* changes: Use PassthruPatchRecord for DIRECT to DIRECT connections Add PassthruPatchRecord for low latency software patches Abstract access to HAL stream via Source in RecordThread Move PlaybackThread::Track::writeFrames to PatchRecord audioflinger: Add tracing of buffer frames to PatchTrack/Record libaudioprocessing: Extract vendor-available part of AudioMixer AudioMixer: Cleanups libaudioprocessing: Trivial dependency cleanupsgugelfrei
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/*
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**
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** Copyright 2019, The Android Open Source Project
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**
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** Licensed under the Apache License, Version 2.0 (the "License");
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** you may not use this file except in compliance with the License.
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** You may obtain a copy of the License at
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**
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** http://www.apache.org/licenses/LICENSE-2.0
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**
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** Unless required by applicable law or agreed to in writing, software
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** distributed under the License is distributed on an "AS IS" BASIS,
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** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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** See the License for the specific language governing permissions and
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** limitations under the License.
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*/
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#ifndef ANDROID_AUDIO_MIXER_BASE_H
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#define ANDROID_AUDIO_MIXER_BASE_H
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#include <map>
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#include <memory>
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#include <string>
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#include <unordered_map>
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#include <vector>
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#include <media/AudioBufferProvider.h>
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#include <media/AudioResampler.h>
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#include <media/AudioResamplerPublic.h>
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#include <system/audio.h>
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#include <utils/Compat.h>
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// This must match frameworks/av/services/audioflinger/Configuration.h
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// when used with the Audio Framework.
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#define FLOAT_AUX
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namespace android {
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// ----------------------------------------------------------------------------
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// AudioMixerBase is functional on its own if only mixing and resampling
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// is needed.
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class AudioMixerBase
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{
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public:
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// Do not change these unless underlying code changes.
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// This mixer has a hard-coded upper limit of 8 channels for output.
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static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
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static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
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static const uint16_t UNITY_GAIN_INT = 0x1000;
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static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
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enum { // names
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// setParameter targets
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TRACK = 0x3000,
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RESAMPLE = 0x3001,
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RAMP_VOLUME = 0x3002, // ramp to new volume
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VOLUME = 0x3003, // don't ramp
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TIMESTRETCH = 0x3004,
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// set Parameter names
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// for target TRACK
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CHANNEL_MASK = 0x4000,
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FORMAT = 0x4001,
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MAIN_BUFFER = 0x4002,
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AUX_BUFFER = 0x4003,
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// 0x4004 reserved
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MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
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MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
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// for target RESAMPLE
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SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
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// parameter 'value' is the new sample rate in Hz.
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// Only creates a sample rate converter the first time that
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// the track sample rate is different from the mix sample rate.
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// If the new sample rate is the same as the mix sample rate,
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// and a sample rate converter already exists,
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// then the sample rate converter remains present but is a no-op.
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RESET = 0x4101, // Reset sample rate converter without changing sample rate.
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// This clears out the resampler's input buffer.
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REMOVE = 0x4102, // Remove the sample rate converter on this track name;
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// the track is restored to the mix sample rate.
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// for target RAMP_VOLUME and VOLUME (8 channels max)
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// FIXME use float for these 3 to improve the dynamic range
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VOLUME0 = 0x4200,
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VOLUME1 = 0x4201,
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AUXLEVEL = 0x4210,
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};
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AudioMixerBase(size_t frameCount, uint32_t sampleRate)
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: mSampleRate(sampleRate)
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, mFrameCount(frameCount) {
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}
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virtual ~AudioMixerBase() {}
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virtual bool isValidFormat(audio_format_t format) const;
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virtual bool isValidChannelMask(audio_channel_mask_t channelMask) const;
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// Create a new track in the mixer.
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//
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// \param name a unique user-provided integer associated with the track.
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// If name already exists, the function will abort.
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// \param channelMask output channel mask.
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// \param format PCM format
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// \param sessionId Session id for the track. Tracks with the same
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// session id will be submixed together.
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//
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// \return OK on success.
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// BAD_VALUE if the format does not satisfy isValidFormat()
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// or the channelMask does not satisfy isValidChannelMask().
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status_t create(
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int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
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bool exists(int name) const {
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return mTracks.count(name) > 0;
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}
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// Free an allocated track by name.
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void destroy(int name);
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// Enable or disable an allocated track by name
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void enable(int name);
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void disable(int name);
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virtual void setParameter(int name, int target, int param, void *value);
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void process() {
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preProcess();
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(this->*mHook)();
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postProcess();
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}
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size_t getUnreleasedFrames(int name) const;
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std::string trackNames() const;
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protected:
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// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
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// original code will be used for stereo sinks, the new mixer for everything else.
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static constexpr bool kUseNewMixer = true;
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// Set kUseFloat to true to allow floating input into the mixer engine.
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// If kUseNewMixer is false, this is ignored or may be overridden internally
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static constexpr bool kUseFloat = true;
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#ifdef FLOAT_AUX
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using TYPE_AUX = float;
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static_assert(kUseNewMixer && kUseFloat,
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"kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
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#else
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using TYPE_AUX = int32_t; // q4.27
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#endif
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/* For multi-format functions (calls template functions
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* in AudioMixerOps.h). The template parameters are as follows:
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*
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* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
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* USEFLOATVOL (set to true if float volume is used)
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* ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
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* TO: int32_t (Q4.27) or float
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* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
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* TA: int32_t (Q4.27)
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*/
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enum {
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// FIXME this representation permits up to 8 channels
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NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
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};
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enum {
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NEEDS_CHANNEL_1 = 0x00000000, // mono
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NEEDS_CHANNEL_2 = 0x00000001, // stereo
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// sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
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NEEDS_MUTE = 0x00000100,
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NEEDS_RESAMPLE = 0x00001000,
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NEEDS_AUX = 0x00010000,
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};
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// hook types
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enum {
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PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
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};
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enum {
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TRACKTYPE_NOP,
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TRACKTYPE_RESAMPLE,
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TRACKTYPE_NORESAMPLE,
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TRACKTYPE_NORESAMPLEMONO,
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};
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// process hook functionality
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using process_hook_t = void(AudioMixerBase::*)();
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struct TrackBase;
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using hook_t = void(TrackBase::*)(
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int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
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struct TrackBase {
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TrackBase()
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: bufferProvider(nullptr)
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{
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// TODO: move additional initialization here.
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}
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virtual ~TrackBase() {}
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virtual uint32_t getOutputChannelCount() { return channelCount; }
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virtual uint32_t getMixerChannelCount() { return mMixerChannelCount; }
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bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
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bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
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bool doesResample() const { return mResampler.get() != nullptr; }
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void recreateResampler(uint32_t devSampleRate);
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void resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
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void adjustVolumeRamp(bool aux, bool useFloat = false);
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size_t getUnreleasedFrames() const { return mResampler.get() != nullptr ?
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mResampler->getUnreleasedFrames() : 0; };
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static hook_t getTrackHook(int trackType, uint32_t channelCount,
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audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
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void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
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typename TO, typename TI, typename TA>
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void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
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uint32_t needs;
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// TODO: Eventually remove legacy integer volume settings
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union {
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int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
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int32_t volumeRL;
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};
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int32_t prevVolume[MAX_NUM_VOLUMES];
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int32_t volumeInc[MAX_NUM_VOLUMES];
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int32_t auxInc;
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int32_t prevAuxLevel;
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int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
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uint16_t frameCount;
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uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
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uint8_t unused_padding; // formerly format, was always 16
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uint16_t enabled; // actually bool
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audio_channel_mask_t channelMask;
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// actual buffer provider used by the track hooks
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AudioBufferProvider* bufferProvider;
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mutable AudioBufferProvider::Buffer buffer; // 8 bytes
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hook_t hook;
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const void *mIn; // current location in buffer
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std::unique_ptr<AudioResampler> mResampler;
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uint32_t sampleRate;
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int32_t* mainBuffer;
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int32_t* auxBuffer;
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int32_t sessionId;
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audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
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audio_format_t mFormat; // input track format
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audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
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// each track must be converted to this format.
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float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
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float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
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float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment
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float mAuxLevel; // floating point set aux level
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float mPrevAuxLevel; // floating point prev aux level
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float mAuxInc; // floating point aux increment
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audio_channel_mask_t mMixerChannelMask;
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uint32_t mMixerChannelCount;
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protected:
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// hooks
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void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
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void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
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// multi-format track hooks
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template <int MIXTYPE, typename TO, typename TI, typename TA>
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void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
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template <int MIXTYPE, typename TO, typename TI, typename TA>
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void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
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};
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// preCreateTrack must create an instance of a proper TrackBase descendant.
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// postCreateTrack is called after filling out fields of TrackBase. It can
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// abort track creation by returning non-OK status. See the implementation
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// of create() for details.
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virtual std::shared_ptr<TrackBase> preCreateTrack();
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virtual status_t postCreateTrack(TrackBase *track __unused) { return OK; }
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// preProcess is called before the process hook, postProcess after,
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// see the implementation of process() method.
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virtual void preProcess() {}
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virtual void postProcess() {}
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virtual bool setChannelMasks(int name,
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audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
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// Called when track info changes and a new process hook should be determined.
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void invalidate() {
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mHook = &AudioMixerBase::process__validate;
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}
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void process__validate();
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void process__nop();
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void process__genericNoResampling();
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void process__genericResampling();
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void process__oneTrack16BitsStereoNoResampling();
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template <int MIXTYPE, typename TO, typename TI, typename TA>
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void process__noResampleOneTrack();
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static process_hook_t getProcessHook(int processType, uint32_t channelCount,
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audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
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static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
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void *in, audio_format_t mixerInFormat, size_t sampleCount);
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// initialization constants
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const uint32_t mSampleRate;
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const size_t mFrameCount;
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process_hook_t mHook = &AudioMixerBase::process__nop; // one of process__*, never nullptr
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// the size of the type (int32_t) should be the largest of all types supported
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// by the mixer.
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std::unique_ptr<int32_t[]> mOutputTemp;
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std::unique_ptr<int32_t[]> mResampleTemp;
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// track names grouped by main buffer, in no particular order of main buffer.
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// however names for a particular main buffer are in order (by construction).
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std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
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// track names that are enabled, in increasing order (by construction).
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std::vector<int /* name */> mEnabled;
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// track smart pointers, by name, in increasing order of name.
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std::map<int /* name */, std::shared_ptr<TrackBase>> mTracks;
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};
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} // namespace android
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#endif // ANDROID_AUDIO_MIXER_BASE_H
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