Support bit-perfect PCM playback.
This is part of USB audio improvement. Bit-perfect PCM playback
indicates that the PCM data will be sent as is down to the audio HAL.
When the track is bit-perfect, there will not be volume control
applied in the audio mixer. Only effects without processing or hw
accelerated effects will be allowed to attach to bit-perfect tracks.
Bug: 239435816
Test: manually
Test: atest audiopolicy_tests
Change-Id: I0bad4d7d78d4eaf741754d01bc5422ba15374782
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 7309cad..9cc0ce0 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -1021,7 +1021,8 @@
audio_port_handle_t* selectedDeviceId,
audio_port_handle_t* portId,
std::vector<audio_io_handle_t>* secondaryOutputs,
- bool *isSpatialized) {
+ bool *isSpatialized,
+ bool *isBitPerfect) {
if (attr == nullptr) {
ALOGE("%s NULL audio attributes", __func__);
return BAD_VALUE;
@@ -1084,6 +1085,7 @@
*secondaryOutputs = VALUE_OR_RETURN_STATUS(convertContainer<std::vector<audio_io_handle_t>>(
responseAidl.secondaryOutputs, aidl2legacy_int32_t_audio_io_handle_t));
*isSpatialized = responseAidl.isSpatialized;
+ *isBitPerfect = responseAidl.isBitPerfect;
return OK;
}
diff --git a/media/libaudioclient/PolicyAidlConversion.cpp b/media/libaudioclient/PolicyAidlConversion.cpp
index 3790000..60b08fa 100644
--- a/media/libaudioclient/PolicyAidlConversion.cpp
+++ b/media/libaudioclient/PolicyAidlConversion.cpp
@@ -477,6 +477,8 @@
switch (aidl) {
case media::AudioMixerBehavior::DEFAULT:
return AUDIO_MIXER_BEHAVIOR_DEFAULT;
+ case media::AudioMixerBehavior::BIT_PERFECT:
+ return AUDIO_MIXER_BEHAVIOR_BIT_PERFECT;
case media::AudioMixerBehavior::INVALID:
return AUDIO_MIXER_BEHAVIOR_INVALID;
}
@@ -487,6 +489,8 @@
switch (legacy) {
case AUDIO_MIXER_BEHAVIOR_DEFAULT:
return media::AudioMixerBehavior::DEFAULT;
+ case AUDIO_MIXER_BEHAVIOR_BIT_PERFECT:
+ return media::AudioMixerBehavior::BIT_PERFECT;
case AUDIO_MIXER_BEHAVIOR_INVALID:
return media::AudioMixerBehavior::INVALID;
}
diff --git a/media/libaudioclient/aidl/android/media/AudioMixerBehavior.aidl b/media/libaudioclient/aidl/android/media/AudioMixerBehavior.aidl
index 0c0c070..38f50d6 100644
--- a/media/libaudioclient/aidl/android/media/AudioMixerBehavior.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioMixerBehavior.aidl
@@ -30,4 +30,8 @@
* different sources.
*/
DEFAULT = 0,
+ /**
+ * The audio data in the mixer will be bit-perfect as long as possible.
+ */
+ BIT_PERFECT = 1,
}
diff --git a/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl b/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
index 5b25d79..385f787 100644
--- a/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
@@ -36,4 +36,5 @@
boolean isSpatialized;
/** The suggested audio config if fails to get an output. **/
AudioConfigBase configBase;
+ boolean isBitPerfect;
}
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 507802d..163adcb 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -300,6 +300,7 @@
* @param[out] portId the generated port id to identify the client
* @param[out] secondaryOutputs collection of io handle for secondary outputs
* @param[out] isSpatialized true if the playback will be spatialized
+ * @param[out] isBitPerfect true if the playback will be bit-perfect
* @return if the call is successful or not
*/
static status_t getOutputForAttr(audio_attributes_t *attr,
@@ -312,7 +313,8 @@
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs,
- bool *isSpatialized);
+ bool *isSpatialized,
+ bool *isBitPerfect);
static status_t startOutput(audio_port_handle_t portId);
static status_t stopOutput(audio_port_handle_t portId);
static void releaseOutput(audio_port_handle_t portId);
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index e6fdb1d..c2b82d1 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -297,6 +297,27 @@
return NO_ERROR;
}
+void AudioMixer::Track::unprepareForTee() {
+ ALOGV("AudioMixer::%s", __func__);
+ if (mTeeBufferProvider.get() != nullptr) {
+ mTeeBufferProvider.reset(nullptr);
+ reconfigureBufferProviders();
+ }
+}
+
+status_t AudioMixer::Track::prepareForTee() {
+ ALOGV("AudioMixer::%s(%p) teeBuffer=%p", __func__, this, teeBuffer);
+ unprepareForTee();
+ if (teeBuffer != nullptr) {
+ const size_t frameSize = audio_bytes_per_frame(channelCount + mHapticChannelCount, mFormat);
+ mTeeBufferProvider.reset(new TeeBufferProvider(
+ frameSize, frameSize, kCopyBufferFrameCount,
+ (uint8_t*)teeBuffer, mTeeBufferFrameCount));
+ reconfigureBufferProviders();
+ }
+ return NO_ERROR;
+}
+
void AudioMixer::Track::clearContractedBuffer()
{
if (mAdjustChannelsBufferProvider.get() != nullptr) {
@@ -305,10 +326,20 @@
}
}
+void AudioMixer::Track::clearTeeFrameCopied() {
+ if (mTeeBufferProvider.get() != nullptr) {
+ static_cast<TeeBufferProvider*>(mTeeBufferProvider.get())->clearFramesCopied();
+ }
+}
+
void AudioMixer::Track::reconfigureBufferProviders()
{
// configure from upstream to downstream buffer providers.
bufferProvider = mInputBufferProvider;
+ if (mTeeBufferProvider != nullptr) {
+ mTeeBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = mTeeBufferProvider.get();
+ }
if (mAdjustChannelsBufferProvider.get() != nullptr) {
mAdjustChannelsBufferProvider->setBufferProvider(bufferProvider);
bufferProvider = mAdjustChannelsBufferProvider.get();
@@ -420,6 +451,20 @@
track->mHapticMaxAmplitude = hapticMaxAmplitude;
}
} break;
+ case TEE_BUFFER:
+ if (track->teeBuffer != valueBuf) {
+ track->teeBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, TEE_BUFFER, %p)", valueBuf);
+ track->prepareForTee();
+ }
+ break;
+ case TEE_BUFFER_FRAME_COUNT:
+ if (track->mTeeBufferFrameCount != valueInt) {
+ track->mTeeBufferFrameCount = valueInt;
+ ALOGV("setParameter(TRACK, TEE_BUFFER_FRAME_COUNT, %i)", valueInt);
+ track->prepareForTee();
+ }
+ break;
default:
LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
}
@@ -500,6 +545,8 @@
track->mReformatBufferProvider->reset();
} else if (track->mAdjustChannelsBufferProvider.get() != nullptr) {
track->mAdjustChannelsBufferProvider->reset();
+ } else if (track->mTeeBufferProvider.get() != nullptr) {
+ track->mTeeBufferProvider->reset();
}
track->mInputBufferProvider = bufferProvider;
@@ -565,6 +612,7 @@
if (t->mKeepContractedChannels) {
t->clearContractedBuffer();
}
+ t->clearTeeFrameCopied();
}
}
diff --git a/media/libaudioprocessing/AudioMixerBase.cpp b/media/libaudioprocessing/AudioMixerBase.cpp
index f30eb54..fd06991 100644
--- a/media/libaudioprocessing/AudioMixerBase.cpp
+++ b/media/libaudioprocessing/AudioMixerBase.cpp
@@ -143,6 +143,7 @@
// setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
t->mainBuffer = NULL;
t->auxBuffer = NULL;
+ t->teeBuffer = nullptr;
t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
t->mFormat = format;
t->mMixerInFormat = kUseFloat && kUseNewMixer ?
@@ -150,6 +151,7 @@
t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
+ t->mTeeBufferFrameCount = 0;
status_t status = postCreateTrack(t.get());
if (status != OK) return status;
mTracks[name] = t;
@@ -401,6 +403,20 @@
invalidate();
}
} break;
+ case TEE_BUFFER:
+ if (track->teeBuffer != valueBuf) {
+ track->teeBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, TEE_BUFFER, %p)", valueBuf);
+ invalidate();
+ }
+ break;
+ case TEE_BUFFER_FRAME_COUNT:
+ if (track->mTeeBufferFrameCount != valueInt) {
+ track->mTeeBufferFrameCount = valueInt;
+ ALOGV("setParameter(TRACK, TEE_BUFFER_FRAME_COUNT, %i)", valueInt);
+ invalidate();
+ }
+ break;
default:
LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
}
diff --git a/media/libaudioprocessing/BufferProviders.cpp b/media/libaudioprocessing/BufferProviders.cpp
index 4658db8..a9944fb 100644
--- a/media/libaudioprocessing/BufferProviders.cpp
+++ b/media/libaudioprocessing/BufferProviders.cpp
@@ -739,5 +739,21 @@
mContractedWrittenFrames = 0;
CopyBufferProvider::reset();
}
+
+void TeeBufferProvider::copyFrames(void *dst, const void *src, size_t frames) {
+ memcpy(dst, src, frames * mInputFrameSize);
+ if (int teeBufferFrameLeft = mTeeBufferFrameCount - mFrameCopied; teeBufferFrameLeft < frames) {
+ ALOGW("Unable to copy all frames to tee buffer, %d frames dropped",
+ (int)frames - teeBufferFrameLeft);
+ frames = teeBufferFrameLeft;
+ }
+ memcpy(mTeeBuffer + mFrameCopied * mInputFrameSize, src, frames * mInputFrameSize);
+ mFrameCopied += frames;
+}
+
+void TeeBufferProvider::clearFramesCopied() {
+ mFrameCopied = 0;
+}
+
// ----------------------------------------------------------------------------
} // namespace android
diff --git a/media/libaudioprocessing/include/media/AudioMixer.h b/media/libaudioprocessing/include/media/AudioMixer.h
index 2993a60..b39fb92 100644
--- a/media/libaudioprocessing/include/media/AudioMixer.h
+++ b/media/libaudioprocessing/include/media/AudioMixer.h
@@ -96,7 +96,10 @@
void unprepareForReformat();
status_t prepareForAdjustChannels(size_t frames);
void unprepareForAdjustChannels();
+ void unprepareForTee();
+ status_t prepareForTee();
void clearContractedBuffer();
+ void clearTeeFrameCopied();
bool setPlaybackRate(const AudioPlaybackRate &playbackRate);
void reconfigureBufferProviders();
@@ -108,20 +111,22 @@
* all pre-mixer track buffer conversions outside the AudioMixer class.
*
* 1) mInputBufferProvider: The AudioTrack buffer provider.
- * 2) mAdjustChannelsBufferProvider: Expands or contracts sample data from one interleaved
+ * 2) mTeeBufferProvider: If not NULL, copy the data to tee buffer.
+ * 3) mAdjustChannelsBufferProvider: Expands or contracts sample data from one interleaved
* channel format to another. Expanded channels are filled with zeros and put at the end
* of each audio frame. Contracted channels are copied to the end of the buffer.
- * 3) mReformatBufferProvider: If not NULL, performs the audio reformat to
+ * 4) mReformatBufferProvider: If not NULL, performs the audio reformat to
* match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
* requires reformat. For example, it may convert floating point input to
* PCM_16_bit if that's required by the downmixer.
- * 4) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match
+ * 5) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match
* the number of channels required by the mixer sink.
- * 5) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
+ * 6) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
* the downmixer requirements to the mixer engine input requirements.
- * 6) mTimestretchBufferProvider: Adds timestretching for playback rate
+ * 7) mTimestretchBufferProvider: Adds timestretching for playback rate
*/
AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
+ std::unique_ptr<PassthruBufferProvider> mTeeBufferProvider;
std::unique_ptr<PassthruBufferProvider> mAdjustChannelsBufferProvider;
std::unique_ptr<PassthruBufferProvider> mReformatBufferProvider;
std::unique_ptr<PassthruBufferProvider> mDownmixerBufferProvider;
diff --git a/media/libaudioprocessing/include/media/AudioMixerBase.h b/media/libaudioprocessing/include/media/AudioMixerBase.h
index 3419816..caccb6a 100644
--- a/media/libaudioprocessing/include/media/AudioMixerBase.h
+++ b/media/libaudioprocessing/include/media/AudioMixerBase.h
@@ -68,6 +68,10 @@
// 0x4004 reserved
MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+ // 0x4007, 0x4008, 0x4009 is defined for haptic stuff in AudioMixer.h
+ TEE_BUFFER = 0x400A,
+ TEE_BUFFER_FORMAT = 0x400B,
+ TEE_BUFFER_FRAME_COUNT = 0x400C,
// for target RESAMPLE
SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
// parameter 'value' is the new sample rate in Hz.
@@ -271,6 +275,7 @@
uint32_t sampleRate;
int32_t* mainBuffer;
int32_t* auxBuffer;
+ int32_t* teeBuffer;
int32_t sessionId;
@@ -290,6 +295,8 @@
audio_channel_mask_t mMixerChannelMask;
uint32_t mMixerChannelCount;
+ int32_t mTeeBufferFrameCount;
+
protected:
// hooks
diff --git a/media/libaudioprocessing/include/media/BufferProviders.h b/media/libaudioprocessing/include/media/BufferProviders.h
index b3ab8a5..7a41002 100644
--- a/media/libaudioprocessing/include/media/BufferProviders.h
+++ b/media/libaudioprocessing/include/media/BufferProviders.h
@@ -279,6 +279,27 @@
size_t mContractedWrittenFrames;
size_t mContractedOutputFrameSize; // contracted output frame size
};
+
+class TeeBufferProvider : public CopyBufferProvider {
+public:
+ TeeBufferProvider(
+ size_t inputFrameSize, size_t outputFrameSize,
+ size_t bufferFrameCount, uint8_t* teeBuffer, int teeBufferFrameCount)
+ : CopyBufferProvider(inputFrameSize, outputFrameSize, bufferFrameCount),
+ mTeeBuffer(teeBuffer), mTeeBufferFrameCount(teeBufferFrameCount),
+ mFrameCopied(0) {};
+
+ void copyFrames(void *dst, const void *src, size_t frames) override;
+
+ void clearFramesCopied();
+
+protected:
+ AudioBufferProvider *mTrackBufferProvider;
+ uint8_t* mTeeBuffer;
+ const int mTeeBufferFrameCount;
+ int mFrameCopied;
+};
+
// ----------------------------------------------------------------------------
} // namespace android
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 474c5be..c307c94 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -621,13 +621,15 @@
fullConfig.format = config->format;
std::vector<audio_io_handle_t> secondaryOutputs;
bool isSpatialized;
+ bool isBitPerfect;
ret = AudioSystem::getOutputForAttr(&localAttr, &io,
actualSessionId,
&streamType, adjAttributionSource,
&fullConfig,
(audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
AUDIO_OUTPUT_FLAG_DIRECT),
- deviceId, &portId, &secondaryOutputs, &isSpatialized);
+ deviceId, &portId, &secondaryOutputs, &isSpatialized,
+ &isBitPerfect);
if (ret != NO_ERROR) {
config->sample_rate = fullConfig.sample_rate;
config->channel_mask = fullConfig.channel_mask;
@@ -1083,7 +1085,8 @@
audio_stream_type_t streamType;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
std::vector<audio_io_handle_t> secondaryOutputs;
- bool isSpatialized = false;;
+ bool isSpatialized = false;
+ bool isBitPerfect = false;
// TODO b/182392553: refactor or make clearer
pid_t clientPid =
@@ -1130,7 +1133,7 @@
lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
adjAttributionSource, &input.config, input.flags,
&output.selectedDeviceId, &portId, &secondaryOutputs,
- &isSpatialized);
+ &isSpatialized, &isBitPerfect);
if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("createTrack() getOutputForAttr() return error %d or invalid output handle", lStatus);
@@ -1196,7 +1199,8 @@
input.notificationsPerBuffer, input.speed,
input.sharedBuffer, sessionId, &output.flags,
callingPid, adjAttributionSource, input.clientInfo.clientTid,
- &lStatus, portId, input.audioTrackCallback, isSpatialized);
+ &lStatus, portId, input.audioTrackCallback, isSpatialized,
+ isBitPerfect);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
@@ -2912,7 +2916,11 @@
return thread;
} else {
sp<PlaybackThread> thread;
- if (flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {
+ if (flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
+ thread = sp<BitPerfectThread>::make(this, outputStream, *output, mSystemReady);
+ ALOGV("%s() created bit-perfect output: ID %d thread %p",
+ __func__, *output, thread.get());
+ } else if (flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {
thread = new SpatializerThread(this, outputStream, *output,
mSystemReady, mixerConfig);
ALOGV("openOutput_l() created spatializer output: ID %d thread %p",
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index a2cde70..333c57b 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -590,6 +590,7 @@
class OffloadThread;
class DuplicatingThread;
class AsyncCallbackThread;
+ class BitPerfectThread;
class Track;
class RecordTrack;
class EffectBase;
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index fb03812..84b9c40 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -2961,6 +2961,9 @@
if ((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0 && !isFastCompatible()) {
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
}
+ if ((*flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != 0 && !isBitPerfectCompatible()) {
+ *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_BIT_PERFECT);
+ }
}
void AudioFlinger::EffectChain::checkInputFlagCompatibility(audio_input_flags_t *flags) const
@@ -2998,6 +3001,18 @@
return true;
}
+bool AudioFlinger::EffectChain::isBitPerfectCompatible() const {
+ Mutex::Autolock _l(mLock);
+ for (const auto &effect : mEffects) {
+ if (effect->isProcessImplemented()
+ && effect->isImplementationSoftware()) {
+ return false;
+ }
+ }
+ // Allow effects without processing or hw accelerated effects.
+ return true;
+}
+
// isCompatibleWithThread_l() must be called with thread->mLock held
bool AudioFlinger::EffectChain::isCompatibleWithThread_l(const sp<ThreadBase>& thread) const
{
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 72ec0e5..7b71a85 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -545,6 +545,9 @@
// Is this EffectChain compatible with the FAST audio flag.
bool isFastCompatible() const;
+ // Is this EffectChain compatible with the bit-perfect audio flag.
+ bool isBitPerfectCompatible() const;
+
// isCompatibleWithThread_l() must be called with thread->mLock held
bool isCompatibleWithThread_l(const sp<ThreadBase>& thread) const;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 0f1373d..ebbdf56 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -83,7 +83,8 @@
* ready as possible (aka. Buffer is full). */
size_t frameCountToBeReady = SIZE_MAX,
float speed = 1.0f,
- bool isSpatialized = false);
+ bool isSpatialized = false,
+ bool isBitPerfect = false);
virtual ~Track();
virtual status_t initCheck() const;
@@ -203,6 +204,7 @@
audio_output_flags_t getOutputFlags() const { return mFlags; }
float getSpeed() const { return mSpeed; }
bool isSpatialized() const override { return mIsSpatialized; }
+ bool isBitPerfect() const override { return mIsBitPerfect; }
/**
* Updates the mute state and notifies the audio service. Call this only when holding player
@@ -361,6 +363,7 @@
TeePatches mTeePatches;
const float mSpeed;
const bool mIsSpatialized;
+ const bool mIsBitPerfect;
// TODO: replace PersistableBundle with own struct
// access these two variables only when holding player thread lock.
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index e06d2cd..f917527 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -536,6 +536,8 @@
return "MMAP_CAPTURE";
case SPATIALIZER:
return "SPATIALIZER";
+ case BIT_PERFECT:
+ return "BIT_PERFECT";
default:
return "unknown";
}
@@ -1519,6 +1521,26 @@
}
}
break;
+ case BIT_PERFECT:
+ if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) {
+ // Allow HW accelerated effects of tunnel type
+ break;
+ }
+ // As bit-perfect tracks will not be allowed to apply audio effect that will touch the audio
+ // data, effects will not be allowed on 1) global effects (AUDIO_SESSION_OUTPUT_MIX),
+ // 2) post-processing effects (AUDIO_SESSION_OUTPUT_STAGE or AUDIO_SESSION_DEVICE) and
+ // 3) there is any bit-perfect track with the given session id.
+ if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE ||
+ sessionId == AUDIO_SESSION_DEVICE) {
+ ALOGW("%s: effect %s not supported on bit-perfect thread %s",
+ __func__, desc->name, mThreadName);
+ return BAD_VALUE;
+ } else if ((hasAudioSession_l(sessionId) & ThreadBase::BIT_PERFECT_SESSION) != 0) {
+ ALOGW("%s: effect %s not supported as there is a bit-perfect track with session as %d",
+ __func__, desc->name, sessionId);
+ return BAD_VALUE;
+ }
+ break;
default:
LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType);
}
@@ -2283,7 +2305,8 @@
status_t *status,
audio_port_handle_t portId,
const sp<media::IAudioTrackCallback>& callback,
- bool isSpatialized)
+ bool isSpatialized,
+ bool isBitPerfect)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
@@ -2315,6 +2338,25 @@
*flags = (audio_output_flags_t)(*flags & outputFlags);
}
+ if (isBitPerfect) {
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
+ if (chain.get() != nullptr) {
+ // Bit-perfect is required according to the configuration and preferred mixer
+ // attributes, but it is not in the output flag from the client's request. Explicitly
+ // adding bit-perfect flag to check the compatibility
+ audio_output_flags_t flagsToCheck =
+ (audio_output_flags_t)(*flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT);
+ chain->checkOutputFlagCompatibility(&flagsToCheck);
+ if ((flagsToCheck & AUDIO_OUTPUT_FLAG_BIT_PERFECT) == AUDIO_OUTPUT_FLAG_NONE) {
+ ALOGE("%s cannot create track as there is data-processing effect attached to "
+ "given session id(%d)", __func__, sessionId);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ *flags = flagsToCheck;
+ }
+ }
+
// client expresses a preference for FAST, but we get the final say
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (
@@ -2495,6 +2537,18 @@
*pNotificationFrameCount = notificationFrameCount;
switch (mType) {
+ case BIT_PERFECT:
+ if (isBitPerfect) {
+ if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
+ ALOGE("%s, bad parameter when request streaming bit-perfect, sampleRate=%u, "
+ "format=%#x, channelMask=%#x, mSampleRate=%u, mFormat=%#x, mChannelMask=%#x",
+ __func__, sampleRate, format, channelMask, mSampleRate, mFormat,
+ mChannelMask);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ }
+ break;
case DIRECT:
if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()?
@@ -2576,7 +2630,7 @@
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
sessionId, creatorPid, attributionSource, trackFlags,
TrackBase::TYPE_DEFAULT, portId, SIZE_MAX /*frameCountToBeReady*/,
- speed, isSpatialized);
+ speed, isSpatialized, isBitPerfect);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
@@ -4020,7 +4074,8 @@
// Either threadLoop_mix() or threadLoop_sleepTime() should have set
// mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0.
// Merge mMixerBuffer data into mEffectBuffer (if any effects are valid)
- // or mSinkBuffer (if there are no effects).
+ // or mSinkBuffer (if there are no effects and there is no data already copied to
+ // mSinkBuffer).
//
// This is done pre-effects computation; if effects change to
// support higher precision, this needs to move.
@@ -4029,7 +4084,7 @@
// TODO use mSleepTimeUs == 0 as an additional condition.
uint32_t mixerChannelCount = mEffectBufferValid ?
audio_channel_count_from_out_mask(mMixerChannelMask) : mChannelCount;
- if (mMixerBufferValid) {
+ if (mMixerBufferValid && (mEffectBufferValid || !mHasDataCopiedToSinkBuffer)) {
void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
@@ -4120,7 +4175,7 @@
// Effects buffer (buffer valid), we need to
// copy into the sink buffer.
// TODO use mSleepTimeUs == 0 as an additional condition.
- if (mEffectBufferValid) {
+ if (mEffectBufferValid && !mHasDataCopiedToSinkBuffer) {
//ALOGV("writing effect buffer to sink buffer format %#x", mFormat);
void *effectBuffer = (mType == SPATIALIZER) ? mPostSpatializerBuffer : mEffectBuffer;
if (requireMonoBlend()) {
@@ -4801,7 +4856,7 @@
// initialize fast mixer depending on configuration
bool initFastMixer;
- if (mType == SPATIALIZER) {
+ if (mType == SPATIALIZER || mType == BIT_PERFECT) {
initFastMixer = false;
} else {
switch (kUseFastMixer) {
@@ -9781,6 +9836,7 @@
audio_port_handle_t deviceId = mDeviceId;
std::vector<audio_io_handle_t> secondaryOutputs;
bool isSpatialized;
+ bool isBitPerfect;
ret = AudioSystem::getOutputForAttr(&mAttr, &io,
mSessionId,
&stream,
@@ -9790,7 +9846,8 @@
&deviceId,
&portId,
&secondaryOutputs,
- &isSpatialized);
+ &isSpatialized,
+ &isBitPerfect);
ALOGD_IF(!secondaryOutputs.empty(),
"MmapThread::start does not support secondary outputs, ignoring them");
} else {
@@ -10737,4 +10794,40 @@
return mInput->getCapturePosition((int64_t*)position, timeNanos);
}
+// ----------------------------------------------------------------------------
+
+AudioFlinger::BitPerfectThread::BitPerfectThread(const sp<AudioFlinger> &audioflinger,
+ AudioStreamOut *output, audio_io_handle_t id, bool systemReady)
+ : MixerThread(audioflinger, output, id, systemReady, BIT_PERFECT) {}
+
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::BitPerfectThread::prepareTracks_l(
+ Vector<sp<Track>> *tracksToRemove) {
+ mixer_state result = MixerThread::prepareTracks_l(tracksToRemove);
+ // If there is only one active track and it is bit-perfect, enable tee buffer.
+ if (mActiveTracks.size() == 1 && mActiveTracks[0]->isBitPerfect()) {
+ const int trackId = mActiveTracks[0]->id();
+ mAudioMixer->setParameter(
+ trackId, AudioMixer::TRACK, AudioMixer::TEE_BUFFER, (void *)mSinkBuffer);
+ mAudioMixer->setParameter(
+ trackId, AudioMixer::TRACK, AudioMixer::TEE_BUFFER_FRAME_COUNT,
+ (void *)(uintptr_t)mNormalFrameCount);
+ mIsBitPerfect = true;
+ } else {
+ mIsBitPerfect = false;
+ // No need to copy bit-perfect data directly to sink buffer given there are multiple tracks
+ // active.
+ for (const auto& track : mActiveTracks) {
+ const int trackId = track->id();
+ mAudioMixer->setParameter(
+ trackId, AudioMixer::TRACK, AudioMixer::TEE_BUFFER, nullptr);
+ }
+ }
+ return result;
+}
+
+void AudioFlinger::BitPerfectThread::threadLoop_mix() {
+ MixerThread::threadLoop_mix();
+ mHasDataCopiedToSinkBuffer = mIsBitPerfect;
+}
+
} // namespace android
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 8bf629b..f1b82e4 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -33,6 +33,7 @@
MMAP_PLAYBACK, // Thread class for MMAP playback stream
MMAP_CAPTURE, // Thread class for MMAP capture stream
SPATIALIZER, //
+ BIT_PERFECT, // Thread class for BitPerfectThread
// If you add any values here, also update ThreadBase::threadTypeToString()
};
@@ -430,8 +431,10 @@
// track
FAST_SESSION = 0x4, // the audio session corresponds to at least one
// fast track
- SPATIALIZED_SESSION = 0x8 // the audio session corresponds to at least one
- // spatialized track
+ SPATIALIZED_SESSION = 0x8, // the audio session corresponds to at least one
+ // spatialized track
+ BIT_PERFECT_SESSION = 0x10 // the audio session corresponds to at least one
+ // bit-perfect track
};
// get effect chain corresponding to session Id.
@@ -497,6 +500,9 @@
if (track->isSpatialized()) {
result |= SPATIALIZED_SESSION; // caution, only first track.
}
+ if (track->isBitPerfect()) {
+ result |= BIT_PERFECT_SESSION;
+ }
break;
}
}
@@ -982,7 +988,8 @@
status_t *status /*non-NULL*/,
audio_port_handle_t portId,
const sp<media::IAudioTrackCallback>& callback,
- bool isSpatialized);
+ bool isSpatialized,
+ bool isBitPerfect);
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
@@ -1163,6 +1170,9 @@
// for any processing (including output processing).
bool mEffectBufferValid;
+ // Set to "true" to enable when data has already copied to sink
+ bool mHasDataCopiedToSinkBuffer = false;
+
// Frame size aligned buffer used as input and output to all post processing effects
// except the Spatializer in a SPATIALIZER thread. Non spatialized tracks are mixed into
// this buffer so that post processing effects can be applied.
@@ -2273,3 +2283,16 @@
AudioStreamIn* mInput;
};
+
+class BitPerfectThread : public MixerThread {
+public:
+ BitPerfectThread(const sp<AudioFlinger>& audioflinger, AudioStreamOut *output,
+ audio_io_handle_t id, bool systemReady);
+
+protected:
+ mixer_state prepareTracks_l(Vector<sp<Track>> *tracksToRemove) override;
+ void threadLoop_mix() override;
+
+private:
+ bool mIsBitPerfect;
+};
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 20bfbb0..f305aa8 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -109,6 +109,8 @@
virtual bool isSpatialized() const { return false; }
+ virtual bool isBitPerfect() const { return false; }
+
#ifdef TEE_SINK
void dumpTee(int fd, const std::string &reason) const {
mTee.dump(fd, reason);
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6493e2a..950d555 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -633,7 +633,8 @@
audio_port_handle_t portId,
size_t frameCountToBeReady,
float speed,
- bool isSpatialized)
+ bool isSpatialized,
+ bool isBitPerfect)
: TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
@@ -668,7 +669,8 @@
mFlushHwPending(false),
mFlags(flags),
mSpeed(speed),
- mIsSpatialized(isSpatialized)
+ mIsSpatialized(isSpatialized),
+ mIsBitPerfect(isBitPerfect)
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 9c4ab80..a5fa78b 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -144,7 +144,8 @@
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs,
output_type_t *outputType,
- bool *isSpatialized) = 0;
+ bool *isSpatialized,
+ bool *isBitPerfect) = 0;
// indicates to the audio policy manager that the output starts being used by corresponding
// stream.
virtual status_t startOutput(audio_port_handle_t portId) = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index 084399d..c489eed 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -66,6 +66,9 @@
if (getRole() == AUDIO_PORT_ROLE_SOURCE) {
mMixerBehaviors.clear();
mMixerBehaviors.insert(AUDIO_MIXER_BEHAVIOR_DEFAULT);
+ if (mFlags.output & AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
+ mMixerBehaviors.insert(AUDIO_MIXER_BEHAVIOR_BIT_PERFECT);
+ }
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 0b3ba4d..c7296e9 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -302,7 +302,9 @@
mDirectClientSession(AUDIO_SESSION_NONE)
{
if (profile != NULL) {
- mFlags = (audio_output_flags_t)profile->getFlags();
+ // By default, opening the output without immutable flags, the bit-perfect flags should be
+ // applied when the apps explicitly request.
+ mFlags = (audio_output_flags_t)(profile->getFlags() & (~AUDIO_OUTPUT_FLAG_BIT_PERFECT));
}
}
diff --git a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
index 28268c9..14f565b 100644
--- a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
+++ b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
@@ -259,13 +259,15 @@
*portId = AUDIO_PORT_HANDLE_NONE;
AudioPolicyInterface::output_type_t outputType;
bool isSpatialized;
+ bool isBitPerfect;
// TODO b/182392769: use attribution source util
AttributionSourceState attributionSource;
attributionSource.uid = 0;
attributionSource.token = sp<BBinder>::make();
if (mManager->getOutputForAttr(&attr, output, AUDIO_SESSION_NONE, &stream, attributionSource,
- &config, &flags, selectedDeviceId, portId, {}, &outputType, &isSpatialized) != OK) {
+ &config, &flags, selectedDeviceId, portId, {}, &outputType, &isSpatialized,
+ &isBitPerfect) != OK) {
return false;
}
if (*output == AUDIO_IO_HANDLE_NONE || *portId == AUDIO_PORT_HANDLE_NONE) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 959dd4f..346191f 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1142,7 +1142,8 @@
bool *isRequestedDeviceForExclusiveUse,
std::vector<sp<AudioPolicyMix>> *secondaryMixes,
output_type_t *outputType,
- bool *isSpatialized)
+ bool *isSpatialized,
+ bool *isBitPerfect)
{
DeviceVector outputDevices;
const audio_port_handle_t requestedPortId = *selectedDeviceId;
@@ -1286,6 +1287,9 @@
}
*output = getOutputForDevices(outputDevices, session, resultAttr, config,
flags, isSpatialized, info, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
+ *isBitPerfect = (info != nullptr
+ && (info->getFlags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE
+ && *output != AUDIO_IO_HANDLE_NONE);
}
if (*output == AUDIO_IO_HANDLE_NONE) {
AudioProfileVector profiles;
@@ -1330,7 +1334,8 @@
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs,
output_type_t *outputType,
- bool *isSpatialized)
+ bool *isSpatialized,
+ bool *isBitPerfect)
{
// The supplied portId must be AUDIO_PORT_HANDLE_NONE
if (*portId != AUDIO_PORT_HANDLE_NONE) {
@@ -1352,7 +1357,8 @@
status_t status = getOutputForAttrInt(&resultAttr, output, session, attr, stream, uid,
config, flags, selectedDeviceId, &isRequestedDeviceForExclusiveUse,
- secondaryOutputs != nullptr ? &secondaryMixes : nullptr, outputType, isSpatialized);
+ secondaryOutputs != nullptr ? &secondaryMixes : nullptr, outputType, isSpatialized,
+ isBitPerfect);
if (status != NO_ERROR) {
return status;
}
@@ -4786,10 +4792,11 @@
bool isRequestedDeviceForExclusiveUse = false;
output_type_t outputType;
bool isSpatialized;
+ bool isBitPerfect;
getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE, &attributes,
&stream, sourceDesc->uid(), &config, &flags,
&selectedDeviceId, &isRequestedDeviceForExclusiveUse,
- nullptr, &outputType, &isSpatialized);
+ nullptr, &outputType, &isSpatialized, &isBitPerfect);
if (output == AUDIO_IO_HANDLE_NONE) {
ALOGV("%s no output for device %s",
__FUNCTION__, sinkDevice->toString().c_str());
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index b7a55e9..f9cf9f5 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -124,7 +124,8 @@
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs,
output_type_t *outputType,
- bool *isSpatialized) override;
+ bool *isSpatialized,
+ bool *isBitPerfect) override;
virtual status_t startOutput(audio_port_handle_t portId);
virtual status_t stopOutput(audio_port_handle_t portId);
virtual bool releaseOutput(audio_port_handle_t portId);
@@ -1080,7 +1081,8 @@
bool *isRequestedDeviceForExclusiveUse,
std::vector<sp<AudioPolicyMix>> *secondaryMixes,
output_type_t *outputType,
- bool *isSpatialized);
+ bool *isSpatialized,
+ bool *isBitPerfect);
// internal method to return the output handle for the given device and format
audio_io_handle_t getOutputForDevices(
const DeviceVector &devices,
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 1088088..4eb5336 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -373,6 +373,7 @@
AutoCallerClear acc;
AudioPolicyInterface::output_type_t outputType;
bool isSpatialized = false;
+ bool isBitPerfect = false;
status_t result = mAudioPolicyManager->getOutputForAttr(&attr, &output, session,
&stream,
attributionSource,
@@ -380,7 +381,8 @@
&flags, &selectedDeviceId, &portId,
&secondaryOutputs,
&outputType,
- &isSpatialized);
+ &isSpatialized,
+ &isBitPerfect);
// FIXME: Introduce a way to check for the the telephony device before opening the output
if (result == NO_ERROR) {
@@ -432,6 +434,7 @@
convertContainer<std::vector<int32_t>>(secondaryOutputs,
legacy2aidl_audio_io_handle_t_int32_t));
_aidl_return->isSpatialized = isSpatialized;
+ _aidl_return->isBitPerfect = isBitPerfect;
} else {
_aidl_return->configBase.format = VALUE_OR_RETURN_BINDER_STATUS(
legacy2aidl_audio_format_t_AudioFormatDescription(config.format));
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 4ad6d7a..a2326b3 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -258,13 +258,14 @@
*portId = AUDIO_PORT_HANDLE_NONE;
AudioPolicyInterface::output_type_t outputType;
bool isSpatialized;
+ bool isBitPerfect;
// TODO b/182392769: use attribution source util
AttributionSourceState attributionSource = AttributionSourceState();
attributionSource.uid = 0;
attributionSource.token = sp<BBinder>::make();
ASSERT_EQ(OK, mManager->getOutputForAttr(
&attr, output, session, &stream, attributionSource, &config, &flags,
- selectedDeviceId, portId, {}, &outputType, &isSpatialized));
+ selectedDeviceId, portId, {}, &outputType, &isSpatialized, &isBitPerfect));
ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
ASSERT_NE(AUDIO_IO_HANDLE_NONE, *output);
}