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author | Glenn Kasten <gkasten@google.com> | 2012-01-27 15:26:23 -0800 |
---|---|---|
committer | Glenn Kasten <gkasten@google.com> | 2012-02-17 09:41:56 -0800 |
commit | 4f22c05eacee669d22251d739fb4c2d2be17501c (patch) | |
tree | 60e25d7d4d4cff16094dc742aa5ed042997aaade /services | |
parent | e13ac73a382465f46a415ef6bdf5c9d7c3700c19 (diff) | |
download | frameworks_base-4f22c05eacee669d22251d739fb4c2d2be17501c.zip frameworks_base-4f22c05eacee669d22251d739fb4c2d2be17501c.tar.gz frameworks_base-4f22c05eacee669d22251d739fb4c2d2be17501c.tar.bz2 |
Remove bit fields to improve performance
uint16_t enabled is (mostly) changed to bool in a separate CL
Change-Id: Ied9f8c034b2479cee9a8778cee7b8ff92ae75b7b
Diffstat (limited to 'services')
-rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 22 | ||||
-rw-r--r-- | services/audioflinger/AudioMixer.cpp | 3 | ||||
-rw-r--r-- | services/audioflinger/AudioMixer.h | 28 |
3 files changed, 32 insertions, 21 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 0292bc3..2e2834c 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -2295,7 +2295,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac uint16_t sendLevel = cblk->getSendLevel_U4_12(); // send level comes from shared memory and so may be corrupt - if (sendLevel >= MAX_GAIN_INT) { + if (sendLevel > MAX_GAIN_INT) { ALOGV("Track send level out of range: %04X", sendLevel); sendLevel = MAX_GAIN_INT; } @@ -2316,25 +2316,21 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac } // Convert volumes from 8.24 to 4.12 format - int16_t left, right, aux; // This additional clamping is needed in case chain->setVolume_l() overshot - uint32_t v_clamped = (vl + (1 << 11)) >> 12; - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - left = int16_t(v_clamped); - v_clamped = (vr + (1 << 11)) >> 12; - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - right = int16_t(v_clamped); + vl = (vl + (1 << 11)) >> 12; + if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; + vr = (vr + (1 << 11)) >> 12; + if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; - if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; - aux = int16_t(va); + if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - // XXX: these things DON'T need to be done each time mAudioMixer->setBufferProvider(name, track); mAudioMixer->enable(name); - mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); - mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); - mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); + mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); + mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); + mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); mAudioMixer->setParameter( name, AudioMixer::TRACK, diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index f3bf953..020d62a 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -257,6 +257,7 @@ void AudioMixer::setParameter(int name, int target, int param, void *value) } break; case AUXLEVEL: + //assert(0 <= valueInt && valueInt <= MAX_GAIN_INT); if (track.auxLevel != valueInt) { ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt); track.prevAuxLevel = track.auxLevel << 16; @@ -565,7 +566,7 @@ void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32 const int16_t vr = t->volume[1]; if (CC_UNLIKELY(aux != NULL)) { - const int16_t va = (int16_t)t->auxLevel; + const int16_t va = t->auxLevel; do { int16_t l = (int16_t)(*temp++ >> 12); int16_t r = (int16_t)(*temp++ >> 12); diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h index c3174ae..b210212 100644 --- a/services/audioflinger/AudioMixer.h +++ b/services/audioflinger/AudioMixer.h @@ -127,32 +127,46 @@ private: int32_t prevVolume[MAX_NUM_CHANNELS]; + // 16-byte boundary + int32_t volumeInc[MAX_NUM_CHANNELS]; - int32_t auxLevel; int32_t auxInc; int32_t prevAuxLevel; + // 16-byte boundary + + int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance uint16_t frameCount; - uint8_t channelCount : 4; - uint8_t enabled : 1; - uint8_t reserved0 : 3; - uint8_t format; - uint32_t channelMask; + uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK) + uint8_t format; // always 16 + uint16_t enabled; // actually bool + uint32_t channelMask; // currently under-used AudioBufferProvider* bufferProvider; - mutable AudioBufferProvider::Buffer buffer; + + // 16-byte boundary + + mutable AudioBufferProvider::Buffer buffer; // 8 bytes hook_t hook; const void* in; // current location in buffer + // 16-byte boundary + AudioResampler* resampler; uint32_t sampleRate; int32_t* mainBuffer; int32_t* auxBuffer; + // 16-byte boundary + uint64_t localTimeFreq; + int64_t padding; + + // 16-byte boundary + bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); bool doesResample() const { return resampler != NULL; } void resetResampler() { if (resampler != NULL) resampler->reset(); } |