
| }; |
| |
| // TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore. |
| static constexpr int BLOCKSIZE = 16; |
| |
| bool setChannelMasks(int name, |
| audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask); |
| |
| // Called when track info changes and a new process hook should be determined. |
| void invalidate() { |
| mHook = &AudioMixer::process__validate; |
| } |
| |
| void process__validate(); |
| void process__nop(); |
| void process__genericNoResampling(); |
| void process__genericResampling(); |
| void process__oneTrack16BitsStereoNoResampling(); |
| |
| template <int MIXTYPE, typename TO, typename TI, typename TA> |
| void process__noResampleOneTrack(); |
| |
| void processHapticData(); |
| |
| static process_hook_t getProcessHook(int processType, uint32_t channelCount, |
| audio_format_t mixerInFormat, audio_format_t mixerOutFormat); |
| |
| static void convertMixerFormat(void *out, audio_format_t mixerOutFormat, |
| void *in, audio_format_t mixerInFormat, size_t sampleCount); |
| |
| static void sInitRoutine(); |
| |
| // initialization constants |
| const uint32_t mSampleRate; |
| const size_t mFrameCount; |
| |
| NBLog::Writer *mNBLogWriter = nullptr; // associated NBLog::Writer |
| |
| process_hook_t mHook = &AudioMixer::process__nop; // one of process__*, never nullptr |
| |
| // the size of the type (int32_t) should be the largest of all types supported |
| // by the mixer. |
| std::unique_ptr<int32_t[]> mOutputTemp; |
| std::unique_ptr<int32_t[]> mResampleTemp; |
| |
| // track names grouped by main buffer, in no particular order of main buffer. |
| // however names for a particular main buffer are in order (by construction). |
| std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups; |
| |
| // track names that are enabled, in increasing order (by construction). |
| std::vector<int /* name */> mEnabled; |
| |
| // track smart pointers, by name, in increasing order of name. |
| std::map<int /* name */, std::shared_ptr<Track>> mTracks; |
| |
| static pthread_once_t sOnceControl; // initialized in constructor by first new |
| }; |
| |
| // ---------------------------------------------------------------------------- |
| } // namespace android |
| |
| #endif // ANDROID_AUDIO_MIXER_H |
