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-rw-r--r--libs/audioflinger/AudioFlinger.cpp1450
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diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
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+++ b/libs/audioflinger/AudioFlinger.cpp
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+/* //device/include/server/AudioFlinger/AudioFlinger.cpp
+**
+** Copyright 2007, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <math.h>
+#include <signal.h>
+#include <sys/time.h>
+#include <sys/resource.h>
+
+#include <utils/IServiceManager.h>
+#include <utils/Log.h>
+#include <utils/Parcel.h>
+#include <utils/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+
+#include <media/AudioTrack.h>
+#include <media/AudioRecord.h>
+
+#include <private/media/AudioTrackShared.h>
+
+#include <hardware/AudioHardwareInterface.h>
+
+#include "AudioMixer.h"
+#include "AudioFlinger.h"
+
+namespace android {
+
+static const nsecs_t kStandbyTimeInNsecs = seconds(3);
+static const unsigned long kBufferRecoveryInUsecs = 2000;
+static const unsigned long kMaxBufferRecoveryInUsecs = 20000;
+static const float MAX_GAIN = 4096.0f;
+
+// retry counts for buffer fill timeout
+// 50 * ~20msecs = 1 second
+static const int8_t kMaxTrackRetries = 50;
+static const int8_t kMaxTrackStartupRetries = 50;
+
+#define AUDIOFLINGER_SECURITY_ENABLED 1
+
+// ----------------------------------------------------------------------------
+
+static bool recordingAllowed() {
+#ifndef HAVE_ANDROID_OS
+ return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+ if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+ bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
+ if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
+ return ok;
+#else
+ if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
+ LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
+ return true;
+#endif
+}
+
+static bool settingsAllowed() {
+#ifndef HAVE_ANDROID_OS
+ return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+ if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+ bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
+ if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+ return ok;
+#else
+ if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
+ LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
+ return true;
+#endif
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioFlinger()
+ : BnAudioFlinger(), Thread(false),
+ mMasterVolume(0), mMasterMute(true),
+ mAudioMixer(0), mAudioHardware(0), mOutput(0), mAudioRecordThread(0),
+ mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0),
+ mMixBuffer(0), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0),
+ mStandby(false), mInWrite(false)
+{
+ mHardwareStatus = AUDIO_HW_IDLE;
+ mAudioHardware = AudioHardwareInterface::create();
+ mHardwareStatus = AUDIO_HW_INIT;
+ if (mAudioHardware->initCheck() == NO_ERROR) {
+ // open 16-bit output stream for s/w mixer
+ mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+ mOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ if (mOutput) {
+ mSampleRate = mOutput->sampleRate();
+ mChannelCount = mOutput->channelCount();
+ mFormat = mOutput->format();
+ mMixBufferSize = mOutput->bufferSize();
+ mFrameCount = mMixBufferSize / mChannelCount / sizeof(int16_t);
+ mMixBuffer = new int16_t[mFrameCount * mChannelCount];
+ memset(mMixBuffer, 0, mMixBufferSize);
+ mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+ // FIXME - this should come from settings
+ setMasterVolume(1.0f);
+ setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
+ setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
+ setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
+ setMode(AudioSystem::MODE_NORMAL);
+ mMasterMute = false;
+ } else {
+ LOGE("Failed to initialize output stream");
+ }
+ } else {
+ LOGE("Couldn't even initialize the stubbed audio hardware!");
+ }
+}
+
+AudioFlinger::~AudioFlinger()
+{
+ delete mOutput;
+ delete mAudioHardware;
+ delete [] mMixBuffer;
+ delete mAudioMixer;
+ mAudioRecordThread.clear();
+}
+
+status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ result.append("Clients:\n");
+ for (size_t i = 0; i < mClients.size(); ++i) {
+ wp<Client> wClient = mClients.valueAt(i);
+ if (wClient != 0) {
+ sp<Client> client = wClient.promote();
+ if (client != 0) {
+ snprintf(buffer, SIZE, " pid: %d\n", client->pid());
+ result.append(buffer);
+ }
+ }
+ }
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::dumpTracks(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ result.append("Tracks:\n");
+ result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ wp<Track> wTrack = mTracks[i];
+ if (wTrack != 0) {
+ sp<Track> track = wTrack.promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ }
+ }
+
+ result.append("Active Tracks:\n");
+ result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+ for (size_t i = 0; i < mActiveTracks.size(); ++i) {
+ wp<Track> wTrack = mTracks[i];
+ if (wTrack != 0) {
+ sp<Track> track = wTrack.promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ }
+ }
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", audioMixer().trackNames());
+ result.append(buffer);
+ snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+ result.append(buffer);
+ snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Hardware status: %d\n", mHardwareStatus);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+ snprintf(buffer, SIZE, "Permission Denial: "
+ "can't dump AudioFlinger from pid=%d, uid=%d\n",
+ IPCThreadState::self()->getCallingPid(),
+ IPCThreadState::self()->getCallingUid());
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
+{
+ if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+ dumpPermissionDenial(fd, args);
+ } else {
+ AutoMutex lock(&mLock);
+
+ dumpClients(fd, args);
+ dumpTracks(fd, args);
+ dumpInternals(fd, args);
+ if (mAudioHardware) {
+ mAudioHardware->dumpState(fd, args);
+ }
+ }
+ return NO_ERROR;
+}
+
+// Thread virtuals
+bool AudioFlinger::threadLoop()
+{
+ nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
+ unsigned long sleepTime = kBufferRecoveryInUsecs;
+ const size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
+ int16_t* curBuf = mMixBuffer;
+ Vector< sp<Track> > tracksToRemove;
+ size_t enabledTracks;
+ nsecs_t standbyTime = systemTime();
+
+ do {
+ enabledTracks = 0;
+ { // scope for the lock
+ Mutex::Autolock _l(mLock);
+ const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+ // put audio hardware into standby after short delay
+ if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
+ // wait until we have something to do...
+ LOGV("Audio hardware entering standby\n");
+ mHardwareStatus = AUDIO_HW_STANDBY;
+ if (!mStandby) {
+ mAudioHardware->standby();
+ mStandby = true;
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
+ mWaitWorkCV.wait(mLock);
+ LOGV("Audio hardware exiting standby\n");
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ continue;
+ }
+
+ // find out which tracks need to be processed
+ size_t count = activeTracks.size();
+ for (size_t i=0 ; i<count ; i++) {
+ sp<Track> t = activeTracks[i].promote();
+ if (t == 0) continue;
+
+ Track* const track = t.get();
+ audio_track_cblk_t* cblk = track->cblk();
+ uint32_t u = cblk->user;
+ uint32_t s = cblk->server;
+
+ // The first time a track is added we wait
+ // for all its buffers to be filled before processing it
+ audioMixer().setActiveTrack(track->name());
+ if ((u > s) && (track->isReady(u, s) || track->isStopped()) &&
+ !track->isPaused())
+ {
+ //LOGD("u=%08x, s=%08x [OK]", u, s);
+
+ // compute volume for this track
+ int16_t left, right;
+ if (track->isMuted() || mMasterMute || track->isPausing()) {
+ left = right = 0;
+ if (track->isPausing()) {
+ LOGV("paused(%d)", track->name());
+ track->setPaused();
+ }
+ } else {
+ float typeVolume = mStreamTypes[track->type()].volume;
+ float v = mMasterVolume * typeVolume;
+ float v_clamped = v * cblk->volume[0];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ left = int16_t(v_clamped);
+ v_clamped = v * cblk->volume[1];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ right = int16_t(v_clamped);
+ }
+
+ // XXX: these things DON'T need to be done each time
+ AudioMixer& mixer(audioMixer());
+ mixer.setBufferProvider(track);
+ mixer.enable(AudioMixer::MIXING);
+
+ int param;
+ if ( track->mFillingUpStatus == Track::FS_FILLED) {
+ // no ramp for the first volume setting
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ param = AudioMixer::RAMP_VOLUME;
+ } else {
+ param = AudioMixer::VOLUME;
+ }
+ } else {
+ param = AudioMixer::RAMP_VOLUME;
+ }
+ mixer.setParameter(param, AudioMixer::VOLUME0, left);
+ mixer.setParameter(param, AudioMixer::VOLUME1, right);
+ mixer.setParameter(
+ AudioMixer::TRACK,
+ AudioMixer::FORMAT, track->format());
+ mixer.setParameter(
+ AudioMixer::TRACK,
+ AudioMixer::CHANNEL_COUNT, track->channelCount());
+ mixer.setParameter(
+ AudioMixer::RESAMPLE,
+ AudioMixer::SAMPLE_RATE,
+ int(cblk->sampleRate));
+
+ // reset retry count
+ track->mRetryCount = kMaxTrackRetries;
+ enabledTracks++;
+ } else {
+ //LOGD("u=%08x, s=%08x [NOT READY]", u, s);
+ if (track->isStopped()) {
+ track->mFillingUpStatus = Track::FS_FILLING;
+ track->mFlags = 0;
+ }
+ if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+ // We have consumed all the buffers of this track.
+ // Remove it from the list of active tracks.
+ LOGV("remove(%d) from active list", track->name());
+ tracksToRemove.add(track);
+ } else {
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+ tracksToRemove.add(track);
+ }
+ }
+ // LOGV("disable(%d)", track->name());
+ audioMixer().disable(AudioMixer::MIXING);
+ }
+ }
+
+ // remove all the tracks that need to be...
+ count = tracksToRemove.size();
+ if (UNLIKELY(count)) {
+ for (size_t i=0 ; i<count ; i++) {
+ const sp<Track>& track = tracksToRemove[i];
+ mActiveTracks.remove(track);
+ if (track->isTerminated()) {
+ mTracks.remove(track);
+ audioMixer().deleteTrackName(track->mName);
+ }
+ }
+ }
+ }
+
+ if (LIKELY(enabledTracks)) {
+ // mix buffers...
+ audioMixer().process(curBuf);
+
+ // output audio to hardware
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ mOutput->write(curBuf, mixBufferSize);
+ mNumWrites++;
+ mInWrite = false;
+ mStandby = false;
+ nsecs_t temp = systemTime();
+ standbyTime = temp + kStandbyTimeInNsecs;
+ nsecs_t delta = temp - mLastWriteTime;
+ if (delta > maxPeriod) {
+ LOGW("write blocked for %llu msecs", ns2ms(delta));
+ mNumDelayedWrites++;
+ }
+ sleepTime = kBufferRecoveryInUsecs;
+ } else {
+ // There was nothing to mix this round, which means all
+ // active tracks were late. Sleep a little bit to give
+ // them another chance. If we're too late, the audio
+ // hardware will zero-fill for us.
+ LOGV("no buffers - usleep(%lu)", sleepTime);
+ usleep(sleepTime);
+ if (sleepTime < kMaxBufferRecoveryInUsecs) {
+ sleepTime += kBufferRecoveryInUsecs;
+ }
+ }
+
+ // finally let go of all our tracks, without the lock held
+ // since we can't guarantee the destructors won't acquire that
+ // same lock.
+ tracksToRemove.clear();
+ } while (true);
+
+ return false;
+}
+
+status_t AudioFlinger::readyToRun()
+{
+ if (mSampleRate == 0) {
+ LOGE("No working audio driver found.");
+ return NO_INIT;
+ }
+ LOGI("AudioFlinger's main thread ready to run.");
+ return NO_ERROR;
+}
+
+void AudioFlinger::onFirstRef()
+{
+ run("AudioFlinger", ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+// IAudioFlinger interface
+sp<IAudioTrack> AudioFlinger::createTrack(
+ pid_t pid,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int bufferCount,
+ uint32_t flags)
+{
+ if (streamType >= AudioTrack::NUM_STREAM_TYPES) {
+ LOGE("invalid stream type");
+ return NULL;
+ }
+
+ if (sampleRate > MAX_SAMPLE_RATE) {
+ LOGE("Sample rate out of range: %d", sampleRate);
+ return NULL;
+ }
+
+ sp<Track> track;
+ sp<TrackHandle> trackHandle;
+ Mutex::Autolock _l(mLock);
+
+ if (mSampleRate == 0) {
+ LOGE("Audio driver not initialized.");
+ return trackHandle;
+ }
+
+ sp<Client> client;
+ wp<Client> wclient = mClients.valueFor(pid);
+
+ if (wclient != NULL) {
+ client = wclient.promote();
+ } else {
+ client = new Client(this, pid);
+ mClients.add(pid, client);
+ }
+
+ // FIXME: Buffer size should be based on sample rate for consistent latency
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelCount, bufferCount, channelCount == 1 ? mMixBufferSize>>1 : mMixBufferSize);
+ mTracks.add(track);
+ trackHandle = new TrackHandle(track);
+ return trackHandle;
+}
+
+uint32_t AudioFlinger::sampleRate() const
+{
+ return mSampleRate;
+}
+
+int AudioFlinger::channelCount() const
+{
+ return mChannelCount;
+}
+
+int AudioFlinger::format() const
+{
+ return mFormat;
+}
+
+size_t AudioFlinger::frameCount() const
+{
+ return mFrameCount;
+}
+
+status_t AudioFlinger::setMasterVolume(float value)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ // when hw supports master volume, don't scale in sw mixer
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
+ mMasterVolume = 1.0f;
+ }
+ else {
+ mMasterVolume = value;
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
+ LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_GET_ROUTING;
+ uint32_t r;
+ uint32_t err = mAudioHardware->getRouting(mode, &r);
+ if (err == NO_ERROR) {
+ r = (r & ~mask) | (routes & mask);
+ mHardwareStatus = AUDIO_HW_SET_ROUTING;
+ err = mAudioHardware->setRouting(mode, r);
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return err;
+}
+
+uint32_t AudioFlinger::getRouting(int mode) const
+{
+ uint32_t routes = 0;
+ if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
+ mHardwareStatus = AUDIO_HW_GET_ROUTING;
+ mAudioHardware->getRouting(mode, &routes);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ } else {
+ LOGW("Illegal value: getRouting(%d)", mode);
+ }
+ return routes;
+}
+
+status_t AudioFlinger::setMode(int mode)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
+ LOGW("Illegal value: setMode(%d)", mode);
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MODE;
+ status_t ret = mAudioHardware->setMode(mode);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return ret;
+}
+
+int AudioFlinger::getMode() const
+{
+ int mode = AudioSystem::MODE_INVALID;
+ mHardwareStatus = AUDIO_HW_SET_MODE;
+ mAudioHardware->getMode(&mode);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return mode;
+}
+
+status_t AudioFlinger::setMicMute(bool state)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
+ status_t ret = mAudioHardware->setMicMute(state);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return ret;
+}
+
+bool AudioFlinger::getMicMute() const
+{
+ bool state = AudioSystem::MODE_INVALID;
+ mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
+ mAudioHardware->getMicMute(&state);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return state;
+}
+
+status_t AudioFlinger::setMasterMute(bool muted)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ mMasterMute = muted;
+ return NO_ERROR;
+}
+
+float AudioFlinger::masterVolume() const
+{
+ return mMasterVolume;
+}
+
+bool AudioFlinger::masterMute() const
+{
+ return mMasterMute;
+}
+
+status_t AudioFlinger::setStreamVolume(int stream, float value)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
+ return BAD_VALUE;
+ }
+
+ mStreamTypes[stream].volume = value;
+ status_t ret = NO_ERROR;
+ if (stream == AudioTrack::VOICE_CALL) {
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
+ ret = mAudioHardware->setVoiceVolume(value);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ }
+ return ret;
+}
+
+status_t AudioFlinger::setStreamMute(int stream, bool muted)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
+ return BAD_VALUE;
+ }
+ mStreamTypes[stream].mute = muted;
+ return NO_ERROR;
+}
+
+float AudioFlinger::streamVolume(int stream) const
+{
+ if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
+ return 0.0f;
+ }
+ return mStreamTypes[stream].volume;
+}
+
+bool AudioFlinger::streamMute(int stream) const
+{
+ if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
+ return true;
+ }
+ return mStreamTypes[stream].mute;
+}
+
+bool AudioFlinger::isMusicActive() const
+{
+ size_t count = mActiveTracks.size();
+ for (size_t i = 0 ; i < count ; ++i) {
+ sp<Track> t = mActiveTracks[i].promote();
+ if (t == 0) continue;
+ Track* const track = t.get();
+ if (t->mStreamType == AudioTrack::MUSIC)
+ return true;
+ }
+ return false;
+}
+
+status_t AudioFlinger::setParameter(const char* key, const char* value)
+{
+ status_t result;
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_PARAMETER;
+ result = mAudioHardware->setParameter(key, value);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return result;
+}
+
+void AudioFlinger::removeClient(pid_t pid)
+{
+ Mutex::Autolock _l(mLock);
+ mClients.removeItem(pid);
+}
+
+status_t AudioFlinger::addTrack(const sp<Track>& track)
+{
+ Mutex::Autolock _l(mLock);
+
+ // here the track could be either new, or restarted
+ // in both cases "unstop" the track
+ if (track->isPaused()) {
+ track->mState = TrackBase::RESUMING;
+ LOGV("PAUSED => RESUMING (%d)", track->name());
+ } else {
+ track->mState = TrackBase::ACTIVE;
+ LOGV("? => ACTIVE (%d)", track->name());
+ }
+ // set retry count for buffer fill
+ track->mRetryCount = kMaxTrackStartupRetries;
+ LOGV("mWaitWorkCV.broadcast");
+ mWaitWorkCV.broadcast();
+
+ if (mActiveTracks.indexOf(track) < 0) {
+ // the track is newly added, make sure it fills up all its
+ // buffers before playing. This is to ensure the client will
+ // effectively get the latency it requested.
+ track->mFillingUpStatus = Track::FS_FILLING;
+ mActiveTracks.add(track);
+ return NO_ERROR;
+ }
+ return ALREADY_EXISTS;
+}
+
+void AudioFlinger::removeTrack(wp<Track> track, int name)
+{
+ Mutex::Autolock _l(mLock);
+ sp<Track> t = track.promote();
+ if (t!=NULL && (t->mState <= TrackBase::STOPPED)) {
+ remove_track_l(track, name);
+ }
+}
+
+void AudioFlinger::remove_track_l(wp<Track> track, int name)
+{
+ sp<Track> t = track.promote();
+ if (t!=NULL) {
+ t->reset();
+ }
+ audioMixer().deleteTrackName(name);
+ mActiveTracks.remove(track);
+ mWaitWorkCV.broadcast();
+}
+
+void AudioFlinger::destroyTrack(const sp<Track>& track)
+{
+ // NOTE: We're acquiring a strong reference on the track before
+ // acquiring the lock, this is to make sure removing it from
+ // mTracks won't cause the destructor to be called while the lock is
+ // held (note that technically, 'track' could be a reference to an item
+ // in mTracks, which is why we need to do this).
+ sp<Track> keep(track);
+ Mutex::Autolock _l(mLock);
+ track->mState = TrackBase::TERMINATED;
+ if (mActiveTracks.indexOf(track) < 0) {
+ LOGV("remove track (%d) and delete from mixer", track->name());
+ mTracks.remove(track);
+ audioMixer().deleteTrackName(keep->name());
+ }
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
+ : RefBase(),
+ mAudioFlinger(audioFlinger),
+ mMemoryDealer(new MemoryDealer(1024*1024)),
+ mPid(pid)
+{
+ // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
+}
+
+AudioFlinger::Client::~Client()
+{
+ mAudioFlinger->removeClient(mPid);
+}
+
+const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+{
+ return mMemoryDealer;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackBase::TrackBase(
+ const sp<AudioFlinger>& audioFlinger,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int bufferCount,
+ int bufferSize)
+ : RefBase(),
+ mAudioFlinger(audioFlinger),
+ mClient(client),
+ mStreamType(streamType),
+ mFormat(format),
+ mChannelCount(channelCount),
+ mBufferCount(bufferCount),
+ mFlags(0),
+ mBufferSize(bufferSize),
+ mState(IDLE),
+ mClientTid(-1)
+{
+ mName = audioFlinger->audioMixer().getTrackName();
+ if (mName < 0) {
+ LOGE("no more track names availlable");
+ return;
+ }
+
+ // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
+ size_t size = sizeof(audio_track_cblk_t) + bufferCount * bufferSize;
+ mCblkMemory = client->heap()->allocate(size);
+ if (mCblkMemory != 0) {
+ mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
+ if (mCblk) { // construct the shared structure in-place.
+ new(mCblk) audio_track_cblk_t();
+ // clear all buffers
+ mCblk->size = bufferSize;
+ mCblk->sampleRate = sampleRate;
+ mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ memset(mBuffers, 0, bufferCount * bufferSize);
+ }
+ } else {
+ LOGE("not enough memory for AudioTrack size=%u", size);
+ client->heap()->dump("AudioTrack");
+ return;
+ }
+}
+
+AudioFlinger::TrackBase::~TrackBase()
+{
+ mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
+ mCblkMemory.clear(); // and free the shared memory
+ mClient.clear();
+}
+
+void AudioFlinger::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ step();
+}
+
+bool AudioFlinger::TrackBase::step() {
+ bool result;
+ audio_track_cblk_t* cblk = this->cblk();
+
+ result = cblk->stepServer(bufferCount());
+ if (!result) {
+ LOGV("stepServer failed acquiring cblk mutex");
+ mFlags |= STEPSERVER_FAILED;
+ }
+ return result;
+}
+
+void AudioFlinger::TrackBase::reset() {
+ audio_track_cblk_t* cblk = this->cblk();
+
+ cblk->user = 0;
+ cblk->server = 0;
+ mFlags = 0;
+}
+
+sp<IMemory> AudioFlinger::TrackBase::getCblk() const
+{
+ return mCblkMemory;
+}
+
+int AudioFlinger::TrackBase::sampleRate() const {
+ return mCblk->sampleRate;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::Track::Track(
+ const sp<AudioFlinger>& audioFlinger,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int bufferCount,
+ int bufferSize)
+ : TrackBase(audioFlinger, client, streamType, sampleRate, format, channelCount, bufferCount, bufferSize)
+{
+ mVolume[0] = 1.0f;
+ mVolume[1] = 1.0f;
+ mMute = false;
+}
+
+AudioFlinger::Track::~Track()
+{
+ wp<Track> weak(this); // never create a strong ref from the dtor
+ mState = TERMINATED;
+ mAudioFlinger->removeTrack(weak, mName);
+}
+
+void AudioFlinger::Track::destroy()
+{
+ mAudioFlinger->destroyTrack(this);
+}
+
+void AudioFlinger::Track::dump(char* buffer, size_t size)
+{
+ snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
+ mName - AudioMixer::TRACK0,
+ mClient->pid(),
+ mStreamType,
+ mFormat,
+ mChannelCount,
+ mBufferCount,
+ mState,
+ mMute,
+ mFillingUpStatus,
+ mCblk->sampleRate,
+ mCblk->volume[0],
+ mCblk->volume[1],
+ mCblk->server,
+ mCblk->user);
+}
+
+status_t AudioFlinger::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ audio_track_cblk_t* cblk = this->cblk();
+ uint32_t u = cblk->user;
+ uint32_t s = cblk->server;
+
+ // Check if last stepServer failed, try to step now
+ if (mFlags & TrackBase::STEPSERVER_FAILED) {
+ if (!step()) goto getNextBuffer_exit;
+ LOGV("stepServer recovered");
+ mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ }
+
+ if (LIKELY(u > s)) {
+ int index = s & audio_track_cblk_t::BUFFER_MASK;
+ buffer->raw = getBuffer(index);
+ buffer->frameCount = mAudioFlinger->frameCount();
+ return NO_ERROR;
+ }
+getNextBuffer_exit:
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+}
+
+bool AudioFlinger::Track::isReady(uint32_t u, int32_t s) const {
+ if (mFillingUpStatus != FS_FILLING) return true;
+ const uint32_t u_seq = u & audio_track_cblk_t::SEQUENCE_MASK;
+ const uint32_t u_buf = u & audio_track_cblk_t::BUFFER_MASK;
+ const uint32_t s_seq = s & audio_track_cblk_t::SEQUENCE_MASK;
+ const uint32_t s_buf = s & audio_track_cblk_t::BUFFER_MASK;
+ if (u_seq > s_seq && u_buf == s_buf) {
+ mFillingUpStatus = FS_FILLED;
+ return true;
+ }
+ return false;
+}
+
+status_t AudioFlinger::Track::start()
+{
+ LOGV("start(%d)", mName);
+ mAudioFlinger->addTrack(this);
+ return NO_ERROR;
+}
+
+void AudioFlinger::Track::stop()
+{
+ LOGV("stop(%d)", mName);
+ Mutex::Autolock _l(mAudioFlinger->mLock);
+ if (mState > STOPPED) {
+ mState = STOPPED;
+ // If the track is not active (PAUSED and buffers full), flush buffers
+ if (mAudioFlinger->mActiveTracks.indexOf(this) < 0) {
+ reset();
+ }
+ LOGV("(> STOPPED) => STOPPED (%d)", mName);
+ }
+}
+
+void AudioFlinger::Track::pause()
+{
+ LOGV("pause(%d)", mName);
+ Mutex::Autolock _l(mAudioFlinger->mLock);
+ if (mState == ACTIVE || mState == RESUMING) {
+ mState = PAUSING;
+ LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
+ }
+}
+
+void AudioFlinger::Track::flush()
+{
+ LOGV("flush(%d)", mName);
+ Mutex::Autolock _l(mAudioFlinger->mLock);
+ if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
+ return;
+ }
+ // No point remaining in PAUSED state after a flush => go to
+ // STOPPED state
+ mState = STOPPED;
+
+ // NOTE: reset() will reset cblk->user and cblk->server with
+ // the risk that at the same time, the AudioMixer is trying to read
+ // data. In this case, getNextBuffer() would return a NULL pointer
+ // as audio buffer => the AudioMixer code MUST always test that pointer
+ // returned by getNextBuffer() is not NULL!
+ reset();
+}
+
+void AudioFlinger::Track::reset()
+{
+ TrackBase::reset();
+ mFillingUpStatus = FS_FILLING;
+}
+
+void AudioFlinger::Track::mute(bool muted)
+{
+ mMute = muted;
+}
+
+void AudioFlinger::Track::setVolume(float left, float right)
+{
+ mVolume[0] = left;
+ mVolume[1] = right;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::Track>& track)
+ : BnAudioTrack(),
+ mTrack(track)
+{
+}
+
+AudioFlinger::TrackHandle::~TrackHandle() {
+ // just stop the track on deletion, associated resources
+ // will be freed from the main thread once all pending buffers have
+ // been played. Unless it's not in the active track list, in which
+ // case we free everything now...
+ mTrack->destroy();
+}
+
+status_t AudioFlinger::TrackHandle::start() {
+ return mTrack->start();
+}
+
+void AudioFlinger::TrackHandle::stop() {
+ mTrack->stop();
+}
+
+void AudioFlinger::TrackHandle::flush() {
+ mTrack->flush();
+}
+
+void AudioFlinger::TrackHandle::mute(bool e) {
+ mTrack->mute(e);
+}
+
+void AudioFlinger::TrackHandle::pause() {
+ mTrack->pause();
+}
+
+void AudioFlinger::TrackHandle::setVolume(float left, float right) {
+ mTrack->setVolume(left, right);
+}
+
+sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
+ return mTrack->getCblk();
+}
+
+status_t AudioFlinger::TrackHandle::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioTrack::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+sp<AudioFlinger::AudioRecordThread> AudioFlinger::audioRecordThread()
+{
+ Mutex::Autolock _l(mLock);
+ return mAudioRecordThread;
+}
+
+void AudioFlinger::endRecord()
+{
+ Mutex::Autolock _l(mLock);
+ mAudioRecordThread.clear();
+}
+
+sp<IAudioRecord> AudioFlinger::openRecord(
+ pid_t pid,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int bufferCount,
+ uint32_t flags)
+{
+ sp<AudioRecordThread> thread;
+ sp<RecordTrack> recordTrack;
+ sp<RecordHandle> recordHandle;
+ sp<Client> client;
+ wp<Client> wclient;
+ AudioStreamIn* input = 0;
+
+ // check calling permissions
+ if (!recordingAllowed()) {
+ goto Exit;
+ }
+
+ if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
+ LOGE("invalid stream type");
+ goto Exit;
+ }
+
+ if (sampleRate > MAX_SAMPLE_RATE) {
+ LOGE("Sample rate out of range");
+ goto Exit;
+ }
+
+ if (mSampleRate == 0) {
+ LOGE("Audio driver not initialized");
+ goto Exit;
+ }
+
+ // Create audio thread - take mutex to prevent race condition
+ {
+ Mutex::Autolock _l(mLock);
+ if (mAudioRecordThread != 0) {
+ LOGE("Record channel already open");
+ goto Exit;
+ }
+ thread = new AudioRecordThread(this);
+ mAudioRecordThread = thread;
+ }
+ // It's safe to release the mutex here since the client doesn't get a
+ // handle until we return from this call
+
+ // open driver, initialize h/w
+ input = mAudioHardware->openInputStream(
+ AudioSystem::PCM_16_BIT, channelCount, sampleRate);
+ if (!input) {
+ LOGE("Error opening input stream");
+ mAudioRecordThread.clear();
+ goto Exit;
+ }
+
+ // add client to list
+ {
+ Mutex::Autolock _l(mLock);
+ wclient = mClients.valueFor(pid);
+ if (wclient != NULL) {
+ client = wclient.promote();
+ } else {
+ client = new Client(this, pid);
+ mClients.add(pid, client);
+ }
+ }
+
+ // create new record track and pass to record thread
+ recordTrack = new RecordTrack(this, client, streamType, sampleRate,
+ format, channelCount, bufferCount, input->bufferSize());
+
+ // spin up record thread
+ thread->open(recordTrack, input);
+ thread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
+
+ // return to handle to client
+ recordHandle = new RecordHandle(recordTrack);
+
+Exit:
+ return recordHandle;
+}
+
+status_t AudioFlinger::startRecord() {
+ sp<AudioRecordThread> t = audioRecordThread();
+ if (t == 0) return NO_INIT;
+ return t->start();
+}
+
+void AudioFlinger::stopRecord() {
+ sp<AudioRecordThread> t = audioRecordThread();
+ if (t != 0) t->stop();
+}
+
+void AudioFlinger::exitRecord()
+{
+ sp<AudioRecordThread> t = audioRecordThread();
+ if (t != 0) t->exit();
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordTrack::RecordTrack(
+ const sp<AudioFlinger>& audioFlinger,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int bufferCount,
+ int bufferSize)
+ : TrackBase(audioFlinger, client, streamType, sampleRate, format,
+ channelCount, bufferCount, bufferSize),
+ mOverflow(false)
+{
+}
+
+AudioFlinger::RecordTrack::~RecordTrack()
+{
+ mAudioFlinger->audioMixer().deleteTrackName(mName);
+ mAudioFlinger->exitRecord();
+}
+
+status_t AudioFlinger::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ audio_track_cblk_t* cblk = this->cblk();
+ const uint32_t u_seq = cblk->user & audio_track_cblk_t::SEQUENCE_MASK;
+ const uint32_t u_buf = cblk->user & audio_track_cblk_t::BUFFER_MASK;
+ const uint32_t s_seq = cblk->server & audio_track_cblk_t::SEQUENCE_MASK;
+ const uint32_t s_buf = cblk->server & audio_track_cblk_t::BUFFER_MASK;
+
+ // Check if last stepServer failed, try to step now
+ if (mFlags & TrackBase::STEPSERVER_FAILED) {
+ if (!step()) goto getNextBuffer_exit;
+ LOGV("stepServer recovered");
+ mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ }
+
+ if (LIKELY(s_seq == u_seq || s_buf != u_buf)) {
+ buffer->raw = getBuffer(s_buf);
+ buffer->frameCount = mAudioFlinger->frameCount();
+ return NO_ERROR;
+ }
+
+getNextBuffer_exit:
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+}
+
+status_t AudioFlinger::RecordTrack::start()
+{
+ return mAudioFlinger->startRecord();
+}
+
+void AudioFlinger::RecordTrack::stop()
+{
+ mAudioFlinger->stopRecord();
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordTrack>& recordTrack)
+ : BnAudioRecord(),
+ mRecordTrack(recordTrack)
+{
+}
+
+AudioFlinger::RecordHandle::~RecordHandle() {}
+
+status_t AudioFlinger::RecordHandle::start() {
+ LOGV("RecordHandle::start()");
+ return mRecordTrack->start();
+}
+
+void AudioFlinger::RecordHandle::stop() {
+ LOGV("RecordHandle::stop()");
+ mRecordTrack->stop();
+}
+
+sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
+ return mRecordTrack->getCblk();
+}
+
+status_t AudioFlinger::RecordHandle::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioRecord::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioRecordThread::AudioRecordThread(const sp<AudioFlinger>& audioFlinger) :
+ mAudioFlinger(audioFlinger),
+ mRecordTrack(0),
+ mInput(0),
+ mActive(false)
+{
+}
+
+AudioFlinger::AudioRecordThread::~AudioRecordThread()
+{
+}
+
+bool AudioFlinger::AudioRecordThread::threadLoop()
+{
+ LOGV("AudioRecordThread: start record loop");
+
+ // start recording
+ while (!exitPending()) {
+ if (!mActive) {
+ mLock.lock();
+ if (!mActive && !exitPending()) {
+ LOGV("AudioRecordThread: loop stopping");
+ mWaitWorkCV.wait(mLock);
+ LOGV("AudioRecordThread: loop starting");
+ }
+ mLock.unlock();
+ } else {
+ // promote strong ref so track isn't deleted while we access it
+ sp<RecordTrack> t = mRecordTrack.promote();
+
+ // if we lose the weak reference, client is gone.
+ if (t == 0) {
+ LOGV("AudioRecordThread: client deleted track");
+ break;
+ }
+
+ if (LIKELY(t->getNextBuffer(&mBuffer) == NO_ERROR)) {
+ if (mInput->read(mBuffer.raw, t->mBufferSize) < 0) {
+ LOGE("Error reading audio input");
+ sleep(1);
+ }
+ t->releaseBuffer(&mBuffer);
+ }
+
+ // client isn't retrieving buffers fast enough
+ else {
+ if (!t->setOverflow())
+ LOGW("AudioRecordThread: buffer overflow");
+ }
+ }
+ };
+
+ // close hardware
+ close();
+
+ // delete this object - no more data references after this call
+ mAudioFlinger->endRecord();
+ return false;
+}
+
+status_t AudioFlinger::AudioRecordThread::open(const sp<RecordTrack>& recordTrack, AudioStreamIn *input) {
+ LOGV("AudioRecordThread::open");
+ // check for record channel already open
+ AutoMutex lock(&mLock);
+ if (mRecordTrack != NULL) {
+ LOGE("Record channel already open");
+ return ALREADY_EXISTS;
+ }
+ mRecordTrack = recordTrack;
+ mInput = input;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::AudioRecordThread::start()
+{
+ LOGV("AudioRecordThread::start");
+ AutoMutex lock(&mLock);
+ if (mActive) return -EBUSY;
+
+ sp<RecordTrack> t = mRecordTrack.promote();
+ if (t == 0) return UNKNOWN_ERROR;
+
+ // signal thread to start
+ LOGV("Signal record thread");
+ mActive = true;
+ mWaitWorkCV.signal();
+ return NO_ERROR;
+}
+
+void AudioFlinger::AudioRecordThread::stop() {
+ LOGV("AudioRecordThread::stop");
+ AutoMutex lock(&mLock);
+ if (mActive) {
+ mActive = false;
+ mWaitWorkCV.signal();
+ }
+}
+
+void AudioFlinger::AudioRecordThread::exit()
+{
+ LOGV("AudioRecordThread::exit");
+ AutoMutex lock(&mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+}
+
+
+status_t AudioFlinger::AudioRecordThread::close()
+{
+ LOGV("AudioRecordThread::close");
+ AutoMutex lock(&mLock);
+ if (!mInput) return NO_INIT;
+ delete mInput;
+ mInput = 0;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioFlinger::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+void AudioFlinger::instantiate() {
+ defaultServiceManager()->addService(
+ String16("media.audio_flinger"), new AudioFlinger());
+}
+
+}; // namespace android
generated by cgit v1.2.3 (git 2.25.1) at 2025-10-03 09:05:13 +0000