diff options
Diffstat (limited to 'services/audioflinger/AudioFlinger.cpp')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 281 |
1 files changed, 216 insertions, 65 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 87eb6aa..a9c9b56 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -19,6 +19,7 @@ #define LOG_TAG "AudioFlinger" //#define LOG_NDEBUG 0 +#include "Configuration.h" #include <dirent.h> #include <math.h> #include <signal.h> @@ -36,10 +37,6 @@ #include <cutils/bitops.h> #include <cutils/properties.h> -#include <cutils/compiler.h> - -//#include <private/media/AudioTrackShared.h> -//#include <private/media/AudioEffectShared.h> #include <system/audio.h> #include <hardware/audio.h> @@ -58,12 +55,13 @@ #include <powermanager/PowerManager.h> #include <common_time/cc_helper.h> -//#include <common_time/local_clock.h> #include <media/IMediaLogService.h> #include <media/nbaio/Pipe.h> #include <media/nbaio/PipeReader.h> +#include <media/AudioParameter.h> +#include <private/android_filesystem_config.h> // ---------------------------------------------------------------------------- @@ -100,6 +98,10 @@ size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; #endif +// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off +// we define a minimum time during which a global effect is considered enabled. +static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); + // ---------------------------------------------------------------------------- static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) @@ -141,7 +143,10 @@ AudioFlinger::AudioFlinger() mMasterMute(false), mNextUniqueId(1), mMode(AUDIO_MODE_INVALID), - mBtNrecIsOff(false) + mBtNrecIsOff(false), + mIsLowRamDevice(true), + mIsDeviceTypeKnown(false), + mGlobalEffectEnableTime(0) { getpid_cached = getpid(); char value[PROPERTY_VALUE_MAX]; @@ -259,6 +264,12 @@ void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) } } + result.append("Notification Clients:\n"); + for (size_t i = 0; i < mNotificationClients.size(); ++i) { + snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); + result.append(buffer); + } + result.append("Global session refs:\n"); result.append(" session pid count\n"); for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { @@ -436,6 +447,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( audio_io_handle_t output, pid_t tid, int *sessionId, + String8& name, status_t *status) { sp<PlaybackThread::Track> track; @@ -524,6 +536,9 @@ sp<IAudioTrack> AudioFlinger::createTrack( } } if (lStatus == NO_ERROR) { + // s for server's pid, n for normal mixer name, f for fast index + name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, + track->fastIndex()); trackHandle = new TrackHandle(track); } else { // remove local strong reference to Client before deleting the Track so that the Client @@ -981,11 +996,12 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t form AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; - struct audio_config config = { - sample_rate: sampleRate, - channel_mask: channelMask, - format: format, - }; + struct audio_config config; + memset(&config, 0, sizeof(config)); + config.sample_rate = sampleRate; + config.channel_mask = channelMask; + config.format = format; + audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); size_t size = dev->get_input_buffer_size(dev, &config); mHardwareStatus = AUDIO_HW_IDLE; @@ -1201,13 +1217,17 @@ void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) // ---------------------------------------------------------------------------- +static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { + return audio_is_remote_submix_device(inDevice); +} + sp<IAudioRecord> AudioFlinger::openRecord( audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, - IAudioFlinger::track_flags_t flags, + IAudioFlinger::track_flags_t *flags, pid_t tid, int *sessionId, status_t *status) @@ -1222,19 +1242,34 @@ sp<IAudioRecord> AudioFlinger::openRecord( // check calling permissions if (!recordingAllowed()) { + ALOGE("openRecord() permission denied: recording not allowed"); lStatus = PERMISSION_DENIED; goto Exit; } + if (format != AUDIO_FORMAT_PCM_16_BIT) { + ALOGE("openRecord() invalid format %d", format); + lStatus = BAD_VALUE; + goto Exit; + } + // add client to list { // scope for mLock Mutex::Autolock _l(mLock); thread = checkRecordThread_l(input); if (thread == NULL) { + ALOGE("openRecord() checkRecordThread_l failed"); lStatus = BAD_VALUE; goto Exit; } + if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) + && !captureAudioOutputAllowed()) { + ALOGE("openRecord() permission denied: capture not allowed"); + lStatus = PERMISSION_DENIED; + goto Exit; + } + pid_t pid = IPCThreadState::self()->getCallingPid(); client = registerPid_l(pid); @@ -1251,6 +1286,7 @@ sp<IAudioRecord> AudioFlinger::openRecord( // The record track uses one track in mHardwareMixerThread by convention. recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, frameCount, lSessionId, flags, tid, &lStatus); + LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR)); } if (lStatus != NO_ERROR) { // remove local strong reference to Client before deleting the RecordTrack so that the @@ -1382,31 +1418,53 @@ size_t AudioFlinger::getPrimaryOutputFrameCount() // ---------------------------------------------------------------------------- +status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) +{ + uid_t uid = IPCThreadState::self()->getCallingUid(); + if (uid != AID_SYSTEM) { + return PERMISSION_DENIED; + } + Mutex::Autolock _l(mLock); + if (mIsDeviceTypeKnown) { + return INVALID_OPERATION; + } + mIsLowRamDevice = isLowRamDevice; + mIsDeviceTypeKnown = true; + return NO_ERROR; +} + +// ---------------------------------------------------------------------------- + audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask, uint32_t *pLatencyMs, - audio_output_flags_t flags) + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) { - status_t status; PlaybackThread *thread = NULL; - struct audio_config config = { - sample_rate: pSamplingRate ? *pSamplingRate : 0, - channel_mask: pChannelMask ? *pChannelMask : 0, - format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, - }; + struct audio_config config; + config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; + config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; + config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; + if (offloadInfo) { + config.offload_info = *offloadInfo; + } + audio_stream_out_t *outStream = NULL; AudioHwDevice *outHwDev; - ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", + ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", module, (pDevices != NULL) ? *pDevices : 0, config.sample_rate, config.format, config.channel_mask, flags); + ALOGV("openOutput(), offloadInfo %p version 0x%04x", + offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); if (pDevices == NULL || *pDevices == 0) { return 0; @@ -1423,7 +1481,7 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; - status = hwDevHal->open_output_stream(hwDevHal, + status_t status = hwDevHal->open_output_stream(hwDevHal, id, *pDevices, (audio_output_flags_t)flags, @@ -1431,7 +1489,7 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, &outStream); mHardwareStatus = AUDIO_HW_IDLE; - ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " + ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " "Channels %x, status %d", outStream, config.sample_rate, @@ -1440,9 +1498,12 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, status); if (status == NO_ERROR && outStream != NULL) { - AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); + AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); - if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || + if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { + thread = new OffloadThread(this, output, id, *pDevices); + ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); + } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || (config.format != AUDIO_FORMAT_PCM_16_BIT) || (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { thread = new DirectOutputThread(this, output, id, *pDevices); @@ -1453,10 +1514,18 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, } mPlaybackThreads.add(id, thread); - if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; - if (pFormat != NULL) *pFormat = config.format; - if (pChannelMask != NULL) *pChannelMask = config.channel_mask; - if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); + if (pSamplingRate != NULL) { + *pSamplingRate = config.sample_rate; + } + if (pFormat != NULL) { + *pFormat = config.format; + } + if (pChannelMask != NULL) { + *pChannelMask = config.channel_mask; + } + if (pLatencyMs != NULL) { + *pLatencyMs = thread->latency(); + } // notify client processes of the new output creation thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); @@ -1524,11 +1593,28 @@ status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); dupThread->removeOutputTrack((MixerThread *)thread.get()); + } } } - audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); + + mPlaybackThreads.removeItem(output); + // save all effects to the default thread + if (mPlaybackThreads.size()) { + PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); + if (dstThread != NULL) { + // audioflinger lock is held here so the acquisition order of thread locks does not + // matter + Mutex::Autolock _dl(dstThread->mLock); + Mutex::Autolock _sl(thread->mLock); + Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); + for (size_t i = 0; i < effectChains.size(); i ++) { + moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); + } + } + } + audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); } thread->exit(); // The thread entity (active unit of execution) is no longer running here, @@ -1583,11 +1669,11 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, { status_t status; RecordThread *thread = NULL; - struct audio_config config = { - sample_rate: pSamplingRate ? *pSamplingRate : 0, - channel_mask: pChannelMask ? *pChannelMask : 0, - format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, - }; + struct audio_config config; + config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; + config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; + config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; + uint32_t reqSamplingRate = config.sample_rate; audio_format_t reqFormat = config.format; audio_channel_mask_t reqChannels = config.channel_mask; @@ -1683,7 +1769,7 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); // Start record thread - // RecorThread require both input and output device indication to forward to audio + // RecordThread requires both input and output device indication to forward to audio // pre processing modules thread = new RecordThread(this, input, @@ -1698,9 +1784,15 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, ); mRecordThreads.add(id, thread); ALOGV("openInput() created record thread: ID %d thread %p", id, thread); - if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; - if (pFormat != NULL) *pFormat = config.format; - if (pChannelMask != NULL) *pChannelMask = reqChannels; + if (pSamplingRate != NULL) { + *pSamplingRate = reqSamplingRate; + } + if (pFormat != NULL) { + *pFormat = config.format; + } + if (pChannelMask != NULL) { + *pChannelMask = reqChannels; + } // notify client processes of the new input creation thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); @@ -1768,6 +1860,16 @@ void AudioFlinger::acquireAudioSessionId(int audioSession) Mutex::Autolock _l(mLock); pid_t caller = IPCThreadState::self()->getCallingPid(); ALOGV("acquiring %d from %d", audioSession, caller); + + // Ignore requests received from processes not known as notification client. The request + // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be + // called from a different pid leaving a stale session reference. Also we don't know how + // to clear this reference if the client process dies. + if (mNotificationClients.indexOfKey(caller) < 0) { + ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); + return; + } + size_t num = mAudioSessionRefs.size(); for (size_t i = 0; i< num; i++) { AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); @@ -1800,7 +1902,9 @@ void AudioFlinger::releaseAudioSessionId(int audioSession) return; } } - ALOGW("session id %d not found for pid %d", audioSession, caller); + // If the caller is mediaserver it is likely that the session being released was acquired + // on behalf of a process not in notification clients and we ignore the warning. + ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); } void AudioFlinger::purgeStaleEffects_l() { @@ -2001,24 +2105,7 @@ sp<IEffect> AudioFlinger::createEffect( goto Exit; } - if (io == 0) { - if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { - // output must be specified by AudioPolicyManager when using session - // AUDIO_SESSION_OUTPUT_STAGE - lStatus = BAD_VALUE; - goto Exit; - } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { - // if the output returned by getOutputForEffect() is removed before we lock the - // mutex below, the call to checkPlaybackThread_l(io) below will detect it - // and we will exit safely - io = AudioSystem::getOutputForEffect(&desc); - } - } - { - Mutex::Autolock _l(mLock); - - if (!EffectIsNullUuid(&pDesc->uuid)) { // if uuid is specified, request effect descriptor lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); @@ -2091,6 +2178,15 @@ sp<IEffect> AudioFlinger::createEffect( // return effect descriptor *pDesc = desc; + if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) { + // if the output returned by getOutputForEffect() is removed before we lock the + // mutex below, the call to checkPlaybackThread_l(io) below will detect it + // and we will exit safely + io = AudioSystem::getOutputForEffect(&desc); + ALOGV("createEffect got output %d", io); + } + + Mutex::Autolock _l(mLock); // If output is not specified try to find a matching audio session ID in one of the // output threads. @@ -2098,6 +2194,12 @@ sp<IEffect> AudioFlinger::createEffect( // because of code checking output when entering the function. // Note: io is never 0 when creating an effect on an input if (io == 0) { + if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { + // output must be specified by AudioPolicyManager when using session + // AUDIO_SESSION_OUTPUT_STAGE + lStatus = BAD_VALUE; + goto Exit; + } // look for the thread where the specified audio session is present for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { @@ -2171,9 +2273,7 @@ status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, Mutex::Autolock _dl(dstThread->mLock); Mutex::Autolock _sl(srcThread->mLock); - moveEffectChain_l(sessionId, srcThread, dstThread, false); - - return NO_ERROR; + return moveEffectChain_l(sessionId, srcThread, dstThread, false); } // moveEffectChain_l must be called with both srcThread and dstThread mLocks held @@ -2200,13 +2300,18 @@ status_t AudioFlinger::moveEffectChain_l(int sessionId, // transfer all effects one by one so that new effect chain is created on new thread with // correct buffer sizes and audio parameters and effect engines reconfigured accordingly - audio_io_handle_t dstOutput = dstThread->id(); sp<EffectChain> dstChain; uint32_t strategy = 0; // prevent compiler warning sp<EffectModule> effect = chain->getEffectFromId_l(0); + Vector< sp<EffectModule> > removed; + status_t status = NO_ERROR; while (effect != 0) { srcThread->removeEffect_l(effect); - dstThread->addEffect_l(effect); + removed.add(effect); + status = dstThread->addEffect_l(effect); + if (status != NO_ERROR) { + break; + } // removeEffect_l() has stopped the effect if it was active so it must be restarted if (effect->state() == EffectModule::ACTIVE || effect->state() == EffectModule::STOPPING) { @@ -2218,15 +2323,15 @@ status_t AudioFlinger::moveEffectChain_l(int sessionId, dstChain = effect->chain().promote(); if (dstChain == 0) { ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); - srcThread->addEffect_l(effect); - return NO_INIT; + status = NO_INIT; + break; } strategy = dstChain->strategy(); } if (reRegister) { AudioSystem::unregisterEffect(effect->id()); AudioSystem::registerEffect(&effect->desc(), - dstOutput, + dstThread->id(), strategy, sessionId, effect->id()); @@ -2234,7 +2339,53 @@ status_t AudioFlinger::moveEffectChain_l(int sessionId, effect = chain->getEffectFromId_l(0); } - return NO_ERROR; + if (status != NO_ERROR) { + for (size_t i = 0; i < removed.size(); i++) { + srcThread->addEffect_l(removed[i]); + if (dstChain != 0 && reRegister) { + AudioSystem::unregisterEffect(removed[i]->id()); + AudioSystem::registerEffect(&removed[i]->desc(), + srcThread->id(), + strategy, + sessionId, + removed[i]->id()); + } + } + } + + return status; +} + +bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() +{ + if (mGlobalEffectEnableTime != 0 && + ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { + return true; + } + + for (size_t i = 0; i < mPlaybackThreads.size(); i++) { + sp<EffectChain> ec = + mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); + if (ec != 0 && ec->isNonOffloadableEnabled()) { + return true; + } + } + return false; +} + +void AudioFlinger::onNonOffloadableGlobalEffectEnable() +{ + Mutex::Autolock _l(mLock); + + mGlobalEffectEnableTime = systemTime(); + + for (size_t i = 0; i < mPlaybackThreads.size(); i++) { + sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); + if (t->mType == ThreadBase::OFFLOAD) { + t->invalidateTracks(AUDIO_STREAM_MUSIC); + } + } + } struct Entry { |