| /* |
| ** |
| ** 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 <media/AudioBufferProvider.h> |
| #include <media/AudioResampler.h> |
| #include <media/AudioResamplerPublic.h> |
| #include <media/BufferProviders.h> |
| #include <media/nbaio/NBLog.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 |
| |
| 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(); |
| |
| 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_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); |
| |
| // 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* mNBLogWriter; // associated NBLog::Writer or &mDummyLog |
| 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 mDummyLogWriter; |
| public: |
| // Called by FastMixer to inform AudioMixer of it's associated NBLog::Writer. |
| // FIXME It would be safer to use TLS for this, so we don't accidentally use wrong one. |
| void setNBLogWriter(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); |
| 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 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 <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL, |
| typename TO, typename TI, typename TA> |
| static void volumeMix(TO *out, size_t outFrames, |
| const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t); |
| |
| // multi-format process hooks |
| template <int MIXTYPE, typename TO, typename TI, typename TA> |
| static void process_NoResampleOneTrack(state_t* state); |
| |
| // multi-format track hooks |
| template <int MIXTYPE, typename TO, typename TI, typename TA> |
| static void track__Resample(track_t* t, TO* out, size_t frameCount, |
| TO* temp __unused, TA* aux); |
| template <int MIXTYPE, typename TO, typename TI, typename TA> |
| 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 |
