diff options
Diffstat (limited to 'services')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 40 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.h | 4 | ||||
| -rw-r--r-- | services/audioflinger/FastMixerDumpState.cpp | 8 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 178 | ||||
| -rw-r--r-- | services/audioflinger/Threads.h | 4 |
5 files changed, 172 insertions, 62 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 940267a..461b5d3 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -1957,18 +1957,18 @@ status_t AudioFlinger::restoreOutput(audio_io_handle_t output) status_t AudioFlinger::openInput(audio_module_handle_t module, audio_io_handle_t *input, audio_config_t *config, - audio_devices_t *device, + audio_devices_t *devices, const String8& address, audio_source_t source, audio_input_flags_t flags) { Mutex::Autolock _l(mLock); - if (*device == AUDIO_DEVICE_NONE) { + if (*devices == AUDIO_DEVICE_NONE) { return BAD_VALUE; } - sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags); + sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); if (thread != 0) { // notify client processes of the new input creation @@ -1981,12 +1981,12 @@ status_t AudioFlinger::openInput(audio_module_handle_t module, sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, audio_io_handle_t *input, audio_config_t *config, - audio_devices_t device, + audio_devices_t devices, const String8& address, audio_source_t source, audio_input_flags_t flags) { - AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device); + AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); if (inHwDev == NULL) { *input = AUDIO_IO_HANDLE_NONE; return 0; @@ -1999,7 +1999,7 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m audio_config_t halconfig = *config; audio_hw_device_t *inHwHal = inHwDev->hwDevice(); audio_stream_in_t *inStream = NULL; - status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, + status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, &inStream, flags, address.string(), source); ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" ", Format %#x, Channels %x, flags %#x, status %d addr %s", @@ -2021,7 +2021,7 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m // FIXME describe the change proposed by HAL (save old values so we can log them here) ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); inStream = NULL; - status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, + status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, &inStream, flags, address.string(), source); // FIXME log this new status; HAL should not propose any further changes } @@ -2086,7 +2086,7 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m inputStream, *input, primaryOutputDevice_l(), - device + devices #ifdef TEE_SINK , teeSink #endif @@ -2809,13 +2809,13 @@ bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule> struct Entry { -#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav - char mName[MAX_NAME]; +#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 + char mFileName[TEE_MAX_FILENAME]; }; int comparEntry(const void *p1, const void *p2) { - return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); + return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); } #ifdef TEE_SINK @@ -2834,11 +2834,11 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand DIR *dir = opendir(teePath); teePath[teePathLen++] = '/'; if (dir != NULL) { -#define MAX_SORT 20 // number of entries to sort -#define MAX_KEEP 10 // number of entries to keep - struct Entry entries[MAX_SORT]; +#define TEE_MAX_SORT 20 // number of entries to sort +#define TEE_MAX_KEEP 10 // number of entries to keep + struct Entry entries[TEE_MAX_SORT]; size_t entryCount = 0; - while (entryCount < MAX_SORT) { + while (entryCount < TEE_MAX_SORT) { struct dirent de; struct dirent *result = NULL; int rc = readdir_r(dir, &de, &result); @@ -2855,17 +2855,17 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand } // ignore non .wav file entries size_t nameLen = strlen(de.d_name); - if (nameLen <= 4 || nameLen >= MAX_NAME || + if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || strcmp(&de.d_name[nameLen - 4], ".wav")) { continue; } - strcpy(entries[entryCount++].mName, de.d_name); + strcpy(entries[entryCount++].mFileName, de.d_name); } (void) closedir(dir); - if (entryCount > MAX_KEEP) { + if (entryCount > TEE_MAX_KEEP) { qsort(entries, entryCount, sizeof(Entry), comparEntry); - for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { - strcpy(&teePath[teePathLen], entries[i].mName); + for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { + strcpy(&teePath[teePathLen], entries[i].mFileName); (void) unlink(teePath); } } diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index f5dec97..7b76185 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -796,6 +796,10 @@ private: #undef INCLUDING_FROM_AUDIOFLINGER_H const char *formatToString(audio_format_t format); +String8 inputFlagsToString(audio_input_flags_t flags); +String8 outputFlagsToString(audio_output_flags_t flags); +String8 devicesToString(audio_devices_t devices); +const char *sourceToString(audio_source_t source); // ---------------------------------------------------------------------------- diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp index 386b29d..65fbf2b 100644 --- a/services/audioflinger/FastMixerDumpState.cpp +++ b/services/audioflinger/FastMixerDumpState.cpp @@ -69,10 +69,10 @@ void FastMixerDumpState::dump(int fd) const " numTracks=%u writeErrors=%u underruns=%u overruns=%u\n" " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n" " mixPeriod=%.2f ms\n", - FastMixerState::commandToString(mCommand), mWriteSequence, mFramesWritten, - mNumTracks, mWriteErrors, mUnderruns, mOverruns, - mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles, - mixPeriodSec * 1e3); + FastMixerState::commandToString(mCommand), mWriteSequence, mFramesWritten, + mNumTracks, mWriteErrors, mUnderruns, mOverruns, + mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles, + mixPeriodSec * 1e3); #ifdef FAST_THREAD_STATISTICS // find the interval of valid samples uint32_t bounds = mBounds; diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 7451245..c1da6bc 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -335,7 +335,90 @@ const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBas } } -static String8 outputFlagsToString(audio_output_flags_t flags) +String8 devicesToString(audio_devices_t devices) +{ + static const struct mapping { + audio_devices_t mDevices; + const char * mString; + } mappingsOut[] = { + AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", + AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", + AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", + AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", + AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", + AUDIO_DEVICE_NONE, "NONE", // must be last + }, mappingsIn[] = { + AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", + AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", + AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", + AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", + AUDIO_DEVICE_NONE, "NONE", // must be last + }; + String8 result; + audio_devices_t allDevices = AUDIO_DEVICE_NONE; + const mapping *entry; + if (devices & AUDIO_DEVICE_BIT_IN) { + devices &= ~AUDIO_DEVICE_BIT_IN; + entry = mappingsIn; + } else { + entry = mappingsOut; + } + for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { + allDevices = (audio_devices_t) (allDevices | entry->mDevices); + if (devices & entry->mDevices) { + if (!result.isEmpty()) { + result.append("|"); + } + result.append(entry->mString); + } + } + if (devices & ~allDevices) { + if (!result.isEmpty()) { + result.append("|"); + } + result.appendFormat("0x%X", devices & ~allDevices); + } + if (result.isEmpty()) { + result.append(entry->mString); + } + return result; +} + +String8 inputFlagsToString(audio_input_flags_t flags) +{ + static const struct mapping { + audio_input_flags_t mFlag; + const char * mString; + } mappings[] = { + AUDIO_INPUT_FLAG_FAST, "FAST", + AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", + AUDIO_INPUT_FLAG_NONE, "NONE", // must be last + }; + String8 result; + audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; + const mapping *entry; + for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { + allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); + if (flags & entry->mFlag) { + if (!result.isEmpty()) { + result.append("|"); + } + result.append(entry->mString); + } + } + if (flags & ~allFlags) { + if (!result.isEmpty()) { + result.append("|"); + } + result.appendFormat("0x%X", flags & ~allFlags); + } + if (result.isEmpty()) { + result.append(entry->mString); + } + return result; +} + +String8 outputFlagsToString(audio_output_flags_t flags) { static const struct mapping { audio_output_flags_t mFlag; @@ -345,7 +428,7 @@ static String8 outputFlagsToString(audio_output_flags_t flags) AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", AUDIO_OUTPUT_FLAG_FAST, "FAST", AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", - AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAAD", + AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last @@ -374,6 +457,24 @@ static String8 outputFlagsToString(audio_output_flags_t flags) return result; } +const char *sourceToString(audio_source_t source) +{ + switch (source) { + case AUDIO_SOURCE_DEFAULT: return "default"; + case AUDIO_SOURCE_MIC: return "mic"; + case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; + case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; + case AUDIO_SOURCE_VOICE_CALL: return "voice call"; + case AUDIO_SOURCE_CAMCORDER: return "camcorder"; + case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; + case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; + case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; + case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; + case AUDIO_SOURCE_HOTWORD: return "hotword"; + default: return "unknown"; + } +} + AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, audio_devices_t outDevice, audio_devices_t inDevice, type_t type) : Thread(false /*canCallJava*/), @@ -640,6 +741,7 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u dprintf(fd, "thread %p may be deadlocked\n", this); } + dprintf(fd, " Thread name: %s\n", mThreadName); dprintf(fd, " I/O handle: %d\n", mId); dprintf(fd, " TID: %d\n", getTid()); dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); @@ -663,6 +765,9 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u } else { dprintf(fd, " none\n"); } + dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); + dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); + dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); if (locked) { mLock.unlock(); @@ -696,19 +801,19 @@ void AudioFlinger::ThreadBase::acquireWakeLock(int uid) String16 AudioFlinger::ThreadBase::getWakeLockTag() { switch (mType) { - case MIXER: - return String16("AudioMix"); - case DIRECT: - return String16("AudioDirectOut"); - case DUPLICATING: - return String16("AudioDup"); - case RECORD: - return String16("AudioIn"); - case OFFLOAD: - return String16("AudioOffload"); - default: - ALOG_ASSERT(false); - return String16("AudioUnknown"); + case MIXER: + return String16("AudioMix"); + case DIRECT: + return String16("AudioDirectOut"); + case DUPLICATING: + return String16("AudioDup"); + case RECORD: + return String16("AudioIn"); + case OFFLOAD: + return String16("AudioOffload"); + default: + ALOG_ASSERT(false); + return String16("AudioUnknown"); } } @@ -735,7 +840,7 @@ void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) if (status == NO_ERROR) { mWakeLockToken = binder; } - ALOGV("acquireWakeLock_l() %s status %d", mName, status); + ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); } } @@ -748,7 +853,7 @@ void AudioFlinger::ThreadBase::releaseWakeLock() void AudioFlinger::ThreadBase::releaseWakeLock_l() { if (mWakeLockToken != 0) { - ALOGV("releaseWakeLock_l() %s", mName); + ALOGV("releaseWakeLock_l() %s", mThreadName); if (mPowerManager != 0) { mPowerManager->releaseWakeLock(mWakeLockToken, 0, true /* FIXME force oneway contrary to .aidl */); @@ -769,7 +874,7 @@ void AudioFlinger::ThreadBase::getPowerManager_l() { sp<IBinder> binder = defaultServiceManager()->checkService(String16("power")); if (binder == 0) { - ALOGW("Thread %s cannot connect to the power manager service", mName); + ALOGW("Thread %s cannot connect to the power manager service", mThreadName); } else { mPowerManager = interface_cast<IPowerManager>(binder); binder->linkToDeath(mDeathRecipient); @@ -789,7 +894,7 @@ void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uid status_t status; status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), true /* FIXME force oneway contrary to .aidl */); - ALOGV("acquireWakeLock_l() %s status %d", mName, status); + ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); } } @@ -973,7 +1078,7 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). if (mType == DIRECT) { ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", - desc->name, mName); + desc->name, mThreadName); lStatus = BAD_VALUE; goto Exit; } @@ -997,7 +1102,8 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( case DUPLICATING: case RECORD: default: - ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); + ALOGW("createEffect_l() Cannot add global effect %s on thread %s", + desc->name, mThreadName); lStatus = BAD_VALUE; goto Exit; } @@ -1262,8 +1368,8 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge // mLatchD, mLatchQ, mLatchDValid(false), mLatchQValid(false) { - snprintf(mName, kNameLength, "AudioOut_%X", id); - mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); + snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); + mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); // Assumes constructor is called by AudioFlinger with it's mLock held, but // it would be safer to explicitly pass initial masterVolume/masterMute as @@ -1377,6 +1483,9 @@ void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& ar void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) { dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); + + dumpBase(fd, args); + dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); dprintf(fd, " Total writes: %d\n", mNumWrites); @@ -1391,15 +1500,13 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; String8 flagsAsString = outputFlagsToString(flags); dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); - - dumpBase(fd, args); } // Thread virtuals void AudioFlinger::PlaybackThread::onFirstRef() { - run(mName, ANDROID_PRIORITY_URGENT_AUDIO); + run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); } // ThreadBase virtuals @@ -1443,9 +1550,10 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac ( (sharedBuffer != 0) ) || - // use case 2: callback handler and frame count is default or at least as large as HAL + // use case 2: frame count is default or at least as large as HAL ( - (tid != -1) && + // we formerly checked for a callback handler (non-0 tid), + // but that is no longer required for TRANSFER_OBTAIN mode ((frameCount == 0) || (frameCount >= mFrameCount)) ) @@ -5059,8 +5167,8 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, // mFastCaptureNBLogWriter , mFastTrackAvail(false) { - snprintf(mName, kNameLength, "AudioIn_%X", id); - mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); + snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); + mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); readInputParameters_l(); @@ -5201,7 +5309,7 @@ AudioFlinger::RecordThread::~RecordThread() void AudioFlinger::RecordThread::onFirstRef() { - run(mName, PRIORITY_URGENT_AUDIO); + run(mThreadName, PRIORITY_URGENT_AUDIO); } bool AudioFlinger::RecordThread::threadLoop() @@ -6049,15 +6157,13 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a { dprintf(fd, "\nInput thread %p:\n", this); - if (mActiveTracks.size() > 0) { - dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); - } else { + dumpBase(fd, args); + + if (mActiveTracks.size() == 0) { dprintf(fd, " No active record clients\n"); } dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); - - dumpBase(fd, args); } void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index a1ac42c..9350e48 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -432,8 +432,8 @@ protected: const audio_io_handle_t mId; Vector< sp<EffectChain> > mEffectChains; - static const int kNameLength = 16; // prctl(PR_SET_NAME) limit - char mName[kNameLength]; + static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit + char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated sp<IPowerManager> mPowerManager; sp<IBinder> mWakeLockToken; const sp<PMDeathRecipient> mDeathRecipient; |