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1793 lines
68 KiB
1793 lines
68 KiB
/*
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**
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** Copyright 2019, The Android Open Source Project
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**
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** Licensed under the Apache License, Version 2.0 (the "License");
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** you may not use this file except in compliance with the License.
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** You may obtain a copy of the License at
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**
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** http://www.apache.org/licenses/LICENSE-2.0
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**
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** Unless required by applicable law or agreed to in writing, software
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** distributed under the License is distributed on an "AS IS" BASIS,
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** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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** See the License for the specific language governing permissions and
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** limitations under the License.
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*/
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#define LOG_TAG "AudioMixer"
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//#define LOG_NDEBUG 0
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#include <array>
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#include <sstream>
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#include <string.h>
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#include <audio_utils/primitives.h>
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#include <cutils/compiler.h>
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#include <media/AudioMixerBase.h>
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#include <utils/Log.h>
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#include "AudioMixerOps.h"
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// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
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#ifndef FCC_2
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#define FCC_2 2
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#endif
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// Look for MONO_HACK for any Mono hack involving legacy mono channel to
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// stereo channel conversion.
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/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
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* being used. This is a considerable amount of log spam, so don't enable unless you
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* are verifying the hook based code.
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*/
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//#define VERY_VERY_VERBOSE_LOGGING
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#ifdef VERY_VERY_VERBOSE_LOGGING
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#define ALOGVV ALOGV
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//define ALOGVV printf // for test-mixer.cpp
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#else
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#define ALOGVV(a...) do { } while (0)
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#endif
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// TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
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static constexpr int BLOCKSIZE = 16;
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namespace android {
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// ----------------------------------------------------------------------------
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bool AudioMixerBase::isValidFormat(audio_format_t format) const
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{
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switch (format) {
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case AUDIO_FORMAT_PCM_8_BIT:
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case AUDIO_FORMAT_PCM_16_BIT:
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case AUDIO_FORMAT_PCM_24_BIT_PACKED:
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case AUDIO_FORMAT_PCM_32_BIT:
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case AUDIO_FORMAT_PCM_FLOAT:
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return true;
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default:
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return false;
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}
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}
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bool AudioMixerBase::isValidChannelMask(audio_channel_mask_t channelMask) const
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{
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return audio_channel_count_from_out_mask(channelMask) <= MAX_NUM_CHANNELS;
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}
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std::shared_ptr<AudioMixerBase::TrackBase> AudioMixerBase::preCreateTrack()
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{
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return std::make_shared<TrackBase>();
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}
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status_t AudioMixerBase::create(
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int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
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{
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LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
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if (!isValidChannelMask(channelMask)) {
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ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
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return BAD_VALUE;
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}
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if (!isValidFormat(format)) {
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ALOGE("%s invalid format: %#x", __func__, format);
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return BAD_VALUE;
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}
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auto t = preCreateTrack();
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{
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// TODO: move initialization to the Track constructor.
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// assume default parameters for the track, except where noted below
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t->needs = 0;
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// Integer volume.
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// Currently integer volume is kept for the legacy integer mixer.
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// Will be removed when the legacy mixer path is removed.
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t->volume[0] = 0;
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t->volume[1] = 0;
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t->prevVolume[0] = 0 << 16;
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t->prevVolume[1] = 0 << 16;
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t->volumeInc[0] = 0;
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t->volumeInc[1] = 0;
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t->auxLevel = 0;
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t->auxInc = 0;
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t->prevAuxLevel = 0;
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// Floating point volume.
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t->mVolume[0] = 0.f;
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t->mVolume[1] = 0.f;
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t->mPrevVolume[0] = 0.f;
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t->mPrevVolume[1] = 0.f;
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t->mVolumeInc[0] = 0.;
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t->mVolumeInc[1] = 0.;
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t->mAuxLevel = 0.;
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t->mAuxInc = 0.;
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t->mPrevAuxLevel = 0.;
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// no initialization needed
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// t->frameCount
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t->channelCount = audio_channel_count_from_out_mask(channelMask);
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t->enabled = false;
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ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
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"Non-stereo channel mask: %d\n", channelMask);
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t->channelMask = channelMask;
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t->sessionId = sessionId;
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// setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
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t->bufferProvider = NULL;
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t->buffer.raw = NULL;
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// no initialization needed
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// t->buffer.frameCount
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t->hook = NULL;
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t->mIn = NULL;
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t->sampleRate = mSampleRate;
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// setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
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t->mainBuffer = NULL;
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t->auxBuffer = NULL;
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t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
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t->mFormat = format;
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t->mMixerInFormat = kUseFloat && kUseNewMixer ?
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AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
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t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
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AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
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t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
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status_t status = postCreateTrack(t.get());
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if (status != OK) return status;
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mTracks[name] = t;
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return OK;
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}
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}
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// Called when channel masks have changed for a track name
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bool AudioMixerBase::setChannelMasks(int name,
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audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask)
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{
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LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
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const std::shared_ptr<TrackBase> &track = mTracks[name];
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if (trackChannelMask == track->channelMask && mixerChannelMask == track->mMixerChannelMask) {
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return false; // no need to change
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}
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// always recompute for both channel masks even if only one has changed.
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const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
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const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
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ALOG_ASSERT(trackChannelCount && mixerChannelCount);
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track->channelMask = trackChannelMask;
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track->channelCount = trackChannelCount;
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track->mMixerChannelMask = mixerChannelMask;
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track->mMixerChannelCount = mixerChannelCount;
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// Resampler channels may have changed.
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track->recreateResampler(mSampleRate);
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return true;
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}
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void AudioMixerBase::destroy(int name)
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{
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LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
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ALOGV("deleteTrackName(%d)", name);
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if (mTracks[name]->enabled) {
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invalidate();
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}
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mTracks.erase(name); // deallocate track
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}
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void AudioMixerBase::enable(int name)
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{
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LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
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const std::shared_ptr<TrackBase> &track = mTracks[name];
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if (!track->enabled) {
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track->enabled = true;
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ALOGV("enable(%d)", name);
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invalidate();
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}
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}
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void AudioMixerBase::disable(int name)
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{
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LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
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const std::shared_ptr<TrackBase> &track = mTracks[name];
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if (track->enabled) {
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track->enabled = false;
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ALOGV("disable(%d)", name);
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invalidate();
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}
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}
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/* Sets the volume ramp variables for the AudioMixer.
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*
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* The volume ramp variables are used to transition from the previous
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* volume to the set volume. ramp controls the duration of the transition.
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* Its value is typically one state framecount period, but may also be 0,
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* meaning "immediate."
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*
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* FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
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* even if there is a nonzero floating point increment (in that case, the volume
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* change is immediate). This restriction should be changed when the legacy mixer
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* is removed (see #2).
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* FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
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* when no longer needed.
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*
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* @param newVolume set volume target in floating point [0.0, 1.0].
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* @param ramp number of frames to increment over. if ramp is 0, the volume
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* should be set immediately. Currently ramp should not exceed 65535 (frames).
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* @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
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* @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
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* @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
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* @param pSetVolume pointer to the float target volume, set on return.
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* @param pPrevVolume pointer to the float previous volume, set on return.
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* @param pVolumeInc pointer to the float increment per output audio frame, set on return.
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* @return true if the volume has changed, false if volume is same.
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*/
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static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
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int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
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float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
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// check floating point volume to see if it is identical to the previously
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// set volume.
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// We do not use a tolerance here (and reject changes too small)
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// as it may be confusing to use a different value than the one set.
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// If the resulting volume is too small to ramp, it is a direct set of the volume.
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if (newVolume == *pSetVolume) {
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return false;
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}
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if (newVolume < 0) {
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newVolume = 0; // should not have negative volumes
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} else {
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switch (fpclassify(newVolume)) {
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case FP_SUBNORMAL:
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case FP_NAN:
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newVolume = 0;
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break;
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case FP_ZERO:
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break; // zero volume is fine
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case FP_INFINITE:
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// Infinite volume could be handled consistently since
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// floating point math saturates at infinities,
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// but we limit volume to unity gain float.
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// ramp = 0; break;
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//
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newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
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break;
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case FP_NORMAL:
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default:
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// Floating point does not have problems with overflow wrap
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// that integer has. However, we limit the volume to
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// unity gain here.
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// TODO: Revisit the volume limitation and perhaps parameterize.
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if (newVolume > AudioMixerBase::UNITY_GAIN_FLOAT) {
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newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
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}
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break;
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}
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}
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// set floating point volume ramp
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if (ramp != 0) {
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// when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
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// is no computational mismatch; hence equality is checked here.
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ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
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" prev:%f set_to:%f", *pPrevVolume, *pSetVolume);
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const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
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// could be inf, cannot be nan, subnormal
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const float maxv = std::max(newVolume, *pPrevVolume);
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if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
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&& maxv + inc != maxv) { // inc must make forward progress
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*pVolumeInc = inc;
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// ramp is set now.
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// Note: if newVolume is 0, then near the end of the ramp,
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// it may be possible that the ramped volume may be subnormal or
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// temporarily negative by a small amount or subnormal due to floating
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// point inaccuracies.
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} else {
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ramp = 0; // ramp not allowed
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}
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}
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// compute and check integer volume, no need to check negative values
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// The integer volume is limited to "unity_gain" to avoid wrapping and other
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// audio artifacts, so it never reaches the range limit of U4.28.
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// We safely use signed 16 and 32 bit integers here.
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const float scaledVolume = newVolume * AudioMixerBase::UNITY_GAIN_INT; // not neg, subnormal, nan
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const int32_t intVolume = (scaledVolume >= (float)AudioMixerBase::UNITY_GAIN_INT) ?
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AudioMixerBase::UNITY_GAIN_INT : (int32_t)scaledVolume;
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// set integer volume ramp
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if (ramp != 0) {
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// integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
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// when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
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// is no computational mismatch; hence equality is checked here.
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ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
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" prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
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const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
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if (inc != 0) { // inc must make forward progress
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*pIntVolumeInc = inc;
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} else {
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ramp = 0; // ramp not allowed
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}
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}
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// if no ramp, or ramp not allowed, then clear float and integer increments
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if (ramp == 0) {
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*pVolumeInc = 0;
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*pPrevVolume = newVolume;
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*pIntVolumeInc = 0;
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*pIntPrevVolume = intVolume << 16;
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}
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*pSetVolume = newVolume;
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*pIntSetVolume = intVolume;
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return true;
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}
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void AudioMixerBase::setParameter(int name, int target, int param, void *value)
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{
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LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
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const std::shared_ptr<TrackBase> &track = mTracks[name];
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int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
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int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
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switch (target) {
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case TRACK:
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switch (param) {
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case CHANNEL_MASK: {
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const audio_channel_mask_t trackChannelMask =
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static_cast<audio_channel_mask_t>(valueInt);
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if (setChannelMasks(name, trackChannelMask, track->mMixerChannelMask)) {
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ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
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invalidate();
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}
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} break;
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case MAIN_BUFFER:
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if (track->mainBuffer != valueBuf) {
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track->mainBuffer = valueBuf;
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ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
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invalidate();
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}
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break;
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case AUX_BUFFER:
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if (track->auxBuffer != valueBuf) {
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track->auxBuffer = valueBuf;
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ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
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invalidate();
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}
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break;
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case FORMAT: {
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audio_format_t format = static_cast<audio_format_t>(valueInt);
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if (track->mFormat != format) {
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ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
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track->mFormat = format;
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ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
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invalidate();
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}
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} break;
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case MIXER_FORMAT: {
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audio_format_t format = static_cast<audio_format_t>(valueInt);
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if (track->mMixerFormat != format) {
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track->mMixerFormat = format;
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ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
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}
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} break;
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case MIXER_CHANNEL_MASK: {
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const audio_channel_mask_t mixerChannelMask =
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static_cast<audio_channel_mask_t>(valueInt);
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if (setChannelMasks(name, track->channelMask, mixerChannelMask)) {
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ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
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invalidate();
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}
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} break;
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default:
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LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
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}
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break;
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case RESAMPLE:
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switch (param) {
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case SAMPLE_RATE:
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ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
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if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
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ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
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uint32_t(valueInt));
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invalidate();
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}
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break;
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case RESET:
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track->resetResampler();
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invalidate();
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break;
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case REMOVE:
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track->mResampler.reset(nullptr);
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track->sampleRate = mSampleRate;
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invalidate();
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break;
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default:
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LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
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}
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break;
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case RAMP_VOLUME:
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case VOLUME:
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switch (param) {
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case AUXLEVEL:
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if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
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target == RAMP_VOLUME ? mFrameCount : 0,
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&track->auxLevel, &track->prevAuxLevel, &track->auxInc,
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&track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
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ALOGV("setParameter(%s, AUXLEVEL: %04x)",
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target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
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invalidate();
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}
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break;
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default:
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if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
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if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
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target == RAMP_VOLUME ? mFrameCount : 0,
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&track->volume[param - VOLUME0],
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&track->prevVolume[param - VOLUME0],
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&track->volumeInc[param - VOLUME0],
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&track->mVolume[param - VOLUME0],
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&track->mPrevVolume[param - VOLUME0],
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&track->mVolumeInc[param - VOLUME0])) {
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ALOGV("setParameter(%s, VOLUME%d: %04x)",
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target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
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track->volume[param - VOLUME0]);
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invalidate();
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}
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} else {
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LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
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}
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}
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break;
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default:
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LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
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}
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}
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|
|
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;
|
|
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1
|
|
&& t->channelMask == AUDIO_CHANNEL_OUT_MONO // MONO_HACK
|
|
&& isAudioChannelPositionMask(t->mMixerChannelMask)) {
|
|
t->hook = TrackBase::getTrackHook(
|
|
TRACKTYPE_RESAMPLEMONO, t->mMixerChannelCount,
|
|
t->mMixerInFormat, t->mMixerFormat);
|
|
} else if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2
|
|
&& t->useStereoVolume()) {
|
|
t->hook = TrackBase::getTrackHook(
|
|
TRACKTYPE_RESAMPLESTEREO, t->mMixerChannelCount,
|
|
t->mMixerInFormat, t->mMixerFormat);
|
|
} else {
|
|
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(
|
|
(isAudioChannelPositionMask(t->mMixerChannelMask) // 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(
|
|
t->useStereoVolume() ? TRACKTYPE_NORESAMPLESTEREO
|
|
: 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,
|
|
t->useStereoVolume());
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
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,
|
|
t->useStereoVolume());
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
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.
|
|
|
|
constexpr int MIXTYPE_MONOVOL(int mixtype, int channels) {
|
|
if (channels <= FCC_2) {
|
|
return mixtype;
|
|
} else if (mixtype == MIXTYPE_MULTI) {
|
|
return MIXTYPE_MULTI_MONOVOL;
|
|
} else if (mixtype == MIXTYPE_MULTI_SAVEONLY) {
|
|
return MIXTYPE_MULTI_SAVEONLY_MONOVOL;
|
|
} else {
|
|
return mixtype;
|
|
}
|
|
}
|
|
|
|
// Helper to make a functional array from volumeRampMulti.
|
|
template <int MIXTYPE, typename TO, typename TI, typename TV, typename TA, typename TAV,
|
|
std::size_t ... Is>
|
|
static constexpr auto makeVRMArray(std::index_sequence<Is...>)
|
|
{
|
|
using F = void(*)(TO*, size_t, const TI*, TA*, TV*, const TV*, TAV*, TAV);
|
|
return std::array<F, sizeof...(Is)>{
|
|
{ &volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE, Is + 1), Is + 1, TO, TI, TV, TA, TAV> ...}
|
|
};
|
|
}
|
|
|
|
/* 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)
|
|
{
|
|
static constexpr auto volumeRampMultiArray =
|
|
makeVRMArray<MIXTYPE, TO, TI, TV, TA, TAV>(std::make_index_sequence<FCC_LIMIT>());
|
|
if (channels > 0 && channels <= volumeRampMultiArray.size()) {
|
|
volumeRampMultiArray[channels - 1](out, frameCount, in, aux, vol, volinc, vola, volainc);
|
|
} else {
|
|
ALOGE("%s: invalid channel count:%d", __func__, channels);
|
|
}
|
|
}
|
|
|
|
// Helper to make a functional array from volumeMulti.
|
|
template <int MIXTYPE, typename TO, typename TI, typename TV, typename TA, typename TAV,
|
|
std::size_t ... Is>
|
|
static constexpr auto makeVMArray(std::index_sequence<Is...>)
|
|
{
|
|
using F = void(*)(TO*, size_t, const TI*, TA*, const TV*, TAV);
|
|
return std::array<F, sizeof...(Is)>{
|
|
{ &volumeMulti<MIXTYPE_MONOVOL(MIXTYPE, Is + 1), Is + 1, TO, TI, TV, TA, TAV> ... }
|
|
};
|
|
}
|
|
|
|
/* 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)
|
|
{
|
|
static constexpr auto volumeMultiArray =
|
|
makeVMArray<MIXTYPE, TO, TI, TV, TA, TAV>(std::make_index_sequence<FCC_LIMIT>());
|
|
if (channels > 0 && channels <= volumeMultiArray.size()) {
|
|
volumeMultiArray[channels - 1](out, frameCount, in, aux, vol, vola);
|
|
} else {
|
|
ALOGE("%s: invalid channel count:%d", __func__, channels);
|
|
}
|
|
}
|
|
|
|
/* 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, std::is_same_v<TI, float> /* 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, std::is_same_v<TI, float>);
|
|
}
|
|
}
|
|
|
|
/* 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 (MIXTYPE == MIXTYPE_MONOEXPAND || MIXTYPE == MIXTYPE_STEREOEXPAND // custom volume handling
|
|
|| 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, std::is_same_v<TI, float> /* 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, std::is_same_v<TI, float> /* 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_RESAMPLESTEREO:
|
|
switch (mixerInFormat) {
|
|
case AUDIO_FORMAT_PCM_FLOAT:
|
|
return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
|
|
MIXTYPE_MULTI_STEREOVOL, float /*TO*/, float /*TI*/,
|
|
TYPE_AUX>;
|
|
case AUDIO_FORMAT_PCM_16_BIT:
|
|
return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
|
|
MIXTYPE_MULTI_STEREOVOL, int32_t /*TO*/, int16_t /*TI*/,
|
|
TYPE_AUX>;
|
|
default:
|
|
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
|
|
break;
|
|
}
|
|
break;
|
|
// RESAMPLEMONO needs MIXTYPE_STEREOEXPAND since resampler will upmix mono
|
|
// track to stereo track
|
|
case TRACKTYPE_RESAMPLEMONO:
|
|
switch (mixerInFormat) {
|
|
case AUDIO_FORMAT_PCM_FLOAT:
|
|
return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
|
|
MIXTYPE_STEREOEXPAND, float /*TO*/, float /*TI*/,
|
|
TYPE_AUX>;
|
|
case AUDIO_FORMAT_PCM_16_BIT:
|
|
return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
|
|
MIXTYPE_STEREOEXPAND, 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;
|
|
case TRACKTYPE_NORESAMPLESTEREO:
|
|
switch (mixerInFormat) {
|
|
case AUDIO_FORMAT_PCM_FLOAT:
|
|
return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
|
|
MIXTYPE_MULTI_STEREOVOL, float /*TO*/, float /*TI*/,
|
|
TYPE_AUX>;
|
|
case AUDIO_FORMAT_PCM_16_BIT:
|
|
return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
|
|
MIXTYPE_MULTI_STEREOVOL, 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,
|
|
bool stereoVolume)
|
|
{
|
|
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);
|
|
|
|
if (stereoVolume) { // templated arguments require explicit values.
|
|
switch (mixerInFormat) {
|
|
case AUDIO_FORMAT_PCM_FLOAT:
|
|
switch (mixerOutFormat) {
|
|
case AUDIO_FORMAT_PCM_FLOAT:
|
|
return &AudioMixerBase::process__noResampleOneTrack<
|
|
MIXTYPE_MULTI_SAVEONLY_STEREOVOL, float /*TO*/,
|
|
float /*TI*/, TYPE_AUX>;
|
|
case AUDIO_FORMAT_PCM_16_BIT:
|
|
return &AudioMixerBase::process__noResampleOneTrack<
|
|
MIXTYPE_MULTI_SAVEONLY_STEREOVOL, 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_STEREOVOL, float /*TO*/,
|
|
int16_t /*TI*/, TYPE_AUX>;
|
|
case AUDIO_FORMAT_PCM_16_BIT:
|
|
return &AudioMixerBase::process__noResampleOneTrack<
|
|
MIXTYPE_MULTI_SAVEONLY_STEREOVOL, 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;
|
|
}
|
|
} else {
|
|
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
|