Add master audio balance
Test: Change Balance through Settings, play audio
Bug: 28390736
Co-author: Ed Savage-Jones <edward.savage-jones@sony.com>
Change-Id: I0169b436ccbaa5628584d9f4954dd7c76d021aae
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 0d6ef46..b8307ce 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -897,6 +897,40 @@
return NO_ERROR;
}
+status_t AudioFlinger::setMasterBalance(float balance)
+{
+ status_t ret = initCheck();
+ if (ret != NO_ERROR) {
+ return ret;
+ }
+
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ // check range
+ if (isnan(balance) || fabs(balance) > 1.f) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock _l(mLock);
+
+ // short cut.
+ if (mMasterBalance == balance) return NO_ERROR;
+
+ mMasterBalance = balance;
+
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
+ continue;
+ }
+ mPlaybackThreads.valueAt(i)->setMasterBalance(balance);
+ }
+
+ return NO_ERROR;
+}
+
status_t AudioFlinger::setMode(audio_mode_t mode)
{
status_t ret = initCheck();
@@ -1036,6 +1070,13 @@
return masterVolume_l();
}
+status_t AudioFlinger::getMasterBalance(float *balance) const
+{
+ Mutex::Autolock _l(mLock);
+ *balance = getMasterBalance_l();
+ return NO_ERROR; // if called through binder, may return a transactional error
+}
+
bool AudioFlinger::masterMute() const
{
Mutex::Autolock _l(mLock);
@@ -1047,6 +1088,11 @@
return mMasterVolume;
}
+float AudioFlinger::getMasterBalance_l() const
+{
+ return mMasterBalance;
+}
+
bool AudioFlinger::masterMute_l() const
{
return mMasterMute;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index c1169d2..7e3a775 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -137,6 +137,10 @@
virtual float masterVolume() const;
virtual bool masterMute() const;
+ // Balance value must be within -1.f (left only) to 1.f (right only) inclusive.
+ status_t setMasterBalance(float balance) override;
+ status_t getMasterBalance(float *balance) const override;
+
virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output);
virtual status_t setStreamMute(audio_stream_type_t stream, bool muted);
@@ -776,6 +780,7 @@
// member variables below are protected by mLock
float mMasterVolume;
bool mMasterMute;
+ float mMasterBalance = 0.f;
// end of variables protected by mLock
DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
@@ -793,6 +798,7 @@
Vector<AudioSessionRef*> mAudioSessionRefs;
float masterVolume_l() const;
+ float getMasterBalance_l() const;
bool masterMute_l() const;
audio_module_handle_t loadHwModule_l(const char *name);
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index f328577..7c9e15a 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -60,7 +60,6 @@
mSinkChannelCount(FCC_2),
mMixerBuffer(NULL),
mMixerBufferSize(0),
- mMixerBufferFormat(AUDIO_FORMAT_PCM_16_BIT),
mMixerBufferState(UNDEFINED),
mFormat(Format_Invalid),
mSampleRate(0),
@@ -161,6 +160,7 @@
mOutputSink = current->mOutputSink;
mOutputSinkGen = current->mOutputSinkGen;
mSinkChannelMask = current->mSinkChannelMask;
+ mBalance.setChannelMask(mSinkChannelMask);
if (mOutputSink == NULL) {
mFormat = Format_Invalid;
mSampleRate = 0;
@@ -191,10 +191,6 @@
free(mSinkBuffer);
mSinkBuffer = NULL;
if (frameCount > 0 && mSampleRate > 0) {
- // The mixer produces either 16 bit PCM or float output, select
- // float output if the HAL supports higher than 16 bit precision.
- mMixerBufferFormat = mFormat.mFormat == AUDIO_FORMAT_PCM_16_BIT ?
- AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_FLOAT;
// FIXME new may block for unbounded time at internal mutex of the heap
// implementation; it would be better to have normal mixer allocate for us
// to avoid blocking here and to prevent possible priority inversion
@@ -471,6 +467,12 @@
mono_blend(mMixerBuffer, mMixerBufferFormat, Format_channelCount(mFormat), frameCount,
true /*limit*/);
}
+
+ // Balance must take effect after mono conversion.
+ // mBalance detects zero balance within the class for speed (not needed here).
+ mBalance.setBalance(mMasterBalance.load());
+ mBalance.process((float *)mMixerBuffer, frameCount);
+
// prepare the buffer used to write to sink
void *buffer = mSinkBuffer != NULL ? mSinkBuffer : mMixerBuffer;
if (mFormat.mFormat != mMixerBufferFormat) { // sink format not the same as mixer format
diff --git a/services/audioflinger/FastMixer.h b/services/audioflinger/FastMixer.h
index 1d332e0..c31d476 100644
--- a/services/audioflinger/FastMixer.h
+++ b/services/audioflinger/FastMixer.h
@@ -18,6 +18,7 @@
#define ANDROID_AUDIO_FAST_MIXER_H
#include <atomic>
+#include <audio_utils/Balance.h>
#include "FastThread.h"
#include "StateQueue.h"
#include "FastMixerState.h"
@@ -41,6 +42,8 @@
FastMixerStateQueue* sq();
virtual void setMasterMono(bool mono) { mMasterMono.store(mono); /* memory_order_seq_cst */ }
+ virtual void setMasterBalance(float balance) { mMasterBalance.store(balance); }
+ virtual float getMasterBalance() const { return mMasterBalance.load(); }
virtual void setBoottimeOffset(int64_t boottimeOffset) {
mBoottimeOffset.store(boottimeOffset); /* memory_order_seq_cst */
}
@@ -74,7 +77,7 @@
audio_channel_mask_t mSinkChannelMask;
void* mMixerBuffer; // mixer output buffer.
size_t mMixerBufferSize;
- audio_format_t mMixerBufferFormat; // mixer output format: AUDIO_FORMAT_PCM_(16_BIT|FLOAT).
+ static constexpr audio_format_t mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT;
uint32_t mAudioChannelCount; // audio channel count, excludes haptic channels.
@@ -89,8 +92,11 @@
ExtendedTimestamp mTimestamp;
int64_t mNativeFramesWrittenButNotPresented;
+ audio_utils::Balance mBalance;
+
// accessed without lock between multiple threads.
std::atomic_bool mMasterMono;
+ std::atomic<float> mMasterBalance{};
std::atomic_int_fast64_t mBoottimeOffset;
const audio_io_handle_t mThreadIoHandle; // parent thread id for debugging purposes
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 9f838a3..1790f11 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -38,6 +38,7 @@
#include <private/media/AudioTrackShared.h>
#include <private/android_filesystem_config.h>
+#include <audio_utils/Balance.h>
#include <audio_utils/channels.h>
#include <audio_utils/mono_blend.h>
#include <audio_utils/primitives.h>
@@ -2271,6 +2272,11 @@
}
}
+void AudioFlinger::PlaybackThread::setMasterBalance(float balance)
+{
+ mMasterBalance.store(balance);
+}
+
void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
if (isDuplicating()) {
@@ -2523,6 +2529,7 @@
mChannelMask);
}
mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
+ mBalance.setChannelMask(mChannelMask);
// Get actual HAL format.
status_t result = mOutput->stream->getFormat(&mHALFormat);
@@ -2642,7 +2649,7 @@
free(mMixerBuffer);
mMixerBuffer = NULL;
if (mMixerBufferEnabled) {
- mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT.
+ mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // no longer valid: AUDIO_FORMAT_PCM_16_BIT.
mMixerBufferSize = mNormalFrameCount * mChannelCount
* audio_bytes_per_sample(mMixerBufferFormat);
(void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
@@ -3531,6 +3538,14 @@
true /*limit*/);
}
+ if (!hasFastMixer()) {
+ // Balance must take effect after mono conversion.
+ // We do it here if there is no FastMixer.
+ // mBalance detects zero balance within the class for speed (not needed here).
+ mBalance.setBalance(mMasterBalance.load());
+ mBalance.process((float *)mMixerBuffer, mNormalFrameCount);
+ }
+
memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
mNormalFrameCount * (mChannelCount + mHapticChannelCount));
@@ -3585,6 +3600,14 @@
true /*limit*/);
}
+ if (!hasFastMixer()) {
+ // Balance must take effect after mono conversion.
+ // We do it here if there is no FastMixer.
+ // mBalance detects zero balance within the class for speed (not needed here).
+ mBalance.setBalance(mMasterBalance.load());
+ mBalance.process((float *)mEffectBuffer, mNormalFrameCount);
+ }
+
memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
mNormalFrameCount * (mChannelCount + mHapticChannelCount));
// The sample data is partially interleaved when haptic channels exist,
@@ -3985,6 +4008,7 @@
// mPipeSink below
// mNormalSink below
{
+ setMasterBalance(audioFlinger->getMasterBalance_l());
ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%#x, mFrameSize=%zu, "
"mFrameCount=%zu, mNormalFrameCount=%zu",
@@ -5266,6 +5290,9 @@
dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
dprintf(fd, " AudioMixer tracks: %s\n", mAudioMixer->trackNames().c_str());
dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off");
+ dprintf(fd, " Master balance: %f (%s)\n", mMasterBalance.load(),
+ (hasFastMixer() ? std::to_string(mFastMixer->getMasterBalance())
+ : mBalance.toString()).c_str());
const double latencyMs = mTimestamp.getOutputServerLatencyMs(mSampleRate);
if (latencyMs != 0.) {
dprintf(fd, " NormalMixer latency ms: %.2lf\n", latencyMs);
@@ -5333,12 +5360,30 @@
ThreadBase::type_t type, bool systemReady)
: PlaybackThread(audioFlinger, output, id, device, type, systemReady)
{
+ setMasterBalance(audioFlinger->getMasterBalance_l());
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
+void AudioFlinger::DirectOutputThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+ PlaybackThread::dumpInternals(fd, args);
+ dprintf(fd, " Master balance: %f Left: %f Right: %f\n",
+ mMasterBalance.load(), mMasterBalanceLeft, mMasterBalanceRight);
+}
+
+void AudioFlinger::DirectOutputThread::setMasterBalance(float balance)
+{
+ Mutex::Autolock _l(mLock);
+ if (mMasterBalance != balance) {
+ mMasterBalance.store(balance);
+ mBalance.computeStereoBalance(balance, &mMasterBalanceLeft, &mMasterBalanceRight);
+ broadcast_l();
+ }
+}
+
void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
{
float left, right;
@@ -5362,12 +5407,12 @@
if (left > GAIN_FLOAT_UNITY) {
left = GAIN_FLOAT_UNITY;
}
- left *= v;
+ left *= v * mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
right = float_from_gain(gain_minifloat_unpack_right(vlr));
if (right > GAIN_FLOAT_UNITY) {
right = GAIN_FLOAT_UNITY;
}
- right *= v;
+ right *= v * mMasterBalanceRight;
}
if (lastTrack) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 8b8222c..1131b26 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -733,6 +733,7 @@
// VolumeInterface
virtual void setMasterVolume(float value);
+ virtual void setMasterBalance(float balance);
virtual void setMasterMute(bool muted);
virtual void setStreamVolume(audio_stream_type_t stream, float value);
virtual void setStreamMute(audio_stream_type_t stream, bool muted);
@@ -1027,6 +1028,8 @@
AudioStreamOut *mOutput;
float mMasterVolume;
+ std::atomic<float> mMasterBalance{};
+ audio_utils::Balance mBalance;
nsecs_t mLastWriteTime;
int mNumWrites;
int mNumDelayedWrites;
@@ -1199,6 +1202,13 @@
// Blending with limiter is not idempotent,
// and blending without limiter is idempotent but inefficient to do twice.
virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); }
+
+ void setMasterBalance(float balance) override {
+ mMasterBalance.store(balance);
+ if (hasFastMixer()) {
+ mFastMixer->setMasterBalance(balance);
+ }
+ }
};
class DirectOutputThread : public PlaybackThread {
@@ -1216,8 +1226,13 @@
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status);
+
+ void dumpInternals(int fd, const Vector<String16>& args) override;
+
virtual void flushHw_l();
+ void setMasterBalance(float balance) override;
+
protected:
virtual uint32_t activeSleepTimeUs() const;
virtual uint32_t idleSleepTimeUs() const;
@@ -1245,6 +1260,10 @@
wp<Track> mPreviousTrack; // used to detect track switch
+ // This must be initialized for initial condition of mMasterBalance = 0 (disabled).
+ float mMasterBalanceLeft = 1.f;
+ float mMasterBalanceRight = 1.f;
+
public:
virtual bool hasFastMixer() const { return false; }