mirror of
https://github.com/ioacademy-jikim/multimedia
synced 2025-06-07 07:56:26 +00:00
471 lines
20 KiB
C++
471 lines
20 KiB
C++
/*
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**
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** Copyright 2007, 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_H
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#define ANDROID_AUDIO_MIXER_H
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#include <stdint.h>
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#include <sys/types.h>
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#include <utils/threads.h>
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#include <media/AudioBufferProvider.h>
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#include "AudioResampler.h"
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#include <hardware/audio_effect.h>
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#include <system/audio.h>
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#include <media/nbaio/NBLog.h>
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// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
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#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
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namespace android {
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// ----------------------------------------------------------------------------
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class AudioMixer
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{
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public:
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AudioMixer(size_t frameCount, uint32_t sampleRate,
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uint32_t maxNumTracks = MAX_NUM_TRACKS);
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/*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
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// This mixer has a hard-coded upper limit of 32 active track inputs.
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// Adding support for > 32 tracks would require more than simply changing this value.
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static const uint32_t MAX_NUM_TRACKS = 32;
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// maximum number of channels supported by the mixer
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// This mixer has a hard-coded upper limit of 8 channels for output.
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static const uint32_t MAX_NUM_CHANNELS = 8;
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static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
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// maximum number of channels supported for the content
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static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
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static const uint16_t UNITY_GAIN_INT = 0x1000;
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static const float UNITY_GAIN_FLOAT = 1.0f;
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enum { // names
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// track names (MAX_NUM_TRACKS units)
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TRACK0 = 0x1000,
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// 0x2000 is unused
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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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// 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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DOWNMIX_TYPE = 0X4004,
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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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// For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
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// Allocate a track name. Returns new track name if successful, -1 on failure.
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// The failure could be because of an invalid channelMask or format, or that
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// the track capacity of the mixer is exceeded.
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int getTrackName(audio_channel_mask_t channelMask,
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audio_format_t format, int sessionId);
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// Free an allocated track by name
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void deleteTrackName(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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void setParameter(int name, int target, int param, void *value);
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void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
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void process(int64_t pts);
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uint32_t trackNames() const { return mTrackNames; }
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size_t getUnreleasedFrames(int name) const;
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static inline bool isValidPcmTrackFormat(audio_format_t format) {
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return format == AUDIO_FORMAT_PCM_16_BIT ||
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format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
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format == AUDIO_FORMAT_PCM_32_BIT ||
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format == AUDIO_FORMAT_PCM_FLOAT;
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}
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private:
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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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struct state_t;
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struct track_t;
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class CopyBufferProvider;
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typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
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int32_t* aux);
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static const int BLOCKSIZE = 16; // 4 cache lines
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struct track_t {
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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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// 16-byte boundary
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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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// 16-byte boundary
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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, see DownmixerBufferProvider below
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// for how the Track buffer provider is wrapped by another one when dowmixing is required
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AudioBufferProvider* bufferProvider;
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// 16-byte boundary
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mutable AudioBufferProvider::Buffer buffer; // 8 bytes
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hook_t hook;
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const void* in; // current location in buffer
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// 16-byte boundary
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AudioResampler* resampler;
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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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// 16-byte boundary
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AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
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CopyBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting.
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CopyBufferProvider* downmixerBufferProvider; // wrapper for channel conversion.
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int32_t sessionId;
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// 16-byte boundary
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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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// 16-byte boundary
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audio_channel_mask_t mMixerChannelMask;
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uint32_t 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 resampler != NULL; }
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void resetResampler() { if (resampler != NULL) resampler->reset(); }
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void adjustVolumeRamp(bool aux, bool useFloat = false);
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size_t getUnreleasedFrames() const { return resampler != NULL ?
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resampler->getUnreleasedFrames() : 0; };
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};
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typedef void (*process_hook_t)(state_t* state, int64_t pts);
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// pad to 32-bytes to fill cache line
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struct state_t {
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uint32_t enabledTracks;
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uint32_t needsChanged;
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size_t frameCount;
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process_hook_t hook; // one of process__*, never NULL
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int32_t *outputTemp;
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int32_t *resampleTemp;
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NBLog::Writer* mLog;
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int32_t reserved[1];
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// FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
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track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
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};
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// Base AudioBufferProvider class used for DownMixerBufferProvider, RemixBufferProvider,
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// and ReformatBufferProvider.
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// It handles a private buffer for use in converting format or channel masks from the
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// input data to a form acceptable by the mixer.
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// TODO: Make a ResamplerBufferProvider when integers are entirely removed from the
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// processing pipeline.
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class CopyBufferProvider : public AudioBufferProvider {
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public:
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// Use a private buffer of bufferFrameCount frames (each frame is outputFrameSize bytes).
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// If bufferFrameCount is 0, no private buffer is created and in-place modification of
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// the upstream buffer provider's buffers is performed by copyFrames().
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CopyBufferProvider(size_t inputFrameSize, size_t outputFrameSize,
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size_t bufferFrameCount);
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virtual ~CopyBufferProvider();
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// Overrides AudioBufferProvider methods
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virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
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virtual void releaseBuffer(Buffer* buffer);
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// Other public methods
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// call this to release the buffer to the upstream provider.
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// treat it as an audio discontinuity for future samples.
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virtual void reset();
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// this function should be supplied by the derived class. It converts
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// #frames in the *src pointer to the *dst pointer. It is public because
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// some providers will allow this to work on arbitrary buffers outside
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// of the internal buffers.
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virtual void copyFrames(void *dst, const void *src, size_t frames) = 0;
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// set the upstream buffer provider. Consider calling "reset" before this function.
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void setBufferProvider(AudioBufferProvider *p) {
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mTrackBufferProvider = p;
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}
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protected:
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AudioBufferProvider* mTrackBufferProvider;
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const size_t mInputFrameSize;
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const size_t mOutputFrameSize;
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private:
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AudioBufferProvider::Buffer mBuffer;
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const size_t mLocalBufferFrameCount;
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void* mLocalBufferData;
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size_t mConsumed;
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};
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// DownmixerBufferProvider wraps a track AudioBufferProvider to provide
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// position dependent downmixing by an Audio Effect.
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class DownmixerBufferProvider : public CopyBufferProvider {
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public:
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DownmixerBufferProvider(audio_channel_mask_t inputChannelMask,
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audio_channel_mask_t outputChannelMask, audio_format_t format,
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uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount);
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virtual ~DownmixerBufferProvider();
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virtual void copyFrames(void *dst, const void *src, size_t frames);
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bool isValid() const { return mDownmixHandle != NULL; }
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static status_t init();
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static bool isMultichannelCapable() { return sIsMultichannelCapable; }
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protected:
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effect_handle_t mDownmixHandle;
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effect_config_t mDownmixConfig;
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// effect descriptor for the downmixer used by the mixer
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static effect_descriptor_t sDwnmFxDesc;
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// indicates whether a downmix effect has been found and is usable by this mixer
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static bool sIsMultichannelCapable;
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// FIXME: should we allow effects outside of the framework?
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// We need to here. A special ioId that must be <= -2 so it does not map to a session.
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static const int32_t SESSION_ID_INVALID_AND_IGNORED = -2;
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};
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// RemixBufferProvider wraps a track AudioBufferProvider to perform an
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// upmix or downmix to the proper channel count and mask.
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class RemixBufferProvider : public CopyBufferProvider {
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public:
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RemixBufferProvider(audio_channel_mask_t inputChannelMask,
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audio_channel_mask_t outputChannelMask, audio_format_t format,
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size_t bufferFrameCount);
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virtual void copyFrames(void *dst, const void *src, size_t frames);
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protected:
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const audio_format_t mFormat;
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const size_t mSampleSize;
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const size_t mInputChannels;
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const size_t mOutputChannels;
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int8_t mIdxAry[sizeof(uint32_t)*8]; // 32 bits => channel indices
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};
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// ReformatBufferProvider wraps a track AudioBufferProvider to convert the input data
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// to an acceptable mixer input format type.
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class ReformatBufferProvider : public CopyBufferProvider {
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public:
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ReformatBufferProvider(int32_t channels,
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audio_format_t inputFormat, audio_format_t outputFormat,
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size_t bufferFrameCount);
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virtual void copyFrames(void *dst, const void *src, size_t frames);
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protected:
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const int32_t mChannels;
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const audio_format_t mInputFormat;
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const audio_format_t mOutputFormat;
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};
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// bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
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uint32_t mTrackNames;
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// bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
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// but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
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const uint32_t mConfiguredNames;
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const uint32_t mSampleRate;
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NBLog::Writer mDummyLog;
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public:
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void setLog(NBLog::Writer* log);
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private:
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state_t mState __attribute__((aligned(32)));
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// Call after changing either the enabled status of a track, or parameters of an enabled track.
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// OK to call more often than that, but unnecessary.
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void invalidateState(uint32_t mask);
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bool setChannelMasks(int name,
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audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
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// TODO: remove unused trackName/trackNum from functions below.
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static status_t initTrackDownmix(track_t* pTrack, int trackName);
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static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
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static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
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static status_t prepareTrackForReformat(track_t* pTrack, int trackNum);
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static void unprepareTrackForReformat(track_t* pTrack, int trackName);
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static void reconfigureBufferProviders(track_t* pTrack);
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static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
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int32_t* aux);
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static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
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static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
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int32_t* aux);
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static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
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int32_t* aux);
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static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
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int32_t* aux);
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static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
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int32_t* aux);
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static void process__validate(state_t* state, int64_t pts);
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static void process__nop(state_t* state, int64_t pts);
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static void process__genericNoResampling(state_t* state, int64_t pts);
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static void process__genericResampling(state_t* state, int64_t pts);
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static void process__OneTrack16BitsStereoNoResampling(state_t* state,
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int64_t pts);
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static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
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int outputFrameIndex);
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static uint64_t sLocalTimeFreq;
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static pthread_once_t sOnceControl;
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static void sInitRoutine();
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/* multi-format volume mixing function (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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template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
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typename TO, typename TI, typename TA>
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static void volumeMix(TO *out, size_t outFrames,
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const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
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// multi-format process hooks
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template <int MIXTYPE, typename TO, typename TI, typename TA>
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static void process_NoResampleOneTrack(state_t* state, int64_t pts);
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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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static void track__Resample(track_t* t, TO* out, size_t frameCount,
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TO* temp __unused, TA* aux);
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template <int MIXTYPE, typename TO, typename TI, typename TA>
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static void track__NoResample(track_t* t, TO* out, size_t frameCount,
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TO* temp __unused, TA* aux);
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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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// hook types
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enum {
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PROCESSTYPE_NORESAMPLEONETRACK,
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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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// functions for determining the proper process and track hooks.
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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 hook_t getTrackHook(int trackType, uint32_t channelCount,
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audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
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};
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// ----------------------------------------------------------------------------
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}; // namespace android
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#endif // ANDROID_AUDIO_MIXER_H
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