AudioFlinger: Split off audio processing library
Test: native AudioResampler test, general playback test
Bug: 31015569
Change-Id: Ifb248f4402a583438d756c014dcd7a4577aef713
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
new file mode 100644
index 0000000..a7d9f0f
--- /dev/null
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -0,0 +1,2085 @@
+/*
+**
+** 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 <stdint.h>
+#include <string.h>
+#include <stdlib.h>
+#include <math.h>
+#include <sys/types.h>
+
+#include <utils/Errors.h>
+#include <utils/Log.h>
+
+#include <cutils/bitops.h>
+#include <cutils/compiler.h>
+#include <utils/Debug.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
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
+#endif
+
+// TODO: Move these macro/inlines to a header file.
+template <typename T>
+static inline
+T max(const T& x, const T& y) {
+ return x > y ? x : y;
+}
+
+// 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 const 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 const bool kUseFloat = true;
+
+// Set to default copy buffer size in frames for input processing.
+static const size_t kCopyBufferFrameCount = 256;
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+template <typename T>
+T min(const T& a, const T& b)
+{
+ return a < b ? a : b;
+}
+
+// ----------------------------------------------------------------------------
+
+// Ensure mConfiguredNames bitmask is initialized properly on all architectures.
+// The value of 1 << x is undefined in C when x >= 32.
+
+AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks)
+ : mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1),
+ mSampleRate(sampleRate)
+{
+ ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u",
+ maxNumTracks, MAX_NUM_TRACKS);
+
+ // AudioMixer is not yet capable of more than 32 active track inputs
+ ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS);
+
+ pthread_once(&sOnceControl, &sInitRoutine);
+
+ mState.enabledTracks= 0;
+ mState.needsChanged = 0;
+ mState.frameCount = frameCount;
+ mState.hook = process__nop;
+ mState.outputTemp = NULL;
+ mState.resampleTemp = NULL;
+ mState.mLog = &mDummyLog;
+ // mState.reserved
+
+ // FIXME Most of the following initialization is probably redundant since
+ // tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0
+ // and mTrackNames is initially 0. However, leave it here until that's verified.
+ track_t* t = mState.tracks;
+ for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
+ t->resampler = NULL;
+ t->downmixerBufferProvider = NULL;
+ t->mReformatBufferProvider = NULL;
+ t->mTimestretchBufferProvider = NULL;
+ t++;
+ }
+
+}
+
+AudioMixer::~AudioMixer()
+{
+ track_t* t = mState.tracks;
+ for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
+ delete t->resampler;
+ delete t->downmixerBufferProvider;
+ delete t->mReformatBufferProvider;
+ delete t->mTimestretchBufferProvider;
+ t++;
+ }
+ delete [] mState.outputTemp;
+ delete [] mState.resampleTemp;
+}
+
+void AudioMixer::setLog(NBLog::Writer *log)
+{
+ mState.mLog = log;
+}
+
+static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
+ return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+}
+
+int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId)
+{
+ if (!isValidPcmTrackFormat(format)) {
+ ALOGE("AudioMixer::getTrackName invalid format (%#x)", format);
+ return -1;
+ }
+ uint32_t names = (~mTrackNames) & mConfiguredNames;
+ if (names != 0) {
+ int n = __builtin_ctz(names);
+ ALOGV("add track (%d)", n);
+ // assume default parameters for the track, except where noted below
+ track_t* t = &mState.tracks[n];
+ 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] = UNITY_GAIN_INT;
+ t->volume[1] = UNITY_GAIN_INT;
+ t->prevVolume[0] = UNITY_GAIN_INT << 16;
+ t->prevVolume[1] = UNITY_GAIN_INT << 16;
+ t->volumeInc[0] = 0;
+ t->volumeInc[1] = 0;
+ t->auxLevel = 0;
+ t->auxInc = 0;
+ t->prevAuxLevel = 0;
+
+ // Floating point volume.
+ t->mVolume[0] = UNITY_GAIN_FLOAT;
+ t->mVolume[1] = UNITY_GAIN_FLOAT;
+ t->mPrevVolume[0] = UNITY_GAIN_FLOAT;
+ t->mPrevVolume[1] = UNITY_GAIN_FLOAT;
+ t->mVolumeInc[0] = 0.;
+ t->mVolumeInc[1] = 0.;
+ t->mAuxLevel = 0.;
+ t->mAuxInc = 0.;
+ t->mPrevAuxLevel = 0.;
+
+ // no initialization needed
+ // t->frameCount
+ 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->in = NULL;
+ t->resampler = 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->mReformatBufferProvider = NULL;
+ t->downmixerBufferProvider = NULL;
+ t->mPostDownmixReformatBufferProvider = NULL;
+ t->mTimestretchBufferProvider = 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;
+ // Check the downmixing (or upmixing) requirements.
+ status_t status = t->prepareForDownmix();
+ if (status != OK) {
+ ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
+ return -1;
+ }
+ // prepareForDownmix() may change mDownmixRequiresFormat
+ ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
+ t->prepareForReformat();
+ mTrackNames |= 1 << n;
+ return TRACK0 + n;
+ }
+ ALOGE("AudioMixer::getTrackName out of available tracks");
+ return -1;
+}
+
+void AudioMixer::invalidateState(uint32_t mask)
+{
+ if (mask != 0) {
+ mState.needsChanged |= mask;
+ mState.hook = process__validate;
+ }
+ }
+
+// 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) {
+ track_t &track = mState.tracks[name];
+
+ if (trackChannelMask == track.channelMask
+ && mixerChannelMask == track.mMixerChannelMask) {
+ return false; // no need to change
+ }
+ // 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 bool mixerChannelCountChanged = track.mMixerChannelCount != mixerChannelCount;
+
+ ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX)
+ && trackChannelCount
+ && mixerChannelCount);
+ track.channelMask = trackChannelMask;
+ track.channelCount = trackChannelCount;
+ track.mMixerChannelMask = mixerChannelMask;
+ track.mMixerChannelCount = mixerChannelCount;
+
+ // 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 audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat;
+ const status_t status = mState.tracks[name].prepareForDownmix();
+ ALOGE_IF(status != OK,
+ "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
+ status, track.channelMask, track.mMixerChannelMask);
+
+ if (prevDownmixerFormat != track.mDownmixRequiresFormat) {
+ track.prepareForReformat(); // because of downmixer, track format may change!
+ }
+
+ if (track.resampler && mixerChannelCountChanged) {
+ // resampler channels may have changed.
+ const uint32_t resetToSampleRate = track.sampleRate;
+ delete track.resampler;
+ track.resampler = NULL;
+ 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*/);
+ }
+ return true;
+}
+
+void AudioMixer::track_t::unprepareForDownmix() {
+ ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
+
+ mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
+ if (downmixerBufferProvider != NULL) {
+ // this track had previously been configured with a downmixer, delete it
+ ALOGV(" deleting old downmixer");
+ delete downmixerBufferProvider;
+ downmixerBufferProvider = NULL;
+ reconfigureBufferProviders();
+ } else {
+ ALOGV(" nothing to do, no downmixer to delete");
+ }
+}
+
+status_t AudioMixer::track_t::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 -> stereo currently is.
+ if (channelMask == mMixerChannelMask
+ || (channelMask == AUDIO_CHANNEL_OUT_MONO
+ && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
+ return NO_ERROR;
+ }
+ // DownmixerBufferProvider is only used for position masks.
+ if (audio_channel_mask_get_representation(channelMask)
+ == AUDIO_CHANNEL_REPRESENTATION_POSITION
+ && DownmixerBufferProvider::isMultichannelCapable()) {
+ DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask,
+ mMixerChannelMask,
+ AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */,
+ sampleRate, sessionId, kCopyBufferFrameCount);
+
+ if (pDbp->isValid()) { // if constructor completed properly
+ mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
+ downmixerBufferProvider = pDbp;
+ reconfigureBufferProviders();
+ return NO_ERROR;
+ }
+ delete pDbp;
+ }
+
+ // Effect downmixer does not accept the channel conversion. Let's use our remixer.
+ RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask,
+ mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount);
+ // Remix always finds a conversion whereas Downmixer effect above may fail.
+ downmixerBufferProvider = pRbp;
+ reconfigureBufferProviders();
+ return NO_ERROR;
+}
+
+void AudioMixer::track_t::unprepareForReformat() {
+ ALOGV("AudioMixer::unprepareForReformat(%p)", this);
+ bool requiresReconfigure = false;
+ if (mReformatBufferProvider != NULL) {
+ delete mReformatBufferProvider;
+ mReformatBufferProvider = NULL;
+ requiresReconfigure = true;
+ }
+ if (mPostDownmixReformatBufferProvider != NULL) {
+ delete mPostDownmixReformatBufferProvider;
+ mPostDownmixReformatBufferProvider = NULL;
+ requiresReconfigure = true;
+ }
+ if (requiresReconfigure) {
+ reconfigureBufferProviders();
+ }
+}
+
+status_t AudioMixer::track_t::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 = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(channelMask),
+ mFormat,
+ targetFormat,
+ kCopyBufferFrameCount);
+ requiresReconfigure = true;
+ }
+ if (targetFormat != mMixerInFormat) {
+ mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(mMixerChannelMask),
+ targetFormat,
+ mMixerInFormat,
+ kCopyBufferFrameCount);
+ requiresReconfigure = true;
+ }
+ if (requiresReconfigure) {
+ reconfigureBufferProviders();
+ }
+ return NO_ERROR;
+}
+
+void AudioMixer::track_t::reconfigureBufferProviders()
+{
+ bufferProvider = mInputBufferProvider;
+ if (mReformatBufferProvider) {
+ mReformatBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = mReformatBufferProvider;
+ }
+ if (downmixerBufferProvider) {
+ downmixerBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = downmixerBufferProvider;
+ }
+ if (mPostDownmixReformatBufferProvider) {
+ mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = mPostDownmixReformatBufferProvider;
+ }
+ if (mTimestretchBufferProvider) {
+ mTimestretchBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = mTimestretchBufferProvider;
+ }
+}
+
+void AudioMixer::deleteTrackName(int name)
+{
+ ALOGV("AudioMixer::deleteTrackName(%d)", name);
+ name -= TRACK0;
+ LOG_ALWAYS_FATAL_IF(name < 0 || name >= (int)MAX_NUM_TRACKS, "bad track name %d", name);
+ ALOGV("deleteTrackName(%d)", name);
+ track_t& track(mState.tracks[ name ]);
+ if (track.enabled) {
+ track.enabled = false;
+ invalidateState(1<<name);
+ }
+ // delete the resampler
+ delete track.resampler;
+ track.resampler = NULL;
+ // delete the downmixer
+ mState.tracks[name].unprepareForDownmix();
+ // delete the reformatter
+ mState.tracks[name].unprepareForReformat();
+ // delete the timestretch provider
+ delete track.mTimestretchBufferProvider;
+ track.mTimestretchBufferProvider = NULL;
+ mTrackNames &= ~(1<<name);
+}
+
+void AudioMixer::enable(int name)
+{
+ name -= TRACK0;
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+ track_t& track = mState.tracks[name];
+
+ if (!track.enabled) {
+ track.enabled = true;
+ ALOGV("enable(%d)", name);
+ invalidateState(1 << name);
+ }
+}
+
+void AudioMixer::disable(int name)
+{
+ name -= TRACK0;
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+ track_t& track = mState.tracks[name];
+
+ if (track.enabled) {
+ track.enabled = false;
+ ALOGV("disable(%d)", name);
+ invalidateState(1 << name);
+ }
+}
+
+/* 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
+ const float maxv = max(newVolume, *pPrevVolume); // could be inf, cannot be nan, subnormal
+
+ 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)
+{
+ name -= TRACK0;
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+ track_t& track = mState.tracks[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, track.mMixerChannelMask)) {
+ ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
+ invalidateState(1 << name);
+ }
+ } break;
+ case MAIN_BUFFER:
+ if (track.mainBuffer != valueBuf) {
+ track.mainBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+ invalidateState(1 << name);
+ }
+ break;
+ case AUX_BUFFER:
+ if (track.auxBuffer != valueBuf) {
+ track.auxBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+ invalidateState(1 << name);
+ }
+ 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();
+ invalidateState(1 << name);
+ }
+ } 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);
+ }
+ } break;
+ case MIXER_CHANNEL_MASK: {
+ const audio_channel_mask_t mixerChannelMask =
+ static_cast<audio_channel_mask_t>(valueInt);
+ if (setChannelMasks(name, track.channelMask, mixerChannelMask)) {
+ ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
+ invalidateState(1 << name);
+ }
+ } break;
+ default:
+ LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
+ }
+ 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));
+ invalidateState(1 << name);
+ }
+ break;
+ case RESET:
+ track.resetResampler();
+ invalidateState(1 << name);
+ break;
+ case REMOVE:
+ delete track.resampler;
+ track.resampler = NULL;
+ track.sampleRate = mSampleRate;
+ invalidateState(1 << name);
+ 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 ? mState.frameCount : 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);
+ invalidateState(1 << name);
+ }
+ break;
+ default:
+ if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
+ if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+ target == RAMP_VOLUME ? mState.frameCount : 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]);
+ invalidateState(1 << name);
+ }
+ } else {
+ LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
+ }
+ }
+ 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);
+ // invalidateState(1 << name);
+ }
+ } break;
+ default:
+ LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
+ }
+ break;
+
+ default:
+ LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
+ }
+}
+
+bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
+{
+ if (trackSampleRate != devSampleRate || resampler != NULL) {
+ if (sampleRate != trackSampleRate) {
+ sampleRate = trackSampleRate;
+ if (resampler == NULL) {
+ 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 = downmixerBufferProvider != NULL
+ ? mMixerChannelCount : channelCount;
+ ALOGVV("Creating resampler:"
+ " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
+ mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
+ resampler = AudioResampler::create(
+ mMixerInFormat,
+ resamplerChannelCount,
+ devSampleRate, quality);
+ }
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioMixer::track_t::setPlaybackRate(const AudioPlaybackRate &playbackRate)
+{
+ if ((mTimestretchBufferProvider == NULL &&
+ 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 == NULL) {
+ // 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 = downmixerBufferProvider != NULL
+ ? mMixerChannelCount : channelCount;
+ mTimestretchBufferProvider = new TimestretchBufferProvider(timestretchChannelCount,
+ mMixerInFormat, sampleRate, playbackRate);
+ reconfigureBufferProviders();
+ } else {
+ reinterpret_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider)
+ ->setPlaybackRate(playbackRate);
+ }
+ 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_t::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]);
+ }
+ }
+ }
+ /* TODO: aux is always integer regardless of output buffer type */
+ if (aux) {
+ if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
+ ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
+ auxInc = 0;
+ prevAuxLevel = auxLevel << 16;
+ mAuxInc = 0.;
+ mPrevAuxLevel = mAuxLevel;
+ } else {
+ //ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc);
+ }
+ }
+}
+
+size_t AudioMixer::getUnreleasedFrames(int name) const
+{
+ name -= TRACK0;
+ if (uint32_t(name) < MAX_NUM_TRACKS) {
+ return mState.tracks[name].getUnreleasedFrames();
+ }
+ return 0;
+}
+
+void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
+{
+ name -= TRACK0;
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+
+ if (mState.tracks[name].mInputBufferProvider == bufferProvider) {
+ return; // don't reset any buffer providers if identical.
+ }
+ if (mState.tracks[name].mReformatBufferProvider != NULL) {
+ mState.tracks[name].mReformatBufferProvider->reset();
+ } else if (mState.tracks[name].downmixerBufferProvider != NULL) {
+ mState.tracks[name].downmixerBufferProvider->reset();
+ } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
+ mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
+ } else if (mState.tracks[name].mTimestretchBufferProvider != NULL) {
+ mState.tracks[name].mTimestretchBufferProvider->reset();
+ }
+
+ mState.tracks[name].mInputBufferProvider = bufferProvider;
+ mState.tracks[name].reconfigureBufferProviders();
+}
+
+
+void AudioMixer::process()
+{
+ mState.hook(&mState);
+}
+
+
+void AudioMixer::process__validate(state_t* state)
+{
+ ALOGW_IF(!state->needsChanged,
+ "in process__validate() but nothing's invalid");
+
+ uint32_t changed = state->needsChanged;
+ state->needsChanged = 0; // clear the validation flag
+
+ // recompute which tracks are enabled / disabled
+ uint32_t enabled = 0;
+ uint32_t disabled = 0;
+ while (changed) {
+ const int i = 31 - __builtin_clz(changed);
+ const uint32_t mask = 1<<i;
+ changed &= ~mask;
+ track_t& t = state->tracks[i];
+ (t.enabled ? enabled : disabled) |= mask;
+ }
+ state->enabledTracks &= ~disabled;
+ state->enabledTracks |= enabled;
+
+ // compute everything we need...
+ int countActiveTracks = 0;
+ // 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;
+ uint32_t en = state->enabledTracks;
+ while (en) {
+ const int i = 31 - __builtin_clz(en);
+ en &= ~(1<<i);
+
+ countActiveTracks++;
+ track_t& t = state->tracks[i];
+ 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__nop;
+ } else {
+ if (n & NEEDS_AUX) {
+ all16BitsStereoNoResample = false;
+ }
+ if (n & NEEDS_RESAMPLE) {
+ all16BitsStereoNoResample = false;
+ resampling = true;
+ t.hook = getTrackHook(TRACKTYPE_RESAMPLE, t.mMixerChannelCount,
+ t.mMixerInFormat, t.mMixerFormat);
+ ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+ "Track %d needs downmix + resample", i);
+ } else {
+ if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
+ t.hook = 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 = getTrackHook(TRACKTYPE_NORESAMPLE, t.mMixerChannelCount,
+ t.mMixerInFormat, t.mMixerFormat);
+ ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+ "Track %d needs downmix", i);
+ }
+ }
+ }
+ }
+
+ // select the processing hooks
+ state->hook = process__nop;
+ if (countActiveTracks > 0) {
+ if (resampling) {
+ if (!state->outputTemp) {
+ state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
+ }
+ if (!state->resampleTemp) {
+ state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
+ }
+ state->hook = process__genericResampling;
+ } else {
+ if (state->outputTemp) {
+ delete [] state->outputTemp;
+ state->outputTemp = NULL;
+ }
+ if (state->resampleTemp) {
+ delete [] state->resampleTemp;
+ state->resampleTemp = NULL;
+ }
+ state->hook = process__genericNoResampling;
+ if (all16BitsStereoNoResample && !volumeRamp) {
+ if (countActiveTracks == 1) {
+ const int i = 31 - __builtin_clz(state->enabledTracks);
+ track_t& t = state->tracks[i];
+ 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.
+ state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+ t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
+ }
+ }
+ }
+ }
+ }
+
+ ALOGV("mixer configuration change: %d activeTracks (%08x) "
+ "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
+ countActiveTracks, state->enabledTracks,
+ all16BitsStereoNoResample, resampling, volumeRamp);
+
+ state->hook(state);
+
+ // Now that the volume ramp has been done, set optimal state and
+ // track hooks for subsequent mixer process
+ if (countActiveTracks > 0) {
+ bool allMuted = true;
+ uint32_t en = state->enabledTracks;
+ while (en) {
+ const int i = 31 - __builtin_clz(en);
+ en &= ~(1<<i);
+ track_t& t = state->tracks[i];
+ if (!t.doesResample() && t.volumeRL == 0) {
+ t.needs |= NEEDS_MUTE;
+ t.hook = track__nop;
+ } else {
+ allMuted = false;
+ }
+ }
+ if (allMuted) {
+ state->hook = process__nop;
+ } else if (all16BitsStereoNoResample) {
+ if (countActiveTracks == 1) {
+ const int i = 31 - __builtin_clz(state->enabledTracks);
+ track_t& t = state->tracks[i];
+ // Muted single tracks handled by allMuted above.
+ state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+ t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
+ }
+ }
+ }
+}
+
+
+void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
+ int32_t* temp, int32_t* aux)
+{
+ ALOGVV("track__genericResample\n");
+ 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
+ t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(int32_t));
+ t->resampler->resample(temp, outFrameCount, t->bufferProvider);
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
+ volumeRampStereo(t, out, outFrameCount, temp, aux);
+ } else {
+ volumeStereo(t, out, outFrameCount, temp, aux);
+ }
+ } else {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
+ t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ 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->mVolume[0], t->mVolume[1]);
+ t->resampler->resample(out, outFrameCount, t->bufferProvider);
+ }
+ }
+}
+
+void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused,
+ size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
+{
+}
+
+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];
+
+ //ALOGD("[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 (CC_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 (CC_UNLIKELY(aux != NULL)) {
+ const int16_t va = 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 __unused, int32_t* aux)
+{
+ ALOGVV("track__16BitsStereo\n");
+ const int16_t *in = static_cast<const int16_t *>(t->in);
+
+ if (CC_UNLIKELY(aux != NULL)) {
+ int32_t l;
+ int32_t r;
+ // ramp gain
+ if (CC_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;
+ // ALOGD("[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<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(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];
+
+ // ALOGD("[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<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);
+ }
+ }
+ t->in = in;
+}
+
+void AudioMixer::track__16BitsMono(track_t* t, 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 *>(t->in);
+
+ if (CC_UNLIKELY(aux != NULL)) {
+ // ramp gain
+ if (CC_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;
+
+ // ALOGD("[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 (CC_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];
+
+ // ALOGD("[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;
+}
+
+// no-op case
+void AudioMixer::process__nop(state_t* state)
+{
+ ALOGVV("process__nop\n");
+ uint32_t e0 = state->enabledTracks;
+ 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 (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
+ e1 &= ~(1<<i);
+ }
+ }
+ e0 &= ~(e1);
+
+ memset(t1.mainBuffer, 0, state->frameCount * t1.mMixerChannelCount
+ * audio_bytes_per_sample(t1.mMixerFormat));
+ }
+
+ while (e1) {
+ i = 31 - __builtin_clz(e1);
+ e1 &= ~(1<<i);
+ {
+ track_t& t3 = state->tracks[i];
+ size_t outFrames = state->frameCount;
+ while (outFrames) {
+ t3.buffer.frameCount = outFrames;
+ t3.bufferProvider->getNextBuffer(&t3.buffer);
+ if (t3.buffer.raw == NULL) break;
+ outFrames -= t3.buffer.frameCount;
+ t3.bufferProvider->releaseBuffer(&t3.buffer);
+ }
+ }
+ }
+ }
+}
+
+// generic code without resampling
+void AudioMixer::process__genericNoResampling(state_t* state)
+{
+ ALOGVV("process__genericNoResampling\n");
+ 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;
+ }
+
+ 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 (CC_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 (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
+ aux = t.auxBuffer + numFrames;
+ }
+ while (outFrames) {
+ // t.in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (t.in == NULL) {
+ enabledTracks &= ~(1<<i);
+ e1 &= ~(1<<i);
+ break;
+ }
+ size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
+ if (inFrames > 0) {
+ t.hook(&t, outTemp + (BLOCKSIZE - outFrames) * t.mMixerChannelCount,
+ inFrames, state->resampleTemp, 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 = (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;
+ }
+ }
+ }
+
+ convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat,
+ BLOCKSIZE * t1.mMixerChannelCount);
+ // TODO: fix ugly casting due to choice of out pointer type
+ out = reinterpret_cast<int32_t*>((uint8_t*)out
+ + BLOCKSIZE * t1.mMixerChannelCount
+ * audio_bytes_per_sample(t1.mMixerFormat));
+ 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)
+{
+ ALOGVV("process__genericResampling\n");
+ // this const just means that local variable outTemp doesn't change
+ int32_t* const outTemp = state->outputTemp;
+ 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 (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
+ e1 &= ~(1<<j);
+ }
+ }
+ e0 &= ~(e1);
+ int32_t *out = t1.mainBuffer;
+ memset(outTemp, 0, sizeof(*outTemp) * t1.mMixerChannelCount * state->frameCount);
+ while (e1) {
+ const int i = 31 - __builtin_clz(e1);
+ e1 &= ~(1<<i);
+ track_t& t = state->tracks[i];
+ 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.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 (CC_UNLIKELY(aux != NULL)) {
+ aux += outFrames;
+ }
+ t.hook(&t, outTemp + outFrames * t.mMixerChannelCount, t.buffer.frameCount,
+ state->resampleTemp, aux);
+ outFrames += t.buffer.frameCount;
+ t.bufferProvider->releaseBuffer(&t.buffer);
+ }
+ }
+ }
+ convertMixerFormat(out, t1.mMixerFormat,
+ outTemp, t1.mMixerInFormat, numFrames * t1.mMixerChannelCount);
+ }
+}
+
+// one track, 16 bits stereo without resampling is the most common case
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
+{
+ ALOGVV("process__OneTrack16BitsStereoNoResampling\n");
+ // This method is only called when state->enabledTracks has exactly
+ // one bit set. The asserts below would verify this, but are commented out
+ // since the whole point of this method is to optimize performance.
+ //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled");
+ const int i = 31 - __builtin_clz(state->enabledTracks);
+ //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
+ const track_t& t = state->tracks[i];
+
+ AudioBufferProvider::Buffer& b(t.buffer);
+
+ int32_t* out = t.mainBuffer;
+ float *fout = reinterpret_cast<float*>(out);
+ 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);
+ 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, i, 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()
+{
+ 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)
+ */
+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)
+{
+ 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;
+ }
+}
+
+/* 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 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)
+{
+ 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;
+ }
+}
+
+/* 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)
+ */
+template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+ typename TO, typename TI, typename TA>
+void AudioMixer::volumeMix(TO *out, size_t outFrames,
+ const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t)
+{
+ if (USEFLOATVOL) {
+ if (ramp) {
+ volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
+ t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc);
+ if (ADJUSTVOL) {
+ t->adjustVolumeRamp(aux != NULL, true);
+ }
+ } else {
+ volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
+ t->mVolume, t->auxLevel);
+ }
+ } else {
+ if (ramp) {
+ volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
+ t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+ if (ADJUSTVOL) {
+ t->adjustVolumeRamp(aux != NULL);
+ }
+ } else {
+ volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
+ t->volume, t->auxLevel);
+ }
+ }
+}
+
+/* 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(state_t* state)
+{
+ ALOGVV("process_NoResampleOneTrack\n");
+ // CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz.
+ const int i = 31 - __builtin_clz(state->enabledTracks);
+ ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
+ track_t *t = &state->tracks[i];
+ 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 = state->frameCount; numFrames; ) {
+ 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;
+ volumeMix<MIXTYPE, is_same<TI, float>::value, false> (
+ out, outFrames, in, aux, ramp, t);
+
+ out += outFrames * channels;
+ if (aux != NULL) {
+ aux += channels;
+ }
+ numFrames -= b.frameCount;
+
+ // release buffer
+ t->bufferProvider->releaseBuffer(&b);
+ }
+ if (ramp) {
+ t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
+ }
+}
+
+/* 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)
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
+{
+ ALOGVV("track__Resample\n");
+ t->resampler->setSampleRate(t->sampleRate);
+ const bool ramp = t->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.
+
+ t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO));
+ t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);
+
+ volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
+ out, outFrameCount, temp, aux, ramp, t);
+
+ } else { // constant volume gain
+ t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
+ t->resampler->resample((int32_t*)out, outFrameCount, t->bufferProvider);
+ }
+}
+
+/* This track hook is called to mix a track, when no resampling is required.
+ * The input buffer should be present in t->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)
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux)
+{
+ ALOGVV("track__NoResample\n");
+ const TI *in = static_cast<const TI *>(t->in);
+
+ volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
+ out, frameCount, in, aux, t->needsRamp(), t);
+
+ // 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 * t->mMixerChannelCount;
+ t->in = 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.
+ */
+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, (int32_t*)in, sampleCount);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ // two int16_t are produced per iteration
+ ditherAndClamp((int32_t*)out, (int32_t*)in, sampleCount >> 1);
+ 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.
+ */
+AudioMixer::hook_t AudioMixer::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__nop;
+ case TRACKTYPE_RESAMPLE:
+ return track__genericResample;
+ case TRACKTYPE_NORESAMPLEMONO:
+ return track__16BitsMono;
+ case TRACKTYPE_NORESAMPLE:
+ return 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__nop;
+ case TRACKTYPE_RESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixer::hook_t)
+ track__Resample<MIXTYPE_MULTI, float /*TO*/, float /*TI*/, int32_t /*TA*/>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)\
+ track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
+ 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__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>;
+ 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__NoResample<MIXTYPE_MULTI, float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
+ 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.
+ */
+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 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 process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
+ float /*TO*/, float /*TI*/, int32_t /*TA*/>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
+ int16_t, float, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
+ float, int16_t, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
+ int16_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ return NULL;
+}
+
+// ----------------------------------------------------------------------------
+} // namespace android