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/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
#include <sstream>
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
#include <math.h>
#include <sys/types.h>
#include <utils/Errors.h>
#include <utils/Log.h>
#include <system/audio.h>
#include <audio_utils/primitives.h>
#include <audio_utils/format.h>
#include <media/AudioMixer.h>
#include "AudioMixerOps.h"
// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
#ifndef FCC_2
#define FCC_2 2
#endif
// Look for MONO_HACK for any Mono hack involving legacy mono channel to
// stereo channel conversion.
/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
* being used. This is a considerable amount of log spam, so don't enable unless you
* are verifying the hook based code.
*/
//#define VERY_VERY_VERBOSE_LOGGING
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
//define ALOGVV printf // for test-mixer.cpp
#else
#define ALOGVV(a...) do { } while (0)
#endif
// Set to default copy buffer size in frames for input processing.
static constexpr size_t kCopyBufferFrameCount = 256;
namespace android {
// ----------------------------------------------------------------------------
bool AudioMixer::isValidChannelMask(audio_channel_mask_t channelMask) const {
return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
}
// Called when channel masks have changed for a track name
// TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format,
// which will simplify this logic.
bool AudioMixer::setChannelMasks(int name,
audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
const std::shared_ptr<Track> &track = getTrack(name);
if (trackChannelMask == (track->channelMask | track->mHapticChannelMask)
&& mixerChannelMask == (track->mMixerChannelMask | track->mMixerHapticChannelMask)) {
return false; // no need to change
}
const audio_channel_mask_t hapticChannelMask =
static_cast<audio_channel_mask_t>(trackChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
trackChannelMask = static_cast<audio_channel_mask_t>(
trackChannelMask & ~AUDIO_CHANNEL_HAPTIC_ALL);
const audio_channel_mask_t mixerHapticChannelMask = static_cast<audio_channel_mask_t>(
mixerChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
mixerChannelMask = static_cast<audio_channel_mask_t>(
mixerChannelMask & ~AUDIO_CHANNEL_HAPTIC_ALL);
// always recompute for both channel masks even if only one has changed.
const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
const uint32_t hapticChannelCount = audio_channel_count_from_out_mask(hapticChannelMask);
const uint32_t mixerHapticChannelCount =
audio_channel_count_from_out_mask(mixerHapticChannelMask);
ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX)
&& trackChannelCount
&& mixerChannelCount);
track->channelMask = trackChannelMask;
track->channelCount = trackChannelCount;
track->mMixerChannelMask = mixerChannelMask;
track->mMixerChannelCount = mixerChannelCount;
track->mHapticChannelMask = hapticChannelMask;
track->mHapticChannelCount = hapticChannelCount;
track->mMixerHapticChannelMask = mixerHapticChannelMask;
track->mMixerHapticChannelCount = mixerHapticChannelCount;
if (track->mHapticChannelCount > 0) {
track->mAdjustInChannelCount = track->channelCount + track->mHapticChannelCount;
track->mAdjustOutChannelCount = track->channelCount;
track->mKeepContractedChannels = track->mHapticPlaybackEnabled;
} else {
track->mAdjustInChannelCount = 0;
track->mAdjustOutChannelCount = 0;
track->mKeepContractedChannels = false;
}
track->mInputFrameSize = audio_bytes_per_frame(
track->channelCount + track->mHapticChannelCount, track->mFormat);
// channel masks have changed, does this track need a downmixer?
// update to try using our desired format (if we aren't already using it)
const status_t status = track->prepareForDownmix();
ALOGE_IF(status != OK,
"prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
status, track->channelMask, track->mMixerChannelMask);
// always do reformat since channel mask changed,
// do it after downmix since track format may change!
track->prepareForReformat();
track->prepareForAdjustChannels(mFrameCount);
// Resampler channels may have changed.
track->recreateResampler(mSampleRate);
return true;
}
void AudioMixer::Track::unprepareForDownmix() {
ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
if (mPostDownmixReformatBufferProvider.get() != nullptr) {
// release any buffers held by the mPostDownmixReformatBufferProvider
// before deallocating the mDownmixerBufferProvider.
mPostDownmixReformatBufferProvider->reset();
}
mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
if (mDownmixerBufferProvider.get() != nullptr) {
// this track had previously been configured with a downmixer, delete it
mDownmixerBufferProvider.reset(nullptr);
reconfigureBufferProviders();
} else {
ALOGV(" nothing to do, no downmixer to delete");
}
}
status_t AudioMixer::Track::prepareForDownmix()
{
ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x",
this, channelMask);
// discard the previous downmixer if there was one
unprepareForDownmix();
// MONO_HACK Only remix (upmix or downmix) if the track and mixer/device channel masks
// are not the same and not handled internally, as mono for channel position masks is.
if (channelMask == mMixerChannelMask
|| (channelMask == AUDIO_CHANNEL_OUT_MONO
&& isAudioChannelPositionMask(mMixerChannelMask))) {
return NO_ERROR;
}
// DownmixerBufferProvider is only used for position masks.
if (audio_channel_mask_get_representation(channelMask)
== AUDIO_CHANNEL_REPRESENTATION_POSITION
&& DownmixerBufferProvider::isMultichannelCapable()) {
// Check if we have a float or int16 downmixer, in that order.
for (const audio_format_t format : { AUDIO_FORMAT_PCM_FLOAT, AUDIO_FORMAT_PCM_16_BIT }) {
mDownmixerBufferProvider.reset(new DownmixerBufferProvider(
channelMask, mMixerChannelMask,
format,
sampleRate, sessionId, kCopyBufferFrameCount));
if (static_cast<DownmixerBufferProvider *>(mDownmixerBufferProvider.get())
->isValid()) {
mDownmixRequiresFormat = format;
reconfigureBufferProviders();
return NO_ERROR;
}
}
// mDownmixerBufferProvider reset below.
}
// See if we should use our built-in non-effect downmixer.
if (mMixerInFormat == AUDIO_FORMAT_PCM_FLOAT
&& ChannelMixBufferProvider::isOutputChannelMaskSupported(mMixerChannelMask)
&& audio_channel_mask_get_representation(channelMask)
== AUDIO_CHANNEL_REPRESENTATION_POSITION) {
mDownmixerBufferProvider.reset(new ChannelMixBufferProvider(channelMask,
mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount));
if (static_cast<ChannelMixBufferProvider *>(mDownmixerBufferProvider.get())
->isValid()) {
mDownmixRequiresFormat = mMixerInFormat;
reconfigureBufferProviders();
ALOGD("%s: Fallback using ChannelMix", __func__);
return NO_ERROR;
} else {
ALOGD("%s: ChannelMix not supported for channel mask %#x", __func__, channelMask);
}
}
// Effect downmixer does not accept the channel conversion. Let's use our remixer.
mDownmixerBufferProvider.reset(new RemixBufferProvider(channelMask,
mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount));
// Remix always finds a conversion whereas Downmixer effect above may fail.
reconfigureBufferProviders();
return NO_ERROR;
}
void AudioMixer::Track::unprepareForReformat() {
ALOGV("AudioMixer::unprepareForReformat(%p)", this);
bool requiresReconfigure = false;
if (mReformatBufferProvider.get() != nullptr) {
mReformatBufferProvider.reset(nullptr);
requiresReconfigure = true;
}
if (mPostDownmixReformatBufferProvider.get() != nullptr) {
mPostDownmixReformatBufferProvider.reset(nullptr);
requiresReconfigure = true;
}
if (requiresReconfigure) {
reconfigureBufferProviders();
}
}
status_t AudioMixer::Track::prepareForReformat()
{
ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
// discard previous reformatters
unprepareForReformat();
// only configure reformatters as needed
const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID
? mDownmixRequiresFormat : mMixerInFormat;
bool requiresReconfigure = false;
if (mFormat != targetFormat) {
mReformatBufferProvider.reset(new ReformatBufferProvider(
audio_channel_count_from_out_mask(channelMask),
mFormat,
targetFormat,
kCopyBufferFrameCount));
requiresReconfigure = true;
} else if (mFormat == AUDIO_FORMAT_PCM_FLOAT) {
// Input and output are floats, make sure application did not provide > 3db samples
// that would break volume application (b/68099072)
// TODO: add a trusted source flag to avoid the overhead
mReformatBufferProvider.reset(new ClampFloatBufferProvider(
audio_channel_count_from_out_mask(channelMask),
kCopyBufferFrameCount));
requiresReconfigure = true;
}
if (targetFormat != mMixerInFormat) {
mPostDownmixReformatBufferProvider.reset(new ReformatBufferProvider(
audio_channel_count_from_out_mask(mMixerChannelMask),
targetFormat,
mMixerInFormat,
kCopyBufferFrameCount));
requiresReconfigure = true;
}
if (requiresReconfigure) {
reconfigureBufferProviders();
}
return NO_ERROR;
}
void AudioMixer::Track::unprepareForAdjustChannels()
{
ALOGV("AUDIOMIXER::unprepareForAdjustChannels");
if (mAdjustChannelsBufferProvider.get() != nullptr) {
mAdjustChannelsBufferProvider.reset(nullptr);
reconfigureBufferProviders();
}
}
status_t AudioMixer::Track::prepareForAdjustChannels(size_t frames)
{
ALOGV("AudioMixer::prepareForAdjustChannels(%p) with inChannelCount: %u, outChannelCount: %u",
this, mAdjustInChannelCount, mAdjustOutChannelCount);
unprepareForAdjustChannels();
if (mAdjustInChannelCount != mAdjustOutChannelCount) {
uint8_t* buffer = mKeepContractedChannels
? (uint8_t*)mainBuffer + frames * audio_bytes_per_frame(
mMixerChannelCount, mMixerFormat)
: nullptr;
mAdjustChannelsBufferProvider.reset(new AdjustChannelsBufferProvider(
mFormat, mAdjustInChannelCount, mAdjustOutChannelCount, frames,
mKeepContractedChannels ? mMixerFormat : AUDIO_FORMAT_INVALID,
buffer, mMixerHapticChannelCount));
reconfigureBufferProviders();
}
return NO_ERROR;
}
void AudioMixer::Track::unprepareForTee() {
ALOGV("AudioMixer::%s", __func__);
if (mTeeBufferProvider.get() != nullptr) {
mTeeBufferProvider.reset(nullptr);
reconfigureBufferProviders();
}
}
status_t AudioMixer::Track::prepareForTee() {
ALOGV("AudioMixer::%s(%p) teeBuffer=%p", __func__, this, teeBuffer);
unprepareForTee();
if (teeBuffer != nullptr) {
mTeeBufferProvider.reset(new TeeBufferProvider(
mInputFrameSize, mInputFrameSize, kCopyBufferFrameCount,
(uint8_t*)teeBuffer, mTeeBufferFrameCount));
reconfigureBufferProviders();
}
return NO_ERROR;
}
void AudioMixer::Track::clearContractedBuffer()
{
if (mAdjustChannelsBufferProvider.get() != nullptr) {
static_cast<AdjustChannelsBufferProvider*>(
mAdjustChannelsBufferProvider.get())->clearContractedFrames();
}
}
void AudioMixer::Track::clearTeeFrameCopied() {
if (mTeeBufferProvider.get() != nullptr) {
static_cast<TeeBufferProvider*>(mTeeBufferProvider.get())->clearFramesCopied();
}
}
void AudioMixer::Track::reconfigureBufferProviders()
{
// configure from upstream to downstream buffer providers.
bufferProvider = mInputBufferProvider;
if (mTeeBufferProvider != nullptr) {
mTeeBufferProvider->setBufferProvider(bufferProvider);
bufferProvider = mTeeBufferProvider.get();
}
if (mAdjustChannelsBufferProvider.get() != nullptr) {
mAdjustChannelsBufferProvider->setBufferProvider(bufferProvider);
bufferProvider = mAdjustChannelsBufferProvider.get();
}
if (mReformatBufferProvider.get() != nullptr) {
mReformatBufferProvider->setBufferProvider(bufferProvider);
bufferProvider = mReformatBufferProvider.get();
}
if (mDownmixerBufferProvider.get() != nullptr) {
mDownmixerBufferProvider->setBufferProvider(bufferProvider);
bufferProvider = mDownmixerBufferProvider.get();
}
if (mPostDownmixReformatBufferProvider.get() != nullptr) {
mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
bufferProvider = mPostDownmixReformatBufferProvider.get();
}
if (mTimestretchBufferProvider.get() != nullptr) {
mTimestretchBufferProvider->setBufferProvider(bufferProvider);
bufferProvider = mTimestretchBufferProvider.get();
}
}
void AudioMixer::setParameter(int name, int target, int param, void *value)
{
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
const std::shared_ptr<Track> &track = getTrack(name);
int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
switch (target) {
case TRACK:
switch (param) {
case CHANNEL_MASK: {
const audio_channel_mask_t trackChannelMask =
static_cast<audio_channel_mask_t>(valueInt);
if (setChannelMasks(name, trackChannelMask,
static_cast<audio_channel_mask_t>(
track->mMixerChannelMask | track->mMixerHapticChannelMask))) {
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
invalidate();
}
} break;
case MAIN_BUFFER:
if (track->mainBuffer != valueBuf) {
track->mainBuffer = valueBuf;
ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
if (track->mKeepContractedChannels) {
track->prepareForAdjustChannels(mFrameCount);
}
invalidate();
}
break;
case AUX_BUFFER:
AudioMixerBase::setParameter(name, target, param, value);
break;
case FORMAT: {
audio_format_t format = static_cast<audio_format_t>(valueInt);
if (track->mFormat != format) {
ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
track->mFormat = format;
ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
track->prepareForReformat();
invalidate();
}
} break;
// FIXME do we want to support setting the downmix type from AudioFlinger?
// for a specific track? or per mixer?
/* case DOWNMIX_TYPE:
break */
case MIXER_FORMAT: {
audio_format_t format = static_cast<audio_format_t>(valueInt);
if (track->mMixerFormat != format) {
track->mMixerFormat = format;
ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
if (track->mKeepContractedChannels) {
track->prepareForAdjustChannels(mFrameCount);
}
}
} break;
case MIXER_CHANNEL_MASK: {
const audio_channel_mask_t mixerChannelMask =
static_cast<audio_channel_mask_t>(valueInt);
if (setChannelMasks(name, static_cast<audio_channel_mask_t>(
track->channelMask | track->mHapticChannelMask),
mixerChannelMask)) {
ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
invalidate();
}
} break;
case HAPTIC_ENABLED: {
const bool hapticPlaybackEnabled = static_cast<bool>(valueInt);
if (track->mHapticPlaybackEnabled != hapticPlaybackEnabled) {
track->mHapticPlaybackEnabled = hapticPlaybackEnabled;
track->mKeepContractedChannels = hapticPlaybackEnabled;
track->prepareForAdjustChannels(mFrameCount);
}
} break;
case HAPTIC_INTENSITY: {
const os::HapticScale hapticIntensity = static_cast<os::HapticScale>(valueInt);
if (track->mHapticIntensity != hapticIntensity) {
track->mHapticIntensity = hapticIntensity;
}
} break;
case HAPTIC_MAX_AMPLITUDE: {
const float hapticMaxAmplitude = *reinterpret_cast<float*>(value);
if (track->mHapticMaxAmplitude != hapticMaxAmplitude) {
track->mHapticMaxAmplitude = hapticMaxAmplitude;
}
} break;
case TEE_BUFFER:
if (track->teeBuffer != valueBuf) {
track->teeBuffer = valueBuf;
ALOGV("setParameter(TRACK, TEE_BUFFER, %p)", valueBuf);
track->prepareForTee();
}
break;
case TEE_BUFFER_FRAME_COUNT:
if (track->mTeeBufferFrameCount != valueInt) {
track->mTeeBufferFrameCount = valueInt;
ALOGV("setParameter(TRACK, TEE_BUFFER_FRAME_COUNT, %i)", valueInt);
track->prepareForTee();
}
break;
default:
LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
}
break;
case RESAMPLE:
case RAMP_VOLUME:
case VOLUME:
AudioMixerBase::setParameter(name, target, param, value);
break;
case TIMESTRETCH:
switch (param) {
case PLAYBACK_RATE: {
const AudioPlaybackRate *playbackRate =
reinterpret_cast<AudioPlaybackRate*>(value);
ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
"bad parameters speed %f, pitch %f",
playbackRate->mSpeed, playbackRate->mPitch);
if (track->setPlaybackRate(*playbackRate)) {
ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
"%f %f %d %d",
playbackRate->mSpeed,
playbackRate->mPitch,
playbackRate->mStretchMode,
playbackRate->mFallbackMode);
// invalidate(); (should not require reconfigure)
}
} break;
default:
LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
}
break;
default:
LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
}
}
bool AudioMixer::Track::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
if ((mTimestretchBufferProvider.get() == nullptr &&
fabs(playbackRate.mSpeed - mPlaybackRate.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA &&
fabs(playbackRate.mPitch - mPlaybackRate.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) ||
isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
return false;
}
mPlaybackRate = playbackRate;
if (mTimestretchBufferProvider.get() == nullptr) {
// TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
// but if none exists, it is the channel count (1 for mono).
const int timestretchChannelCount = getOutputChannelCount();
mTimestretchBufferProvider.reset(new TimestretchBufferProvider(timestretchChannelCount,
mMixerInFormat, sampleRate, playbackRate));
reconfigureBufferProviders();
} else {
static_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider.get())
->setPlaybackRate(playbackRate);
}
return true;
}
void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
{
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
const std::shared_ptr<Track> &track = getTrack(name);
if (track->mInputBufferProvider == bufferProvider) {
return; // don't reset any buffer providers if identical.
}
// reset order from downstream to upstream buffer providers.
if (track->mTimestretchBufferProvider.get() != nullptr) {
track->mTimestretchBufferProvider->reset();
} else if (track->mPostDownmixReformatBufferProvider.get() != nullptr) {
track->mPostDownmixReformatBufferProvider->reset();
} else if (track->mDownmixerBufferProvider != nullptr) {
track->mDownmixerBufferProvider->reset();
} else if (track->mReformatBufferProvider.get() != nullptr) {
track->mReformatBufferProvider->reset();
} else if (track->mAdjustChannelsBufferProvider.get() != nullptr) {
track->mAdjustChannelsBufferProvider->reset();
} else if (track->mTeeBufferProvider.get() != nullptr) {
track->mTeeBufferProvider->reset();
}
track->mInputBufferProvider = bufferProvider;
track->reconfigureBufferProviders();
}
/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
/*static*/ void AudioMixer::sInitRoutine()
{
DownmixerBufferProvider::init(); // for the downmixer
}
std::shared_ptr<AudioMixerBase::TrackBase> AudioMixer::preCreateTrack()
{
return std::make_shared<Track>();
}
status_t AudioMixer::postCreateTrack(TrackBase *track)
{
Track* t = static_cast<Track*>(track);
audio_channel_mask_t channelMask = t->channelMask;
t->mHapticChannelMask = static_cast<audio_channel_mask_t>(
channelMask & AUDIO_CHANNEL_HAPTIC_ALL);
t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
channelMask = static_cast<audio_channel_mask_t>(channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL);
t->channelCount = audio_channel_count_from_out_mask(channelMask);
ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
"Non-stereo channel mask: %d\n", channelMask);
t->channelMask = channelMask;
t->mInputBufferProvider = NULL;
t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// haptic
t->mHapticPlaybackEnabled = false;
t->mHapticIntensity = os::HapticScale::NONE;
t->mHapticMaxAmplitude = NAN;
t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
t->mMixerHapticChannelCount = 0;
t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
t->mAdjustOutChannelCount = t->channelCount;
t->mKeepContractedChannels = false;
t->mInputFrameSize = audio_bytes_per_frame(
t->channelCount + t->mHapticChannelCount, t->mFormat);
// Check the downmixing (or upmixing) requirements.
status_t status = t->prepareForDownmix();
if (status != OK) {
ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
return BAD_VALUE;
}
// prepareForDownmix() may change mDownmixRequiresFormat
ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
t->prepareForReformat();
t->prepareForAdjustChannels(mFrameCount);
return OK;
}
void AudioMixer::preProcess()
{
for (const auto &pair : mTracks) {
// Clear contracted buffer before processing if contracted channels are saved
const std::shared_ptr<TrackBase> &tb = pair.second;
Track *t = static_cast<Track*>(tb.get());
if (t->mKeepContractedChannels) {
t->clearContractedBuffer();
}
t->clearTeeFrameCopied();
}
}
void AudioMixer::postProcess()
{
// Process haptic data.
// Need to keep consistent with VibrationEffect.scale(int, float, int)
for (const auto &pair : mGroups) {
// process by group of tracks with same output main buffer.
const auto &group = pair.second;
for (const int name : group) {
const std::shared_ptr<Track> &t = getTrack(name);
if (t->mHapticPlaybackEnabled) {
size_t sampleCount = mFrameCount * t->mMixerHapticChannelCount;
uint8_t* buffer = (uint8_t*)pair.first + mFrameCount * audio_bytes_per_frame(
t->mMixerChannelCount, t->mMixerFormat);
switch (t->mMixerFormat) {
// Mixer format should be AUDIO_FORMAT_PCM_FLOAT.
case AUDIO_FORMAT_PCM_FLOAT: {
os::scaleHapticData((float*) buffer, sampleCount, t->mHapticIntensity,
t->mHapticMaxAmplitude);
} break;
default:
LOG_ALWAYS_FATAL("bad mMixerFormat: %#x", t->mMixerFormat);
break;
}
break;
}
if (t->teeBuffer != nullptr && t->volumeRL == 0) {
// Need to mute tee
memset(t->teeBuffer, 0, t->mTeeBufferFrameCount * t->mInputFrameSize);
}
}
}
}
// ----------------------------------------------------------------------------
} // namespace android