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-rw-r--r--services/audioflinger/AudioFlinger.cpp6
-rw-r--r--services/audioflinger/AudioFlinger.h3
-rw-r--r--services/audioflinger/AudioMixer.cpp19
-rw-r--r--services/audioflinger/AudioMixer.h4
-rw-r--r--services/audioflinger/Effects.h1
-rw-r--r--services/audioflinger/FastMixer.cpp18
-rw-r--r--services/audioflinger/FastMixerState.h6
-rw-r--r--services/audioflinger/PlaybackTracks.h2
-rw-r--r--services/audioflinger/Threads.cpp57
-rw-r--r--services/audioflinger/Tracks.cpp35
10 files changed, 84 insertions, 67 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 11170c2..45e17f8 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1837,7 +1837,8 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
if (status == BAD_VALUE &&
reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
(config.sample_rate <= 2 * reqSamplingRate) &&
- (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
+ (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) &&
+ (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) {
// FIXME describe the change proposed by HAL (save old values so we can log them here)
ALOGV("openInput() reopening with proposed sampling rate and channel mask");
inStream = NULL;
@@ -1857,7 +1858,8 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
TEE_SINK_OLD, // copy input using an existing pipe
} kind;
NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
- popcount(inStream->common.get_channels(&inStream->common)));
+ audio_channel_count_from_in_mask(
+ inStream->common.get_channels(&inStream->common)));
if (!mTeeSinkInputEnabled) {
kind = TEE_SINK_NO;
} else if (!Format_isValid(format)) {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index d69d6a2..d2ded9a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -82,9 +82,6 @@ class ServerProxy;
static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
-#define MAX_GAIN 4096.0f
-#define MAX_GAIN_INT 0x1000
-
#define INCLUDING_FROM_AUDIOFLINGER_H
class AudioFlinger :
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 2d67efb..805eaa4 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -159,7 +159,6 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
if (names != 0) {
int n = __builtin_ctz(names);
ALOGV("add track (%d)", n);
- mTrackNames |= 1 << n;
// assume default parameters for the track, except where noted below
track_t* t = &mState.tracks[n];
t->needs = 0;
@@ -175,10 +174,10 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
// no initialization needed
// t->prevAuxLevel
// t->frameCount
- t->channelCount = 2;
+ t->channelCount = audio_channel_count_from_out_mask(channelMask);
t->enabled = false;
t->format = 16;
- t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
+ t->channelMask = channelMask;
t->sessionId = sessionId;
// setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
t->bufferProvider = NULL;
@@ -196,12 +195,14 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
- if (status == OK) {
- return TRACK0 + n;
+ if (status != OK) {
+ ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
+ return -1;
}
- ALOGE("AudioMixer::getTrackName(0x%x) failed, error preparing track for downmix",
- channelMask);
+ mTrackNames |= 1 << n;
+ return TRACK0 + n;
}
+ ALOGE("AudioMixer::getTrackName out of available tracks");
return -1;
}
@@ -215,7 +216,7 @@ void AudioMixer::invalidateState(uint32_t mask)
status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask)
{
- uint32_t channelCount = popcount(mask);
+ uint32_t channelCount = audio_channel_count_from_out_mask(mask);
ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
status_t status = OK;
if (channelCount > MAX_NUM_CHANNELS) {
@@ -410,7 +411,7 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
audio_channel_mask_t mask =
static_cast<audio_channel_mask_t>(reinterpret_cast<uintptr_t>(value));
if (track.channelMask != mask) {
- uint32_t channelCount = popcount(mask);
+ uint32_t channelCount = audio_channel_count_from_out_mask(mask);
ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
track.channelMask = mask;
track.channelCount = channelCount;
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index e5e120c..09e63a6 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -30,6 +30,9 @@
#include <system/audio.h>
#include <media/nbaio/NBLog.h>
+// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN
+
namespace android {
// ----------------------------------------------------------------------------
@@ -91,6 +94,7 @@ public:
REMOVE = 0x4102, // Remove the sample rate converter on this track name;
// the track is restored to the mix sample rate.
// for target RAMP_VOLUME and VOLUME (8 channels max)
+ // FIXME use float for these 3 to improve the dynamic range
VOLUME0 = 0x4200,
VOLUME1 = 0x4201,
AUXLEVEL = 0x4210,
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index ccc4825..4170fd4 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -270,6 +270,7 @@ public:
sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
sp<EffectModule> getEffectFromId_l(int id);
sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
+ // FIXME use float to improve the dynamic range
bool setVolume_l(uint32_t *left, uint32_t *right);
void setDevice_l(audio_devices_t device);
void setMode_l(audio_mode_t mode);
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 5cb42cc..1caed11 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -224,17 +224,13 @@ void FastMixer::onStateChange()
AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;
ALOG_ASSERT(bufferProvider != NULL && fastTrackNames[i] == -1);
if (mixer != NULL) {
- // calling getTrackName with default channel mask and a random invalid
- // sessionId (no effects here)
- name = mixer->getTrackName(AUDIO_CHANNEL_OUT_STEREO, -555);
+ name = mixer->getTrackName(fastTrack->mChannelMask, AUDIO_SESSION_OUTPUT_MIX);
ALOG_ASSERT(name >= 0);
fastTrackNames[i] = name;
mixer->setBufferProvider(name, bufferProvider);
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(void *) mixBuffer);
// newly allocated track names default to full scale volume
- mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
- (void *)(uintptr_t)fastTrack->mChannelMask);
mixer->enable(name);
}
generations[i] = fastTrack->mGeneration;
@@ -257,9 +253,9 @@ void FastMixer::onStateChange()
mixer->setBufferProvider(name, bufferProvider);
if (fastTrack->mVolumeProvider == NULL) {
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
- (void *)0x1000);
+ (void *) MAX_GAIN_INT);
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
- (void *)0x1000);
+ (void *) MAX_GAIN_INT);
}
mixer->setParameter(name, AudioMixer::RESAMPLE,
AudioMixer::REMOVE, NULL);
@@ -312,11 +308,13 @@ void FastMixer::onWork()
int name = fastTrackNames[i];
ALOG_ASSERT(name >= 0);
if (fastTrack->mVolumeProvider != NULL) {
- uint32_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
+ gain_minifloat_packed_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
- (void *)(uintptr_t)(vlr & 0xFFFF));
+ (void *) (uintptr_t)
+ (float_from_gain(gain_minifloat_unpack_left(vlr)) * MAX_GAIN_INT));
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
- (void *)(uintptr_t)(vlr >> 16));
+ (void *) (uintptr_t)
+ (float_from_gain(gain_minifloat_unpack_right(vlr)) * MAX_GAIN_INT));
}
// FIXME The current implementation of framesReady() for fast tracks
// takes a tryLock, which can block
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index be1a376..e388fb3 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -17,6 +17,7 @@
#ifndef ANDROID_AUDIO_FAST_MIXER_STATE_H
#define ANDROID_AUDIO_FAST_MIXER_STATE_H
+#include <audio_utils/minifloat.h>
#include <system/audio.h>
#include <media/ExtendedAudioBufferProvider.h>
#include <media/nbaio/NBAIO.h>
@@ -29,9 +30,8 @@ struct FastMixerDumpState;
class VolumeProvider {
public:
- // Return the track volume in U4_12 format: left in lower half, right in upper half. The
- // provider implementation is responsible for validating that the return value is in range.
- virtual uint32_t getVolumeLR() = 0;
+ // The provider implementation is responsible for validating that the return value is in range.
+ virtual gain_minifloat_packed_t getVolumeLR() = 0;
protected:
VolumeProvider() { }
virtual ~VolumeProvider() { }
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 08b1728..6f1f293 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -65,7 +65,7 @@ public:
void signal();
// implement FastMixerState::VolumeProvider interface
- virtual uint32_t getVolumeLR();
+ virtual gain_minifloat_packed_t getVolumeLR();
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 2d4e025..ce08ff1 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -35,6 +35,7 @@
#include <audio_effects/effect_aec.h>
#include <audio_utils/primitives.h>
#include <audio_utils/format.h>
+#include <audio_utils/minifloat.h>
// NBAIO implementations
#include <media/nbaio/AudioStreamOutSink.h>
@@ -1730,7 +1731,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; "
"must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask);
}
- mChannelCount = popcount(mChannelMask);
+ mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
if (!audio_is_valid_format(mFormat)) {
LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
@@ -3255,21 +3256,23 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
- uint32_t vlr = proxy->getVolumeLR();
- vl = vlr & 0xFFFF;
- vr = vlr >> 16;
+ gain_minifloat_packed_t vlr = proxy->getVolumeLR();
+ float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
+ float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
// track volumes come from shared memory, so can't be trusted and must be clamped
- if (vl > MAX_GAIN_INT) {
- ALOGV("Track left volume out of range: %04X", vl);
- vl = MAX_GAIN_INT;
+ if (vlf > GAIN_FLOAT_UNITY) {
+ ALOGV("Track left volume out of range: %.3g", vlf);
+ vlf = GAIN_FLOAT_UNITY;
}
- if (vr > MAX_GAIN_INT) {
- ALOGV("Track right volume out of range: %04X", vr);
- vr = MAX_GAIN_INT;
+ if (vrf > GAIN_FLOAT_UNITY) {
+ ALOGV("Track right volume out of range: %.3g", vrf);
+ vrf = GAIN_FLOAT_UNITY;
}
// now apply the master volume and stream type volume
- vl = (uint32_t)(v * vl) << 12;
- vr = (uint32_t)(v * vr) << 12;
+ // FIXME we're losing the wonderful dynamic range in the minifloat representation
+ float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT);
+ vl = (uint32_t) (v8_24 * vlf);
+ vr = (uint32_t) (v8_24 * vrf);
// assuming master volume and stream type volume each go up to 1.0,
// vl and vr are now in 8.24 format
@@ -3296,6 +3299,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
track->mHasVolumeController = false;
}
+ // FIXME Use float
// Convert volumes from 8.24 to 4.12 format
// This additional clamping is needed in case chain->setVolume_l() overshot
vl = (vl + (1 << 11)) >> 12;
@@ -3750,13 +3754,17 @@ void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTr
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
- uint32_t vlr = proxy->getVolumeLR();
- float v_clamped = v * (vlr & 0xFFFF);
- if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
- left = v_clamped/MAX_GAIN;
- v_clamped = v * (vlr >> 16);
- if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
- right = v_clamped/MAX_GAIN;
+ gain_minifloat_packed_t vlr = proxy->getVolumeLR();
+ left = float_from_gain(gain_minifloat_unpack_left(vlr));
+ if (left > GAIN_FLOAT_UNITY) {
+ left = GAIN_FLOAT_UNITY;
+ }
+ left *= v;
+ right = float_from_gain(gain_minifloat_unpack_right(vlr));
+ if (right > GAIN_FLOAT_UNITY) {
+ right = GAIN_FLOAT_UNITY;
+ }
+ right *= v;
}
if (lastTrack) {
@@ -4152,7 +4160,10 @@ void AudioFlinger::OffloadThread::threadLoop_exit()
mMixerStatus = MIXER_DRAIN_ALL;
threadLoop_drain();
}
- mCallbackThread->exit();
+ if (mUseAsyncWrite) {
+ ALOG_ASSERT(mCallbackThread != 0);
+ mCallbackThread->exit();
+ }
PlaybackThread::threadLoop_exit();
}
@@ -5564,8 +5575,8 @@ bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValueP
reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
(mInput->stream->common.get_sample_rate(&mInput->stream->common)
<= (2 * samplingRate)) &&
- popcount(mInput->stream->common.get_channels(&mInput->stream->common))
- <= FCC_2 &&
+ audio_channel_count_from_in_mask(
+ mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
(channelMask == AUDIO_CHANNEL_IN_MONO ||
channelMask == AUDIO_CHANNEL_IN_STEREO)) {
status = NO_ERROR;
@@ -5619,7 +5630,7 @@ void AudioFlinger::RecordThread::readInputParameters_l()
{
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
- mChannelCount = popcount(mChannelMask);
+ mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6dc7f30..de1782d 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -34,6 +34,7 @@
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
+#include <audio_utils/minifloat.h>
// ----------------------------------------------------------------------------
@@ -81,7 +82,9 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
mSampleRate(sampleRate),
mFormat(format),
mChannelMask(channelMask),
- mChannelCount(popcount(channelMask)),
+ mChannelCount(isOut ?
+ audio_channel_count_from_out_mask(channelMask) :
+ audio_channel_count_from_in_mask(channelMask)),
mFrameSize(audio_is_linear_pcm(format) ?
mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
mFrameCount(frameCount),
@@ -459,7 +462,7 @@ void AudioFlinger::PlaybackThread::Track::destroy()
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
{
- uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
+ gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
if (isFastTrack()) {
sprintf(buffer, " F %2d", mFastIndex);
} else if (mName >= AudioMixer::TRACK0) {
@@ -532,8 +535,8 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool a
stateChar,
mFillingUpStatus,
mAudioTrackServerProxy->getSampleRate(),
- 20.0 * log10((vlr & 0xFFFF) / 4096.0),
- 20.0 * log10((vlr >> 16) / 4096.0),
+ 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))),
+ 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
mCblk->mServer,
mMainBuffer,
mAuxBuffer,
@@ -959,27 +962,27 @@ void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_
// implement VolumeBufferProvider interface
-uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
+gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
{
// called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
- uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
- uint32_t vl = vlr & 0xFFFF;
- uint32_t vr = vlr >> 16;
+ gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
+ float vl = float_from_gain(gain_minifloat_unpack_left(vlr));
+ float vr = float_from_gain(gain_minifloat_unpack_right(vlr));
// track volumes come from shared memory, so can't be trusted and must be clamped
- if (vl > MAX_GAIN_INT) {
- vl = MAX_GAIN_INT;
+ if (vl > GAIN_FLOAT_UNITY) {
+ vl = GAIN_FLOAT_UNITY;
}
- if (vr > MAX_GAIN_INT) {
- vr = MAX_GAIN_INT;
+ if (vr > GAIN_FLOAT_UNITY) {
+ vr = GAIN_FLOAT_UNITY;
}
// now apply the cached master volume and stream type volume;
// this is trusted but lacks any synchronization or barrier so may be stale
float v = mCachedVolume;
vl *= v;
vr *= v;
- // re-combine into U4.16
- vlr = (vr << 16) | (vl & 0xFFFF);
+ // re-combine into packed minifloat
+ vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr));
// FIXME look at mute, pause, and stop flags
return vlr;
}
@@ -1590,7 +1593,7 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
// since client and server are in the same process,
// the buffer has the same virtual address on both sides
mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
- mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
+ mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
mClientProxy->setSendLevel(0.0);
mClientProxy->setSampleRate(sampleRate);
mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
@@ -1849,7 +1852,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
- uint32_t channelCount = popcount(channelMask);
+ uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
// FIXME I don't understand either of the channel count checks
if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
channelCount <= FCC_2) {
generated by cgit v1.1 at 2025-08-04 19:30:52 +0000