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-rw-r--r--include/private/media/AudioTrackShared.h14
-rw-r--r--media/libmedia/AudioTrack.cpp13
-rw-r--r--media/libmedia/AudioTrackShared.cpp2
-rw-r--r--media/mediaserver/Android.mk3
-rw-r--r--services/audioflinger/AudioFlinger.h3
-rw-r--r--services/audioflinger/AudioMixer.h4
-rw-r--r--services/audioflinger/Effects.h1
-rw-r--r--services/audioflinger/FastMixer.cpp12
-rw-r--r--services/audioflinger/FastMixerState.h6
-rw-r--r--services/audioflinger/PlaybackTracks.h2
-rw-r--r--services/audioflinger/Threads.cpp44
-rw-r--r--services/audioflinger/Tracks.cpp29
12 files changed, 74 insertions, 59 deletions
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 3901e79..5116d1e 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -20,6 +20,7 @@
#include <stdint.h>
#include <sys/types.h>
+#include <audio_utils/minifloat.h>
#include <utils/threads.h>
#include <utils/Log.h>
#include <utils/RefBase.h>
@@ -110,11 +111,8 @@ private:
// force to 32-bit. The client and server may have different typedefs for size_t.
uint32_t mMinimum; // server wakes up client if available >= mMinimum
- // Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
- // Left channel is in [0:15], right channel is in [16:31].
- // Always read and write the combined pair atomically.
- // For AudioTrack only, not used by AudioRecord.
- uint32_t mVolumeLR;
+ // Stereo gains for AudioTrack only, not used by AudioRecord.
+ gain_minifloat_packed_t mVolumeLR;
uint32_t mSampleRate; // AudioTrack only: client's requested sample rate in Hz
// or 0 == default. Write-only client, read-only server.
@@ -285,8 +283,8 @@ public:
mCblk->mSendLevel = uint16_t(sendLevel * 0x1000);
}
- // caller must limit to 0 <= volumeLR <= 0x10001000
- void setVolumeLR(uint32_t volumeLR) {
+ // set stereo gains
+ void setVolumeLR(gain_minifloat_packed_t volumeLR) {
mCblk->mVolumeLR = volumeLR;
}
@@ -405,7 +403,7 @@ public:
// return value of these methods must be validated by the caller
uint32_t getSampleRate() const { return mCblk->mSampleRate; }
uint16_t getSendLevel_U4_12() const { return mCblk->mSendLevel; }
- uint32_t getVolumeLR() const { return mCblk->mVolumeLR; }
+ gain_minifloat_packed_t getVolumeLR() const { return mCblk->mVolumeLR; }
// estimated total number of filled frames available to server to read,
// which may include non-contiguous frames
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index aaaa3f1..120b28e 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -19,6 +19,7 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioTrack"
+#include <math.h>
#include <sys/resource.h>
#include <audio_utils/primitives.h>
#include <binder/IPCThreadState.h>
@@ -566,7 +567,9 @@ void AudioTrack::pause()
status_t AudioTrack::setVolume(float left, float right)
{
- if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
+ // This duplicates a test by AudioTrack JNI, but that is not the only caller
+ if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
+ isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
return BAD_VALUE;
}
@@ -574,7 +577,7 @@ status_t AudioTrack::setVolume(float left, float right)
mVolume[AUDIO_INTERLEAVE_LEFT] = left;
mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
- mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
+ mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
if (isOffloaded_l()) {
mAudioTrack->signal();
@@ -589,7 +592,8 @@ status_t AudioTrack::setVolume(float volume)
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
- if (level < 0.0f || level > 1.0f) {
+ // This duplicates a test by AudioTrack JNI, but that is not the only caller
+ if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
return BAD_VALUE;
}
@@ -1137,8 +1141,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
mProxy = mStaticProxy;
}
- mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[AUDIO_INTERLEAVE_RIGHT] * 0x1000)) << 16) |
- uint16_t(mVolume[AUDIO_INTERLEAVE_LEFT] * 0x1000));
+ mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
mProxy->setSendLevel(mSendLevel);
mProxy->setSampleRate(mSampleRate);
mProxy->setEpoch(epoch);
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 323b675..27a3718 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -27,7 +27,7 @@ namespace android {
audio_track_cblk_t::audio_track_cblk_t()
: mServer(0), mFutex(0), mMinimum(0),
- mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mFlags(0)
+ mVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY), mSampleRate(0), mSendLevel(0), mFlags(0)
{
memset(&u, 0, sizeof(u));
}
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index d3e546a..5bc3f2f 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -35,7 +35,8 @@ LOCAL_C_INCLUDES := \
frameworks/av/services/medialog \
frameworks/av/services/audioflinger \
frameworks/av/services/audiopolicy \
- frameworks/av/services/camera/libcameraservice
+ frameworks/av/services/camera/libcameraservice \
+ $(call include-path-for, audio-utils)
LOCAL_MODULE:= mediaserver
LOCAL_32_BIT_ONLY := true
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.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..42ba791 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -257,9 +257,9 @@ 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 +312,13 @@ 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..5cb6a09 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>
@@ -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) {
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6dc7f30..08687a2 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>
// ----------------------------------------------------------------------------
@@ -459,7 +460,7 @@ 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 +533,8 @@ 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 +960,27 @@ 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 +1591,7 @@ 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,
generated by cgit v1.1 at 2024-05-04 11:54:46 +0000