1196 lines
38 KiB
C++
1196 lines
38 KiB
C++
/* //device/include/server/AudioFlinger/AudioMixer.cpp
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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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#define LOG_TAG "AudioMixer"
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//#define LOG_NDEBUG 0
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#include <stdint.h>
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#include <string.h>
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#include <stdlib.h>
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#include <sys/types.h>
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#include <utils/Errors.h>
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#include <utils/Log.h>
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#include "AudioMixer.h"
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namespace android {
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// ----------------------------------------------------------------------------
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static inline int16_t clamp16(int32_t sample)
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{
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if ((sample>>15) ^ (sample>>31))
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sample = 0x7FFF ^ (sample>>31);
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return sample;
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}
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// ----------------------------------------------------------------------------
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AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
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: mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
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{
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mState.enabledTracks= 0;
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mState.needsChanged = 0;
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mState.frameCount = frameCount;
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mState.outputTemp = 0;
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mState.resampleTemp = 0;
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mState.hook = process__nop;
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track_t* t = mState.tracks;
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for (int i=0 ; i<32 ; i++) {
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t->needs = 0;
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t->volume[0] = UNITY_GAIN;
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t->volume[1] = UNITY_GAIN;
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t->volumeInc[0] = 0;
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t->volumeInc[1] = 0;
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t->auxLevel = 0;
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t->auxInc = 0;
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t->channelCount = 2;
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t->enabled = 0;
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t->format = 16;
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t->buffer.raw = 0;
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t->bufferProvider = 0;
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t->hook = 0;
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t->resampler = 0;
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t->sampleRate = mSampleRate;
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t->in = 0;
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t->mainBuffer = NULL;
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t->auxBuffer = NULL;
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t++;
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}
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}
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AudioMixer::~AudioMixer()
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{
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track_t* t = mState.tracks;
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for (int i=0 ; i<32 ; i++) {
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delete t->resampler;
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t++;
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}
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delete [] mState.outputTemp;
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delete [] mState.resampleTemp;
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}
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int AudioMixer::getTrackName()
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{
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uint32_t names = mTrackNames;
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uint32_t mask = 1;
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int n = 0;
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while (names & mask) {
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mask <<= 1;
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n++;
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}
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if (mask) {
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LOGV("add track (%d)", n);
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mTrackNames |= mask;
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return TRACK0 + n;
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}
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return -1;
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}
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void AudioMixer::invalidateState(uint32_t mask)
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{
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if (mask) {
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mState.needsChanged |= mask;
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mState.hook = process__validate;
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}
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}
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void AudioMixer::deleteTrackName(int name)
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{
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name -= TRACK0;
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if (uint32_t(name) < MAX_NUM_TRACKS) {
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LOGV("deleteTrackName(%d)", name);
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track_t& track(mState.tracks[ name ]);
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if (track.enabled != 0) {
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track.enabled = 0;
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invalidateState(1<<name);
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}
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if (track.resampler) {
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// delete the resampler
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delete track.resampler;
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track.resampler = 0;
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track.sampleRate = mSampleRate;
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invalidateState(1<<name);
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}
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track.volumeInc[0] = 0;
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track.volumeInc[1] = 0;
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mTrackNames &= ~(1<<name);
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}
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}
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status_t AudioMixer::enable(int name)
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{
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switch (name) {
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case MIXING: {
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if (mState.tracks[ mActiveTrack ].enabled != 1) {
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mState.tracks[ mActiveTrack ].enabled = 1;
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LOGV("enable(%d)", mActiveTrack);
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invalidateState(1<<mActiveTrack);
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}
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} break;
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default:
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return NAME_NOT_FOUND;
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}
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return NO_ERROR;
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}
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status_t AudioMixer::disable(int name)
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{
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switch (name) {
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case MIXING: {
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if (mState.tracks[ mActiveTrack ].enabled != 0) {
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mState.tracks[ mActiveTrack ].enabled = 0;
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LOGV("disable(%d)", mActiveTrack);
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invalidateState(1<<mActiveTrack);
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}
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} break;
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default:
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return NAME_NOT_FOUND;
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}
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return NO_ERROR;
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}
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status_t AudioMixer::setActiveTrack(int track)
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{
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if (uint32_t(track-TRACK0) >= MAX_NUM_TRACKS) {
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return BAD_VALUE;
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}
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mActiveTrack = track - TRACK0;
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return NO_ERROR;
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}
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status_t AudioMixer::setParameter(int target, int name, void *value)
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{
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int valueInt = (int)value;
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int32_t *valueBuf = (int32_t *)value;
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switch (target) {
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case TRACK:
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if (name == CHANNEL_COUNT) {
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if ((uint32_t(valueInt) <= MAX_NUM_CHANNELS) && (valueInt)) {
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if (mState.tracks[ mActiveTrack ].channelCount != valueInt) {
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mState.tracks[ mActiveTrack ].channelCount = valueInt;
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LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", valueInt);
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invalidateState(1<<mActiveTrack);
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}
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return NO_ERROR;
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}
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}
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if (name == MAIN_BUFFER) {
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if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) {
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mState.tracks[ mActiveTrack ].mainBuffer = valueBuf;
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LOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
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invalidateState(1<<mActiveTrack);
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}
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return NO_ERROR;
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}
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if (name == AUX_BUFFER) {
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if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) {
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mState.tracks[ mActiveTrack ].auxBuffer = valueBuf;
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LOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
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invalidateState(1<<mActiveTrack);
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}
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return NO_ERROR;
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}
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break;
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case RESAMPLE:
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if (name == SAMPLE_RATE) {
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if (valueInt > 0) {
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track_t& track = mState.tracks[ mActiveTrack ];
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if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
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LOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
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uint32_t(valueInt));
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invalidateState(1<<mActiveTrack);
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}
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return NO_ERROR;
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}
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}
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break;
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case RAMP_VOLUME:
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case VOLUME:
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if ((uint32_t(name-VOLUME0) < MAX_NUM_CHANNELS)) {
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track_t& track = mState.tracks[ mActiveTrack ];
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if (track.volume[name-VOLUME0] != valueInt) {
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LOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
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track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16;
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track.volume[name-VOLUME0] = valueInt;
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if (target == VOLUME) {
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track.prevVolume[name-VOLUME0] = valueInt << 16;
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track.volumeInc[name-VOLUME0] = 0;
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} else {
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int32_t d = (valueInt<<16) - track.prevVolume[name-VOLUME0];
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int32_t volInc = d / int32_t(mState.frameCount);
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track.volumeInc[name-VOLUME0] = volInc;
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if (volInc == 0) {
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track.prevVolume[name-VOLUME0] = valueInt << 16;
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}
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}
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invalidateState(1<<mActiveTrack);
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}
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return NO_ERROR;
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} else if (name == AUXLEVEL) {
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track_t& track = mState.tracks[ mActiveTrack ];
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if (track.auxLevel != valueInt) {
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LOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
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track.prevAuxLevel = track.auxLevel << 16;
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track.auxLevel = valueInt;
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if (target == VOLUME) {
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track.prevAuxLevel = valueInt << 16;
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track.auxInc = 0;
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} else {
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int32_t d = (valueInt<<16) - track.prevAuxLevel;
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int32_t volInc = d / int32_t(mState.frameCount);
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track.auxInc = volInc;
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if (volInc == 0) {
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track.prevAuxLevel = valueInt << 16;
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}
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}
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invalidateState(1<<mActiveTrack);
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}
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return NO_ERROR;
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}
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break;
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}
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return BAD_VALUE;
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}
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bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
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{
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if (value!=devSampleRate || resampler) {
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if (sampleRate != value) {
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sampleRate = value;
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if (resampler == 0) {
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resampler = AudioResampler::create(
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format, channelCount, devSampleRate);
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}
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return true;
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}
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}
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return false;
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}
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bool AudioMixer::track_t::doesResample() const
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{
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return resampler != 0;
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}
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inline
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void AudioMixer::track_t::adjustVolumeRamp(bool aux)
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{
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for (int i=0 ; i<2 ; i++) {
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if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
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((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
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volumeInc[i] = 0;
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prevVolume[i] = volume[i]<<16;
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}
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}
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if (aux) {
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if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
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((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
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auxInc = 0;
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prevAuxLevel = auxLevel<<16;
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}
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}
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}
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status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
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{
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mState.tracks[ mActiveTrack ].bufferProvider = buffer;
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return NO_ERROR;
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}
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void AudioMixer::process()
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{
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mState.hook(&mState);
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}
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void AudioMixer::process__validate(state_t* state)
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{
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LOGW_IF(!state->needsChanged,
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"in process__validate() but nothing's invalid");
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uint32_t changed = state->needsChanged;
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state->needsChanged = 0; // clear the validation flag
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// recompute which tracks are enabled / disabled
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uint32_t enabled = 0;
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uint32_t disabled = 0;
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while (changed) {
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const int i = 31 - __builtin_clz(changed);
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const uint32_t mask = 1<<i;
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changed &= ~mask;
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track_t& t = state->tracks[i];
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(t.enabled ? enabled : disabled) |= mask;
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}
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state->enabledTracks &= ~disabled;
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state->enabledTracks |= enabled;
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// compute everything we need...
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int countActiveTracks = 0;
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int all16BitsStereoNoResample = 1;
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int resampling = 0;
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int volumeRamp = 0;
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uint32_t en = state->enabledTracks;
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while (en) {
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const int i = 31 - __builtin_clz(en);
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en &= ~(1<<i);
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countActiveTracks++;
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track_t& t = state->tracks[i];
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uint32_t n = 0;
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n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
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n |= NEEDS_FORMAT_16;
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n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED;
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if (t.auxLevel != 0 && t.auxBuffer != NULL) {
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n |= NEEDS_AUX_ENABLED;
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}
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if (t.volumeInc[0]|t.volumeInc[1]) {
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volumeRamp = 1;
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} else if (!t.doesResample() && t.volumeRL == 0) {
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n |= NEEDS_MUTE_ENABLED;
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}
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t.needs = n;
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if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) {
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t.hook = track__nop;
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} else {
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if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
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all16BitsStereoNoResample = 0;
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}
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if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
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all16BitsStereoNoResample = 0;
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resampling = 1;
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t.hook = track__genericResample;
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} else {
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if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
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t.hook = track__16BitsMono;
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all16BitsStereoNoResample = 0;
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}
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if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_2){
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t.hook = track__16BitsStereo;
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}
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}
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}
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}
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// select the processing hooks
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state->hook = process__nop;
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if (countActiveTracks) {
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if (resampling) {
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if (!state->outputTemp) {
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state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
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}
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if (!state->resampleTemp) {
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state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
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}
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state->hook = process__genericResampling;
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} else {
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if (state->outputTemp) {
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delete [] state->outputTemp;
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state->outputTemp = 0;
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}
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if (state->resampleTemp) {
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delete [] state->resampleTemp;
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state->resampleTemp = 0;
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}
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state->hook = process__genericNoResampling;
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if (all16BitsStereoNoResample && !volumeRamp) {
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if (countActiveTracks == 1) {
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state->hook = process__OneTrack16BitsStereoNoResampling;
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}
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}
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}
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}
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LOGV("mixer configuration change: %d activeTracks (%08x) "
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"all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
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countActiveTracks, state->enabledTracks,
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all16BitsStereoNoResample, resampling, volumeRamp);
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state->hook(state);
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// Now that the volume ramp has been done, set optimal state and
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// track hooks for subsequent mixer process
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if (countActiveTracks) {
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int allMuted = 1;
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uint32_t en = state->enabledTracks;
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while (en) {
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const int i = 31 - __builtin_clz(en);
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en &= ~(1<<i);
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track_t& t = state->tracks[i];
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if (!t.doesResample() && t.volumeRL == 0)
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{
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t.needs |= NEEDS_MUTE_ENABLED;
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t.hook = track__nop;
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} else {
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allMuted = 0;
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}
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}
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if (allMuted) {
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state->hook = process__nop;
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} else if (all16BitsStereoNoResample) {
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if (countActiveTracks == 1) {
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state->hook = process__OneTrack16BitsStereoNoResampling;
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}
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}
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}
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}
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static inline
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int32_t mulAdd(int16_t in, int16_t v, int32_t a)
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{
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#if defined(__arm__) && !defined(__thumb__)
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int32_t out;
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asm( "smlabb %[out], %[in], %[v], %[a] \n"
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: [out]"=r"(out)
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: [in]"%r"(in), [v]"r"(v), [a]"r"(a)
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: );
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return out;
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#else
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return a + in * int32_t(v);
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#endif
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}
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static inline
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int32_t mul(int16_t in, int16_t v)
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{
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#if defined(__arm__) && !defined(__thumb__)
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int32_t out;
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asm( "smulbb %[out], %[in], %[v] \n"
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: [out]"=r"(out)
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: [in]"%r"(in), [v]"r"(v)
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: );
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return out;
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#else
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return in * int32_t(v);
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#endif
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}
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static inline
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int32_t mulAddRL(int left, uint32_t inRL, uint32_t vRL, int32_t a)
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{
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#if defined(__arm__) && !defined(__thumb__)
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int32_t out;
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if (left) {
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asm( "smlabb %[out], %[inRL], %[vRL], %[a] \n"
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: [out]"=r"(out)
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: [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
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: );
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} else {
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asm( "smlatt %[out], %[inRL], %[vRL], %[a] \n"
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: [out]"=r"(out)
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: [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
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: );
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}
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return out;
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#else
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if (left) {
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return a + int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
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} else {
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return a + int16_t(inRL>>16) * int16_t(vRL>>16);
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}
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#endif
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}
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static inline
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int32_t mulRL(int left, uint32_t inRL, uint32_t vRL)
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{
|
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#if defined(__arm__) && !defined(__thumb__)
|
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int32_t out;
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if (left) {
|
|
asm( "smulbb %[out], %[inRL], %[vRL] \n"
|
|
: [out]"=r"(out)
|
|
: [inRL]"%r"(inRL), [vRL]"r"(vRL)
|
|
: );
|
|
} else {
|
|
asm( "smultt %[out], %[inRL], %[vRL] \n"
|
|
: [out]"=r"(out)
|
|
: [inRL]"%r"(inRL), [vRL]"r"(vRL)
|
|
: );
|
|
}
|
|
return out;
|
|
#else
|
|
if (left) {
|
|
return int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
|
|
} else {
|
|
return int16_t(inRL>>16) * int16_t(vRL>>16);
|
|
}
|
|
#endif
|
|
}
|
|
|
|
|
|
void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
|
|
{
|
|
t->resampler->setSampleRate(t->sampleRate);
|
|
|
|
// ramp gain - resample to temp buffer and scale/mix in 2nd step
|
|
if (aux != NULL) {
|
|
// always resample with unity gain when sending to auxiliary buffer to be able
|
|
// to apply send level after resampling
|
|
// TODO: modify each resampler to support aux channel?
|
|
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
|
|
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
|
|
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
|
|
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
|
|
volumeRampStereo(t, out, outFrameCount, temp, aux);
|
|
} else {
|
|
volumeStereo(t, out, outFrameCount, temp, aux);
|
|
}
|
|
} else {
|
|
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
|
|
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
|
|
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
|
|
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
|
|
volumeRampStereo(t, out, outFrameCount, temp, aux);
|
|
}
|
|
|
|
// constant gain
|
|
else {
|
|
t->resampler->setVolume(t->volume[0], t->volume[1]);
|
|
t->resampler->resample(out, outFrameCount, t->bufferProvider);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
|
|
{
|
|
}
|
|
|
|
void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
|
|
{
|
|
int32_t vl = t->prevVolume[0];
|
|
int32_t vr = t->prevVolume[1];
|
|
const int32_t vlInc = t->volumeInc[0];
|
|
const int32_t vrInc = t->volumeInc[1];
|
|
|
|
//LOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
|
|
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
|
|
// (vl + vlInc*frameCount)/65536.0f, frameCount);
|
|
|
|
// ramp volume
|
|
if UNLIKELY(aux != NULL) {
|
|
int32_t va = t->prevAuxLevel;
|
|
const int32_t vaInc = t->auxInc;
|
|
int32_t l;
|
|
int32_t r;
|
|
|
|
do {
|
|
l = (*temp++ >> 12);
|
|
r = (*temp++ >> 12);
|
|
*out++ += (vl >> 16) * l;
|
|
*out++ += (vr >> 16) * r;
|
|
*aux++ += (va >> 17) * (l + r);
|
|
vl += vlInc;
|
|
vr += vrInc;
|
|
va += vaInc;
|
|
} while (--frameCount);
|
|
t->prevAuxLevel = va;
|
|
} else {
|
|
do {
|
|
*out++ += (vl >> 16) * (*temp++ >> 12);
|
|
*out++ += (vr >> 16) * (*temp++ >> 12);
|
|
vl += vlInc;
|
|
vr += vrInc;
|
|
} while (--frameCount);
|
|
}
|
|
t->prevVolume[0] = vl;
|
|
t->prevVolume[1] = vr;
|
|
t->adjustVolumeRamp((aux != NULL));
|
|
}
|
|
|
|
void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
|
|
{
|
|
const int16_t vl = t->volume[0];
|
|
const int16_t vr = t->volume[1];
|
|
|
|
if UNLIKELY(aux != NULL) {
|
|
const int16_t va = (int16_t)t->auxLevel;
|
|
do {
|
|
int16_t l = (int16_t)(*temp++ >> 12);
|
|
int16_t r = (int16_t)(*temp++ >> 12);
|
|
out[0] = mulAdd(l, vl, out[0]);
|
|
int16_t a = (int16_t)(((int32_t)l + r) >> 1);
|
|
out[1] = mulAdd(r, vr, out[1]);
|
|
out += 2;
|
|
aux[0] = mulAdd(a, va, aux[0]);
|
|
aux++;
|
|
} while (--frameCount);
|
|
} else {
|
|
do {
|
|
int16_t l = (int16_t)(*temp++ >> 12);
|
|
int16_t r = (int16_t)(*temp++ >> 12);
|
|
out[0] = mulAdd(l, vl, out[0]);
|
|
out[1] = mulAdd(r, vr, out[1]);
|
|
out += 2;
|
|
} while (--frameCount);
|
|
}
|
|
}
|
|
|
|
void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
|
|
{
|
|
int16_t const *in = static_cast<int16_t const *>(t->in);
|
|
|
|
if UNLIKELY(aux != NULL) {
|
|
int32_t l;
|
|
int32_t r;
|
|
// ramp gain
|
|
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
|
|
int32_t vl = t->prevVolume[0];
|
|
int32_t vr = t->prevVolume[1];
|
|
int32_t va = t->prevAuxLevel;
|
|
const int32_t vlInc = t->volumeInc[0];
|
|
const int32_t vrInc = t->volumeInc[1];
|
|
const int32_t vaInc = t->auxInc;
|
|
// LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
|
|
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
|
|
// (vl + vlInc*frameCount)/65536.0f, frameCount);
|
|
|
|
do {
|
|
l = (int32_t)*in++;
|
|
r = (int32_t)*in++;
|
|
*out++ += (vl >> 16) * l;
|
|
*out++ += (vr >> 16) * r;
|
|
*aux++ += (va >> 17) * (l + r);
|
|
vl += vlInc;
|
|
vr += vrInc;
|
|
va += vaInc;
|
|
} while (--frameCount);
|
|
|
|
t->prevVolume[0] = vl;
|
|
t->prevVolume[1] = vr;
|
|
t->prevAuxLevel = va;
|
|
t->adjustVolumeRamp(true);
|
|
}
|
|
|
|
// constant gain
|
|
else {
|
|
const uint32_t vrl = t->volumeRL;
|
|
const int16_t va = (int16_t)t->auxLevel;
|
|
do {
|
|
uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
|
|
int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
|
|
in += 2;
|
|
out[0] = mulAddRL(1, rl, vrl, out[0]);
|
|
out[1] = mulAddRL(0, rl, vrl, out[1]);
|
|
out += 2;
|
|
aux[0] = mulAdd(a, va, aux[0]);
|
|
aux++;
|
|
} while (--frameCount);
|
|
}
|
|
} else {
|
|
// ramp gain
|
|
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
|
|
int32_t vl = t->prevVolume[0];
|
|
int32_t vr = t->prevVolume[1];
|
|
const int32_t vlInc = t->volumeInc[0];
|
|
const int32_t vrInc = t->volumeInc[1];
|
|
|
|
// LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
|
|
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
|
|
// (vl + vlInc*frameCount)/65536.0f, frameCount);
|
|
|
|
do {
|
|
*out++ += (vl >> 16) * (int32_t) *in++;
|
|
*out++ += (vr >> 16) * (int32_t) *in++;
|
|
vl += vlInc;
|
|
vr += vrInc;
|
|
} while (--frameCount);
|
|
|
|
t->prevVolume[0] = vl;
|
|
t->prevVolume[1] = vr;
|
|
t->adjustVolumeRamp(false);
|
|
}
|
|
|
|
// constant gain
|
|
else {
|
|
const uint32_t vrl = t->volumeRL;
|
|
do {
|
|
uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
|
|
in += 2;
|
|
out[0] = mulAddRL(1, rl, vrl, out[0]);
|
|
out[1] = mulAddRL(0, rl, vrl, out[1]);
|
|
out += 2;
|
|
} while (--frameCount);
|
|
}
|
|
}
|
|
t->in = in;
|
|
}
|
|
|
|
void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
|
|
{
|
|
int16_t const *in = static_cast<int16_t const *>(t->in);
|
|
|
|
if UNLIKELY(aux != NULL) {
|
|
// ramp gain
|
|
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
|
|
int32_t vl = t->prevVolume[0];
|
|
int32_t vr = t->prevVolume[1];
|
|
int32_t va = t->prevAuxLevel;
|
|
const int32_t vlInc = t->volumeInc[0];
|
|
const int32_t vrInc = t->volumeInc[1];
|
|
const int32_t vaInc = t->auxInc;
|
|
|
|
// LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
|
|
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
|
|
// (vl + vlInc*frameCount)/65536.0f, frameCount);
|
|
|
|
do {
|
|
int32_t l = *in++;
|
|
*out++ += (vl >> 16) * l;
|
|
*out++ += (vr >> 16) * l;
|
|
*aux++ += (va >> 16) * l;
|
|
vl += vlInc;
|
|
vr += vrInc;
|
|
va += vaInc;
|
|
} while (--frameCount);
|
|
|
|
t->prevVolume[0] = vl;
|
|
t->prevVolume[1] = vr;
|
|
t->prevAuxLevel = va;
|
|
t->adjustVolumeRamp(true);
|
|
}
|
|
// constant gain
|
|
else {
|
|
const int16_t vl = t->volume[0];
|
|
const int16_t vr = t->volume[1];
|
|
const int16_t va = (int16_t)t->auxLevel;
|
|
do {
|
|
int16_t l = *in++;
|
|
out[0] = mulAdd(l, vl, out[0]);
|
|
out[1] = mulAdd(l, vr, out[1]);
|
|
out += 2;
|
|
aux[0] = mulAdd(l, va, aux[0]);
|
|
aux++;
|
|
} while (--frameCount);
|
|
}
|
|
} else {
|
|
// ramp gain
|
|
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
|
|
int32_t vl = t->prevVolume[0];
|
|
int32_t vr = t->prevVolume[1];
|
|
const int32_t vlInc = t->volumeInc[0];
|
|
const int32_t vrInc = t->volumeInc[1];
|
|
|
|
// LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
|
|
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
|
|
// (vl + vlInc*frameCount)/65536.0f, frameCount);
|
|
|
|
do {
|
|
int32_t l = *in++;
|
|
*out++ += (vl >> 16) * l;
|
|
*out++ += (vr >> 16) * l;
|
|
vl += vlInc;
|
|
vr += vrInc;
|
|
} while (--frameCount);
|
|
|
|
t->prevVolume[0] = vl;
|
|
t->prevVolume[1] = vr;
|
|
t->adjustVolumeRamp(false);
|
|
}
|
|
// constant gain
|
|
else {
|
|
const int16_t vl = t->volume[0];
|
|
const int16_t vr = t->volume[1];
|
|
do {
|
|
int16_t l = *in++;
|
|
out[0] = mulAdd(l, vl, out[0]);
|
|
out[1] = mulAdd(l, vr, out[1]);
|
|
out += 2;
|
|
} while (--frameCount);
|
|
}
|
|
}
|
|
t->in = in;
|
|
}
|
|
|
|
void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
|
|
{
|
|
for (size_t i=0 ; i<c ; i++) {
|
|
int32_t l = *sums++;
|
|
int32_t r = *sums++;
|
|
int32_t nl = l >> 12;
|
|
int32_t nr = r >> 12;
|
|
l = clamp16(nl);
|
|
r = clamp16(nr);
|
|
*out++ = (r<<16) | (l & 0xFFFF);
|
|
}
|
|
}
|
|
|
|
// no-op case
|
|
void AudioMixer::process__nop(state_t* state)
|
|
{
|
|
uint32_t e0 = state->enabledTracks;
|
|
size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
|
|
while (e0) {
|
|
// process by group of tracks with same output buffer to
|
|
// avoid multiple memset() on same buffer
|
|
uint32_t e1 = e0, e2 = e0;
|
|
int i = 31 - __builtin_clz(e1);
|
|
track_t& t1 = state->tracks[i];
|
|
e2 &= ~(1<<i);
|
|
while (e2) {
|
|
i = 31 - __builtin_clz(e2);
|
|
e2 &= ~(1<<i);
|
|
track_t& t2 = state->tracks[i];
|
|
if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
|
|
e1 &= ~(1<<i);
|
|
}
|
|
}
|
|
e0 &= ~(e1);
|
|
|
|
memset(t1.mainBuffer, 0, bufSize);
|
|
|
|
while (e1) {
|
|
i = 31 - __builtin_clz(e1);
|
|
e1 &= ~(1<<i);
|
|
t1 = state->tracks[i];
|
|
size_t outFrames = state->frameCount;
|
|
while (outFrames) {
|
|
t1.buffer.frameCount = outFrames;
|
|
t1.bufferProvider->getNextBuffer(&t1.buffer);
|
|
if (!t1.buffer.raw) break;
|
|
outFrames -= t1.buffer.frameCount;
|
|
t1.bufferProvider->releaseBuffer(&t1.buffer);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// generic code without resampling
|
|
void AudioMixer::process__genericNoResampling(state_t* state)
|
|
{
|
|
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
|
|
|
|
// acquire each track's buffer
|
|
uint32_t enabledTracks = state->enabledTracks;
|
|
uint32_t e0 = enabledTracks;
|
|
while (e0) {
|
|
const int i = 31 - __builtin_clz(e0);
|
|
e0 &= ~(1<<i);
|
|
track_t& t = state->tracks[i];
|
|
t.buffer.frameCount = state->frameCount;
|
|
t.bufferProvider->getNextBuffer(&t.buffer);
|
|
t.frameCount = t.buffer.frameCount;
|
|
t.in = t.buffer.raw;
|
|
// t.in == NULL can happen if the track was flushed just after having
|
|
// been enabled for mixing.
|
|
if (t.in == NULL)
|
|
enabledTracks &= ~(1<<i);
|
|
}
|
|
|
|
e0 = enabledTracks;
|
|
while (e0) {
|
|
// process by group of tracks with same output buffer to
|
|
// optimize cache use
|
|
uint32_t e1 = e0, e2 = e0;
|
|
int j = 31 - __builtin_clz(e1);
|
|
track_t& t1 = state->tracks[j];
|
|
e2 &= ~(1<<j);
|
|
while (e2) {
|
|
j = 31 - __builtin_clz(e2);
|
|
e2 &= ~(1<<j);
|
|
track_t& t2 = state->tracks[j];
|
|
if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
|
|
e1 &= ~(1<<j);
|
|
}
|
|
}
|
|
e0 &= ~(e1);
|
|
// this assumes output 16 bits stereo, no resampling
|
|
int32_t *out = t1.mainBuffer;
|
|
size_t numFrames = 0;
|
|
do {
|
|
memset(outTemp, 0, sizeof(outTemp));
|
|
e2 = e1;
|
|
while (e2) {
|
|
const int i = 31 - __builtin_clz(e2);
|
|
e2 &= ~(1<<i);
|
|
track_t& t = state->tracks[i];
|
|
size_t outFrames = BLOCKSIZE;
|
|
int32_t *aux = NULL;
|
|
if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
|
|
aux = t.auxBuffer + numFrames;
|
|
}
|
|
while (outFrames) {
|
|
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
|
|
if (inFrames) {
|
|
(t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
|
|
t.frameCount -= inFrames;
|
|
outFrames -= inFrames;
|
|
if UNLIKELY(aux != NULL) {
|
|
aux += inFrames;
|
|
}
|
|
}
|
|
if (t.frameCount == 0 && outFrames) {
|
|
t.bufferProvider->releaseBuffer(&t.buffer);
|
|
t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
|
|
t.bufferProvider->getNextBuffer(&t.buffer);
|
|
t.in = t.buffer.raw;
|
|
if (t.in == NULL) {
|
|
enabledTracks &= ~(1<<i);
|
|
e1 &= ~(1<<i);
|
|
break;
|
|
}
|
|
t.frameCount = t.buffer.frameCount;
|
|
}
|
|
}
|
|
}
|
|
ditherAndClamp(out, outTemp, BLOCKSIZE);
|
|
out += BLOCKSIZE;
|
|
numFrames += BLOCKSIZE;
|
|
} while (numFrames < state->frameCount);
|
|
}
|
|
|
|
// release each track's buffer
|
|
e0 = enabledTracks;
|
|
while (e0) {
|
|
const int i = 31 - __builtin_clz(e0);
|
|
e0 &= ~(1<<i);
|
|
track_t& t = state->tracks[i];
|
|
t.bufferProvider->releaseBuffer(&t.buffer);
|
|
}
|
|
}
|
|
|
|
|
|
// generic code with resampling
|
|
void AudioMixer::process__genericResampling(state_t* state)
|
|
{
|
|
int32_t* const outTemp = state->outputTemp;
|
|
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
|
|
memset(outTemp, 0, size);
|
|
|
|
size_t numFrames = state->frameCount;
|
|
|
|
uint32_t e0 = state->enabledTracks;
|
|
while (e0) {
|
|
// process by group of tracks with same output buffer
|
|
// to optimize cache use
|
|
uint32_t e1 = e0, e2 = e0;
|
|
int j = 31 - __builtin_clz(e1);
|
|
track_t& t1 = state->tracks[j];
|
|
e2 &= ~(1<<j);
|
|
while (e2) {
|
|
j = 31 - __builtin_clz(e2);
|
|
e2 &= ~(1<<j);
|
|
track_t& t2 = state->tracks[j];
|
|
if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
|
|
e1 &= ~(1<<j);
|
|
}
|
|
}
|
|
e0 &= ~(e1);
|
|
int32_t *out = t1.mainBuffer;
|
|
while (e1) {
|
|
const int i = 31 - __builtin_clz(e1);
|
|
e1 &= ~(1<<i);
|
|
track_t& t = state->tracks[i];
|
|
int32_t *aux = NULL;
|
|
if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
|
|
aux = t.auxBuffer;
|
|
}
|
|
|
|
// this is a little goofy, on the resampling case we don't
|
|
// acquire/release the buffers because it's done by
|
|
// the resampler.
|
|
if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
|
|
(t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux);
|
|
} else {
|
|
|
|
size_t outFrames = 0;
|
|
|
|
while (outFrames < numFrames) {
|
|
t.buffer.frameCount = numFrames - outFrames;
|
|
t.bufferProvider->getNextBuffer(&t.buffer);
|
|
t.in = t.buffer.raw;
|
|
// t.in == NULL can happen if the track was flushed just after having
|
|
// been enabled for mixing.
|
|
if (t.in == NULL) break;
|
|
|
|
if UNLIKELY(aux != NULL) {
|
|
aux += outFrames;
|
|
}
|
|
(t.hook)(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
|
|
outFrames += t.buffer.frameCount;
|
|
t.bufferProvider->releaseBuffer(&t.buffer);
|
|
}
|
|
}
|
|
}
|
|
ditherAndClamp(out, outTemp, numFrames);
|
|
}
|
|
}
|
|
|
|
// one track, 16 bits stereo without resampling is the most common case
|
|
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
|
|
{
|
|
const int i = 31 - __builtin_clz(state->enabledTracks);
|
|
const track_t& t = state->tracks[i];
|
|
|
|
AudioBufferProvider::Buffer& b(t.buffer);
|
|
|
|
int32_t* out = t.mainBuffer;
|
|
size_t numFrames = state->frameCount;
|
|
|
|
const int16_t vl = t.volume[0];
|
|
const int16_t vr = t.volume[1];
|
|
const uint32_t vrl = t.volumeRL;
|
|
while (numFrames) {
|
|
b.frameCount = numFrames;
|
|
t.bufferProvider->getNextBuffer(&b);
|
|
int16_t const *in = b.i16;
|
|
|
|
// in == NULL can happen if the track was flushed just after having
|
|
// been enabled for mixing.
|
|
if (in == NULL || ((unsigned long)in & 3)) {
|
|
memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t));
|
|
LOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: buffer %p track %d, channels %d, needs %08x",
|
|
in, i, t.channelCount, t.needs);
|
|
return;
|
|
}
|
|
size_t outFrames = b.frameCount;
|
|
|
|
if (UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
|
|
// volume is boosted, so we might need to clamp even though
|
|
// we process only one track.
|
|
do {
|
|
uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
|
|
in += 2;
|
|
int32_t l = mulRL(1, rl, vrl) >> 12;
|
|
int32_t r = mulRL(0, rl, vrl) >> 12;
|
|
// clamping...
|
|
l = clamp16(l);
|
|
r = clamp16(r);
|
|
*out++ = (r<<16) | (l & 0xFFFF);
|
|
} while (--outFrames);
|
|
} else {
|
|
do {
|
|
uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
|
|
in += 2;
|
|
int32_t l = mulRL(1, rl, vrl) >> 12;
|
|
int32_t r = mulRL(0, rl, vrl) >> 12;
|
|
*out++ = (r<<16) | (l & 0xFFFF);
|
|
} while (--outFrames);
|
|
}
|
|
numFrames -= b.frameCount;
|
|
t.bufferProvider->releaseBuffer(&b);
|
|
}
|
|
}
|
|
|
|
// 2 tracks is also a common case
|
|
// NEVER used in current implementation of process__validate()
|
|
// only use if the 2 tracks have the same output buffer
|
|
void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
|
|
{
|
|
int i;
|
|
uint32_t en = state->enabledTracks;
|
|
|
|
i = 31 - __builtin_clz(en);
|
|
const track_t& t0 = state->tracks[i];
|
|
AudioBufferProvider::Buffer& b0(t0.buffer);
|
|
|
|
en &= ~(1<<i);
|
|
i = 31 - __builtin_clz(en);
|
|
const track_t& t1 = state->tracks[i];
|
|
AudioBufferProvider::Buffer& b1(t1.buffer);
|
|
|
|
int16_t const *in0;
|
|
const int16_t vl0 = t0.volume[0];
|
|
const int16_t vr0 = t0.volume[1];
|
|
size_t frameCount0 = 0;
|
|
|
|
int16_t const *in1;
|
|
const int16_t vl1 = t1.volume[0];
|
|
const int16_t vr1 = t1.volume[1];
|
|
size_t frameCount1 = 0;
|
|
|
|
//FIXME: only works if two tracks use same buffer
|
|
int32_t* out = t0.mainBuffer;
|
|
size_t numFrames = state->frameCount;
|
|
int16_t const *buff = NULL;
|
|
|
|
|
|
while (numFrames) {
|
|
|
|
if (frameCount0 == 0) {
|
|
b0.frameCount = numFrames;
|
|
t0.bufferProvider->getNextBuffer(&b0);
|
|
if (b0.i16 == NULL) {
|
|
if (buff == NULL) {
|
|
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
|
|
}
|
|
in0 = buff;
|
|
b0.frameCount = numFrames;
|
|
} else {
|
|
in0 = b0.i16;
|
|
}
|
|
frameCount0 = b0.frameCount;
|
|
}
|
|
if (frameCount1 == 0) {
|
|
b1.frameCount = numFrames;
|
|
t1.bufferProvider->getNextBuffer(&b1);
|
|
if (b1.i16 == NULL) {
|
|
if (buff == NULL) {
|
|
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
|
|
}
|
|
in1 = buff;
|
|
b1.frameCount = numFrames;
|
|
} else {
|
|
in1 = b1.i16;
|
|
}
|
|
frameCount1 = b1.frameCount;
|
|
}
|
|
|
|
size_t outFrames = frameCount0 < frameCount1?frameCount0:frameCount1;
|
|
|
|
numFrames -= outFrames;
|
|
frameCount0 -= outFrames;
|
|
frameCount1 -= outFrames;
|
|
|
|
do {
|
|
int32_t l0 = *in0++;
|
|
int32_t r0 = *in0++;
|
|
l0 = mul(l0, vl0);
|
|
r0 = mul(r0, vr0);
|
|
int32_t l = *in1++;
|
|
int32_t r = *in1++;
|
|
l = mulAdd(l, vl1, l0) >> 12;
|
|
r = mulAdd(r, vr1, r0) >> 12;
|
|
// clamping...
|
|
l = clamp16(l);
|
|
r = clamp16(r);
|
|
*out++ = (r<<16) | (l & 0xFFFF);
|
|
} while (--outFrames);
|
|
|
|
if (frameCount0 == 0) {
|
|
t0.bufferProvider->releaseBuffer(&b0);
|
|
}
|
|
if (frameCount1 == 0) {
|
|
t1.bufferProvider->releaseBuffer(&b1);
|
|
}
|
|
}
|
|
|
|
if (buff != NULL) {
|
|
delete [] buff;
|
|
}
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
}; // namespace android
|
|
|