libaudioprocessing: Extract vendor-available part of AudioMixer
Split AudioMixer into the base part which doesn't rely on
the framework and can be used in vendor code and the derived part
which is intended to be used by Audioflinger.
Test: A/B compare output from test scripts, manual testing on the phone
Merged-In: I24c390f67f20baa8109902099359ca6e34eebcfd
Change-Id: I24c390f67f20baa8109902099359ca6e34eebcfd
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index 2d740ab..c0b11a4 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -28,9 +28,6 @@
#include <utils/Errors.h>
#include <utils/Log.h>
-#include <cutils/compiler.h>
-#include <utils/Debug.h>
-
#include <system/audio.h>
#include <audio_utils/primitives.h>
@@ -59,26 +56,6 @@
#define ALOGVV(a...) do { } while (0)
#endif
-// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
-// original code will be used for stereo sinks, the new mixer for multichannel.
-static constexpr bool kUseNewMixer = true;
-
-// Set kUseFloat to true to allow floating input into the mixer engine.
-// If kUseNewMixer is false, this is ignored or may be overridden internally
-// because of downmix/upmix support.
-static constexpr bool kUseFloat = true;
-
-#ifdef FLOAT_AUX
-using TYPE_AUX = float;
-static_assert(kUseNewMixer && kUseFloat,
- "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
-#else
-using TYPE_AUX = int32_t; // q4.27
-#endif
-
-// TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
-static constexpr int BLOCKSIZE = 16;
-
// Set to default copy buffer size in frames for input processing.
static constexpr size_t kCopyBufferFrameCount = 256;
@@ -86,110 +63,8 @@
// ----------------------------------------------------------------------------
-static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
- return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
-}
-
-status_t AudioMixer::create(
- int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
-{
- LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
-
- if (!isValidChannelMask(channelMask)) {
- ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
- return BAD_VALUE;
- }
- if (!isValidFormat(format)) {
- ALOGE("%s invalid format: %#x", __func__, format);
- return BAD_VALUE;
- }
-
- auto t = std::make_shared<Track>();
- {
- // TODO: move initialization to the Track constructor.
- // assume default parameters for the track, except where noted below
- t->needs = 0;
-
- // Integer volume.
- // Currently integer volume is kept for the legacy integer mixer.
- // Will be removed when the legacy mixer path is removed.
- t->volume[0] = 0;
- t->volume[1] = 0;
- t->prevVolume[0] = 0 << 16;
- t->prevVolume[1] = 0 << 16;
- t->volumeInc[0] = 0;
- t->volumeInc[1] = 0;
- t->auxLevel = 0;
- t->auxInc = 0;
- t->prevAuxLevel = 0;
-
- // Floating point volume.
- t->mVolume[0] = 0.f;
- t->mVolume[1] = 0.f;
- t->mPrevVolume[0] = 0.f;
- t->mPrevVolume[1] = 0.f;
- t->mVolumeInc[0] = 0.;
- t->mVolumeInc[1] = 0.;
- t->mAuxLevel = 0.;
- t->mAuxInc = 0.;
- t->mPrevAuxLevel = 0.;
-
- // no initialization needed
- // t->frameCount
- t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
- t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
- channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
- t->channelCount = audio_channel_count_from_out_mask(channelMask);
- t->enabled = false;
- ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
- "Non-stereo channel mask: %d\n", channelMask);
- t->channelMask = channelMask;
- t->sessionId = sessionId;
- // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
- t->bufferProvider = NULL;
- t->buffer.raw = NULL;
- // no initialization needed
- // t->buffer.frameCount
- t->hook = NULL;
- t->mIn = NULL;
- t->sampleRate = mSampleRate;
- // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
- t->mainBuffer = NULL;
- t->auxBuffer = NULL;
- t->mInputBufferProvider = NULL;
- t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
- t->mFormat = format;
- t->mMixerInFormat = selectMixerInFormat(format);
- t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
- t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
- AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
- t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
- t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
- // haptic
- t->mHapticPlaybackEnabled = false;
- t->mHapticIntensity = HAPTIC_SCALE_NONE;
- t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
- t->mMixerHapticChannelCount = 0;
- t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
- t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
- t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
- t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
- t->mKeepContractedChannels = false;
- // 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->prepareForAdjustChannelsNonDestructive(mFrameCount);
- t->prepareForAdjustChannels();
-
- mTracks[name] = t;
- return OK;
- }
+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
@@ -198,7 +73,7 @@
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 = mTracks[name];
+ const std::shared_ptr<Track> &track = getTrack(name);
if (trackChannelMask == (track->channelMask | track->mHapticChannelMask)
&& mixerChannelMask == (track->mMixerChannelMask | track->mMixerHapticChannelMask)) {
@@ -255,14 +130,8 @@
track->prepareForAdjustChannelsNonDestructive(mFrameCount);
track->prepareForAdjustChannels();
- if (track->mResampler.get() != nullptr) {
- // resampler channels may have changed.
- const uint32_t resetToSampleRate = track->sampleRate;
- track->mResampler.reset(nullptr);
- track->sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
- // recreate the resampler with updated format, channels, saved sampleRate.
- track->setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
- }
+ // Resampler channels may have changed.
+ track->recreateResampler(mSampleRate);
return true;
}
@@ -477,171 +346,10 @@
}
}
-void AudioMixer::destroy(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- ALOGV("deleteTrackName(%d)", name);
-
- if (mTracks[name]->enabled) {
- invalidate();
- }
- mTracks.erase(name); // deallocate track
-}
-
-void AudioMixer::enable(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
-
- if (!track->enabled) {
- track->enabled = true;
- ALOGV("enable(%d)", name);
- invalidate();
- }
-}
-
-void AudioMixer::disable(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
-
- if (track->enabled) {
- track->enabled = false;
- ALOGV("disable(%d)", name);
- invalidate();
- }
-}
-
-/* Sets the volume ramp variables for the AudioMixer.
- *
- * The volume ramp variables are used to transition from the previous
- * volume to the set volume. ramp controls the duration of the transition.
- * Its value is typically one state framecount period, but may also be 0,
- * meaning "immediate."
- *
- * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
- * even if there is a nonzero floating point increment (in that case, the volume
- * change is immediate). This restriction should be changed when the legacy mixer
- * is removed (see #2).
- * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
- * when no longer needed.
- *
- * @param newVolume set volume target in floating point [0.0, 1.0].
- * @param ramp number of frames to increment over. if ramp is 0, the volume
- * should be set immediately. Currently ramp should not exceed 65535 (frames).
- * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
- * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
- * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
- * @param pSetVolume pointer to the float target volume, set on return.
- * @param pPrevVolume pointer to the float previous volume, set on return.
- * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
- * @return true if the volume has changed, false if volume is same.
- */
-static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
- int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
- float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
- // check floating point volume to see if it is identical to the previously
- // set volume.
- // We do not use a tolerance here (and reject changes too small)
- // as it may be confusing to use a different value than the one set.
- // If the resulting volume is too small to ramp, it is a direct set of the volume.
- if (newVolume == *pSetVolume) {
- return false;
- }
- if (newVolume < 0) {
- newVolume = 0; // should not have negative volumes
- } else {
- switch (fpclassify(newVolume)) {
- case FP_SUBNORMAL:
- case FP_NAN:
- newVolume = 0;
- break;
- case FP_ZERO:
- break; // zero volume is fine
- case FP_INFINITE:
- // Infinite volume could be handled consistently since
- // floating point math saturates at infinities,
- // but we limit volume to unity gain float.
- // ramp = 0; break;
- //
- newVolume = AudioMixer::UNITY_GAIN_FLOAT;
- break;
- case FP_NORMAL:
- default:
- // Floating point does not have problems with overflow wrap
- // that integer has. However, we limit the volume to
- // unity gain here.
- // TODO: Revisit the volume limitation and perhaps parameterize.
- if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) {
- newVolume = AudioMixer::UNITY_GAIN_FLOAT;
- }
- break;
- }
- }
-
- // set floating point volume ramp
- if (ramp != 0) {
- // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
- // is no computational mismatch; hence equality is checked here.
- ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
- " prev:%f set_to:%f", *pPrevVolume, *pSetVolume);
- const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
- // could be inf, cannot be nan, subnormal
- const float maxv = std::max(newVolume, *pPrevVolume);
-
- if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
- && maxv + inc != maxv) { // inc must make forward progress
- *pVolumeInc = inc;
- // ramp is set now.
- // Note: if newVolume is 0, then near the end of the ramp,
- // it may be possible that the ramped volume may be subnormal or
- // temporarily negative by a small amount or subnormal due to floating
- // point inaccuracies.
- } else {
- ramp = 0; // ramp not allowed
- }
- }
-
- // compute and check integer volume, no need to check negative values
- // The integer volume is limited to "unity_gain" to avoid wrapping and other
- // audio artifacts, so it never reaches the range limit of U4.28.
- // We safely use signed 16 and 32 bit integers here.
- const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan
- const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ?
- AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume;
-
- // set integer volume ramp
- if (ramp != 0) {
- // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
- // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
- // is no computational mismatch; hence equality is checked here.
- ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
- " prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
- const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
-
- if (inc != 0) { // inc must make forward progress
- *pIntVolumeInc = inc;
- } else {
- ramp = 0; // ramp not allowed
- }
- }
-
- // if no ramp, or ramp not allowed, then clear float and integer increments
- if (ramp == 0) {
- *pVolumeInc = 0;
- *pPrevVolume = newVolume;
- *pIntVolumeInc = 0;
- *pIntPrevVolume = intVolume << 16;
- }
- *pSetVolume = newVolume;
- *pIntSetVolume = intVolume;
- return true;
-}
-
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 = mTracks[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);
@@ -670,11 +378,7 @@
}
break;
case AUX_BUFFER:
- if (track->auxBuffer != valueBuf) {
- track->auxBuffer = valueBuf;
- ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
- invalidate();
- }
+ AudioMixerBase::setParameter(name, target, param, value);
break;
case FORMAT: {
audio_format_t format = static_cast<audio_format_t>(valueInt);
@@ -730,61 +434,9 @@
break;
case RESAMPLE:
- switch (param) {
- case SAMPLE_RATE:
- ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
- if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
- ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
- uint32_t(valueInt));
- invalidate();
- }
- break;
- case RESET:
- track->resetResampler();
- invalidate();
- break;
- case REMOVE:
- track->mResampler.reset(nullptr);
- track->sampleRate = mSampleRate;
- invalidate();
- break;
- default:
- LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
- }
- break;
-
case RAMP_VOLUME:
case VOLUME:
- switch (param) {
- case AUXLEVEL:
- if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
- target == RAMP_VOLUME ? mFrameCount : 0,
- &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
- &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
- ALOGV("setParameter(%s, AUXLEVEL: %04x)",
- target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
- invalidate();
- }
- break;
- default:
- if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
- if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
- target == RAMP_VOLUME ? mFrameCount : 0,
- &track->volume[param - VOLUME0],
- &track->prevVolume[param - VOLUME0],
- &track->volumeInc[param - VOLUME0],
- &track->mVolume[param - VOLUME0],
- &track->mPrevVolume[param - VOLUME0],
- &track->mVolumeInc[param - VOLUME0])) {
- ALOGV("setParameter(%s, VOLUME%d: %04x)",
- target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
- track->volume[param - VOLUME0]);
- invalidate();
- }
- } else {
- LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
- }
- }
+ AudioMixerBase::setParameter(name, target, param, value);
break;
case TIMESTRETCH:
switch (param) {
@@ -814,43 +466,6 @@
}
}
-bool AudioMixer::Track::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
-{
- if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
- if (sampleRate != trackSampleRate) {
- sampleRate = trackSampleRate;
- if (mResampler.get() == nullptr) {
- ALOGV("Creating resampler from track %d Hz to device %d Hz",
- trackSampleRate, devSampleRate);
- AudioResampler::src_quality quality;
- // force lowest quality level resampler if use case isn't music or video
- // FIXME this is flawed for dynamic sample rates, as we choose the resampler
- // quality level based on the initial ratio, but that could change later.
- // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
- if (isMusicRate(trackSampleRate)) {
- quality = AudioResampler::DEFAULT_QUALITY;
- } else {
- quality = AudioResampler::DYN_LOW_QUALITY;
- }
-
- // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
- // but if none exists, it is the channel count (1 for mono).
- const int resamplerChannelCount = mDownmixerBufferProvider.get() != nullptr
- ? mMixerChannelCount : channelCount;
- ALOGVV("Creating resampler:"
- " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
- mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
- mResampler.reset(AudioResampler::create(
- mMixerInFormat,
- resamplerChannelCount,
- devSampleRate, quality));
- }
- return true;
- }
- }
- return false;
-}
-
bool AudioMixer::Track::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
if ((mTimestretchBufferProvider.get() == nullptr &&
@@ -863,8 +478,7 @@
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 = mDownmixerBufferProvider.get() != nullptr
- ? mMixerChannelCount : channelCount;
+ const int timestretchChannelCount = getOutputChannelCount();
mTimestretchBufferProvider.reset(new TimestretchBufferProvider(timestretchChannelCount,
mMixerInFormat, sampleRate, playbackRate));
reconfigureBufferProviders();
@@ -875,92 +489,10 @@
return true;
}
-/* Checks to see if the volume ramp has completed and clears the increment
- * variables appropriately.
- *
- * FIXME: There is code to handle int/float ramp variable switchover should it not
- * complete within a mixer buffer processing call, but it is preferred to avoid switchover
- * due to precision issues. The switchover code is included for legacy code purposes
- * and can be removed once the integer volume is removed.
- *
- * It is not sufficient to clear only the volumeInc integer variable because
- * if one channel requires ramping, all channels are ramped.
- *
- * There is a bit of duplicated code here, but it keeps backward compatibility.
- */
-inline void AudioMixer::Track::adjustVolumeRamp(bool aux, bool useFloat)
-{
- if (useFloat) {
- for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
- if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
- (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
- volumeInc[i] = 0;
- prevVolume[i] = volume[i] << 16;
- mVolumeInc[i] = 0.;
- mPrevVolume[i] = mVolume[i];
- } else {
- //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
- prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
- }
- }
- } else {
- for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
- if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
- ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
- volumeInc[i] = 0;
- prevVolume[i] = volume[i] << 16;
- mVolumeInc[i] = 0.;
- mPrevVolume[i] = mVolume[i];
- } else {
- //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
- mPrevVolume[i] = float_from_u4_28(prevVolume[i]);
- }
- }
- }
-
- if (aux) {
-#ifdef FLOAT_AUX
- if (useFloat) {
- if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
- (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
- auxInc = 0;
- prevAuxLevel = auxLevel << 16;
- mAuxInc = 0.f;
- mPrevAuxLevel = mAuxLevel;
- }
- } else
-#endif
- if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
- (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
- auxInc = 0;
- prevAuxLevel = auxLevel << 16;
- mAuxInc = 0.f;
- mPrevAuxLevel = mAuxLevel;
- }
- }
-}
-
-size_t AudioMixer::getUnreleasedFrames(int name) const
-{
- const auto it = mTracks.find(name);
- if (it != mTracks.end()) {
- return it->second->getUnreleasedFrames();
- }
- return 0;
-}
-
-std::string AudioMixer::trackNames() const {
- std::stringstream ss;
- for (const auto &pair : mTracks) {
- ss << pair.first << " ";
- }
- return ss.str();
-}
-
void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
{
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
+ const std::shared_ptr<Track> &track = getTrack(name);
if (track->mInputBufferProvider == bufferProvider) {
return; // don't reset any buffer providers if identical.
@@ -984,679 +516,6 @@
track->reconfigureBufferProviders();
}
-void AudioMixer::process__validate()
-{
- // TODO: fix all16BitsStereNoResample logic to
- // either properly handle muted tracks (it should ignore them)
- // or remove altogether as an obsolete optimization.
- bool all16BitsStereoNoResample = true;
- bool resampling = false;
- bool volumeRamp = false;
-
- mEnabled.clear();
- mGroups.clear();
- for (const auto &pair : mTracks) {
- const int name = pair.first;
- const std::shared_ptr<Track> &t = pair.second;
- if (!t->enabled) continue;
-
- mEnabled.emplace_back(name); // we add to mEnabled in order of name.
- mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
-
- uint32_t n = 0;
- // FIXME can overflow (mask is only 3 bits)
- n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
- if (t->doesResample()) {
- n |= NEEDS_RESAMPLE;
- }
- if (t->auxLevel != 0 && t->auxBuffer != NULL) {
- n |= NEEDS_AUX;
- }
-
- if (t->volumeInc[0]|t->volumeInc[1]) {
- volumeRamp = true;
- } else if (!t->doesResample() && t->volumeRL == 0) {
- n |= NEEDS_MUTE;
- }
- t->needs = n;
-
- if (n & NEEDS_MUTE) {
- t->hook = &Track::track__nop;
- } else {
- if (n & NEEDS_AUX) {
- all16BitsStereoNoResample = false;
- }
- if (n & NEEDS_RESAMPLE) {
- all16BitsStereoNoResample = false;
- resampling = true;
- t->hook = Track::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
- "Track %d needs downmix + resample", name);
- } else {
- if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
- t->hook = Track::getTrackHook(
- (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO // TODO: MONO_HACK
- && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
- ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
- t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- all16BitsStereoNoResample = false;
- }
- if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
- t->hook = Track::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
- "Track %d needs downmix", name);
- }
- }
- }
- }
-
- // select the processing hooks
- mHook = &AudioMixer::process__nop;
- if (mEnabled.size() > 0) {
- if (resampling) {
- if (mOutputTemp.get() == nullptr) {
- mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
- }
- if (mResampleTemp.get() == nullptr) {
- mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
- }
- mHook = &AudioMixer::process__genericResampling;
- } else {
- // we keep temp arrays around.
- mHook = &AudioMixer::process__genericNoResampling;
- if (all16BitsStereoNoResample && !volumeRamp) {
- if (mEnabled.size() == 1) {
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- if ((t->needs & NEEDS_MUTE) == 0) {
- // The check prevents a muted track from acquiring a process hook.
- //
- // This is dangerous if the track is MONO as that requires
- // special case handling due to implicit channel duplication.
- // Stereo or Multichannel should actually be fine here.
- mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
- t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
- }
- }
- }
- }
- }
-
- ALOGV("mixer configuration change: %zu "
- "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
- mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
-
- process();
-
- // Now that the volume ramp has been done, set optimal state and
- // track hooks for subsequent mixer process
- if (mEnabled.size() > 0) {
- bool allMuted = true;
-
- for (const int name : mEnabled) {
- const std::shared_ptr<Track> &t = mTracks[name];
- if (!t->doesResample() && t->volumeRL == 0) {
- t->needs |= NEEDS_MUTE;
- t->hook = &Track::track__nop;
- } else {
- allMuted = false;
- }
- }
- if (allMuted) {
- mHook = &AudioMixer::process__nop;
- } else if (all16BitsStereoNoResample) {
- if (mEnabled.size() == 1) {
- //const int i = 31 - __builtin_clz(enabledTracks);
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- // Muted single tracks handled by allMuted above.
- mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
- t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
- }
- }
- }
-}
-
-void AudioMixer::Track::track__genericResample(
- int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
-{
- ALOGVV("track__genericResample\n");
- mResampler->setSampleRate(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
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
- mResampler->resample(temp, outFrameCount, bufferProvider);
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- volumeRampStereo(out, outFrameCount, temp, aux);
- } else {
- volumeStereo(out, outFrameCount, temp, aux);
- }
- } else {
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
- mResampler->resample(temp, outFrameCount, bufferProvider);
- volumeRampStereo(out, outFrameCount, temp, aux);
- }
-
- // constant gain
- else {
- mResampler->setVolume(mVolume[0], mVolume[1]);
- mResampler->resample(out, outFrameCount, bufferProvider);
- }
- }
-}
-
-void AudioMixer::Track::track__nop(int32_t* out __unused,
- size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
-{
-}
-
-void AudioMixer::Track::volumeRampStereo(
- int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- // ramp volume
- if (CC_UNLIKELY(aux != NULL)) {
- int32_t va = prevAuxLevel;
- const int32_t vaInc = 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);
- prevAuxLevel = va;
- } else {
- do {
- *out++ += (vl >> 16) * (*temp++ >> 12);
- *out++ += (vr >> 16) * (*temp++ >> 12);
- vl += vlInc;
- vr += vrInc;
- } while (--frameCount);
- }
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(aux != NULL);
-}
-
-void AudioMixer::Track::volumeStereo(
- int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
- const int16_t vl = volume[0];
- const int16_t vr = volume[1];
-
- if (CC_UNLIKELY(aux != NULL)) {
- const int16_t va = 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::track__16BitsStereo(
- int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
- ALOGVV("track__16BitsStereo\n");
- const int16_t *in = static_cast<const int16_t *>(mIn);
-
- if (CC_UNLIKELY(aux != NULL)) {
- int32_t l;
- int32_t r;
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- int32_t va = prevAuxLevel;
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
- const int32_t vaInc = auxInc;
- // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, 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);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- prevAuxLevel = va;
- adjustVolumeRamp(true);
- }
-
- // constant gain
- else {
- const uint32_t vrl = volumeRL;
- const int16_t va = (int16_t)auxLevel;
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(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 (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, 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);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(false);
- }
-
- // constant gain
- else {
- const uint32_t vrl = volumeRL;
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- out[0] = mulAddRL(1, rl, vrl, out[0]);
- out[1] = mulAddRL(0, rl, vrl, out[1]);
- out += 2;
- } while (--frameCount);
- }
- }
- mIn = in;
-}
-
-void AudioMixer::Track::track__16BitsMono(
- int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
- ALOGVV("track__16BitsMono\n");
- const int16_t *in = static_cast<int16_t const *>(mIn);
-
- if (CC_UNLIKELY(aux != NULL)) {
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- int32_t va = prevAuxLevel;
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
- const int32_t vaInc = auxInc;
-
- // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, 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);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- prevAuxLevel = va;
- adjustVolumeRamp(true);
- }
- // constant gain
- else {
- const int16_t vl = volume[0];
- const int16_t vr = volume[1];
- const int16_t va = (int16_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 (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, 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);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(false);
- }
- // constant gain
- else {
- const int16_t vl = volume[0];
- const int16_t vr = 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);
- }
- }
- mIn = in;
-}
-
-// no-op case
-void AudioMixer::process__nop()
-{
- ALOGVV("process__nop\n");
-
- for (const auto &pair : mGroups) {
- // process by group of tracks with same output buffer to
- // avoid multiple memset() on same buffer
- const auto &group = pair.second;
-
- const std::shared_ptr<Track> &t = mTracks[group[0]];
- memset(t->mainBuffer, 0,
- mFrameCount * audio_bytes_per_frame(
- t->mMixerChannelCount + t->mMixerHapticChannelCount, t->mMixerFormat));
-
- // now consume data
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- size_t outFrames = mFrameCount;
- while (outFrames) {
- t->buffer.frameCount = outFrames;
- t->bufferProvider->getNextBuffer(&t->buffer);
- if (t->buffer.raw == NULL) break;
- outFrames -= t->buffer.frameCount;
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
- }
-}
-
-// generic code without resampling
-void AudioMixer::process__genericNoResampling()
-{
- ALOGVV("process__genericNoResampling\n");
- int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
-
- for (const auto &pair : mGroups) {
- // process by group of tracks with same output main buffer to
- // avoid multiple memset() on same buffer
- const auto &group = pair.second;
-
- // acquire buffer
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- t->buffer.frameCount = mFrameCount;
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->frameCount = t->buffer.frameCount;
- t->mIn = t->buffer.raw;
- }
-
- int32_t *out = (int *)pair.first;
- size_t numFrames = 0;
- do {
- const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
- memset(outTemp, 0, sizeof(outTemp));
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- int32_t *aux = NULL;
- if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
- aux = t->auxBuffer + numFrames;
- }
- for (int outFrames = frameCount; outFrames > 0; ) {
- // t->in == nullptr can happen if the track was flushed just after having
- // been enabled for mixing.
- if (t->mIn == nullptr) {
- break;
- }
- size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
- if (inFrames > 0) {
- (t.get()->*t->hook)(
- outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
- inFrames, mResampleTemp.get() /* naked ptr */, aux);
- t->frameCount -= inFrames;
- outFrames -= inFrames;
- if (CC_UNLIKELY(aux != NULL)) {
- aux += inFrames;
- }
- }
- if (t->frameCount == 0 && outFrames) {
- t->bufferProvider->releaseBuffer(&t->buffer);
- t->buffer.frameCount = (mFrameCount - numFrames) -
- (frameCount - outFrames);
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->mIn = t->buffer.raw;
- if (t->mIn == nullptr) {
- break;
- }
- t->frameCount = t->buffer.frameCount;
- }
- }
- }
-
- const std::shared_ptr<Track> &t1 = mTracks[group[0]];
- convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
- frameCount * t1->mMixerChannelCount);
- // TODO: fix ugly casting due to choice of out pointer type
- out = reinterpret_cast<int32_t*>((uint8_t*)out
- + frameCount * t1->mMixerChannelCount
- * audio_bytes_per_sample(t1->mMixerFormat));
- numFrames += frameCount;
- } while (numFrames < mFrameCount);
-
- // release each track's buffer
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
-}
-
-// generic code with resampling
-void AudioMixer::process__genericResampling()
-{
- ALOGVV("process__genericResampling\n");
- int32_t * const outTemp = mOutputTemp.get(); // naked ptr
- size_t numFrames = mFrameCount;
-
- for (const auto &pair : mGroups) {
- const auto &group = pair.second;
- const std::shared_ptr<Track> &t1 = mTracks[group[0]];
-
- // clear temp buffer
- memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- int32_t *aux = NULL;
- if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
- 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) {
- (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
- } else {
-
- size_t outFrames = 0;
-
- while (outFrames < numFrames) {
- t->buffer.frameCount = numFrames - outFrames;
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->mIn = t->buffer.raw;
- // t->mIn == nullptr can happen if the track was flushed just after having
- // been enabled for mixing.
- if (t->mIn == nullptr) break;
-
- (t.get()->*t->hook)(
- outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
- mResampleTemp.get() /* naked ptr */,
- aux != nullptr ? aux + outFrames : nullptr);
- outFrames += t->buffer.frameCount;
-
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
- }
- convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
- outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
- }
-}
-
-// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__oneTrack16BitsStereoNoResampling()
-{
- ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
- LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
- "%zu != 1 tracks enabled", mEnabled.size());
- const int name = mEnabled[0];
- const std::shared_ptr<Track> &t = mTracks[name];
-
- AudioBufferProvider::Buffer& b(t->buffer);
-
- int32_t* out = t->mainBuffer;
- float *fout = reinterpret_cast<float*>(out);
- size_t numFrames = mFrameCount;
-
- 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);
- const int16_t *in = b.i16;
-
- // in == NULL can happen if the track was flushed just after having
- // been enabled for mixing.
- if (in == NULL || (((uintptr_t)in) & 3)) {
- if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
- memset((char*)fout, 0, numFrames
- * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
- } else {
- memset((char*)out, 0, numFrames
- * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
- }
- ALOGE_IF((((uintptr_t)in) & 3),
- "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
- " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
- in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
- return;
- }
- size_t outFrames = b.frameCount;
-
- switch (t->mMixerFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl);
- int32_t r = mulRL(0, rl, vrl);
- *fout++ = float_from_q4_27(l);
- *fout++ = float_from_q4_27(r);
- // Note: In case of later int16_t sink output,
- // conversion and clamping is done by memcpy_to_i16_from_float().
- } while (--outFrames);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
- // volume is boosted, so we might need to clamp even though
- // we process only one track.
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(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<const uint32_t *>(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);
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
- }
- numFrames -= b.frameCount;
- t->bufferProvider->releaseBuffer(&b);
- }
-}
-
/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
/*static*/ void AudioMixer::sInitRoutine()
@@ -1664,211 +523,71 @@
DownmixerBufferProvider::init(); // for the downmixer
}
-/* TODO: consider whether this level of optimization is necessary.
- * Perhaps just stick with a single for loop.
- */
-
-// Needs to derive a compile time constant (constexpr). Could be targeted to go
-// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
-#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
- (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
-
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
- typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
- const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixer::preCreateTrack()
{
- switch (channels) {
- case 1:
- volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 2:
- volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 3:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 4:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 5:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 6:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 7:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 8:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- }
+ return std::make_shared<Track>();
}
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
- typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
- const TI* in, TA* aux, const TV *vol, TAV vola)
+status_t AudioMixer::postCreateTrack(TrackBase *track)
{
- switch (channels) {
- case 1:
- volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
- break;
- case 2:
- volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
- break;
- case 3:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
- break;
- case 4:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
- break;
- case 5:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
- break;
- case 6:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
- break;
- case 7:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
- break;
- case 8:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
- break;
+ Track* t = static_cast<Track*>(track);
+
+ audio_channel_mask_t channelMask = t->channelMask;
+ t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
+ t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
+ 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 = HAPTIC_SCALE_NONE;
+ t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
+ t->mMixerHapticChannelCount = 0;
+ t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
+ t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
+ t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
+ t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
+ t->mKeepContractedChannels = false;
+ // 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->prepareForAdjustChannelsNonDestructive(mFrameCount);
+ t->prepareForAdjustChannels();
+ return OK;
}
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * USEFLOATVOL (set to true if float volume is used)
- * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
- typename TO, typename TI, typename TA>
-void AudioMixer::Track::volumeMix(TO *out, size_t outFrames,
- const TI *in, TA *aux, bool ramp)
+void AudioMixer::preProcess()
{
- if (USEFLOATVOL) {
- if (ramp) {
- volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- mPrevVolume, mVolumeInc,
-#ifdef FLOAT_AUX
- &mPrevAuxLevel, mAuxInc
-#else
- &prevAuxLevel, auxInc
-#endif
- );
- if (ADJUSTVOL) {
- adjustVolumeRamp(aux != NULL, true);
- }
- } else {
- volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- mVolume,
-#ifdef FLOAT_AUX
- mAuxLevel
-#else
- auxLevel
-#endif
- );
- }
- } else {
- if (ramp) {
- volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- prevVolume, volumeInc, &prevAuxLevel, auxInc);
- if (ADJUSTVOL) {
- adjustVolumeRamp(aux != NULL);
- }
- } else {
- volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- volume, auxLevel);
+ 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();
}
}
}
-/* This process hook is called when there is a single track without
- * aux buffer, volume ramp, or resampling.
- * TODO: Update the hook selection: this can properly handle aux and ramp.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::process__noResampleOneTrack()
+void AudioMixer::postProcess()
{
- ALOGVV("process__noResampleOneTrack\n");
- LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
- "%zu != 1 tracks enabled", mEnabled.size());
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- const uint32_t channels = t->mMixerChannelCount;
- TO* out = reinterpret_cast<TO*>(t->mainBuffer);
- TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
- const bool ramp = t->needsRamp();
-
- for (size_t numFrames = mFrameCount; numFrames > 0; ) {
- AudioBufferProvider::Buffer& b(t->buffer);
- // get input buffer
- b.frameCount = numFrames;
- t->bufferProvider->getNextBuffer(&b);
- const TI *in = reinterpret_cast<TI*>(b.raw);
-
- // in == NULL can happen if the track was flushed just after having
- // been enabled for mixing.
- if (in == NULL || (((uintptr_t)in) & 3)) {
- memset(out, 0, numFrames
- * channels * audio_bytes_per_sample(t->mMixerFormat));
- ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
- "buffer %p track %p, channels %d, needs %#x",
- in, &t, t->channelCount, t->needs);
- return;
- }
-
- const size_t outFrames = b.frameCount;
- t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
- out, outFrames, in, aux, ramp);
-
- out += outFrames * channels;
- if (aux != NULL) {
- aux += outFrames;
- }
- numFrames -= b.frameCount;
-
- // release buffer
- t->bufferProvider->releaseBuffer(&b);
- }
- if (ramp) {
- t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
- }
-}
-
-void AudioMixer::processHapticData()
-{
+ // 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 = mTracks[name];
+ const std::shared_ptr<Track> &t = getTrack(name);
if (t->mHapticPlaybackEnabled) {
size_t sampleCount = mFrameCount * t->mMixerHapticChannelCount;
float gamma = t->getHapticScaleGamma();
@@ -1895,225 +614,5 @@
}
}
-/* This track hook is called to do resampling then mixing,
- * pulling from the track's upstream AudioBufferProvider.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
-{
- ALOGVV("track__Resample\n");
- mResampler->setSampleRate(sampleRate);
- const bool ramp = needsRamp();
- if (ramp || aux != NULL) {
- // if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step.
- // if aux != NULL: resample with unity gain to temp buffer then apply send level.
-
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
- mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
-
- volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
- out, outFrameCount, temp, aux, ramp);
-
- } else { // constant volume gain
- mResampler->setVolume(mVolume[0], mVolume[1]);
- mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
- }
-}
-
-/* This track hook is called to mix a track, when no resampling is required.
- * The input buffer should be present in in.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux)
-{
- ALOGVV("track__NoResample\n");
- const TI *in = static_cast<const TI *>(mIn);
-
- volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
- out, frameCount, in, aux, needsRamp());
-
- // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
- // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
- in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
- mIn = in;
-}
-
-/* The Mixer engine generates either int32_t (Q4_27) or float data.
- * We use this function to convert the engine buffers
- * to the desired mixer output format, either int16_t (Q.15) or float.
- */
-/* static */
-void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
- void *in, audio_format_t mixerInFormat, size_t sampleCount)
-{
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
-}
-
-/* Returns the proper track hook to use for mixing the track into the output buffer.
- */
-/* static */
-AudioMixer::hook_t AudioMixer::Track::getTrackHook(int trackType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
-{
- if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
- switch (trackType) {
- case TRACKTYPE_NOP:
- return &Track::track__nop;
- case TRACKTYPE_RESAMPLE:
- return &Track::track__genericResample;
- case TRACKTYPE_NORESAMPLEMONO:
- return &Track::track__16BitsMono;
- case TRACKTYPE_NORESAMPLE:
- return &Track::track__16BitsStereo;
- default:
- LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
- break;
- }
- }
- LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
- switch (trackType) {
- case TRACKTYPE_NOP:
- return &Track::track__nop;
- case TRACKTYPE_RESAMPLE:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__Resample<
- MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__Resample<
- MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- case TRACKTYPE_NORESAMPLEMONO:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- case TRACKTYPE_NORESAMPLE:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
- break;
- }
- return NULL;
-}
-
-/* Returns the proper process hook for mixing tracks. Currently works only for
- * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
- *
- * TODO: Due to the special mixing considerations of duplicating to
- * a stereo output track, the input track cannot be MONO. This should be
- * prevented by the caller.
- */
-/* static */
-AudioMixer::process_hook_t AudioMixer::getProcessHook(
- int processType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
-{
- if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
- LOG_ALWAYS_FATAL("bad processType: %d", processType);
- return NULL;
- }
- if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
- return &AudioMixer::process__oneTrack16BitsStereoNoResampling;
- }
- LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- return NULL;
-}
-
// ----------------------------------------------------------------------------
} // namespace android