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axmol/cocos/audio/android/AudioMixer.cpp

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[big refactoring] Audio latency fix for Android. Support to preload effects on Android now. (#15875) * Audio latency fix for Android. Support preload effects on Android now. Squashed commits: [b6d80fe] log fix [a0a918e] Fixes assetFd didn't be released while PcmData is returned from cache. [4b956ba] Potential crash fix for PcmAudioPlayer while pause / resume. [398ab8c] Updates LOG_TAG position in AudioEngine-inl.cpp [e3634e7] include stdlib.h for posix_memalign [9004074] fixes setVolume logical error. [c96df46] Don't use another thread for mixing, enqueue is in a seperated thread, therefore doing mixing in another thread will waste more time. [0a4c1a8] Adds setLoop, setVolume, setPostion support for Track [c35fb20] Fixed include. [cdd9d32] Do mixing by ourself. (TO BE POLISHED) [6447025] µ -> u since µ could not be shown on some android devices. [97be0c6] Don't send a silence clip. [c1607ed] Make linter.py happy. [0898b54] Puts enqueue & SetPlayState in PcmAudioPlayer::play to thread pool. [b79fc01] Adds getDuration, getPosition support for PcmAudioPlayer [80fa2ab] minor fix of the code position of resetting state to State::INITIALIZED [d9c62f1] underrun fix for PcmAudioPlayer. [9c2212a] UrlAudioPlayer, playOverMutex should be static, and should be used in update method. [1519d2e] static variables [19da936] _pcmAudioPlayer Null pointer check in AudioPlayerProvider. [e6b0d14] Updates audio performance test. [fc01dd4] Registers foreground & background event in AudioEngine-inl.cpp(android), the callback should invoke `provider`'s pause & resume method. [e00a886] TBD: Pause & resume support for PcmAudioPlayerPool. Since OpenSLES audio resources are expensive and device shared, we should delete all unused PcmAudioPlayers in pool while pause and re-create them while resume. But this commit isn't finished yet, I don't find a better way to register pause&resume event in AudioEngine module. [9e42ea3] Interleave mono audio to stereo audio. PcmAudioPlayerPool only contains PcmAudioPlayers with 2 channels. [3f18d05] Adds a strategy for checking small size of different file formats. [753ff49] Adds performance test for AudioEngine. [09d3045] Releases an extra PcmAudioPlayer for UrlAudioPlayer while allocating PcmAudioPlayer fails. [9dd4477] Using std::move for PcmData move constructor & move assignment. [6ca3bcb] some fixes: 1) new -> new (std::nothrow) 2) break if allocate PcmAudioPlayer fails 3) renames 'initForPlayPcmData' to 'init' 4) PcmAudioPlayer destructor deadlock if 'init' failed [54675b6] include path fix. [a1903ca] More refactorings. [19b9498] Makes linter.py happy. :) [923c530] Fixes: 1) Avoid getFileInfo to be invoked twice 2) A critical bug fix for UrlAudioPlayer and adds detailed comments 3) __clang__ compiler option fix for AudioResamplerSinc.cpp. [5ec4faf] minor fix. [faaa0f3] output a log in the destructor of UrlAudioPlayer. [9c20355] NewAudioEngineTest,TestControll crash fix. [f114464] fixes an unused import. [1dc5dab] Better algorithm for allocating PcmAudioPlayer. [331a213] minor fix. [e54084a] null -> nullptr [f9a0389] Support uncache. [89a364f] Removes unused update, and TODO uncache functionality. [1732bf9] Supports AudioEngineImpl::setFinishCallback for android. [43d1596] UrlAudioPlayer::stop fix. [e2ee941] Test case fix in NewAudioEngineTest/AudioIssue11143Test [5c5ba01] More fixes for making cpp-tests/New Audio Engine Test happy. [8b554a3] Adds log while remove player from map. [ed71322] If original file is larger than 30k bytes, consider it's a large audio file. [fb1845a] Updates project.properties [6f3839f] minor log output fix in AudioEngine-inl.cpp [c68bc6c] Don't resample if the sample rate of the decoded pcm data matchs the device's. [43ca45f] PcmAudioPlayers also need to be removed while they play over, but should not be deleted since their lifecycle is managed by PcmAudioPlayerPool. [f5e63c9] Audio latency fix for Android. Support preload effects on Android now. * Supports to loading audio files asynchronously. * Crash fix for stop audio right after play2d. * Minor fix for logic in AudioMixerController.cpp * Adds missing files (CCThreadPool.h/.cpp). * Minor fix for including. * Minor fix for missing include <functional> in Track.h * update license information in audio.h * Don't use std::future/std::promise anymore since ndk counldn't support it well in armeabi arch. * isSmallFile postion updated, fixes large audio file goto the checking logic of cache. * std::atomic<int> isn't supported by ndk-r10e while compiling with `armeabi` arch, using a int with a mutex instead. * fixes __isnanf & posix_memalign doesn't exist on low api (<=16) devices. * namespace updated: cocos2d -> cocos2d::experimental * Removes commented code in AudioMixerController.h/.cpp * Removes unused code again, and fixes a memory leak of `Track` instance. * Oops, namespace changed. * Only outputs log in debug mode. * Uses ALOGV for outputing logs in AudioEngine-inl.cpp * const PcmData& -> PcmData for Track * Fixes a protential crash in NewAudioEngineTest * Adds `COCOS` prefix in header #ifndef COCOS_BALABALA #define COCOS_BALABALA * Uses _ prefix for cocos code style instead of `m` prefix. * Deletes AudioResamplerSinc related files. * Bug fix from @minggo's reply on github. * Don't need to invoke pause after in UrlAudioPlayer::prepare. * Updates ThreadPool class, uses enum class and adds const keyword.
2016-07-18 10:22:40 +08:00
/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioMixer"
#define LOG_NDEBUG 1
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
#include <math.h>
#include <sys/types.h>
#include "audio/android/audio.h"
#include "audio/android/audio_utils/include/audio_utils/primitives.h"
#include "audio/android/AudioMixerOps.h"
#include "audio/android/AudioMixer.h"
// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
#ifndef FCC_2
#define FCC_2 2
#endif
// Look for MONO_HACK for any Mono hack involving legacy mono channel to
// stereo channel conversion.
/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
* being used. This is a considerable amount of log spam, so don't enable unless you
* are verifying the hook based code.
*/
//#define VERY_VERY_VERBOSE_LOGGING
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
//define ALOGVV printf // for test-mixer.cpp
#else
#define ALOGVV(a...) do { } while (0)
#endif
#ifndef ARRAY_SIZE
#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
#endif
// TODO: Move these macro/inlines to a header file.
template <typename T>
static inline
T max(const T& x, const T& y) {
return x > y ? x : y;
}
// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
// original code will be used for stereo sinks, the new mixer for multichannel.
static const bool kUseNewMixer = false;
// Set kUseFloat to true to allow floating input into the mixer engine.
// If kUseNewMixer is false, this is ignored or may be overridden internally
// because of downmix/upmix support.
static const bool kUseFloat = false;
// Set to default copy buffer size in frames for input processing.
static const size_t kCopyBufferFrameCount = 256;
Remove experimental namespace (#20229)
2019-10-23 14:58:31 +08:00
namespace cocos2d {
[big refactoring] Audio latency fix for Android. Support to preload effects on Android now. (#15875) * Audio latency fix for Android. Support preload effects on Android now. Squashed commits: [b6d80fe] log fix [a0a918e] Fixes assetFd didn't be released while PcmData is returned from cache. [4b956ba] Potential crash fix for PcmAudioPlayer while pause / resume. [398ab8c] Updates LOG_TAG position in AudioEngine-inl.cpp [e3634e7] include stdlib.h for posix_memalign [9004074] fixes setVolume logical error. [c96df46] Don't use another thread for mixing, enqueue is in a seperated thread, therefore doing mixing in another thread will waste more time. [0a4c1a8] Adds setLoop, setVolume, setPostion support for Track [c35fb20] Fixed include. [cdd9d32] Do mixing by ourself. (TO BE POLISHED) [6447025] µ -> u since µ could not be shown on some android devices. [97be0c6] Don't send a silence clip. [c1607ed] Make linter.py happy. [0898b54] Puts enqueue & SetPlayState in PcmAudioPlayer::play to thread pool. [b79fc01] Adds getDuration, getPosition support for PcmAudioPlayer [80fa2ab] minor fix of the code position of resetting state to State::INITIALIZED [d9c62f1] underrun fix for PcmAudioPlayer. [9c2212a] UrlAudioPlayer, playOverMutex should be static, and should be used in update method. [1519d2e] static variables [19da936] _pcmAudioPlayer Null pointer check in AudioPlayerProvider. [e6b0d14] Updates audio performance test. [fc01dd4] Registers foreground & background event in AudioEngine-inl.cpp(android), the callback should invoke `provider`'s pause & resume method. [e00a886] TBD: Pause & resume support for PcmAudioPlayerPool. Since OpenSLES audio resources are expensive and device shared, we should delete all unused PcmAudioPlayers in pool while pause and re-create them while resume. But this commit isn't finished yet, I don't find a better way to register pause&resume event in AudioEngine module. [9e42ea3] Interleave mono audio to stereo audio. PcmAudioPlayerPool only contains PcmAudioPlayers with 2 channels. [3f18d05] Adds a strategy for checking small size of different file formats. [753ff49] Adds performance test for AudioEngine. [09d3045] Releases an extra PcmAudioPlayer for UrlAudioPlayer while allocating PcmAudioPlayer fails. [9dd4477] Using std::move for PcmData move constructor & move assignment. [6ca3bcb] some fixes: 1) new -> new (std::nothrow) 2) break if allocate PcmAudioPlayer fails 3) renames 'initForPlayPcmData' to 'init' 4) PcmAudioPlayer destructor deadlock if 'init' failed [54675b6] include path fix. [a1903ca] More refactorings. [19b9498] Makes linter.py happy. :) [923c530] Fixes: 1) Avoid getFileInfo to be invoked twice 2) A critical bug fix for UrlAudioPlayer and adds detailed comments 3) __clang__ compiler option fix for AudioResamplerSinc.cpp. [5ec4faf] minor fix. [faaa0f3] output a log in the destructor of UrlAudioPlayer. [9c20355] NewAudioEngineTest,TestControll crash fix. [f114464] fixes an unused import. [1dc5dab] Better algorithm for allocating PcmAudioPlayer. [331a213] minor fix. [e54084a] null -> nullptr [f9a0389] Support uncache. [89a364f] Removes unused update, and TODO uncache functionality. [1732bf9] Supports AudioEngineImpl::setFinishCallback for android. [43d1596] UrlAudioPlayer::stop fix. [e2ee941] Test case fix in NewAudioEngineTest/AudioIssue11143Test [5c5ba01] More fixes for making cpp-tests/New Audio Engine Test happy. [8b554a3] Adds log while remove player from map. [ed71322] If original file is larger than 30k bytes, consider it's a large audio file. [fb1845a] Updates project.properties [6f3839f] minor log output fix in AudioEngine-inl.cpp [c68bc6c] Don't resample if the sample rate of the decoded pcm data matchs the device's. [43ca45f] PcmAudioPlayers also need to be removed while they play over, but should not be deleted since their lifecycle is managed by PcmAudioPlayerPool. [f5e63c9] Audio latency fix for Android. Support preload effects on Android now. * Supports to loading audio files asynchronously. * Crash fix for stop audio right after play2d. * Minor fix for logic in AudioMixerController.cpp * Adds missing files (CCThreadPool.h/.cpp). * Minor fix for including. * Minor fix for missing include <functional> in Track.h * update license information in audio.h * Don't use std::future/std::promise anymore since ndk counldn't support it well in armeabi arch. * isSmallFile postion updated, fixes large audio file goto the checking logic of cache. * std::atomic<int> isn't supported by ndk-r10e while compiling with `armeabi` arch, using a int with a mutex instead. * fixes __isnanf & posix_memalign doesn't exist on low api (<=16) devices. * namespace updated: cocos2d -> cocos2d::experimental * Removes commented code in AudioMixerController.h/.cpp * Removes unused code again, and fixes a memory leak of `Track` instance. * Oops, namespace changed. * Only outputs log in debug mode. * Uses ALOGV for outputing logs in AudioEngine-inl.cpp * const PcmData& -> PcmData for Track * Fixes a protential crash in NewAudioEngineTest * Adds `COCOS` prefix in header #ifndef COCOS_BALABALA #define COCOS_BALABALA * Uses _ prefix for cocos code style instead of `m` prefix. * Deletes AudioResamplerSinc related files. * Bug fix from @minggo's reply on github. * Don't need to invoke pause after in UrlAudioPlayer::prepare. * Updates ThreadPool class, uses enum class and adds const keyword.
2016-07-18 10:22:40 +08:00
// ----------------------------------------------------------------------------
template <typename T>
T min(const T& a, const T& b)
{
return a < b ? a : b;
}
// ----------------------------------------------------------------------------
// Ensure mConfiguredNames bitmask is initialized properly on all architectures.
// The value of 1 << x is undefined in C when x >= 32.
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks)
: mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1),
mSampleRate(sampleRate)
{
ALOGVV("AudioMixer constructed, frameCount: %d, sampleRate: %d", (int)frameCount, (int)sampleRate);
ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u",
maxNumTracks, MAX_NUM_TRACKS);
// AudioMixer is not yet capable of more than 32 active track inputs
ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS);
pthread_once(&sOnceControl, &sInitRoutine);
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
mState.hook = process__nop;
mState.outputTemp = NULL;
mState.resampleTemp = NULL;
//cjh mState.mLog = &mDummyLog;
// mState.reserved
// FIXME Most of the following initialization is probably redundant since
// tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0
// and mTrackNames is initially 0. However, leave it here until that's verified.
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
t->resampler = NULL;
//cjh t->downmixerBufferProvider = NULL;
// t->mReformatBufferProvider = NULL;
// t->mTimestretchBufferProvider = NULL;
t++;
}
}
AudioMixer::~AudioMixer()
{
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
delete t->resampler;
//cjh delete t->downmixerBufferProvider;
// delete t->mReformatBufferProvider;
// delete t->mTimestretchBufferProvider;
t++;
}
delete [] mState.outputTemp;
delete [] mState.resampleTemp;
}
//cjh void AudioMixer::setLog(NBLog::Writer *log)
//{
// mState.mLog = log;
//}
static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
}
int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
audio_format_t format, int sessionId)
{
if (!isValidPcmTrackFormat(format)) {
ALOGE("AudioMixer::getTrackName invalid format (%#x)", format);
return -1;
}
uint32_t names = (~mTrackNames) & mConfiguredNames;
if (names != 0) {
int n = __builtin_ctz(names);
ALOGV("add track (%d)", n);
// assume default parameters for the track, except where noted below
track_t* t = &mState.tracks[n];
t->needs = 0;
// Integer volume.
// Currently integer volume is kept for the legacy integer mixer.
// Will be removed when the legacy mixer path is removed.
t->volume[0] = UNITY_GAIN_INT;
t->volume[1] = UNITY_GAIN_INT;
t->prevVolume[0] = UNITY_GAIN_INT << 16;
t->prevVolume[1] = UNITY_GAIN_INT << 16;
t->volumeInc[0] = 0;
t->volumeInc[1] = 0;
t->auxLevel = 0;
t->auxInc = 0;
t->prevAuxLevel = 0;
// Floating point volume.
t->mVolume[0] = UNITY_GAIN_FLOAT;
t->mVolume[1] = UNITY_GAIN_FLOAT;
t->mPrevVolume[0] = UNITY_GAIN_FLOAT;
t->mPrevVolume[1] = UNITY_GAIN_FLOAT;
t->mVolumeInc[0] = 0.;
t->mVolumeInc[1] = 0.;
t->mAuxLevel = 0.;
t->mAuxInc = 0.;
t->mPrevAuxLevel = 0.;
// no initialization needed
// t->frameCount
t->channelCount = audio_channel_count_from_out_mask(channelMask);
t->enabled = false;
ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
"Non-stereo channel mask: %d\n", channelMask);
t->channelMask = channelMask;
t->sessionId = sessionId;
// setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
t->bufferProvider = NULL;
t->buffer.raw = NULL;
// no initialization needed
// t->buffer.frameCount
t->hook = NULL;
t->in = NULL;
t->resampler = NULL;
t->sampleRate = mSampleRate;
// setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t->mInputBufferProvider = NULL;
//cjh t->mReformatBufferProvider = NULL;
// t->downmixerBufferProvider = NULL;
// t->mPostDownmixReformatBufferProvider = NULL;
// t->mTimestretchBufferProvider = NULL;
t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
t->mFormat = format;
t->mMixerInFormat = selectMixerInFormat(format);
t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
ALOGVV("t->mMixerChannelCount: %d", t->mMixerChannelCount);
t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// Check the downmixing (or upmixing) requirements.
status_t status = t->prepareForDownmix();
if (status != OK) {
ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
return -1;
}
// prepareForDownmix() may change mDownmixRequiresFormat
ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
t->prepareForReformat();
mTrackNames |= 1 << n;
ALOGVV("getTrackName return: %d", TRACK0 + n);
return TRACK0 + n;
}
ALOGE("AudioMixer::getTrackName out of available tracks");
return -1;
}
void AudioMixer::invalidateState(uint32_t mask)
{
if (mask != 0) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
}
// Called when channel masks have changed for a track name
// TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format,
// which will simplify this logic.
bool AudioMixer::setChannelMasks(int name,
audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
track_t &track = mState.tracks[name];
ALOGVV("AudioMixer::setChannelMask ...");
if (trackChannelMask == track.channelMask
&& mixerChannelMask == track.mMixerChannelMask) {
ALOGVV("No need to change channel mask ...");
return false; // no need to change
}
// always recompute for both channel masks even if only one has changed.
const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
const bool mixerChannelCountChanged = track.mMixerChannelCount != mixerChannelCount;
ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX)
&& trackChannelCount
&& mixerChannelCount);
track.channelMask = trackChannelMask;
track.channelCount = trackChannelCount;
track.mMixerChannelMask = mixerChannelMask;
track.mMixerChannelCount = mixerChannelCount;
// channel masks have changed, does this track need a downmixer?
// update to try using our desired format (if we aren't already using it)
const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat;
const status_t status = mState.tracks[name].prepareForDownmix();
ALOGE_IF(status != OK,
"prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
status, track.channelMask, track.mMixerChannelMask);
if (prevDownmixerFormat != track.mDownmixRequiresFormat) {
track.prepareForReformat(); // because of downmixer, track format may change!
}
if (track.resampler && mixerChannelCountChanged) {
// resampler channels may have changed.
const uint32_t resetToSampleRate = track.sampleRate;
delete track.resampler;
track.resampler = NULL;
track.sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
// recreate the resampler with updated format, channels, saved sampleRate.
track.setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
}
return true;
}
void AudioMixer::track_t::unprepareForDownmix() {
ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
//cjh if (downmixerBufferProvider != NULL) {
// // this track had previously been configured with a downmixer, delete it
// ALOGV(" deleting old downmixer");
// delete downmixerBufferProvider;
// downmixerBufferProvider = NULL;
// reconfigureBufferProviders();
// } else
{
ALOGV(" nothing to do, no downmixer to delete");
}
}
status_t AudioMixer::track_t::prepareForDownmix()
{
ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x",
this, channelMask);
// discard the previous downmixer if there was one
unprepareForDownmix();
// MONO_HACK Only remix (upmix or downmix) if the track and mixer/device channel masks
// are not the same and not handled internally, as mono -> stereo currently is.
if (channelMask == mMixerChannelMask
|| (channelMask == AUDIO_CHANNEL_OUT_MONO
&& mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
return NO_ERROR;
}
// DownmixerBufferProvider is only used for position masks.
//cjh if (audio_channel_mask_get_representation(channelMask)
// == AUDIO_CHANNEL_REPRESENTATION_POSITION
// && DownmixerBufferProvider::isMultichannelCapable()) {
// DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask,
// mMixerChannelMask,
// AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */,
// sampleRate, sessionId, kCopyBufferFrameCount);
//
// if (pDbp->isValid()) { // if constructor completed properly
// mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
// downmixerBufferProvider = pDbp;
// reconfigureBufferProviders();
// return NO_ERROR;
// }
// delete pDbp;
// }
//
// // Effect downmixer does not accept the channel conversion. Let's use our remixer.
// RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask,
// mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount);
// // Remix always finds a conversion whereas Downmixer effect above may fail.
// downmixerBufferProvider = pRbp;
// reconfigureBufferProviders();
return NO_ERROR;
}
void AudioMixer::track_t::unprepareForReformat() {
ALOGV("AudioMixer::unprepareForReformat(%p)", this);
bool requiresReconfigure = false;
//cjh if (mReformatBufferProvider != NULL) {
// delete mReformatBufferProvider;
// mReformatBufferProvider = NULL;
// requiresReconfigure = true;
// }
// if (mPostDownmixReformatBufferProvider != NULL) {
// delete mPostDownmixReformatBufferProvider;
// mPostDownmixReformatBufferProvider = NULL;
// requiresReconfigure = true;
// }
if (requiresReconfigure) {
reconfigureBufferProviders();
}
}
status_t AudioMixer::track_t::prepareForReformat()
{
ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
// discard previous reformatters
unprepareForReformat();
// only configure reformatters as needed
const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID
? mDownmixRequiresFormat : mMixerInFormat;
bool requiresReconfigure = false;
//cjh if (mFormat != targetFormat) {
// mReformatBufferProvider = new ReformatBufferProvider(
// audio_channel_count_from_out_mask(channelMask),
// mFormat,
// targetFormat,
// kCopyBufferFrameCount);
// requiresReconfigure = true;
// }
// if (targetFormat != mMixerInFormat) {
// mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
// audio_channel_count_from_out_mask(mMixerChannelMask),
// targetFormat,
// mMixerInFormat,
// kCopyBufferFrameCount);
// requiresReconfigure = true;
// }
if (requiresReconfigure) {
reconfigureBufferProviders();
}
ALOGVV("prepareForReformat return ...");
return NO_ERROR;
}
void AudioMixer::track_t::reconfigureBufferProviders()
{
bufferProvider = mInputBufferProvider;
//cjh if (mReformatBufferProvider) {
// mReformatBufferProvider->setBufferProvider(bufferProvider);
// bufferProvider = mReformatBufferProvider;
// }
// if (downmixerBufferProvider) {
// downmixerBufferProvider->setBufferProvider(bufferProvider);
// bufferProvider = downmixerBufferProvider;
// }
// if (mPostDownmixReformatBufferProvider) {
// mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
// bufferProvider = mPostDownmixReformatBufferProvider;
// }
// if (mTimestretchBufferProvider) {
// mTimestretchBufferProvider->setBufferProvider(bufferProvider);
// bufferProvider = mTimestretchBufferProvider;
// }
}
void AudioMixer::deleteTrackName(int name)
{
ALOGV("AudioMixer::deleteTrackName(%d)", name);
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
ALOGV("deleteTrackName(%d)", name);
track_t& track(mState.tracks[ name ]);
if (track.enabled) {
track.enabled = false;
invalidateState(1<<name);
}
// delete the resampler
delete track.resampler;
track.resampler = NULL;
// delete the downmixer
mState.tracks[name].unprepareForDownmix();
// delete the reformatter
mState.tracks[name].unprepareForReformat();
// delete the timestretch provider
//cjh delete track.mTimestretchBufferProvider;
// track.mTimestretchBufferProvider = NULL;
mTrackNames &= ~(1<<name);
}
void AudioMixer::enable(int name)
{
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
track_t& track = mState.tracks[name];
if (!track.enabled) {
track.enabled = true;
ALOGV("enable(%d)", name);
invalidateState(1 << name);
}
}
void AudioMixer::disable(int name)
{
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
track_t& track = mState.tracks[name];
if (track.enabled) {
track.enabled = false;
ALOGV("disable(%d)", name);
invalidateState(1 << name);
}
}
/* Sets the volume ramp variables for the AudioMixer.
*
* The volume ramp variables are used to transition from the previous
* volume to the set volume. ramp controls the duration of the transition.
* Its value is typically one state framecount period, but may also be 0,
* meaning "immediate."
*
* FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
* even if there is a nonzero floating point increment (in that case, the volume
* change is immediate). This restriction should be changed when the legacy mixer
* is removed (see #2).
* FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
* when no longer needed.
*
* @param newVolume set volume target in floating point [0.0, 1.0].
* @param ramp number of frames to increment over. if ramp is 0, the volume
* should be set immediately. Currently ramp should not exceed 65535 (frames).
* @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
* @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
* @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
* @param pSetVolume pointer to the float target volume, set on return.
* @param pPrevVolume pointer to the float previous volume, set on return.
* @param pVolumeInc pointer to the float increment per output audio frame, set on return.
* @return true if the volume has changed, false if volume is same.
*/
static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
// check floating point volume to see if it is identical to the previously
// set volume.
// We do not use a tolerance here (and reject changes too small)
// as it may be confusing to use a different value than the one set.
// If the resulting volume is too small to ramp, it is a direct set of the volume.
if (newVolume == *pSetVolume) {
return false;
}
if (newVolume < 0) {
newVolume = 0; // should not have negative volumes
} else {
switch (fpclassify(newVolume)) {
case FP_SUBNORMAL:
case FP_NAN:
newVolume = 0;
break;
case FP_ZERO:
break; // zero volume is fine
case FP_INFINITE:
// Infinite volume could be handled consistently since
// floating point math saturates at infinities,
// but we limit volume to unity gain float.
// ramp = 0; break;
//
newVolume = AudioMixer::UNITY_GAIN_FLOAT;
break;
case FP_NORMAL:
default:
// Floating point does not have problems with overflow wrap
// that integer has. However, we limit the volume to
// unity gain here.
// TODO: Revisit the volume limitation and perhaps parameterize.
if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) {
newVolume = AudioMixer::UNITY_GAIN_FLOAT;
}
break;
}
}
// set floating point volume ramp
if (ramp != 0) {
// when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
// is no computational mismatch; hence equality is checked here.
ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
" prev:%f set_to:%f", *pPrevVolume, *pSetVolume);
const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
const float maxv = max(newVolume, *pPrevVolume); // could be inf, cannot be nan, subnormal
if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
&& maxv + inc != maxv) { // inc must make forward progress
*pVolumeInc = inc;
// ramp is set now.
// Note: if newVolume is 0, then near the end of the ramp,
// it may be possible that the ramped volume may be subnormal or
// temporarily negative by a small amount or subnormal due to floating
// point inaccuracies.
} else {
ramp = 0; // ramp not allowed
}
}
// compute and check integer volume, no need to check negative values
// The integer volume is limited to "unity_gain" to avoid wrapping and other
// audio artifacts, so it never reaches the range limit of U4.28.
// We safely use signed 16 and 32 bit integers here.
const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan
const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ?
AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume;
// set integer volume ramp
if (ramp != 0) {
// integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
// when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
// is no computational mismatch; hence equality is checked here.
ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
" prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
if (inc != 0) { // inc must make forward progress
*pIntVolumeInc = inc;
} else {
ramp = 0; // ramp not allowed
}
}
// if no ramp, or ramp not allowed, then clear float and integer increments
if (ramp == 0) {
*pVolumeInc = 0;
*pPrevVolume = newVolume;
*pIntVolumeInc = 0;
*pIntPrevVolume = intVolume << 16;
}
*pSetVolume = newVolume;
*pIntSetVolume = intVolume;
return true;
}
void AudioMixer::setParameter(int name, int target, int param, void *value)
{
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
track_t& track = mState.tracks[name];
int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
switch (target) {
case TRACK:
switch (param) {
case CHANNEL_MASK: {
const audio_channel_mask_t trackChannelMask =
static_cast<audio_channel_mask_t>(valueInt);
if (setChannelMasks(name, trackChannelMask, track.mMixerChannelMask)) {
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
invalidateState(1 << name);
}
} break;
case MAIN_BUFFER:
if (track.mainBuffer != valueBuf) {
track.mainBuffer = valueBuf;
ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
invalidateState(1 << name);
}
break;
case AUX_BUFFER:
if (track.auxBuffer != valueBuf) {
track.auxBuffer = valueBuf;
ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
invalidateState(1 << name);
}
break;
case FORMAT: {
audio_format_t format = static_cast<audio_format_t>(valueInt);
if (track.mFormat != format) {
ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
track.mFormat = format;
ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
track.prepareForReformat();
invalidateState(1 << name);
}
} break;
// FIXME do we want to support setting the downmix type from AudioMixerController?
// for a specific track? or per mixer?
/* case DOWNMIX_TYPE:
break */
case MIXER_FORMAT: {
audio_format_t format = static_cast<audio_format_t>(valueInt);
if (track.mMixerFormat != format) {
track.mMixerFormat = format;
ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
}
} break;
case MIXER_CHANNEL_MASK: {
const audio_channel_mask_t mixerChannelMask =
static_cast<audio_channel_mask_t>(valueInt);
if (setChannelMasks(name, track.channelMask, mixerChannelMask)) {
ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
invalidateState(1 << name);
}
} break;
default:
LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
}
break;
case RESAMPLE:
switch (param) {
case SAMPLE_RATE:
ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
uint32_t(valueInt));
invalidateState(1 << name);
}
break;
case RESET:
track.resetResampler();
invalidateState(1 << name);
break;
case REMOVE:
delete track.resampler;
track.resampler = NULL;
track.sampleRate = mSampleRate;
invalidateState(1 << name);
break;
default:
LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
}
break;
case RAMP_VOLUME:
case VOLUME:
switch (param) {
case AUXLEVEL:
if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
target == RAMP_VOLUME ? mState.frameCount : 0,
&track.auxLevel, &track.prevAuxLevel, &track.auxInc,
&track.mAuxLevel, &track.mPrevAuxLevel, &track.mAuxInc)) {
ALOGV("setParameter(%s, AUXLEVEL: %04x)",
target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track.auxLevel);
invalidateState(1 << name);
}
break;
default:
if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
target == RAMP_VOLUME ? mState.frameCount : 0,
&track.volume[param - VOLUME0], &track.prevVolume[param - VOLUME0],
&track.volumeInc[param - VOLUME0],
&track.mVolume[param - VOLUME0], &track.mPrevVolume[param - VOLUME0],
&track.mVolumeInc[param - VOLUME0])) {
ALOGV("setParameter(%s, VOLUME%d: %04x)",
target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
track.volume[param - VOLUME0]);
invalidateState(1 << name);
}
} else {
LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
}
}
break;
case TIMESTRETCH:
switch (param) {
case PLAYBACK_RATE: {
const AudioPlaybackRate *playbackRate =
reinterpret_cast<AudioPlaybackRate*>(value);
ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
"bad parameters speed %f, pitch %f",playbackRate->mSpeed,
playbackRate->mPitch);
if (track.setPlaybackRate(*playbackRate)) {
ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
"%f %f %d %d",
playbackRate->mSpeed,
playbackRate->mPitch,
playbackRate->mStretchMode,
playbackRate->mFallbackMode);
// invalidateState(1 << name);
}
} break;
default:
LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
}
break;
default:
LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
}
}
bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
{
if (trackSampleRate != devSampleRate || resampler != NULL) {
if (sampleRate != trackSampleRate) {
sampleRate = trackSampleRate;
if (resampler == NULL) {
ALOGV("Creating resampler from track %d Hz to device %d Hz",
trackSampleRate, devSampleRate);
AudioResampler::src_quality quality;
// force lowest quality level resampler if use case isn't music or video
// FIXME this is flawed for dynamic sample rates, as we choose the resampler
// quality level based on the initial ratio, but that could change later.
// Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
//cjh if (isMusicRate(trackSampleRate)) {
quality = AudioResampler::DEFAULT_QUALITY;
//cjh } 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 = false/*downmixerBufferProvider != NULL*/
? mMixerChannelCount : channelCount;
ALOGVV("Creating resampler:"
" format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
resampler = AudioResampler::create(
mMixerInFormat,
resamplerChannelCount,
devSampleRate, quality);
resampler->setLocalTimeFreq(sLocalTimeFreq);
}
return true;
}
}
return false;
}
bool AudioMixer::track_t::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
//cjh if ((mTimestretchBufferProvider == NULL &&
// fabs(playbackRate.mSpeed - mPlaybackRate.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA &&
// fabs(playbackRate.mPitch - mPlaybackRate.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) ||
// isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
// return false;
// }
mPlaybackRate = playbackRate;
// if (mTimestretchBufferProvider == NULL) {
// // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
// // but if none exists, it is the channel count (1 for mono).
// const int timestretchChannelCount = downmixerBufferProvider != NULL
// ? mMixerChannelCount : channelCount;
// mTimestretchBufferProvider = new TimestretchBufferProvider(timestretchChannelCount,
// mMixerInFormat, sampleRate, playbackRate);
// reconfigureBufferProviders();
// } else {
// reinterpret_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider)
// ->setPlaybackRate(playbackRate);
// }
return true;
}
/* Checks to see if the volume ramp has completed and clears the increment
* variables appropriately.
*
* FIXME: There is code to handle int/float ramp variable switchover should it not
* complete within a mixer buffer processing call, but it is preferred to avoid switchover
* due to precision issues. The switchover code is included for legacy code purposes
* and can be removed once the integer volume is removed.
*
* It is not sufficient to clear only the volumeInc integer variable because
* if one channel requires ramping, all channels are ramped.
*
* There is a bit of duplicated code here, but it keeps backward compatibility.
*/
inline void AudioMixer::track_t::adjustVolumeRamp(bool aux, bool useFloat)
{
if (useFloat) {
for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
(mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
volumeInc[i] = 0;
prevVolume[i] = volume[i] << 16;
mVolumeInc[i] = 0.;
mPrevVolume[i] = mVolume[i];
} else {
//ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
}
}
} else {
for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
volumeInc[i] = 0;
prevVolume[i] = volume[i] << 16;
mVolumeInc[i] = 0.;
mPrevVolume[i] = mVolume[i];
} else {
//ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
mPrevVolume[i] = float_from_u4_28(prevVolume[i]);
}
}
}
/* TODO: aux is always integer regardless of output buffer type */
if (aux) {
if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
auxInc = 0;
prevAuxLevel = auxLevel << 16;
mAuxInc = 0.;
mPrevAuxLevel = mAuxLevel;
} else {
//ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc);
}
}
}
size_t AudioMixer::getUnreleasedFrames(int name) const
{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
return mState.tracks[name].getUnreleasedFrames();
}
return 0;
}
void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
{
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
if (mState.tracks[name].mInputBufferProvider == bufferProvider) {
return; // don't reset any buffer providers if identical.
}
//cjh if (mState.tracks[name].mReformatBufferProvider != NULL) {
// mState.tracks[name].mReformatBufferProvider->reset();
// } else if (mState.tracks[name].downmixerBufferProvider != NULL) {
// mState.tracks[name].downmixerBufferProvider->reset();
// } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
// mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
// } else if (mState.tracks[name].mTimestretchBufferProvider != NULL) {
// mState.tracks[name].mTimestretchBufferProvider->reset();
// }
mState.tracks[name].mInputBufferProvider = bufferProvider;
mState.tracks[name].reconfigureBufferProviders();
}
void AudioMixer::process(int64_t pts)
{
mState.hook(&mState, pts);
}
void AudioMixer::process__validate(state_t* state, int64_t pts)
{
ALOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
uint32_t changed = state->needsChanged;
state->needsChanged = 0; // clear the validation flag
// recompute which tracks are enabled / disabled
uint32_t enabled = 0;
uint32_t disabled = 0;
while (changed) {
const int i = 31 - __builtin_clz(changed);
const uint32_t mask = 1<<i;
changed &= ~mask;
track_t& t = state->tracks[i];
(t.enabled ? enabled : disabled) |= mask;
}
state->enabledTracks &= ~disabled;
state->enabledTracks |= enabled;
// compute everything we need...
int countActiveTracks = 0;
// TODO: fix all16BitsStereNoResample logic to
// either properly handle muted tracks (it should ignore them)
// or remove altogether as an obsolete optimization.
bool all16BitsStereoNoResample = true;
bool resampling = false;
bool volumeRamp = false;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
en &= ~(1<<i);
countActiveTracks++;
track_t& t = state->tracks[i];
uint32_t n = 0;
// FIXME can overflow (mask is only 3 bits)
n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
if (t.doesResample()) {
n |= NEEDS_RESAMPLE;
}
if (t.auxLevel != 0 && t.auxBuffer != NULL) {
n |= NEEDS_AUX;
}
if (t.volumeInc[0]|t.volumeInc[1]) {
volumeRamp = true;
} else if (!t.doesResample() && t.volumeRL == 0) {
n |= NEEDS_MUTE;
}
t.needs = n;
if (n & NEEDS_MUTE) {
t.hook = track__nop;
} else {
if (n & NEEDS_AUX) {
all16BitsStereoNoResample = false;
}
if (n & NEEDS_RESAMPLE) {
all16BitsStereoNoResample = false;
resampling = true;
t.hook = getTrackHook(TRACKTYPE_RESAMPLE, t.mMixerChannelCount,
t.mMixerInFormat, t.mMixerFormat);
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix + resample", i);
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
t.hook = getTrackHook(
(t.mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO // TODO: MONO_HACK
&& t.channelMask == AUDIO_CHANNEL_OUT_MONO)
? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
t.mMixerChannelCount,
t.mMixerInFormat, t.mMixerFormat);
all16BitsStereoNoResample = false;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
t.hook = getTrackHook(TRACKTYPE_NORESAMPLE, t.mMixerChannelCount,
t.mMixerInFormat, t.mMixerFormat);
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix", i);
}
}
}
}
// select the processing hooks
state->hook = process__nop;
if (countActiveTracks > 0) {
if (resampling) {
if (!state->outputTemp) {
state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
}
if (!state->resampleTemp) {
state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
}
state->hook = process__genericResampling;
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
state->outputTemp = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
if (countActiveTracks == 1) {
const int i = 31 - __builtin_clz(state->enabledTracks);
track_t& t = state->tracks[i];
if ((t.needs & NEEDS_MUTE) == 0) {
// The check prevents a muted track from acquiring a process hook.
//
// This is dangerous if the track is MONO as that requires
// special case handling due to implicit channel duplication.
// Stereo or Multichannel should actually be fine here.
state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
}
}
}
}
}
ALOGV("mixer configuration change: %d activeTracks (%08x) "
"all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
state->hook(state, pts);
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
if (countActiveTracks > 0) {
bool allMuted = true;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
en &= ~(1<<i);
track_t& t = state->tracks[i];
if (!t.doesResample() && t.volumeRL == 0) {
t.needs |= NEEDS_MUTE;
t.hook = track__nop;
} else {
allMuted = false;
}
}
if (allMuted) {
state->hook = process__nop;
} else if (all16BitsStereoNoResample) {
if (countActiveTracks == 1) {
const int i = 31 - __builtin_clz(state->enabledTracks);
track_t& t = state->tracks[i];
// Muted single tracks handled by allMuted above.
state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
}
}
}
}
void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
int32_t* temp, int32_t* aux)
{
ALOGVV("track__genericResample\n");
t->resampler->setSampleRate(t->sampleRate);
// ramp gain - resample to temp buffer and scale/mix in 2nd step
if (aux != NULL) {
// always resample with unity gain when sending to auxiliary buffer to be able
// to apply send level after resampling
t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
volumeRampStereo(t, out, outFrameCount, temp, aux);
} else {
volumeStereo(t, out, outFrameCount, temp, aux);
}
} else {
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
volumeRampStereo(t, out, outFrameCount, temp, aux);
}
// constant gain
else {
t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
t->resampler->resample(out, outFrameCount, t->bufferProvider);
}
}
}
void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused,
size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
{
}
void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux)
{
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
//ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
// ramp volume
if (CC_UNLIKELY(aux != NULL)) {
int32_t va = t->prevAuxLevel;
const int32_t vaInc = t->auxInc;
int32_t l;
int32_t r;
do {
l = (*temp++ >> 12);
r = (*temp++ >> 12);
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * r;
*aux++ += (va >> 17) * (l + r);
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevAuxLevel = va;
} else {
do {
*out++ += (vl >> 16) * (*temp++ >> 12);
*out++ += (vr >> 16) * (*temp++ >> 12);
vl += vlInc;
vr += vrInc;
} while (--frameCount);
}
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp(aux != NULL);
}
void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux)
{
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
if (CC_UNLIKELY(aux != NULL)) {
const int16_t va = t->auxLevel;
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
out[0] = mulAdd(l, vl, out[0]);
int16_t a = (int16_t)(((int32_t)l + r) >> 1);
out[1] = mulAdd(r, vr, out[1]);
out += 2;
aux[0] = mulAdd(a, va, aux[0]);
aux++;
} while (--frameCount);
} else {
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(r, vr, out[1]);
out += 2;
} while (--frameCount);
}
}
void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount,
int32_t* temp __unused, int32_t* aux)
{
ALOGVV("track__16BitsStereo\n");
const int16_t *in = static_cast<const int16_t *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
int32_t l;
int32_t r;
// ramp gain
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
const int32_t vaInc = t->auxInc;
// ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
l = (int32_t)*in++;
r = (int32_t)*in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * r;
*aux++ += (va >> 17) * (l + r);
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->prevAuxLevel = va;
t->adjustVolumeRamp(true);
}
// constant gain
else {
const uint32_t vrl = t->volumeRL;
const int16_t va = (int16_t)t->auxLevel;
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
out[1] = mulAddRL(0, rl, vrl, out[1]);
out += 2;
aux[0] = mulAdd(a, va, aux[0]);
aux++;
} while (--frameCount);
}
} else {
// ramp gain
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
// ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
*out++ += (vl >> 16) * (int32_t) *in++;
*out++ += (vr >> 16) * (int32_t) *in++;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp(false);
}
// constant gain
else {
const uint32_t vrl = t->volumeRL;
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
out[1] = mulAddRL(0, rl, vrl, out[1]);
out += 2;
} while (--frameCount);
}
}
t->in = in;
}
void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
int32_t* temp __unused, int32_t* aux)
{
ALOGVV("track__16BitsMono\n");
const int16_t *in = static_cast<int16_t const *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
// ramp gain
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
const int32_t vaInc = t->auxInc;
// ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
int32_t l = *in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * l;
*aux++ += (va >> 16) * l;
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->prevAuxLevel = va;
t->adjustVolumeRamp(true);
}
// constant gain
else {
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
const int16_t va = (int16_t)t->auxLevel;
do {
int16_t l = *in++;
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(l, vr, out[1]);
out += 2;
aux[0] = mulAdd(l, va, aux[0]);
aux++;
} while (--frameCount);
}
} else {
// ramp gain
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
// ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
int32_t l = *in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * l;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp(false);
}
// constant gain
else {
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
do {
int16_t l = *in++;
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(l, vr, out[1]);
out += 2;
} while (--frameCount);
}
}
t->in = in;
}
// no-op case
void AudioMixer::process__nop(state_t* state, int64_t pts)
{
ALOGVV("process__nop\n");
uint32_t e0 = state->enabledTracks;
while (e0) {
// process by group of tracks with same output buffer to
// avoid multiple memset() on same buffer
uint32_t e1 = e0, e2 = e0;
int i = 31 - __builtin_clz(e1);
{
track_t& t1 = state->tracks[i];
e2 &= ~(1<<i);
while (e2) {
i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t2 = state->tracks[i];
if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<i);
}
}
e0 &= ~(e1);
memset(t1.mainBuffer, 0, state->frameCount * t1.mMixerChannelCount
* audio_bytes_per_sample(t1.mMixerFormat));
}
while (e1) {
i = 31 - __builtin_clz(e1);
e1 &= ~(1<<i);
{
track_t& t3 = state->tracks[i];
size_t outFrames = state->frameCount;
while (outFrames) {
t3.buffer.frameCount = outFrames;
int64_t outputPTS = calculateOutputPTS(
t3, pts, state->frameCount - outFrames);
t3.bufferProvider->getNextBuffer(&t3.buffer, outputPTS);
if (t3.buffer.raw == NULL) break;
outFrames -= t3.buffer.frameCount;
t3.bufferProvider->releaseBuffer(&t3.buffer);
}
}
}
}
}
// generic code without resampling
void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
{
ALOGVV("process__genericNoResampling\n");
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
// acquire each track's buffer
uint32_t enabledTracks = state->enabledTracks;
uint32_t e0 = enabledTracks;
while (e0) {
const int i = 31 - __builtin_clz(e0);
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.buffer.frameCount = state->frameCount;
t.bufferProvider->getNextBuffer(&t.buffer, pts);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
}
e0 = enabledTracks;
while (e0) {
// process by group of tracks with same output buffer to
// optimize cache use
uint32_t e1 = e0, e2 = e0;
int j = 31 - __builtin_clz(e1);
track_t& t1 = state->tracks[j];
e2 &= ~(1<<j);
while (e2) {
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
e0 &= ~(e1);
// this assumes output 16 bits stereo, no resampling
int32_t *out = t1.mainBuffer;
size_t numFrames = 0;
do {
memset(outTemp, 0, sizeof(outTemp));
e2 = e1;
while (e2) {
const int i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t = state->tracks[i];
size_t outFrames = BLOCKSIZE;
int32_t *aux = NULL;
if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
e1 &= ~(1<<i);
break;
}
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
if (inFrames > 0) {
t.hook(&t, outTemp + (BLOCKSIZE - outFrames) * t.mMixerChannelCount,
inFrames, state->resampleTemp, aux);
t.frameCount -= inFrames;
outFrames -= inFrames;
if (CC_UNLIKELY(aux != NULL)) {
aux += inFrames;
}
}
if (t.frameCount == 0 && outFrames) {
t.bufferProvider->releaseBuffer(&t.buffer);
t.buffer.frameCount = (state->frameCount - numFrames) -
(BLOCKSIZE - outFrames);
int64_t outputPTS = calculateOutputPTS(
t, pts, numFrames + (BLOCKSIZE - outFrames));
t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
e1 &= ~(1<<i);
break;
}
t.frameCount = t.buffer.frameCount;
}
}
}
convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat,
BLOCKSIZE * t1.mMixerChannelCount);
// TODO: fix ugly casting due to choice of out pointer type
out = reinterpret_cast<int32_t*>((uint8_t*)out
+ BLOCKSIZE * t1.mMixerChannelCount
* audio_bytes_per_sample(t1.mMixerFormat));
numFrames += BLOCKSIZE;
} while (numFrames < state->frameCount);
}
// release each track's buffer
e0 = enabledTracks;
while (e0) {
const int i = 31 - __builtin_clz(e0);
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.bufferProvider->releaseBuffer(&t.buffer);
}
}
// generic code with resampling
void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
ALOGVV("process__genericResampling\n");
// this const just means that local variable outTemp doesn't change
int32_t* const outTemp = state->outputTemp;
size_t numFrames = state->frameCount;
uint32_t e0 = state->enabledTracks;
while (e0) {
// process by group of tracks with same output buffer
// to optimize cache use
uint32_t e1 = e0, e2 = e0;
int j = 31 - __builtin_clz(e1);
track_t& t1 = state->tracks[j];
e2 &= ~(1<<j);
while (e2) {
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
e0 &= ~(e1);
int32_t *out = t1.mainBuffer;
memset(outTemp, 0, sizeof(*outTemp) * t1.mMixerChannelCount * state->frameCount);
while (e1) {
const int i = 31 - __builtin_clz(e1);
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
aux = t.auxBuffer;
}
// this is a little goofy, on the resampling case we don't
// acquire/release the buffers because it's done by
// the resampler.
if (t.needs & NEEDS_RESAMPLE) {
t.resampler->setPTS(pts);
t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
size_t outFrames = 0;
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL) break;
if (CC_UNLIKELY(aux != NULL)) {
aux += outFrames;
}
t.hook(&t, outTemp + outFrames * t.mMixerChannelCount, t.buffer.frameCount,
state->resampleTemp, aux);
outFrames += t.buffer.frameCount;
t.bufferProvider->releaseBuffer(&t.buffer);
}
}
}
convertMixerFormat(out, t1.mMixerFormat,
outTemp, t1.mMixerInFormat, numFrames * t1.mMixerChannelCount);
}
}
// one track, 16 bits stereo without resampling is the most common case
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
int64_t pts)
{
ALOGVV("process__OneTrack16BitsStereoNoResampling\n");
// This method is only called when state->enabledTracks has exactly
// one bit set. The asserts below would verify this, but are commented out
// since the whole point of this method is to optimize performance.
//ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled");
const int i = 31 - __builtin_clz(state->enabledTracks);
//ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
const track_t& t = state->tracks[i];
AudioBufferProvider::Buffer& b(t.buffer);
int32_t* out = t.mainBuffer;
float *fout = reinterpret_cast<float*>(out);
size_t numFrames = state->frameCount;
const int16_t vl = t.volume[0];
const int16_t vr = t.volume[1];
const uint32_t vrl = t.volumeRL;
while (numFrames) {
b.frameCount = numFrames;
int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
t.bufferProvider->getNextBuffer(&b, outputPTS);
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)) {
memset(out, 0, numFrames
* t.mMixerChannelCount * audio_bytes_per_sample(t.mMixerFormat));
ALOGE_IF((((uintptr_t)in) & 3),
"process__OneTrack16BitsStereoNoResampling: misaligned buffer"
" %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
in, i, t.channelCount, t.needs, vrl, t.mVolume[0], t.mVolume[1]);
return;
}
size_t outFrames = b.frameCount;
switch (t.mMixerFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl);
int32_t r = mulRL(0, rl, vrl);
*fout++ = float_from_q4_27(l);
*fout++ = float_from_q4_27(r);
// Note: In case of later int16_t sink output,
// conversion and clamping is done by memcpy_to_i16_from_float().
} while (--outFrames);
break;
case AUDIO_FORMAT_PCM_16_BIT:
if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
// volume is boosted, so we might need to clamp even though
// we process only one track.
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
// clamping...
l = clamp16(l);
r = clamp16(r);
*out++ = (r<<16) | (l & 0xFFFF);
} while (--outFrames);
} else {
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
*out++ = (r<<16) | (l & 0xFFFF);
} while (--outFrames);
}
break;
default:
LOG_ALWAYS_FATAL("bad mixer format: %d", t.mMixerFormat);
}
numFrames -= b.frameCount;
t.bufferProvider->releaseBuffer(&b);
}
}
int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
int outputFrameIndex)
{
if (AudioBufferProvider::kInvalidPTS == basePTS) {
return AudioBufferProvider::kInvalidPTS;
}
return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate);
}
/*static*/ uint64_t AudioMixer::sLocalTimeFreq;
/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
/*static*/ void AudioMixer::sInitRoutine()
{
//cjh LocalClock lc;
// sLocalTimeFreq = lc.getLocalFreq(); // for the resampler
//
// DownmixerBufferProvider::init(); // for the downmixer
}
/* TODO: consider whether this level of optimization is necessary.
* Perhaps just stick with a single for loop.
*/
// Needs to derive a compile time constant (constexpr). Could be targeted to go
// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
#define MIXTYPE_MONOVOL(mixtype) (mixtype == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
mixtype == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : mixtype)
/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
*/
template <int MIXTYPE,
typename TO, typename TI, typename TV, typename TA, typename TAV>
static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
{
switch (channels) {
case 1:
volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
break;
case 2:
volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
break;
case 3:
volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
frameCount, in, aux, vol, volinc, vola, volainc);
break;
case 4:
volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
frameCount, in, aux, vol, volinc, vola, volainc);
break;
case 5:
volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
frameCount, in, aux, vol, volinc, vola, volainc);
break;
case 6:
volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
frameCount, in, aux, vol, volinc, vola, volainc);
break;
case 7:
volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
frameCount, in, aux, vol, volinc, vola, volainc);
break;
case 8:
volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
frameCount, in, aux, vol, volinc, vola, volainc);
break;
}
}
/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
*/
template <int MIXTYPE,
typename TO, typename TI, typename TV, typename TA, typename TAV>
static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
const TI* in, TA* aux, const TV *vol, TAV vola)
{
switch (channels) {
case 1:
volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
break;
case 2:
volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
break;
case 3:
volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
break;
case 4:
volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
break;
case 5:
volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
break;
case 6:
volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
break;
case 7:
volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
break;
case 8:
volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
break;
}
}
/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* USEFLOATVOL (set to true if float volume is used)
* ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
*/
template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
typename TO, typename TI, typename TA>
void AudioMixer::volumeMix(TO *out, size_t outFrames,
const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t)
{
if (USEFLOATVOL) {
if (ramp) {
volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc);
if (ADJUSTVOL) {
t->adjustVolumeRamp(aux != NULL, true);
}
} else {
volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
t->mVolume, t->auxLevel);
}
} else {
if (ramp) {
volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
if (ADJUSTVOL) {
t->adjustVolumeRamp(aux != NULL);
}
} else {
volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
t->volume, t->auxLevel);
}
}
}
/* This process hook is called when there is a single track without
* aux buffer, volume ramp, or resampling.
* TODO: Update the hook selection: this can properly handle aux and ramp.
*
* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
*/
template <int MIXTYPE, typename TO, typename TI, typename TA>
void AudioMixer::process_NoResampleOneTrack(state_t* state, int64_t pts)
{
ALOGVV("process_NoResampleOneTrack\n");
// CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz.
const int i = 31 - __builtin_clz(state->enabledTracks);
ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
track_t *t = &state->tracks[i];
const uint32_t channels = t->mMixerChannelCount;
TO* out = reinterpret_cast<TO*>(t->mainBuffer);
TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
const bool ramp = t->needsRamp();
for (size_t numFrames = state->frameCount; numFrames; ) {
AudioBufferProvider::Buffer& b(t->buffer);
// get input buffer
b.frameCount = numFrames;
const int64_t outputPTS = calculateOutputPTS(*t, pts, state->frameCount - numFrames);
t->bufferProvider->getNextBuffer(&b, outputPTS);
const TI *in = reinterpret_cast<TI*>(b.raw);
// in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (in == NULL || (((uintptr_t)in) & 3)) {
memset(out, 0, numFrames
* channels * audio_bytes_per_sample(t->mMixerFormat));
ALOGE_IF((((uintptr_t)in) & 3), "process_NoResampleOneTrack: bus error: "
"buffer %p track %p, channels %d, needs %#x",
in, t, t->channelCount, t->needs);
return;
}
const size_t outFrames = b.frameCount;
volumeMix<MIXTYPE, is_same<TI, float>::value, false> (
out, outFrames, in, aux, ramp, t);
out += outFrames * channels;
if (aux != NULL) {
aux += channels;
}
numFrames -= b.frameCount;
// release buffer
t->bufferProvider->releaseBuffer(&b);
}
if (ramp) {
t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
}
}
/* This track hook is called to do resampling then mixing,
* pulling from the track's upstream AudioBufferProvider.
*
* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
*/
template <int MIXTYPE, typename TO, typename TI, typename TA>
void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
{
ALOGVV("track__Resample\n");
t->resampler->setSampleRate(t->sampleRate);
const bool ramp = t->needsRamp();
if (ramp || aux != NULL) {
// if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step.
// if aux != NULL: resample with unity gain to temp buffer then apply send level.
t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO));
t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);
volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
out, outFrameCount, temp, aux, ramp, t);
} else { // constant volume gain
t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
t->resampler->resample((int32_t*)out, outFrameCount, t->bufferProvider);
}
}
/* This track hook is called to mix a track, when no resampling is required.
* The input buffer should be present in t->in.
*
* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
*/
template <int MIXTYPE, typename TO, typename TI, typename TA>
void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
TO* temp __unused, TA* aux)
{
ALOGVV("track__NoResample\n");
const TI *in = static_cast<const TI *>(t->in);
volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
out, frameCount, in, aux, t->needsRamp(), t);
// MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
// MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * t->mMixerChannelCount;
t->in = in;
}
/* The Mixer engine generates either int32_t (Q4_27) or float data.
* We use this function to convert the engine buffers
* to the desired mixer output format, either int16_t (Q.15) or float.
*/
void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
void *in, audio_format_t mixerInFormat, size_t sampleCount)
{
switch (mixerInFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
switch (mixerOutFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
break;
case AUDIO_FORMAT_PCM_16_BIT:
memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
break;
default:
LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
break;
}
break;
case AUDIO_FORMAT_PCM_16_BIT:
switch (mixerOutFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
memcpy_to_float_from_q4_27((float*)out, (int32_t*)in, sampleCount);
break;
case AUDIO_FORMAT_PCM_16_BIT:
// two int16_t are produced per iteration
ditherAndClamp((int32_t*)out, (int32_t*)in, sampleCount >> 1);
break;
default:
LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
break;
}
break;
default:
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
break;
}
}
/* Returns the proper track hook to use for mixing the track into the output buffer.
*/
AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount,
audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
{
if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
switch (trackType) {
case TRACKTYPE_NOP:
return track__nop;
case TRACKTYPE_RESAMPLE:
return track__genericResample;
case TRACKTYPE_NORESAMPLEMONO:
return track__16BitsMono;
case TRACKTYPE_NORESAMPLE:
return track__16BitsStereo;
default:
LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
break;
}
}
LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
switch (trackType) {
case TRACKTYPE_NOP:
return track__nop;
case TRACKTYPE_RESAMPLE:
switch (mixerInFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return (AudioMixer::hook_t)
track__Resample<MIXTYPE_MULTI, float /*TO*/, float /*TI*/, int32_t /*TA*/>;
case AUDIO_FORMAT_PCM_16_BIT:
return (AudioMixer::hook_t)\
track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
default:
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
break;
}
break;
case TRACKTYPE_NORESAMPLEMONO:
switch (mixerInFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return (AudioMixer::hook_t)
track__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>;
case AUDIO_FORMAT_PCM_16_BIT:
return (AudioMixer::hook_t)
track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>;
default:
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
break;
}
break;
case TRACKTYPE_NORESAMPLE:
switch (mixerInFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return (AudioMixer::hook_t)
track__NoResample<MIXTYPE_MULTI, float, float, int32_t>;
case AUDIO_FORMAT_PCM_16_BIT:
return (AudioMixer::hook_t)
track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
default:
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
break;
}
break;
default:
LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
break;
}
return NULL;
}
/* Returns the proper process hook for mixing tracks. Currently works only for
* PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
*
* TODO: Due to the special mixing considerations of duplicating to
* a stereo output track, the input track cannot be MONO. This should be
* prevented by the caller.
*/
AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t channelCount,
audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
{
if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
LOG_ALWAYS_FATAL("bad processType: %d", processType);
return NULL;
}
if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
return process__OneTrack16BitsStereoNoResampling;
}
LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
switch (mixerInFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
switch (mixerOutFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
float /*TO*/, float /*TI*/, int32_t /*TA*/>;
case AUDIO_FORMAT_PCM_16_BIT:
return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
int16_t, float, int32_t>;
default:
LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
break;
}
break;
case AUDIO_FORMAT_PCM_16_BIT:
switch (mixerOutFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
float, int16_t, int32_t>;
case AUDIO_FORMAT_PCM_16_BIT:
return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
int16_t, int16_t, int32_t>;
default:
LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
break;
}
break;
default:
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
break;
}
return NULL;
}
// ----------------------------------------------------------------------------
Remove experimental namespace (#20229)
2019-10-23 14:58:31 +08:00
} // namespace cocos2d {