Split AudioMixer into the base part which doesn't rely on the framework and can be used in vendor code and the derived part which is intended to be used by Audioflinger. Test: A/B compare output from test scripts, manual testing on the phone Change-Id: I24c390f67f20baa8109902099359ca6e34eebcfdgugelfrei
Mikhail Naganov
5 years ago
parent
3b73e994a0
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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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