diff options
author | Glenn Kasten <gkasten@google.com> | 2013-02-15 23:55:04 +0000 |
---|---|---|
committer | Android (Google) Code Review <android-gerrit@google.com> | 2013-02-15 23:55:04 +0000 |
commit | 7f5d335f7b4caecd0dfb8f1085f352f1d2da5d2e (patch) | |
tree | dda17a3e4149815397c7bef198e64bc81dee3552 /services/audioflinger/AudioMixer.cpp | |
parent | 32584a7d672864b20ab8b83a3cb23c1858e908b7 (diff) | |
download | frameworks_av-7f5d335f7b4caecd0dfb8f1085f352f1d2da5d2e.zip frameworks_av-7f5d335f7b4caecd0dfb8f1085f352f1d2da5d2e.tar.gz frameworks_av-7f5d335f7b4caecd0dfb8f1085f352f1d2da5d2e.tar.bz2 |
Revert "Temporary additional logging to investigate bug"
This reverts commit 32584a7d672864b20ab8b83a3cb23c1858e908b7
Change-Id: I9dc680578b955b1af462eeb7a49d61a0d45eb81b
Diffstat (limited to 'services/audioflinger/AudioMixer.cpp')
-rw-r--r-- | services/audioflinger/AudioMixer.cpp | 72 |
1 files changed, 4 insertions, 68 deletions
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index 2d7894d..08325ad 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -16,7 +16,7 @@ */ #define LOG_TAG "AudioMixer" -#define LOG_NDEBUG 0 +//#define LOG_NDEBUG 0 #include <stdint.h> #include <string.h> @@ -25,8 +25,6 @@ #include <utils/Errors.h> #include <utils/Log.h> -#undef ALOGV -#define ALOGV(a...) do { } while (0) #include <cutils/bitops.h> #include <cutils/compiler.h> @@ -100,7 +98,7 @@ effect_descriptor_t AudioMixer::dwnmFxDesc; AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) : mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1), - mSampleRate(sampleRate), mLog(&mDummyLog) + mSampleRate(sampleRate) { // AudioMixer is not yet capable of multi-channel beyond stereo COMPILE_TIME_ASSERT_FUNCTION_SCOPE(2 == MAX_NUM_CHANNELS); @@ -124,7 +122,6 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr mState.hook = process__nop; mState.outputTemp = NULL; mState.resampleTemp = NULL; - mState.mLog = &mDummyLog; // mState.reserved // FIXME Most of the following initialization is probably redundant since @@ -134,7 +131,6 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { t->resampler = NULL; t->downmixerBufferProvider = NULL; - t->magic = track_t::kMagic; t++; } @@ -173,12 +169,6 @@ AudioMixer::~AudioMixer() delete [] mState.resampleTemp; } -void AudioMixer::setLog(NBLog::Writer *log) -{ - mLog = log; - mState.mLog = log; -} - int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) { uint32_t names = (~mTrackNames) & mConfiguredNames; @@ -219,12 +209,9 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) t->mainBuffer = NULL; t->auxBuffer = NULL; t->downmixerBufferProvider = NULL; - t->fastIndex = -1; - // t->magic unchanged status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask); if (status == OK) { - mLog->logf("getTrackName %d", n); return TRACK0 + n; } ALOGE("AudioMixer::getTrackName(0x%x) failed, error preparing track for downmix", @@ -379,11 +366,9 @@ void AudioMixer::deleteTrackName(int name) { ALOGV("AudioMixer::deleteTrackName(%d)", name); name -= TRACK0; - mLog->logf("deleteTrackName %d", name); ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); ALOGV("deleteTrackName(%d)", name); track_t& track(mState.tracks[ name ]); - track.checkMagic(); if (track.enabled) { track.enabled = false; invalidateState(1<<name); @@ -402,10 +387,8 @@ void AudioMixer::enable(int name) name -= TRACK0; ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); track_t& track = mState.tracks[name]; - track.checkMagic(); if (!track.enabled) { - mLog->logf("enable %d", name); track.enabled = true; ALOGV("enable(%d)", name); invalidateState(1 << name); @@ -417,36 +400,19 @@ void AudioMixer::disable(int name) name -= TRACK0; ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); track_t& track = mState.tracks[name]; - track.checkMagic(); if (track.enabled) { - mLog->logf("disable %d", name); track.enabled = false; ALOGV("disable(%d)", name); invalidateState(1 << name); } } -bool AudioMixer::enabled(int name) -{ - name -= TRACK0; - ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); - track_t& track = mState.tracks[name]; - track.checkMagic(); -#if 0 - // can't do this because mState.enabledTracks is updated lazily - ALOG_ASSERT(track.enabled == ((mState.enabledTracks & (1 << name)) != 0)); -#endif - - return track.enabled; -} - void AudioMixer::setParameter(int name, int target, int param, void *value) { name -= TRACK0; ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); track_t& track = mState.tracks[name]; - track.checkMagic(); int valueInt = (int)value; int32_t *valueBuf = (int32_t *)value; @@ -489,9 +455,6 @@ void AudioMixer::setParameter(int name, int target, int param, void *value) // for a specific track? or per mixer? /* case DOWNMIX_TYPE: break */ - case FAST_INDEX: - track.fastIndex = valueInt; - break; default: LOG_FATAL("bad param"); } @@ -577,7 +540,6 @@ void AudioMixer::setParameter(int name, int target, int param, void *value) bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) { - checkMagic(); if (value != devSampleRate || resampler != NULL) { if (sampleRate != value) { sampleRate = value; @@ -610,7 +572,6 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) inline void AudioMixer::track_t::adjustVolumeRamp(bool aux) { - checkMagic(); for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; i++) { if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) || ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) { @@ -639,10 +600,8 @@ size_t AudioMixer::getUnreleasedFrames(int name) const void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider) { name -= TRACK0; - mLog->logf("set bp %d=%p", name, bufferProvider); ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); - mState.tracks[name].checkMagic(); if (mState.tracks[name].downmixerBufferProvider != NULL) { // update required? if (mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider != bufferProvider) { @@ -660,27 +619,10 @@ void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider } } -AudioBufferProvider* AudioMixer::getBufferProvider(int name) -{ - name -= TRACK0; - ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); - mState.tracks[name].checkMagic(); - return mState.tracks[name].bufferProvider; -} -int AudioMixer::getFastIndex(int name) -{ - name -= TRACK0; - ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); - mState.tracks[name].checkMagic(); - return mState.tracks[name].fastIndex; -} void AudioMixer::process(int64_t pts) { - if (mState.needsChanged) { - mLog->logf("process needs=%#x", mState.needsChanged); - } mState.hook(&mState, pts); } @@ -705,7 +647,6 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) } state->enabledTracks &= ~disabled; state->enabledTracks |= enabled; - state->mLog->logf("process_validate ena=%#x", state->enabledTracks); // compute everything we need... int countActiveTracks = 0; @@ -1117,7 +1058,6 @@ void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, void AudioMixer::process__nop(state_t* state, int64_t pts) { uint32_t e0 = state->enabledTracks; - state->mLog->logf("process_nop ena=%#x", e0); size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS; while (e0) { // process by group of tracks with same output buffer to @@ -1163,7 +1103,6 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) // acquire each track's buffer uint32_t enabledTracks = state->enabledTracks; - state->mLog->logf("process_gNR ena=%#x", enabledTracks); uint32_t e0 = enabledTracks; while (e0) { const int i = 31 - __builtin_clz(e0); @@ -1172,8 +1111,8 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) t.buffer.frameCount = state->frameCount; int valid = t.bufferProvider->getValid(); if (valid != AudioBufferProvider::kValid) { - ALOGE("invalid bufferProvider=%p name=%d fastIndex=%d frameCount=%d valid=%#x enabledTracks=%#x", - t.bufferProvider, i, t.fastIndex, t.buffer.frameCount, valid, enabledTracks); + ALOGE("invalid bufferProvider=%p name=%d frameCount=%d valid=%#x enabledTracks=%#x", + t.bufferProvider, i, t.buffer.frameCount, valid, enabledTracks); // expect to crash } t.bufferProvider->getNextBuffer(&t.buffer, pts); @@ -1272,7 +1211,6 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) size_t numFrames = state->frameCount; uint32_t e0 = state->enabledTracks; - state->mLog->logf("process_gR ena=%#x", e0); while (e0) { // process by group of tracks with same output buffer // to optimize cache use @@ -1337,7 +1275,6 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, int64_t pts) { - state->mLog->logf("process_1TSNR ena=%#x", state->enabledTracks); // This method is only called when state->enabledTracks has exactly // one bit set. The asserts below would verify this, but are commented out // since the whole point of this method is to optimize performance. @@ -1407,7 +1344,6 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, { int i; uint32_t en = state->enabledTracks; - state->mLog->logf("process_2TSNR ena=%#x", en); i = 31 - __builtin_clz(en); const track_t& t0 = state->tracks[i]; |