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/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+ mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // no longer valid: AUDIO_FORMAT_PCM_16_BIT.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) {