| /* |
| ** |
| ** Copyright 2019, The Android Open Source Project |
| ** |
| ** Licensed under the Apache License, Version 2.0 (the "License"); |
| ** you may not use this file except in compliance with the License. |
| ** You may obtain a copy of the License at |
| ** |
| ** http://www.apache.org/licenses/LICENSE-2.0 |
| ** |
| ** Unless required by applicable law or agreed to in writing, software |
| ** distributed under the License is distributed on an "AS IS" BASIS, |
| ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| ** See the License for the specific language governing permissions and |
| ** limitations under the License. |
| */ |
| |
| #define LOG_TAG "AudioMixer" |
| //#define LOG_NDEBUG 0 |
| |
| #include <sstream> |
| #include <string.h> |
| |
| #include <audio_utils/primitives.h> |
| #include <cutils/compiler.h> |
| #include <media/AudioMixerBase.h> |
| #include <utils/Log.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 |
| |
| // TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore. |
| static constexpr int BLOCKSIZE = 16; |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| |
| bool AudioMixerBase::isValidFormat(audio_format_t format) const |
| { |
| switch (format) { |
| case AUDIO_FORMAT_PCM_8_BIT: |
| case AUDIO_FORMAT_PCM_16_BIT: |
| case AUDIO_FORMAT_PCM_24_BIT_PACKED: |
| case AUDIO_FORMAT_PCM_32_BIT: |
| case AUDIO_FORMAT_PCM_FLOAT: |
| return true; |
| default: |
| return false; |
| } |
| } |
| |
| bool AudioMixerBase::isValidChannelMask(audio_channel_mask_t channelMask) const |
| { |
| return audio_channel_count_from_out_mask(channelMask) <= MAX_NUM_CHANNELS; |
| } |
| |
| std::shared_ptr<AudioMixerBase::TrackBase> AudioMixerBase::preCreateTrack() |
| { |
| return std::make_shared<TrackBase>(); |
| } |
| |
| status_t AudioMixerBase::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 = preCreateTrack(); |
| { |
| // 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->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->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT; |
| t->mFormat = format; |
| t->mMixerInFormat = kUseFloat && kUseNewMixer ? |
| AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; |
| 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); |
| status_t status = postCreateTrack(t.get()); |
| if (status != OK) return status; |
| mTracks[name] = t; |
| return OK; |
| } |
| } |
| |
| // Called when channel masks have changed for a track name |
| bool AudioMixerBase::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<TrackBase> &track = mTracks[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); |
| |
| ALOG_ASSERT(trackChannelCount && mixerChannelCount); |
| track->channelMask = trackChannelMask; |
| track->channelCount = trackChannelCount; |
| track->mMixerChannelMask = mixerChannelMask; |
| track->mMixerChannelCount = mixerChannelCount; |
| |
| // Resampler channels may have changed. |
| track->recreateResampler(mSampleRate); |
| return true; |
| } |
| |
| void AudioMixerBase::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 AudioMixerBase::enable(int name) |
| { |
| LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name); |
| const std::shared_ptr<TrackBase> &track = mTracks[name]; |
| |
| if (!track->enabled) { |
| track->enabled = true; |
| ALOGV("enable(%d)", name); |
| invalidate(); |
| } |
| } |
| |
| void AudioMixerBase::disable(int name) |
| { |
| LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name); |
| const std::shared_ptr<TrackBase> &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 = AudioMixerBase::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 > AudioMixerBase::UNITY_GAIN_FLOAT) { |
| newVolume = AudioMixerBase::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 * AudioMixerBase::UNITY_GAIN_INT; // not neg, subnormal, nan |
| const int32_t intVolume = (scaledVolume >= (float)AudioMixerBase::UNITY_GAIN_INT) ? |
| AudioMixerBase::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 AudioMixerBase::setParameter(int name, int target, int param, void *value) |
| { |
| LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name); |
| const std::shared_ptr<TrackBase> &track = mTracks[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); |
| invalidate(); |
| } |
| } break; |
| case MAIN_BUFFER: |
| if (track->mainBuffer != valueBuf) { |
| track->mainBuffer = valueBuf; |
| ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf); |
| invalidate(); |
| } |
| break; |
| case AUX_BUFFER: |
| if (track->auxBuffer != valueBuf) { |
| track->auxBuffer = valueBuf; |
| ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf); |
| invalidate(); |
| } |
| 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); |
| invalidate(); |
| } |
| } 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); |
| invalidate(); |
| } |
| } 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)); |
| 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); |
| } |
| } |
| break; |
| |
| default: |
| LOG_ALWAYS_FATAL("setParameter: bad target %d", target); |
| } |
| } |
| |
| bool AudioMixerBase::TrackBase::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 = getOutputChannelCount(); |
| 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; |
| } |
| |
| /* 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. |
| */ |
| void AudioMixerBase::TrackBase::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; |
| } |
| } |
| } |
| |
| void AudioMixerBase::TrackBase::recreateResampler(uint32_t devSampleRate) |
| { |
| if (mResampler.get() != nullptr) { |
| const uint32_t resetToSampleRate = sampleRate; |
| mResampler.reset(nullptr); |
| sampleRate = devSampleRate; // without resampler, track rate is device sample rate. |
| // recreate the resampler with updated format, channels, saved sampleRate. |
| setResampler(resetToSampleRate /*trackSampleRate*/, devSampleRate); |
| } |
| } |
| |
| size_t AudioMixerBase::getUnreleasedFrames(int name) const |
| { |
| const auto it = mTracks.find(name); |
| if (it != mTracks.end()) { |
| return it->second->getUnreleasedFrames(); |
| } |
| return 0; |
| } |
| |
| std::string AudioMixerBase::trackNames() const |
| { |
| std::stringstream ss; |
| for (const auto &pair : mTracks) { |
| ss << pair.first << " "; |
| } |
| return ss.str(); |
| } |
| |
| void AudioMixerBase::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<TrackBase> &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 = &TrackBase::track__nop; |
| } else { |
| if (n & NEEDS_AUX) { |
| all16BitsStereoNoResample = false; |
| } |
| if (n & NEEDS_RESAMPLE) { |
| all16BitsStereoNoResample = false; |
| resampling = true; |
| t->hook = TrackBase::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 = TrackBase::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 = TrackBase::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 = &AudioMixerBase::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 = &AudioMixerBase::process__genericResampling; |
| } else { |
| // we keep temp arrays around. |
| mHook = &AudioMixerBase::process__genericNoResampling; |
| if (all16BitsStereoNoResample && !volumeRamp) { |
| if (mEnabled.size() == 1) { |
| const std::shared_ptr<TrackBase> &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<TrackBase> &t = mTracks[name]; |
| if (!t->doesResample() && t->volumeRL == 0) { |
| t->needs |= NEEDS_MUTE; |
| t->hook = &TrackBase::track__nop; |
| } else { |
| allMuted = false; |
| } |
| } |
| if (allMuted) { |
| mHook = &AudioMixerBase::process__nop; |
| } else if (all16BitsStereoNoResample) { |
| if (mEnabled.size() == 1) { |
| //const int i = 31 - __builtin_clz(enabledTracks); |
| const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]]; |
| // Muted single tracks handled by allMuted above. |
| mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK, |
| t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat); |
| } |
| } |
| } |
| } |
| |
| void AudioMixerBase::TrackBase::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 AudioMixerBase::TrackBase::track__nop(int32_t* out __unused, |
| size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused) |
| { |
| } |
| |
| void AudioMixerBase::TrackBase::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 AudioMixerBase::TrackBase::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 AudioMixerBase::TrackBase::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 AudioMixerBase::TrackBase::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 AudioMixerBase::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<TrackBase> &t = mTracks[group[0]]; |
| memset(t->mainBuffer, 0, |
| mFrameCount * audio_bytes_per_frame(t->getMixerChannelCount(), t->mMixerFormat)); |
| |
| // now consume data |
| for (const int name : group) { |
| const std::shared_ptr<TrackBase> &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 AudioMixerBase::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<TrackBase> &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<TrackBase> &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<TrackBase> &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<TrackBase> &t = mTracks[name]; |
| t->bufferProvider->releaseBuffer(&t->buffer); |
| } |
| } |
| } |
| |
| // generic code with resampling |
| void AudioMixerBase::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<TrackBase> &t1 = mTracks[group[0]]; |
| |
| // clear temp buffer |
| memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount); |
| for (const int name : group) { |
| const std::shared_ptr<TrackBase> &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 AudioMixerBase::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<TrackBase> &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); |
| } |
| } |
| |
| /* 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) |
| { |
| 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) 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) |
| { |
| 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) or float |
| */ |
| template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL, |
| typename TO, typename TI, typename TA> |
| void AudioMixerBase::TrackBase::volumeMix(TO *out, size_t outFrames, |
| const TI *in, TA *aux, bool ramp) |
| { |
| 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); |
| } |
| } |
| } |
| |
| /* 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 AudioMixerBase::process__noResampleOneTrack() |
| { |
| ALOGVV("process__noResampleOneTrack\n"); |
| LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1, |
| "%zu != 1 tracks enabled", mEnabled.size()); |
| const std::shared_ptr<TrackBase> &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); |
| } |
| } |
| |
| /* 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 AudioMixerBase::TrackBase::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 AudioMixerBase::TrackBase::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 AudioMixerBase::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 */ |
| AudioMixerBase::hook_t AudioMixerBase::TrackBase::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 &TrackBase::track__nop; |
| case TRACKTYPE_RESAMPLE: |
| return &TrackBase::track__genericResample; |
| case TRACKTYPE_NORESAMPLEMONO: |
| return &TrackBase::track__16BitsMono; |
| case TRACKTYPE_NORESAMPLE: |
| return &TrackBase::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 &TrackBase::track__nop; |
| case TRACKTYPE_RESAMPLE: |
| switch (mixerInFormat) { |
| case AUDIO_FORMAT_PCM_FLOAT: |
| return (AudioMixerBase::hook_t) &TrackBase::track__Resample< |
| MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>; |
| case AUDIO_FORMAT_PCM_16_BIT: |
| return (AudioMixerBase::hook_t) &TrackBase::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 (AudioMixerBase::hook_t) &TrackBase::track__NoResample< |
| MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>; |
| case AUDIO_FORMAT_PCM_16_BIT: |
| return (AudioMixerBase::hook_t) &TrackBase::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 (AudioMixerBase::hook_t) &TrackBase::track__NoResample< |
| MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>; |
| case AUDIO_FORMAT_PCM_16_BIT: |
| return (AudioMixerBase::hook_t) &TrackBase::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 */ |
| AudioMixerBase::process_hook_t AudioMixerBase::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 &AudioMixerBase::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 &AudioMixerBase::process__noResampleOneTrack< |
| MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>; |
| case AUDIO_FORMAT_PCM_16_BIT: |
| return &AudioMixerBase::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 &AudioMixerBase::process__noResampleOneTrack< |
| MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>; |
| case AUDIO_FORMAT_PCM_16_BIT: |
| return &AudioMixerBase::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 |
