| /* |
| ** |
| ** 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 <utils/threads.h> |
| |
| #include <media/AudioBufferProvider.h> |
| #include "AudioResampler.h" |
| |
| #include <hardware/audio_effect.h> |
| #include <system/audio.h> |
| #include <media/nbaio/NBLog.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 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 |
| |
| // 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 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) { |
| return format == AUDIO_FORMAT_PCM_16_BIT || |
| format == AUDIO_FORMAT_PCM_24_BIT_PACKED || |
| format == AUDIO_FORMAT_PCM_32_BIT || |
| format == AUDIO_FORMAT_PCM_FLOAT; |
| } |
| |
| 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; |
| class CopyBufferProvider; |
| |
| 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 |
| AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider. |
| CopyBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting. |
| CopyBufferProvider* downmixerBufferProvider; // wrapper for channel conversion. |
| |
| int32_t sessionId; |
| |
| // 16-byte boundary |
| 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. |
| |
| 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 |
| |
| // 16-byte boundary |
| audio_channel_mask_t mMixerChannelMask; |
| uint32_t mMixerChannelCount; |
| |
| 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; }; |
| }; |
| |
| 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))); |
| }; |
| |
| // Base AudioBufferProvider class used for DownMixerBufferProvider, RemixBufferProvider, |
| // and ReformatBufferProvider. |
| // It handles a private buffer for use in converting format or channel masks from the |
| // input data to a form acceptable by the mixer. |
| // TODO: Make a ResamplerBufferProvider when integers are entirely removed from the |
| // processing pipeline. |
| class CopyBufferProvider : public AudioBufferProvider { |
| public: |
| // Use a private buffer of bufferFrameCount frames (each frame is outputFrameSize bytes). |
| // If bufferFrameCount is 0, no private buffer is created and in-place modification of |
| // the upstream buffer provider's buffers is performed by copyFrames(). |
| CopyBufferProvider(size_t inputFrameSize, size_t outputFrameSize, |
| size_t bufferFrameCount); |
| virtual ~CopyBufferProvider(); |
| |
| // Overrides AudioBufferProvider methods |
| virtual status_t getNextBuffer(Buffer* buffer, int64_t pts); |
| virtual void releaseBuffer(Buffer* buffer); |
| |
| // Other public methods |
| |
| // call this to release the buffer to the upstream provider. |
| // treat it as an audio discontinuity for future samples. |
| virtual void reset(); |
| |
| // this function should be supplied by the derived class. It converts |
| // #frames in the *src pointer to the *dst pointer. It is public because |
| // some providers will allow this to work on arbitrary buffers outside |
| // of the internal buffers. |
| virtual void copyFrames(void *dst, const void *src, size_t frames) = 0; |
| |
| // set the upstream buffer provider. Consider calling "reset" before this function. |
| void setBufferProvider(AudioBufferProvider *p) { |
| mTrackBufferProvider = p; |
| } |
| |
| protected: |
| AudioBufferProvider* mTrackBufferProvider; |
| const size_t mInputFrameSize; |
| const size_t mOutputFrameSize; |
| private: |
| AudioBufferProvider::Buffer mBuffer; |
| const size_t mLocalBufferFrameCount; |
| void* mLocalBufferData; |
| size_t mConsumed; |
| }; |
| |
| // DownmixerBufferProvider wraps a track AudioBufferProvider to provide |
| // position dependent downmixing by an Audio Effect. |
| class DownmixerBufferProvider : public CopyBufferProvider { |
| public: |
| DownmixerBufferProvider(audio_channel_mask_t inputChannelMask, |
| audio_channel_mask_t outputChannelMask, audio_format_t format, |
| uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount); |
| virtual ~DownmixerBufferProvider(); |
| virtual void copyFrames(void *dst, const void *src, size_t frames); |
| bool isValid() const { return mDownmixHandle != NULL; } |
| |
| static status_t init(); |
| static bool isMultichannelCapable() { return sIsMultichannelCapable; } |
| |
| protected: |
| effect_handle_t mDownmixHandle; |
| effect_config_t mDownmixConfig; |
| |
| // effect descriptor for the downmixer used by the mixer |
| static effect_descriptor_t sDwnmFxDesc; |
| // indicates whether a downmix effect has been found and is usable by this mixer |
| static bool sIsMultichannelCapable; |
| // FIXME: should we allow effects outside of the framework? |
| // We need to here. A special ioId that must be <= -2 so it does not map to a session. |
| static const int32_t SESSION_ID_INVALID_AND_IGNORED = -2; |
| }; |
| |
| // RemixBufferProvider wraps a track AudioBufferProvider to perform an |
| // upmix or downmix to the proper channel count and mask. |
| class RemixBufferProvider : public CopyBufferProvider { |
| public: |
| RemixBufferProvider(audio_channel_mask_t inputChannelMask, |
| audio_channel_mask_t outputChannelMask, audio_format_t format, |
| size_t bufferFrameCount); |
| virtual void copyFrames(void *dst, const void *src, size_t frames); |
| |
| protected: |
| const audio_format_t mFormat; |
| const size_t mSampleSize; |
| const size_t mInputChannels; |
| const size_t mOutputChannels; |
| int8_t mIdxAry[sizeof(uint32_t)*8]; // 32 bits => channel indices |
| }; |
| |
| // ReformatBufferProvider wraps a track AudioBufferProvider to convert the input data |
| // to an acceptable mixer input format type. |
| class ReformatBufferProvider : public CopyBufferProvider { |
| public: |
| ReformatBufferProvider(int32_t channels, |
| audio_format_t inputFormat, audio_format_t outputFormat, |
| size_t bufferFrameCount); |
| virtual void copyFrames(void *dst, const void *src, size_t frames); |
| |
| protected: |
| const int32_t mChannels; |
| const audio_format_t mInputFormat; |
| const audio_format_t mOutputFormat; |
| }; |
| |
| // 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); |
| |
| // TODO: remove unused trackName/trackNum from functions below. |
| static status_t initTrackDownmix(track_t* pTrack, int trackName); |
| static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); |
| static void unprepareTrackForDownmix(track_t* pTrack, int trackName); |
| static status_t prepareTrackForReformat(track_t* pTrack, int trackNum); |
| static void unprepareTrackForReformat(track_t* pTrack, int trackName); |
| static void reconfigureBufferProviders(track_t* pTrack); |
| |
| 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 <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, int64_t pts); |
| |
| // 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 |
