17500c2031
Merge commit '3893da46f0a97d59a7687ae2bd71ba855eb5ffe3' * commit '3893da46f0a97d59a7687ae2bd71ba855eb5ffe3': Fix issue 1970108: crash in AudioFlinger::isMusicActive()
2548 lines
81 KiB
C++
2548 lines
81 KiB
C++
/* //device/include/server/AudioFlinger/AudioFlinger.cpp
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**
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** Copyright 2007, The Android Open Source Project
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**
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** Licensed under the Apache License, Version 2.0 (the "License");
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** you may not use this file except in compliance with the License.
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** You may obtain a copy of the License at
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**
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** http://www.apache.org/licenses/LICENSE-2.0
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**
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** Unless required by applicable law or agreed to in writing, software
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** distributed under the License is distributed on an "AS IS" BASIS,
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** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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** See the License for the specific language governing permissions and
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** limitations under the License.
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*/
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#define LOG_TAG "AudioFlinger"
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//#define LOG_NDEBUG 0
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#include <math.h>
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#include <signal.h>
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#include <sys/time.h>
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#include <sys/resource.h>
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#include <binder/IServiceManager.h>
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#include <utils/Log.h>
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#include <binder/Parcel.h>
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#include <binder/IPCThreadState.h>
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#include <utils/String16.h>
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#include <utils/threads.h>
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#include <cutils/properties.h>
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#include <media/AudioTrack.h>
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#include <media/AudioRecord.h>
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#include <private/media/AudioTrackShared.h>
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#include <hardware_legacy/AudioHardwareInterface.h>
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#include "AudioMixer.h"
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#include "AudioFlinger.h"
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#ifdef WITH_A2DP
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#include "A2dpAudioInterface.h"
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#endif
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// ----------------------------------------------------------------------------
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// the sim build doesn't have gettid
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#ifndef HAVE_GETTID
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# define gettid getpid
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#endif
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// ----------------------------------------------------------------------------
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namespace android {
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static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
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static const char* kHardwareLockedString = "Hardware lock is taken\n";
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//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
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static const unsigned long kBufferRecoveryInUsecs = 2000;
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static const unsigned long kMaxBufferRecoveryInUsecs = 20000;
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static const float MAX_GAIN = 4096.0f;
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// retry counts for buffer fill timeout
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// 50 * ~20msecs = 1 second
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static const int8_t kMaxTrackRetries = 50;
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static const int8_t kMaxTrackStartupRetries = 50;
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static const int kStartSleepTime = 30000;
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static const int kStopSleepTime = 30000;
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static const int kDumpLockRetries = 50;
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static const int kDumpLockSleep = 20000;
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// Maximum number of pending buffers allocated by OutputTrack::write()
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static const uint8_t kMaxOutputTrackBuffers = 5;
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#define AUDIOFLINGER_SECURITY_ENABLED 1
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// ----------------------------------------------------------------------------
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static bool recordingAllowed() {
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#ifndef HAVE_ANDROID_OS
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return true;
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#endif
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#if AUDIOFLINGER_SECURITY_ENABLED
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if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
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bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
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if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
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return ok;
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#else
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if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
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LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
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return true;
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#endif
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}
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static bool settingsAllowed() {
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#ifndef HAVE_ANDROID_OS
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return true;
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#endif
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#if AUDIOFLINGER_SECURITY_ENABLED
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if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
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bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
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if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
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return ok;
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#else
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if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
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LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
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return true;
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#endif
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}
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// ----------------------------------------------------------------------------
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AudioFlinger::AudioFlinger()
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: BnAudioFlinger(),
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mAudioHardware(0), mA2dpAudioInterface(0), mA2dpEnabled(false), mNotifyA2dpChange(false),
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mForcedSpeakerCount(0), mA2dpDisableCount(0), mA2dpSuppressed(false), mForcedRoute(0),
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mRouteRestoreTime(0), mMusicMuteSaved(false)
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{
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mHardwareStatus = AUDIO_HW_IDLE;
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mAudioHardware = AudioHardwareInterface::create();
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mHardwareStatus = AUDIO_HW_INIT;
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if (mAudioHardware->initCheck() == NO_ERROR) {
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// open 16-bit output stream for s/w mixer
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mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
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status_t status;
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AudioStreamOut *hwOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
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mHardwareStatus = AUDIO_HW_IDLE;
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if (hwOutput) {
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mHardwareMixerThread = new MixerThread(this, hwOutput, AudioSystem::AUDIO_OUTPUT_HARDWARE);
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} else {
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LOGE("Failed to initialize hardware output stream, status: %d", status);
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}
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#ifdef WITH_A2DP
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// Create A2DP interface
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mA2dpAudioInterface = new A2dpAudioInterface();
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AudioStreamOut *a2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
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if (a2dpOutput) {
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mA2dpMixerThread = new MixerThread(this, a2dpOutput, AudioSystem::AUDIO_OUTPUT_A2DP);
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if (hwOutput) {
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uint32_t frameCount = ((a2dpOutput->bufferSize()/a2dpOutput->frameSize()) * hwOutput->sampleRate()) / a2dpOutput->sampleRate();
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MixerThread::OutputTrack *a2dpOutTrack = new MixerThread::OutputTrack(mA2dpMixerThread,
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hwOutput->sampleRate(),
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AudioSystem::PCM_16_BIT,
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hwOutput->channelCount(),
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frameCount);
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mHardwareMixerThread->setOuputTrack(a2dpOutTrack);
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}
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} else {
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LOGE("Failed to initialize A2DP output stream, status: %d", status);
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}
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#endif
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// FIXME - this should come from settings
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setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
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setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
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setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
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setMode(AudioSystem::MODE_NORMAL);
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setMasterVolume(1.0f);
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setMasterMute(false);
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// Start record thread
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mAudioRecordThread = new AudioRecordThread(mAudioHardware, this);
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if (mAudioRecordThread != 0) {
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mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
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}
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} else {
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LOGE("Couldn't even initialize the stubbed audio hardware!");
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}
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}
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AudioFlinger::~AudioFlinger()
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{
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if (mAudioRecordThread != 0) {
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mAudioRecordThread->exit();
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mAudioRecordThread.clear();
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}
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mHardwareMixerThread.clear();
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delete mAudioHardware;
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// deleting mA2dpAudioInterface also deletes mA2dpOutput;
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#ifdef WITH_A2DP
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mA2dpMixerThread.clear();
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delete mA2dpAudioInterface;
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#endif
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}
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#ifdef WITH_A2DP
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// setA2dpEnabled_l() must be called with AudioFlinger::mLock held
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void AudioFlinger::setA2dpEnabled_l(bool enable)
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{
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SortedVector < sp<MixerThread::Track> > tracks;
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SortedVector < wp<MixerThread::Track> > activeTracks;
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LOGV_IF(enable, "set output to A2DP\n");
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LOGV_IF(!enable, "set output to hardware audio\n");
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// Transfer tracks playing on MUSIC stream from one mixer to the other
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if (enable) {
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mHardwareMixerThread->getTracks_l(tracks, activeTracks);
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mA2dpMixerThread->putTracks_l(tracks, activeTracks);
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} else {
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mA2dpMixerThread->getTracks_l(tracks, activeTracks);
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mHardwareMixerThread->putTracks_l(tracks, activeTracks);
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}
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mA2dpEnabled = enable;
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mNotifyA2dpChange = true;
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mWaitWorkCV.broadcast();
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}
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// checkA2dpEnabledChange_l() must be called with AudioFlinger::mLock held
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void AudioFlinger::checkA2dpEnabledChange_l()
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{
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if (mNotifyA2dpChange) {
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// Notify AudioSystem of the A2DP activation/deactivation
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size_t size = mNotificationClients.size();
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for (size_t i = 0; i < size; i++) {
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sp<IBinder> binder = mNotificationClients.itemAt(i).promote();
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if (binder != NULL) {
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LOGV("Notifying output change to client %p", binder.get());
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sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
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client->a2dpEnabledChanged(mA2dpEnabled);
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}
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}
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mNotifyA2dpChange = false;
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}
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}
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#endif // WITH_A2DP
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bool AudioFlinger::streamForcedToSpeaker(int streamType)
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{
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// NOTE that streams listed here must not be routed to A2DP by default:
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// AudioSystem::routedToA2dpOutput(streamType) == false
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return (streamType == AudioSystem::RING ||
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streamType == AudioSystem::ALARM ||
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streamType == AudioSystem::NOTIFICATION ||
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streamType == AudioSystem::ENFORCED_AUDIBLE);
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}
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status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
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{
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const size_t SIZE = 256;
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char buffer[SIZE];
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String8 result;
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result.append("Clients:\n");
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for (size_t i = 0; i < mClients.size(); ++i) {
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wp<Client> wClient = mClients.valueAt(i);
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if (wClient != 0) {
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sp<Client> client = wClient.promote();
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if (client != 0) {
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snprintf(buffer, SIZE, " pid: %d\n", client->pid());
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result.append(buffer);
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}
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}
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}
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write(fd, result.string(), result.size());
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return NO_ERROR;
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}
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status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
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{
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const size_t SIZE = 256;
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char buffer[SIZE];
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String8 result;
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int hardwareStatus = mHardwareStatus;
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if (hardwareStatus == AUDIO_HW_IDLE && mHardwareMixerThread->mStandby) {
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hardwareStatus = AUDIO_HW_STANDBY;
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}
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snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
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result.append(buffer);
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write(fd, result.string(), result.size());
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return NO_ERROR;
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}
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status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
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{
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const size_t SIZE = 256;
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char buffer[SIZE];
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String8 result;
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snprintf(buffer, SIZE, "Permission Denial: "
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"can't dump AudioFlinger from pid=%d, uid=%d\n",
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IPCThreadState::self()->getCallingPid(),
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IPCThreadState::self()->getCallingUid());
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result.append(buffer);
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write(fd, result.string(), result.size());
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return NO_ERROR;
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}
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static bool tryLock(Mutex& mutex)
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{
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bool locked = false;
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for (int i = 0; i < kDumpLockRetries; ++i) {
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if (mutex.tryLock() == NO_ERROR) {
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locked = true;
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break;
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}
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usleep(kDumpLockSleep);
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}
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return locked;
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}
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status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
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{
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if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
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dumpPermissionDenial(fd, args);
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} else {
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// get state of hardware lock
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bool hardwareLocked = tryLock(mHardwareLock);
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if (!hardwareLocked) {
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String8 result(kHardwareLockedString);
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write(fd, result.string(), result.size());
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} else {
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mHardwareLock.unlock();
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}
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bool locked = tryLock(mLock);
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// failed to lock - AudioFlinger is probably deadlocked
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if (!locked) {
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String8 result(kDeadlockedString);
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write(fd, result.string(), result.size());
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}
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dumpClients(fd, args);
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dumpInternals(fd, args);
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mHardwareMixerThread->dump(fd, args);
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#ifdef WITH_A2DP
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mA2dpMixerThread->dump(fd, args);
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#endif
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// dump record client
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if (mAudioRecordThread != 0) mAudioRecordThread->dump(fd, args);
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if (mAudioHardware) {
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mAudioHardware->dumpState(fd, args);
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}
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if (locked) mLock.unlock();
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}
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return NO_ERROR;
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}
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// IAudioFlinger interface
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sp<IAudioTrack> AudioFlinger::createTrack(
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pid_t pid,
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int streamType,
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uint32_t sampleRate,
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int format,
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int channelCount,
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int frameCount,
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uint32_t flags,
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const sp<IMemory>& sharedBuffer,
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status_t *status)
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{
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sp<MixerThread::Track> track;
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sp<TrackHandle> trackHandle;
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sp<Client> client;
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wp<Client> wclient;
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status_t lStatus;
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if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
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LOGE("invalid stream type");
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lStatus = BAD_VALUE;
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goto Exit;
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}
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{
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Mutex::Autolock _l(mLock);
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wclient = mClients.valueFor(pid);
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if (wclient != NULL) {
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client = wclient.promote();
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} else {
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client = new Client(this, pid);
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mClients.add(pid, client);
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}
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#ifdef WITH_A2DP
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if (isA2dpEnabled() && AudioSystem::routedToA2dpOutput(streamType)) {
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track = mA2dpMixerThread->createTrack_l(client, streamType, sampleRate, format,
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channelCount, frameCount, sharedBuffer, &lStatus);
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} else
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#endif
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{
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track = mHardwareMixerThread->createTrack_l(client, streamType, sampleRate, format,
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channelCount, frameCount, sharedBuffer, &lStatus);
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}
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}
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if (lStatus == NO_ERROR) {
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trackHandle = new TrackHandle(track);
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} else {
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track.clear();
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}
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Exit:
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if(status) {
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*status = lStatus;
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}
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return trackHandle;
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}
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uint32_t AudioFlinger::sampleRate(int output) const
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{
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#ifdef WITH_A2DP
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if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
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return mA2dpMixerThread->sampleRate();
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}
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#endif
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return mHardwareMixerThread->sampleRate();
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}
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int AudioFlinger::channelCount(int output) const
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{
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#ifdef WITH_A2DP
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if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
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return mA2dpMixerThread->channelCount();
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}
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#endif
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return mHardwareMixerThread->channelCount();
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}
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int AudioFlinger::format(int output) const
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{
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#ifdef WITH_A2DP
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if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
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return mA2dpMixerThread->format();
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}
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#endif
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return mHardwareMixerThread->format();
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}
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size_t AudioFlinger::frameCount(int output) const
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{
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#ifdef WITH_A2DP
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if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
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return mA2dpMixerThread->frameCount();
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}
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#endif
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return mHardwareMixerThread->frameCount();
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}
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uint32_t AudioFlinger::latency(int output) const
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{
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#ifdef WITH_A2DP
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if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
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return mA2dpMixerThread->latency();
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}
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#endif
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return mHardwareMixerThread->latency();
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}
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status_t AudioFlinger::setMasterVolume(float value)
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{
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// check calling permissions
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if (!settingsAllowed()) {
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return PERMISSION_DENIED;
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}
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// when hw supports master volume, don't scale in sw mixer
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AutoMutex lock(mHardwareLock);
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mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
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if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
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value = 1.0f;
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}
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mHardwareStatus = AUDIO_HW_IDLE;
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mHardwareMixerThread->setMasterVolume(value);
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#ifdef WITH_A2DP
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mA2dpMixerThread->setMasterVolume(value);
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#endif
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return NO_ERROR;
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}
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status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
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{
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status_t err = NO_ERROR;
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// check calling permissions
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if (!settingsAllowed()) {
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return PERMISSION_DENIED;
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}
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if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
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LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
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return BAD_VALUE;
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}
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#ifdef WITH_A2DP
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LOGV("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(),
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IPCThreadState::self()->getCallingPid());
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if (mode == AudioSystem::MODE_NORMAL &&
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(mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
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AutoMutex lock(&mLock);
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bool enableA2dp = false;
|
|
if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) {
|
|
enableA2dp = true;
|
|
}
|
|
if (mA2dpDisableCount > 0) {
|
|
mA2dpSuppressed = enableA2dp;
|
|
} else {
|
|
setA2dpEnabled_l(enableA2dp);
|
|
}
|
|
LOGV("setOutput done\n");
|
|
}
|
|
// setRouting() is always called at least for mode == AudioSystem::MODE_IN_CALL when
|
|
// SCO is enabled, whatever current mode is so we can safely handle A2DP disabling only
|
|
// in this case to avoid doing it several times.
|
|
if (mode == AudioSystem::MODE_IN_CALL &&
|
|
(mask & AudioSystem::ROUTE_BLUETOOTH_SCO)) {
|
|
AutoMutex lock(&mLock);
|
|
handleRouteDisablesA2dp_l(routes);
|
|
}
|
|
#endif
|
|
|
|
// do nothing if only A2DP routing is affected
|
|
mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP;
|
|
if (mask) {
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_HW_GET_ROUTING;
|
|
uint32_t r;
|
|
err = mAudioHardware->getRouting(mode, &r);
|
|
if (err == NO_ERROR) {
|
|
r = (r & ~mask) | (routes & mask);
|
|
if (mode == AudioSystem::MODE_NORMAL ||
|
|
(mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
|
|
mSavedRoute = r;
|
|
r |= mForcedRoute;
|
|
LOGV("setRouting mSavedRoute %08x mForcedRoute %08x\n", mSavedRoute, mForcedRoute);
|
|
}
|
|
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)) {
|
|
if (mode == AudioSystem::MODE_NORMAL ||
|
|
(mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
|
|
routes = mSavedRoute;
|
|
} else {
|
|
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;
|
|
}
|
|
mHardwareMixerThread->setMasterMute(muted);
|
|
#ifdef WITH_A2DP
|
|
mA2dpMixerThread->setMasterMute(muted);
|
|
#endif
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::masterVolume() const
|
|
{
|
|
return mHardwareMixerThread->masterVolume();
|
|
}
|
|
|
|
bool AudioFlinger::masterMute() const
|
|
{
|
|
return mHardwareMixerThread->masterMute();
|
|
}
|
|
|
|
status_t AudioFlinger::setStreamVolume(int stream, float value)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
|
|
uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
status_t ret = NO_ERROR;
|
|
|
|
if (stream == AudioSystem::VOICE_CALL ||
|
|
stream == AudioSystem::BLUETOOTH_SCO) {
|
|
float hwValue;
|
|
if (stream == AudioSystem::VOICE_CALL) {
|
|
hwValue = (float)AudioSystem::logToLinear(value)/100.0f;
|
|
// offset value to reflect actual hardware volume that never reaches 0
|
|
// 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
|
|
value = 0.01 + 0.99 * value;
|
|
} else { // (type == AudioSystem::BLUETOOTH_SCO)
|
|
hwValue = 1.0f;
|
|
}
|
|
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
|
|
ret = mAudioHardware->setVoiceVolume(hwValue);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
|
|
}
|
|
|
|
mHardwareMixerThread->setStreamVolume(stream, value);
|
|
#ifdef WITH_A2DP
|
|
mA2dpMixerThread->setStreamVolume(stream, value);
|
|
#endif
|
|
|
|
mHardwareMixerThread->setStreamVolume(stream, value);
|
|
#ifdef WITH_A2DP
|
|
mA2dpMixerThread->setStreamVolume(stream, value);
|
|
#endif
|
|
|
|
return ret;
|
|
}
|
|
|
|
status_t AudioFlinger::setStreamMute(int stream, bool muted)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
|
|
uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
#ifdef WITH_A2DP
|
|
mA2dpMixerThread->setStreamMute(stream, muted);
|
|
#endif
|
|
if (stream == AudioSystem::MUSIC)
|
|
{
|
|
AutoMutex lock(&mHardwareLock);
|
|
if (mForcedRoute != 0)
|
|
mMusicMuteSaved = muted;
|
|
else
|
|
mHardwareMixerThread->setStreamMute(stream, muted);
|
|
} else {
|
|
mHardwareMixerThread->setStreamMute(stream, muted);
|
|
}
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::streamVolume(int stream) const
|
|
{
|
|
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
|
|
return 0.0f;
|
|
}
|
|
|
|
float volume = mHardwareMixerThread->streamVolume(stream);
|
|
// remove correction applied by setStreamVolume()
|
|
if (stream == AudioSystem::VOICE_CALL) {
|
|
volume = (volume - 0.01) / 0.99 ;
|
|
}
|
|
|
|
return volume;
|
|
}
|
|
|
|
bool AudioFlinger::streamMute(int stream) const
|
|
{
|
|
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
|
|
return true;
|
|
}
|
|
|
|
if (stream == AudioSystem::MUSIC && mForcedRoute != 0)
|
|
{
|
|
return mMusicMuteSaved;
|
|
}
|
|
return mHardwareMixerThread->streamMute(stream);
|
|
}
|
|
|
|
bool AudioFlinger::isMusicActive() const
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
#ifdef WITH_A2DP
|
|
if (isA2dpEnabled()) {
|
|
return mA2dpMixerThread->isMusicActive_l();
|
|
}
|
|
#endif
|
|
return mHardwareMixerThread->isMusicActive_l();
|
|
}
|
|
|
|
status_t AudioFlinger::setParameter(const char* key, const char* value)
|
|
{
|
|
status_t result, result2;
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_SET_PARAMETER;
|
|
|
|
LOGV("setParameter() key %s, value %s, tid %d, calling tid %d", key, value, gettid(), IPCThreadState::self()->getCallingPid());
|
|
result = mAudioHardware->setParameter(key, value);
|
|
if (mA2dpAudioInterface) {
|
|
result2 = mA2dpAudioInterface->setParameter(key, value);
|
|
if (result2)
|
|
result = result2;
|
|
}
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return result;
|
|
}
|
|
|
|
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
|
|
{
|
|
return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
|
|
}
|
|
|
|
void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
|
|
{
|
|
|
|
LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
sp<IBinder> binder = client->asBinder();
|
|
if (mNotificationClients.indexOf(binder) < 0) {
|
|
LOGV("Adding notification client %p", binder.get());
|
|
binder->linkToDeath(this);
|
|
mNotificationClients.add(binder);
|
|
client->a2dpEnabledChanged(isA2dpEnabled());
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::binderDied(const wp<IBinder>& who) {
|
|
|
|
LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
IBinder *binder = who.unsafe_get();
|
|
|
|
if (binder != NULL) {
|
|
int index = mNotificationClients.indexOf(binder);
|
|
if (index >= 0) {
|
|
LOGV("Removing notification client %p", binder);
|
|
mNotificationClients.removeAt(index);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::removeClient(pid_t pid)
|
|
{
|
|
LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
|
|
Mutex::Autolock _l(mLock);
|
|
mClients.removeItem(pid);
|
|
}
|
|
|
|
bool AudioFlinger::isA2dpEnabled() const
|
|
{
|
|
return mA2dpEnabled;
|
|
}
|
|
|
|
void AudioFlinger::handleForcedSpeakerRoute(int command)
|
|
{
|
|
switch(command) {
|
|
case ACTIVE_TRACK_ADDED:
|
|
{
|
|
AutoMutex lock(mHardwareLock);
|
|
if (mForcedSpeakerCount++ == 0) {
|
|
if (mForcedRoute == 0) {
|
|
mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
|
|
LOGV("++mForcedSpeakerCount == 0, mMusicMuteSaved = %d, mRouteRestoreTime = %d", mMusicMuteSaved, mRouteRestoreTime);
|
|
if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
|
|
LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
|
|
mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
|
|
usleep(mHardwareMixerThread->latency()*1000);
|
|
mHardwareStatus = AUDIO_HW_SET_ROUTING;
|
|
mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
// delay track start so that audio hardware has time to siwtch routes
|
|
usleep(kStartSleepTime);
|
|
}
|
|
}
|
|
mForcedRoute = AudioSystem::ROUTE_SPEAKER;
|
|
mRouteRestoreTime = 0;
|
|
}
|
|
LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount);
|
|
}
|
|
break;
|
|
case ACTIVE_TRACK_REMOVED:
|
|
{
|
|
AutoMutex lock(mHardwareLock);
|
|
if (mForcedSpeakerCount > 0){
|
|
if (--mForcedSpeakerCount == 0) {
|
|
mRouteRestoreTime = systemTime() + milliseconds(kStopSleepTime/1000);
|
|
}
|
|
LOGV("mForcedSpeakerCount decremented to %d", mForcedSpeakerCount);
|
|
} else {
|
|
LOGE("mForcedSpeakerCount is already zero");
|
|
}
|
|
}
|
|
break;
|
|
case CHECK_ROUTE_RESTORE_TIME:
|
|
case FORCE_ROUTE_RESTORE:
|
|
if (mRouteRestoreTime) {
|
|
AutoMutex lock(mHardwareLock);
|
|
if (mRouteRestoreTime &&
|
|
(systemTime() > mRouteRestoreTime || command == FORCE_ROUTE_RESTORE)) {
|
|
mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, mMusicMuteSaved);
|
|
mForcedRoute = 0;
|
|
if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
|
|
mHardwareStatus = AUDIO_HW_SET_ROUTING;
|
|
mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
LOGV("Route forced to Speaker OFF %08x", mSavedRoute);
|
|
}
|
|
mRouteRestoreTime = 0;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
#ifdef WITH_A2DP
|
|
// handleRouteDisablesA2dp_l() must be called with AudioFlinger::mLock held
|
|
void AudioFlinger::handleRouteDisablesA2dp_l(int routes)
|
|
{
|
|
if (routes & AudioSystem::ROUTE_BLUETOOTH_SCO) {
|
|
if (mA2dpDisableCount++ == 0) {
|
|
if (mA2dpEnabled) {
|
|
setA2dpEnabled_l(false);
|
|
mA2dpSuppressed = true;
|
|
}
|
|
}
|
|
LOGV("mA2dpDisableCount incremented to %d", mA2dpDisableCount);
|
|
} else {
|
|
if (mA2dpDisableCount > 0) {
|
|
if (--mA2dpDisableCount == 0) {
|
|
if (mA2dpSuppressed) {
|
|
setA2dpEnabled_l(true);
|
|
mA2dpSuppressed = false;
|
|
}
|
|
}
|
|
LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount);
|
|
} else {
|
|
LOGV("mA2dpDisableCount is already zero");
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int outputType)
|
|
: Thread(false),
|
|
mAudioFlinger(audioFlinger), mAudioMixer(0), mOutput(output), mOutputType(outputType),
|
|
mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0),
|
|
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false),
|
|
mInWrite(false)
|
|
{
|
|
mSampleRate = output->sampleRate();
|
|
mChannelCount = output->channelCount();
|
|
|
|
// FIXME - Current mixer implementation only supports stereo output
|
|
if (mChannelCount == 1) {
|
|
LOGE("Invalid audio hardware channel count");
|
|
}
|
|
|
|
mFormat = output->format();
|
|
mFrameCount = output->bufferSize() / output->channelCount() / sizeof(int16_t);
|
|
mAudioMixer = new AudioMixer(mFrameCount, output->sampleRate());
|
|
|
|
// FIXME - Current mixer implementation only supports stereo output: Always
|
|
// Allocate a stereo buffer even if HW output is mono.
|
|
mMixBuffer = new int16_t[mFrameCount * 2];
|
|
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
|
|
}
|
|
|
|
AudioFlinger::MixerThread::~MixerThread()
|
|
{
|
|
delete [] mMixBuffer;
|
|
delete mAudioMixer;
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::dump(int fd, const Vector<String16>& args)
|
|
{
|
|
dumpInternals(fd, args);
|
|
dumpTracks(fd, args);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector<String16>& args)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, "Output %d mixer thread tracks\n", mOutputType);
|
|
result.append(buffer);
|
|
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) {
|
|
sp<Track> track = mTracks[i];
|
|
if (track != 0) {
|
|
track->dump(buffer, SIZE);
|
|
result.append(buffer);
|
|
}
|
|
}
|
|
|
|
snprintf(buffer, SIZE, "Output %d mixer thread active tracks\n", mOutputType);
|
|
result.append(buffer);
|
|
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 = mActiveTracks[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::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, "Output %d mixer thread internals\n", mOutputType);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->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);
|
|
write(fd, result.string(), result.size());
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// Thread virtuals
|
|
bool AudioFlinger::MixerThread::threadLoop()
|
|
{
|
|
unsigned long sleepTime = kBufferRecoveryInUsecs;
|
|
int16_t* curBuf = mMixBuffer;
|
|
Vector< sp<Track> > tracksToRemove;
|
|
size_t enabledTracks = 0;
|
|
nsecs_t standbyTime = systemTime();
|
|
size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
|
|
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
|
|
|
|
#ifdef WITH_A2DP
|
|
bool outputTrackActive = false;
|
|
#endif
|
|
|
|
do {
|
|
enabledTracks = 0;
|
|
{ // scope for the AudioFlinger::mLock
|
|
|
|
Mutex::Autolock _l(mAudioFlinger->mLock);
|
|
|
|
#ifdef WITH_A2DP
|
|
if (mOutputTrack != NULL && !mAudioFlinger->isA2dpEnabled()) {
|
|
if (outputTrackActive) {
|
|
mAudioFlinger->mLock.unlock();
|
|
mOutputTrack->stop();
|
|
mAudioFlinger->mLock.lock();
|
|
outputTrackActive = false;
|
|
}
|
|
}
|
|
mAudioFlinger->checkA2dpEnabledChange_l();
|
|
#endif
|
|
|
|
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, output %d\n", mOutputType);
|
|
if (!mStandby) {
|
|
mOutput->standby();
|
|
mStandby = true;
|
|
}
|
|
|
|
#ifdef WITH_A2DP
|
|
if (outputTrackActive) {
|
|
mAudioFlinger->mLock.unlock();
|
|
mOutputTrack->stop();
|
|
mAudioFlinger->mLock.lock();
|
|
outputTrackActive = false;
|
|
}
|
|
#endif
|
|
if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
|
|
mAudioFlinger->handleForcedSpeakerRoute(FORCE_ROUTE_RESTORE);
|
|
}
|
|
// we're about to wait, flush the binder command buffer
|
|
IPCThreadState::self()->flushCommands();
|
|
mAudioFlinger->mWaitWorkCV.wait(mAudioFlinger->mLock);
|
|
LOGV("Audio hardware exiting standby, output %d\n", mOutputType);
|
|
|
|
if (mMasterMute == false) {
|
|
char value[PROPERTY_VALUE_MAX];
|
|
property_get("ro.audio.silent", value, "0");
|
|
if (atoi(value)) {
|
|
LOGD("Silence is golden");
|
|
setMasterMute(true);
|
|
}
|
|
}
|
|
|
|
standbyTime = systemTime() + kStandbyTimeInNsecs;
|
|
continue;
|
|
}
|
|
|
|
// Forced route to speaker is handled by hardware mixer thread
|
|
if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
|
|
mAudioFlinger->handleForcedSpeakerRoute(CHECK_ROUTE_RESTORE_TIME);
|
|
}
|
|
|
|
// 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();
|
|
|
|
// The first time a track is added we wait
|
|
// for all its buffers to be filled before processing it
|
|
mAudioMixer->setActiveTrack(track->name());
|
|
if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
|
|
!track->isPaused())
|
|
{
|
|
//LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
|
|
|
|
// 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
|
|
mAudioMixer->setBufferProvider(track);
|
|
mAudioMixer->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;
|
|
}
|
|
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
|
|
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
|
|
mAudioMixer->setParameter(
|
|
AudioMixer::TRACK,
|
|
AudioMixer::FORMAT, track->format());
|
|
mAudioMixer->setParameter(
|
|
AudioMixer::TRACK,
|
|
AudioMixer::CHANNEL_COUNT, track->channelCount());
|
|
mAudioMixer->setParameter(
|
|
AudioMixer::RESAMPLE,
|
|
AudioMixer::SAMPLE_RATE,
|
|
int(cblk->sampleRate));
|
|
|
|
// reset retry count
|
|
track->mRetryCount = kMaxTrackRetries;
|
|
enabledTracks++;
|
|
} else {
|
|
//LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
|
|
if (track->isStopped()) {
|
|
track->reset();
|
|
}
|
|
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());
|
|
mAudioMixer->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];
|
|
removeActiveTrack_l(track);
|
|
if (track->isTerminated()) {
|
|
mTracks.remove(track);
|
|
deleteTrackName_l(track->mName);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (LIKELY(enabledTracks)) {
|
|
// mix buffers...
|
|
mAudioMixer->process(curBuf);
|
|
|
|
#ifdef WITH_A2DP
|
|
if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
|
|
if (!outputTrackActive) {
|
|
LOGV("starting output track in mixer for output %d", mOutputType);
|
|
mOutputTrack->start();
|
|
outputTrackActive = true;
|
|
}
|
|
mOutputTrack->write(curBuf, mFrameCount);
|
|
}
|
|
#endif
|
|
|
|
// 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 {
|
|
#ifdef WITH_A2DP
|
|
if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
|
|
if (outputTrackActive) {
|
|
mOutputTrack->write(curBuf, 0);
|
|
if (mOutputTrack->bufferQueueEmpty()) {
|
|
mOutputTrack->stop();
|
|
outputTrackActive = false;
|
|
} else {
|
|
standbyTime = systemTime() + kStandbyTimeInNsecs;
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
// 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::MixerThread::readyToRun()
|
|
{
|
|
if (mSampleRate == 0) {
|
|
LOGE("No working audio driver found.");
|
|
return NO_INIT;
|
|
}
|
|
LOGI("AudioFlinger's thread ready to run for output %d", mOutputType);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::onFirstRef()
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
|
|
snprintf(buffer, SIZE, "Mixer Thread for output %d", mOutputType);
|
|
|
|
run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
|
|
}
|
|
|
|
// MixerThread::createTrack_l() must be called with AudioFlinger::mLock held
|
|
sp<AudioFlinger::MixerThread::Track> AudioFlinger::MixerThread::createTrack_l(
|
|
const sp<AudioFlinger::Client>& client,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
const sp<IMemory>& sharedBuffer,
|
|
status_t *status)
|
|
{
|
|
sp<Track> track;
|
|
status_t lStatus;
|
|
|
|
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
|
|
if (sampleRate > mSampleRate*2) {
|
|
LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
|
|
lStatus = BAD_VALUE;
|
|
goto Exit;
|
|
}
|
|
|
|
|
|
if (mSampleRate == 0) {
|
|
LOGE("Audio driver not initialized.");
|
|
lStatus = NO_INIT;
|
|
goto Exit;
|
|
}
|
|
|
|
track = new Track(this, client, streamType, sampleRate, format,
|
|
channelCount, frameCount, sharedBuffer);
|
|
if (track->getCblk() == NULL) {
|
|
lStatus = NO_MEMORY;
|
|
goto Exit;
|
|
}
|
|
mTracks.add(track);
|
|
lStatus = NO_ERROR;
|
|
|
|
Exit:
|
|
if(status) {
|
|
*status = lStatus;
|
|
}
|
|
return track;
|
|
}
|
|
|
|
// getTracks_l() must be called with AudioFlinger::mLock held
|
|
void AudioFlinger::MixerThread::getTracks_l(
|
|
SortedVector < sp<Track> >& tracks,
|
|
SortedVector < wp<Track> >& activeTracks)
|
|
{
|
|
size_t size = mTracks.size();
|
|
LOGV ("MixerThread::getTracks_l() for output %d, mTracks.size %d, mActiveTracks.size %d", mOutputType, mTracks.size(), mActiveTracks.size());
|
|
for (size_t i = 0; i < size; i++) {
|
|
sp<Track> t = mTracks[i];
|
|
if (AudioSystem::routedToA2dpOutput(t->mStreamType)) {
|
|
tracks.add(t);
|
|
int j = mActiveTracks.indexOf(t);
|
|
if (j >= 0) {
|
|
t = mActiveTracks[j].promote();
|
|
if (t != NULL) {
|
|
activeTracks.add(t);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
size = activeTracks.size();
|
|
for (size_t i = 0; i < size; i++) {
|
|
removeActiveTrack_l(activeTracks[i]);
|
|
}
|
|
|
|
size = tracks.size();
|
|
for (size_t i = 0; i < size; i++) {
|
|
sp<Track> t = tracks[i];
|
|
mTracks.remove(t);
|
|
deleteTrackName_l(t->name());
|
|
}
|
|
}
|
|
|
|
// putTracks_l() must be called with AudioFlinger::mLock held
|
|
void AudioFlinger::MixerThread::putTracks_l(
|
|
SortedVector < sp<Track> >& tracks,
|
|
SortedVector < wp<Track> >& activeTracks)
|
|
{
|
|
|
|
LOGV ("MixerThread::putTracks_l() for output %d, tracks.size %d, activeTracks.size %d", mOutputType, tracks.size(), activeTracks.size());
|
|
|
|
size_t size = tracks.size();
|
|
for (size_t i = 0; i < size ; i++) {
|
|
sp<Track> t = tracks[i];
|
|
int name = getTrackName_l();
|
|
|
|
if (name < 0) return;
|
|
|
|
t->mName = name;
|
|
t->mMixerThread = this;
|
|
mTracks.add(t);
|
|
|
|
int j = activeTracks.indexOf(t);
|
|
if (j >= 0) {
|
|
addActiveTrack_l(t);
|
|
}
|
|
}
|
|
}
|
|
|
|
uint32_t AudioFlinger::MixerThread::sampleRate() const
|
|
{
|
|
return mSampleRate;
|
|
}
|
|
|
|
int AudioFlinger::MixerThread::channelCount() const
|
|
{
|
|
return mChannelCount;
|
|
}
|
|
|
|
int AudioFlinger::MixerThread::format() const
|
|
{
|
|
return mFormat;
|
|
}
|
|
|
|
size_t AudioFlinger::MixerThread::frameCount() const
|
|
{
|
|
return mFrameCount;
|
|
}
|
|
|
|
uint32_t AudioFlinger::MixerThread::latency() const
|
|
{
|
|
if (mOutput) {
|
|
return mOutput->latency();
|
|
}
|
|
else {
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::setMasterVolume(float value)
|
|
{
|
|
mMasterVolume = value;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::setMasterMute(bool muted)
|
|
{
|
|
mMasterMute = muted;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::MixerThread::masterVolume() const
|
|
{
|
|
return mMasterVolume;
|
|
}
|
|
|
|
bool AudioFlinger::MixerThread::masterMute() const
|
|
{
|
|
return mMasterMute;
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::setStreamVolume(int stream, float value)
|
|
{
|
|
mStreamTypes[stream].volume = value;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::setStreamMute(int stream, bool muted)
|
|
{
|
|
mStreamTypes[stream].mute = muted;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::MixerThread::streamVolume(int stream) const
|
|
{
|
|
return mStreamTypes[stream].volume;
|
|
}
|
|
|
|
bool AudioFlinger::MixerThread::streamMute(int stream) const
|
|
{
|
|
return mStreamTypes[stream].mute;
|
|
}
|
|
|
|
// isMusicActive_l() must be called with AudioFlinger::mLock held
|
|
bool AudioFlinger::MixerThread::isMusicActive_l() 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 == AudioSystem::MUSIC)
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
// addTrack_l() must be called with AudioFlinger::mLock held
|
|
status_t AudioFlinger::MixerThread::addTrack_l(const sp<Track>& track)
|
|
{
|
|
status_t status = ALREADY_EXISTS;
|
|
|
|
// 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;
|
|
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;
|
|
track->mResetDone = false;
|
|
addActiveTrack_l(track);
|
|
status = NO_ERROR;
|
|
}
|
|
|
|
LOGV("mWaitWorkCV.broadcast");
|
|
mAudioFlinger->mWaitWorkCV.broadcast();
|
|
|
|
return status;
|
|
}
|
|
|
|
// destroyTrack_l() must be called with AudioFlinger::mLock held
|
|
void AudioFlinger::MixerThread::destroyTrack_l(const sp<Track>& track)
|
|
{
|
|
track->mState = TrackBase::TERMINATED;
|
|
if (mActiveTracks.indexOf(track) < 0) {
|
|
LOGV("remove track (%d) and delete from mixer", track->name());
|
|
mTracks.remove(track);
|
|
deleteTrackName_l(track->name());
|
|
}
|
|
}
|
|
|
|
// addActiveTrack_l() must be called with AudioFlinger::mLock held
|
|
void AudioFlinger::MixerThread::addActiveTrack_l(const wp<Track>& t)
|
|
{
|
|
mActiveTracks.add(t);
|
|
|
|
// Force routing to speaker for certain stream types
|
|
// The forced routing to speaker is managed by hardware mixer
|
|
if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
|
|
sp<Track> track = t.promote();
|
|
if (track == NULL) return;
|
|
|
|
if (streamForcedToSpeaker(track->type())) {
|
|
mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_ADDED);
|
|
}
|
|
}
|
|
}
|
|
|
|
// removeActiveTrack_l() must be called with AudioFlinger::mLock held
|
|
void AudioFlinger::MixerThread::removeActiveTrack_l(const wp<Track>& t)
|
|
{
|
|
mActiveTracks.remove(t);
|
|
|
|
// Force routing to speaker for certain stream types
|
|
// The forced routing to speaker is managed by hardware mixer
|
|
if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
|
|
sp<Track> track = t.promote();
|
|
if (track == NULL) return;
|
|
|
|
if (streamForcedToSpeaker(track->type())) {
|
|
mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_REMOVED);
|
|
}
|
|
}
|
|
}
|
|
|
|
// getTrackName_l() must be called with AudioFlinger::mLock held
|
|
int AudioFlinger::MixerThread::getTrackName_l()
|
|
{
|
|
return mAudioMixer->getTrackName();
|
|
}
|
|
|
|
// deleteTrackName_l() must be called with AudioFlinger::mLock held
|
|
void AudioFlinger::MixerThread::deleteTrackName_l(int name)
|
|
{
|
|
mAudioMixer->deleteTrackName(name);
|
|
}
|
|
|
|
size_t AudioFlinger::MixerThread::getOutputFrameCount()
|
|
{
|
|
return mOutput->bufferSize() / mOutput->channelCount() / sizeof(int16_t);
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
// TrackBase constructor must be called with AudioFlinger::mLock held
|
|
AudioFlinger::MixerThread::TrackBase::TrackBase(
|
|
const sp<MixerThread>& mixerThread,
|
|
const sp<Client>& client,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags,
|
|
const sp<IMemory>& sharedBuffer)
|
|
: RefBase(),
|
|
mMixerThread(mixerThread),
|
|
mClient(client),
|
|
mFrameCount(0),
|
|
mState(IDLE),
|
|
mClientTid(-1),
|
|
mFormat(format),
|
|
mFlags(flags & ~SYSTEM_FLAGS_MASK)
|
|
{
|
|
mName = mixerThread->getTrackName_l();
|
|
LOGV("TrackBase contructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
|
|
if (mName < 0) {
|
|
LOGE("no more track names availlable");
|
|
return;
|
|
}
|
|
|
|
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
|
|
|
|
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
|
|
size_t size = sizeof(audio_track_cblk_t);
|
|
size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
|
|
if (sharedBuffer == 0) {
|
|
size += bufferSize;
|
|
}
|
|
|
|
if (client != NULL) {
|
|
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->frameCount = frameCount;
|
|
mCblk->sampleRate = sampleRate;
|
|
mCblk->channels = (uint8_t)channelCount;
|
|
if (sharedBuffer == 0) {
|
|
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
|
|
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
|
|
// Force underrun condition to avoid false underrun callback until first data is
|
|
// written to buffer
|
|
mCblk->flowControlFlag = 1;
|
|
} else {
|
|
mBuffer = sharedBuffer->pointer();
|
|
}
|
|
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
|
|
}
|
|
} else {
|
|
LOGE("not enough memory for AudioTrack size=%u", size);
|
|
client->heap()->dump("AudioTrack");
|
|
return;
|
|
}
|
|
} else {
|
|
mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
|
|
if (mCblk) { // construct the shared structure in-place.
|
|
new(mCblk) audio_track_cblk_t();
|
|
// clear all buffers
|
|
mCblk->frameCount = frameCount;
|
|
mCblk->sampleRate = sampleRate;
|
|
mCblk->channels = (uint8_t)channelCount;
|
|
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
|
|
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
|
|
// Force underrun condition to avoid false underrun callback until first data is
|
|
// written to buffer
|
|
mCblk->flowControlFlag = 1;
|
|
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
|
|
}
|
|
}
|
|
}
|
|
|
|
AudioFlinger::MixerThread::TrackBase::~TrackBase()
|
|
{
|
|
if (mCblk) {
|
|
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
|
|
}
|
|
mCblkMemory.clear(); // and free the shared memory
|
|
mClient.clear();
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
buffer->raw = 0;
|
|
mFrameCount = buffer->frameCount;
|
|
step();
|
|
buffer->frameCount = 0;
|
|
}
|
|
|
|
bool AudioFlinger::MixerThread::TrackBase::step() {
|
|
bool result;
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
|
|
result = cblk->stepServer(mFrameCount);
|
|
if (!result) {
|
|
LOGV("stepServer failed acquiring cblk mutex");
|
|
mFlags |= STEPSERVER_FAILED;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::TrackBase::reset() {
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
|
|
cblk->user = 0;
|
|
cblk->server = 0;
|
|
cblk->userBase = 0;
|
|
cblk->serverBase = 0;
|
|
mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
|
|
LOGV("TrackBase::reset");
|
|
}
|
|
|
|
sp<IMemory> AudioFlinger::MixerThread::TrackBase::getCblk() const
|
|
{
|
|
return mCblkMemory;
|
|
}
|
|
|
|
int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
|
|
return (int)mCblk->sampleRate;
|
|
}
|
|
|
|
int AudioFlinger::MixerThread::TrackBase::channelCount() const {
|
|
return (int)mCblk->channels;
|
|
}
|
|
|
|
void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels;
|
|
int16_t *bufferEnd = bufferStart + frames * cblk->channels;
|
|
|
|
// Check validity of returned pointer in case the track control block would have been corrupted.
|
|
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
|
|
(cblk->channels == 2 && ((unsigned long)bufferStart & 3))) {
|
|
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
|
|
server %d, serverBase %d, user %d, userBase %d, channels %d",
|
|
bufferStart, bufferEnd, mBuffer, mBufferEnd,
|
|
cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channels);
|
|
return 0;
|
|
}
|
|
|
|
return bufferStart;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
// Track constructor must be called with AudioFlinger::mLock held
|
|
AudioFlinger::MixerThread::Track::Track(
|
|
const sp<MixerThread>& mixerThread,
|
|
const sp<Client>& client,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
const sp<IMemory>& sharedBuffer)
|
|
: TrackBase(mixerThread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
|
|
{
|
|
mVolume[0] = 1.0f;
|
|
mVolume[1] = 1.0f;
|
|
mMute = false;
|
|
mSharedBuffer = sharedBuffer;
|
|
mStreamType = streamType;
|
|
}
|
|
|
|
AudioFlinger::MixerThread::Track::~Track()
|
|
{
|
|
wp<Track> weak(this); // never create a strong ref from the dtor
|
|
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
|
|
mState = TERMINATED;
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::Track::destroy()
|
|
{
|
|
// NOTE: destroyTrack_l() can remove a strong reference to this Track
|
|
// by removing it from mTracks vector, so there is a risk that this Tracks's
|
|
// desctructor is called. As the destructor needs to lock AudioFlinger::mLock,
|
|
// we must acquire a strong reference on this Track before locking AudioFlinger::mLock
|
|
// here so that the destructor is called only when exiting this function.
|
|
// On the other hand, as long as Track::destroy() is only called by
|
|
// TrackHandle destructor, the TrackHandle still holds a strong ref on
|
|
// this Track with its member mTrack.
|
|
sp<Track> keep(this);
|
|
{ // scope for AudioFlinger::mLock
|
|
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
|
|
mMixerThread->destroyTrack_l(this);
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::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 == NULL) ? getpid() : mClient->pid(),
|
|
mStreamType,
|
|
mFormat,
|
|
mCblk->channels,
|
|
mFrameCount,
|
|
mState,
|
|
mMute,
|
|
mFillingUpStatus,
|
|
mCblk->sampleRate,
|
|
mCblk->volume[0],
|
|
mCblk->volume[1],
|
|
mCblk->server,
|
|
mCblk->user);
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
uint32_t framesReady;
|
|
uint32_t framesReq = buffer->frameCount;
|
|
|
|
// 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;
|
|
}
|
|
|
|
framesReady = cblk->framesReady();
|
|
|
|
if (LIKELY(framesReady)) {
|
|
uint32_t s = cblk->server;
|
|
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
|
|
|
|
bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
|
|
if (framesReq > framesReady) {
|
|
framesReq = framesReady;
|
|
}
|
|
if (s + framesReq > bufferEnd) {
|
|
framesReq = bufferEnd - s;
|
|
}
|
|
|
|
buffer->raw = getBuffer(s, framesReq);
|
|
if (buffer->raw == 0) goto getNextBuffer_exit;
|
|
|
|
buffer->frameCount = framesReq;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
getNextBuffer_exit:
|
|
buffer->raw = 0;
|
|
buffer->frameCount = 0;
|
|
return NOT_ENOUGH_DATA;
|
|
}
|
|
|
|
bool AudioFlinger::MixerThread::Track::isReady() const {
|
|
if (mFillingUpStatus != FS_FILLING) return true;
|
|
|
|
if (mCblk->framesReady() >= mCblk->frameCount ||
|
|
mCblk->forceReady) {
|
|
mFillingUpStatus = FS_FILLED;
|
|
mCblk->forceReady = 0;
|
|
LOGV("Track::isReady() track %d for output %d", mName, mMixerThread->mOutputType);
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::Track::start()
|
|
{
|
|
LOGV("start(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
|
|
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
|
|
mMixerThread->addTrack_l(this);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::Track::stop()
|
|
{
|
|
LOGV("stop(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
|
|
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
|
|
if (mState > STOPPED) {
|
|
mState = STOPPED;
|
|
// If the track is not active (PAUSED and buffers full), flush buffers
|
|
if (mMixerThread->mActiveTracks.indexOf(this) < 0) {
|
|
reset();
|
|
}
|
|
LOGV("(> STOPPED) => STOPPED (%d)", mName);
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::Track::pause()
|
|
{
|
|
LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
|
|
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
|
|
if (mState == ACTIVE || mState == RESUMING) {
|
|
mState = PAUSING;
|
|
LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::Track::flush()
|
|
{
|
|
LOGV("flush(%d)", mName);
|
|
Mutex::Autolock _l(mMixerThread->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;
|
|
|
|
mCblk->lock.lock();
|
|
// 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();
|
|
mCblk->lock.unlock();
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::Track::reset()
|
|
{
|
|
// Do not reset twice to avoid discarding data written just after a flush and before
|
|
// the audioflinger thread detects the track is stopped.
|
|
if (!mResetDone) {
|
|
TrackBase::reset();
|
|
// Force underrun condition to avoid false underrun callback until first data is
|
|
// written to buffer
|
|
mCblk->flowControlFlag = 1;
|
|
mCblk->forceReady = 0;
|
|
mFillingUpStatus = FS_FILLING;
|
|
mResetDone = true;
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::Track::mute(bool muted)
|
|
{
|
|
mMute = muted;
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
|
|
{
|
|
mVolume[0] = left;
|
|
mVolume[1] = right;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
// RecordTrack constructor must be called with AudioFlinger::mLock held
|
|
AudioFlinger::MixerThread::RecordTrack::RecordTrack(
|
|
const sp<MixerThread>& mixerThread,
|
|
const sp<Client>& client,
|
|
int inputSource,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags)
|
|
: TrackBase(mixerThread, client, sampleRate, format,
|
|
channelCount, frameCount, flags, 0),
|
|
mOverflow(false), mInputSource(inputSource)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::MixerThread::RecordTrack::~RecordTrack()
|
|
{
|
|
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
|
|
mMixerThread->deleteTrackName_l(mName);
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
uint32_t framesAvail;
|
|
uint32_t framesReq = buffer->frameCount;
|
|
|
|
// 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;
|
|
}
|
|
|
|
framesAvail = cblk->framesAvailable_l();
|
|
|
|
if (LIKELY(framesAvail)) {
|
|
uint32_t s = cblk->server;
|
|
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
|
|
|
|
if (framesReq > framesAvail) {
|
|
framesReq = framesAvail;
|
|
}
|
|
if (s + framesReq > bufferEnd) {
|
|
framesReq = bufferEnd - s;
|
|
}
|
|
|
|
buffer->raw = getBuffer(s, framesReq);
|
|
if (buffer->raw == 0) goto getNextBuffer_exit;
|
|
|
|
buffer->frameCount = framesReq;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
getNextBuffer_exit:
|
|
buffer->raw = 0;
|
|
buffer->frameCount = 0;
|
|
return NOT_ENOUGH_DATA;
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::RecordTrack::start()
|
|
{
|
|
return mMixerThread->mAudioFlinger->startRecord(this);
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::RecordTrack::stop()
|
|
{
|
|
mMixerThread->mAudioFlinger->stopRecord(this);
|
|
TrackBase::reset();
|
|
// Force overerrun condition to avoid false overrun callback until first data is
|
|
// read from buffer
|
|
mCblk->flowControlFlag = 1;
|
|
}
|
|
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::MixerThread::OutputTrack::OutputTrack(
|
|
const sp<MixerThread>& mixerThread,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount)
|
|
: Track(mixerThread, NULL, AudioSystem::SYSTEM, sampleRate, format, channelCount, frameCount, NULL),
|
|
mOutputMixerThread(mixerThread)
|
|
{
|
|
|
|
mCblk->out = 1;
|
|
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
|
|
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
|
|
mOutBuffer.frameCount = 0;
|
|
mCblk->bufferTimeoutMs = 10;
|
|
|
|
LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p",
|
|
mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd);
|
|
|
|
}
|
|
|
|
AudioFlinger::MixerThread::OutputTrack::~OutputTrack()
|
|
{
|
|
stop();
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::OutputTrack::start()
|
|
{
|
|
status_t status = Track::start();
|
|
|
|
mRetryCount = 127;
|
|
return status;
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::OutputTrack::stop()
|
|
{
|
|
Track::stop();
|
|
clearBufferQueue();
|
|
mOutBuffer.frameCount = 0;
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frames)
|
|
{
|
|
Buffer *pInBuffer;
|
|
Buffer inBuffer;
|
|
uint32_t channels = mCblk->channels;
|
|
|
|
inBuffer.frameCount = frames;
|
|
inBuffer.i16 = data;
|
|
|
|
if (mCblk->user == 0) {
|
|
mOutputMixerThread->mAudioFlinger->mLock.lock();
|
|
bool isMusicActive = mOutputMixerThread->isMusicActive_l();
|
|
mOutputMixerThread->mAudioFlinger->mLock.unlock();
|
|
if (isMusicActive) {
|
|
mCblk->forceReady = 1;
|
|
LOGV("OutputTrack::start() force ready");
|
|
} else if (mCblk->frameCount > frames){
|
|
if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
|
|
uint32_t startFrames = (mCblk->frameCount - frames);
|
|
LOGV("OutputTrack::start() write %d frames", startFrames);
|
|
pInBuffer = new Buffer;
|
|
pInBuffer->mBuffer = new int16_t[startFrames * channels];
|
|
pInBuffer->frameCount = startFrames;
|
|
pInBuffer->i16 = pInBuffer->mBuffer;
|
|
memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
|
|
mBufferQueue.add(pInBuffer);
|
|
} else {
|
|
LOGW ("OutputTrack::write() no more buffers");
|
|
}
|
|
}
|
|
}
|
|
|
|
while (1) {
|
|
// First write pending buffers, then new data
|
|
if (mBufferQueue.size()) {
|
|
pInBuffer = mBufferQueue.itemAt(0);
|
|
} else {
|
|
pInBuffer = &inBuffer;
|
|
}
|
|
|
|
if (pInBuffer->frameCount == 0) {
|
|
break;
|
|
}
|
|
|
|
if (mOutBuffer.frameCount == 0) {
|
|
mOutBuffer.frameCount = pInBuffer->frameCount;
|
|
if (obtainBuffer(&mOutBuffer) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
|
|
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));
|
|
mCblk->stepUser(outFrames);
|
|
pInBuffer->frameCount -= outFrames;
|
|
pInBuffer->i16 += outFrames * channels;
|
|
mOutBuffer.frameCount -= outFrames;
|
|
mOutBuffer.i16 += outFrames * channels;
|
|
|
|
if (pInBuffer->frameCount == 0) {
|
|
if (mBufferQueue.size()) {
|
|
mBufferQueue.removeAt(0);
|
|
delete [] pInBuffer->mBuffer;
|
|
delete pInBuffer;
|
|
} else {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
// If we could not write all frames, allocate a buffer and queue it for next time.
|
|
if (inBuffer.frameCount) {
|
|
if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
|
|
pInBuffer = new Buffer;
|
|
pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels];
|
|
pInBuffer->frameCount = inBuffer.frameCount;
|
|
pInBuffer->i16 = pInBuffer->mBuffer;
|
|
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t));
|
|
mBufferQueue.add(pInBuffer);
|
|
} else {
|
|
LOGW("OutputTrack::write() no more buffers");
|
|
}
|
|
}
|
|
|
|
// Calling write() with a 0 length buffer, means that no more data will be written:
|
|
// If no more buffers are pending, fill output track buffer to make sure it is started
|
|
// by output mixer.
|
|
if (frames == 0 && mBufferQueue.size() == 0 && mCblk->user < mCblk->frameCount) {
|
|
frames = mCblk->frameCount - mCblk->user;
|
|
pInBuffer = new Buffer;
|
|
pInBuffer->mBuffer = new int16_t[frames * channels];
|
|
pInBuffer->frameCount = frames;
|
|
pInBuffer->i16 = pInBuffer->mBuffer;
|
|
memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
|
|
mBufferQueue.add(pInBuffer);
|
|
}
|
|
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
int active;
|
|
int timeout = 0;
|
|
status_t result;
|
|
audio_track_cblk_t* cblk = mCblk;
|
|
uint32_t framesReq = buffer->frameCount;
|
|
|
|
LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
|
|
buffer->frameCount = 0;
|
|
|
|
uint32_t framesAvail = cblk->framesAvailable();
|
|
|
|
if (framesAvail == 0) {
|
|
return AudioTrack::NO_MORE_BUFFERS;
|
|
}
|
|
|
|
if (framesReq > framesAvail) {
|
|
framesReq = framesAvail;
|
|
}
|
|
|
|
uint32_t u = cblk->user;
|
|
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
|
|
|
|
if (u + framesReq > bufferEnd) {
|
|
framesReq = bufferEnd - u;
|
|
}
|
|
|
|
buffer->frameCount = framesReq;
|
|
buffer->raw = (void *)cblk->buffer(u);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
|
|
void AudioFlinger::MixerThread::OutputTrack::clearBufferQueue()
|
|
{
|
|
size_t size = mBufferQueue.size();
|
|
Buffer *pBuffer;
|
|
|
|
for (size_t i = 0; i < size; i++) {
|
|
pBuffer = mBufferQueue.itemAt(i);
|
|
delete [] pBuffer->mBuffer;
|
|
delete pBuffer;
|
|
}
|
|
mBufferQueue.clear();
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
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::TrackHandle::TrackHandle(const sp<AudioFlinger::MixerThread::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<IAudioRecord> AudioFlinger::openRecord(
|
|
pid_t pid,
|
|
int inputSource,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags,
|
|
status_t *status)
|
|
{
|
|
sp<MixerThread::RecordTrack> recordTrack;
|
|
sp<RecordHandle> recordHandle;
|
|
sp<Client> client;
|
|
wp<Client> wclient;
|
|
AudioStreamIn* input = 0;
|
|
int inFrameCount;
|
|
size_t inputBufferSize;
|
|
status_t lStatus;
|
|
|
|
// check calling permissions
|
|
if (!recordingAllowed()) {
|
|
lStatus = PERMISSION_DENIED;
|
|
goto Exit;
|
|
}
|
|
|
|
if (uint32_t(inputSource) >= AudioRecord::NUM_INPUT_SOURCES) {
|
|
LOGE("invalid stream type");
|
|
lStatus = BAD_VALUE;
|
|
goto Exit;
|
|
}
|
|
|
|
if (mAudioRecordThread == 0) {
|
|
LOGE("Audio record thread not started");
|
|
lStatus = NO_INIT;
|
|
goto Exit;
|
|
}
|
|
|
|
|
|
// Check that audio input stream accepts requested audio parameters
|
|
inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
|
|
if (inputBufferSize == 0) {
|
|
lStatus = BAD_VALUE;
|
|
LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount);
|
|
goto Exit;
|
|
}
|
|
|
|
// add client to list
|
|
{ // scope for mLock
|
|
Mutex::Autolock _l(mLock);
|
|
wclient = mClients.valueFor(pid);
|
|
if (wclient != NULL) {
|
|
client = wclient.promote();
|
|
} else {
|
|
client = new Client(this, pid);
|
|
mClients.add(pid, client);
|
|
}
|
|
|
|
// frameCount must be a multiple of input buffer size
|
|
inFrameCount = inputBufferSize/channelCount/sizeof(short);
|
|
frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
|
|
|
|
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
|
|
recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, inputSource, sampleRate,
|
|
format, channelCount, frameCount, flags);
|
|
}
|
|
if (recordTrack->getCblk() == NULL) {
|
|
recordTrack.clear();
|
|
lStatus = NO_MEMORY;
|
|
goto Exit;
|
|
}
|
|
|
|
// return to handle to client
|
|
recordHandle = new RecordHandle(recordTrack);
|
|
lStatus = NO_ERROR;
|
|
|
|
Exit:
|
|
if (status) {
|
|
*status = lStatus;
|
|
}
|
|
return recordHandle;
|
|
}
|
|
|
|
status_t AudioFlinger::startRecord(MixerThread::RecordTrack* recordTrack) {
|
|
if (mAudioRecordThread != 0) {
|
|
return mAudioRecordThread->start(recordTrack);
|
|
}
|
|
return NO_INIT;
|
|
}
|
|
|
|
void AudioFlinger::stopRecord(MixerThread::RecordTrack* recordTrack) {
|
|
if (mAudioRecordThread != 0) {
|
|
mAudioRecordThread->stop(recordTrack);
|
|
}
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::MixerThread::RecordTrack>& recordTrack)
|
|
: BnAudioRecord(),
|
|
mRecordTrack(recordTrack)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::RecordHandle::~RecordHandle() {
|
|
stop();
|
|
}
|
|
|
|
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(AudioHardwareInterface* audioHardware,
|
|
const sp<AudioFlinger>& audioFlinger) :
|
|
mAudioHardware(audioHardware),
|
|
mAudioFlinger(audioFlinger),
|
|
mActive(false)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::AudioRecordThread::~AudioRecordThread()
|
|
{
|
|
}
|
|
|
|
bool AudioFlinger::AudioRecordThread::threadLoop()
|
|
{
|
|
LOGV("AudioRecordThread: start record loop");
|
|
AudioBufferProvider::Buffer buffer;
|
|
int inBufferSize = 0;
|
|
int inFrameCount = 0;
|
|
AudioStreamIn* input = 0;
|
|
|
|
mActive = 0;
|
|
|
|
// start recording
|
|
while (!exitPending()) {
|
|
if (!mActive) {
|
|
mLock.lock();
|
|
if (!mActive && !exitPending()) {
|
|
LOGV("AudioRecordThread: loop stopping");
|
|
if (input) {
|
|
delete input;
|
|
input = 0;
|
|
}
|
|
mRecordTrack.clear();
|
|
mStopped.signal();
|
|
|
|
mWaitWorkCV.wait(mLock);
|
|
|
|
LOGV("AudioRecordThread: loop starting");
|
|
if (mRecordTrack != 0) {
|
|
input = mAudioHardware->openInputStream(
|
|
mRecordTrack->inputSource(),
|
|
mRecordTrack->format(),
|
|
mRecordTrack->channelCount(),
|
|
mRecordTrack->sampleRate(),
|
|
&mStartStatus,
|
|
(AudioSystem::audio_in_acoustics)(mRecordTrack->mFlags >> 16));
|
|
if (input != 0) {
|
|
inBufferSize = input->bufferSize();
|
|
inFrameCount = inBufferSize/input->frameSize();
|
|
}
|
|
} else {
|
|
mStartStatus = NO_INIT;
|
|
}
|
|
if (mStartStatus !=NO_ERROR) {
|
|
LOGW("record start failed, status %d", mStartStatus);
|
|
mActive = false;
|
|
mRecordTrack.clear();
|
|
}
|
|
mWaitWorkCV.signal();
|
|
}
|
|
mLock.unlock();
|
|
} else if (mRecordTrack != 0) {
|
|
|
|
buffer.frameCount = inFrameCount;
|
|
if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR &&
|
|
(int)buffer.frameCount == inFrameCount)) {
|
|
LOGV("AudioRecordThread read: %d frames", buffer.frameCount);
|
|
ssize_t bytesRead = input->read(buffer.raw, inBufferSize);
|
|
if (bytesRead < 0) {
|
|
LOGE("Error reading audio input");
|
|
sleep(1);
|
|
}
|
|
mRecordTrack->releaseBuffer(&buffer);
|
|
mRecordTrack->overflow();
|
|
}
|
|
|
|
// client isn't retrieving buffers fast enough
|
|
else {
|
|
if (!mRecordTrack->setOverflow())
|
|
LOGW("AudioRecordThread: buffer overflow");
|
|
// Release the processor for a while before asking for a new buffer.
|
|
// This will give the application more chance to read from the buffer and
|
|
// clear the overflow.
|
|
usleep(5000);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
if (input) {
|
|
delete input;
|
|
}
|
|
mRecordTrack.clear();
|
|
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::AudioRecordThread::start(MixerThread::RecordTrack* recordTrack)
|
|
{
|
|
LOGV("AudioRecordThread::start");
|
|
AutoMutex lock(&mLock);
|
|
mActive = true;
|
|
// If starting the active track, just reset mActive in case a stop
|
|
// was pending and exit
|
|
if (recordTrack == mRecordTrack.get()) return NO_ERROR;
|
|
|
|
if (mRecordTrack != 0) return -EBUSY;
|
|
|
|
mRecordTrack = recordTrack;
|
|
|
|
// signal thread to start
|
|
LOGV("Signal record thread");
|
|
mWaitWorkCV.signal();
|
|
mWaitWorkCV.wait(mLock);
|
|
LOGV("Record started, status %d", mStartStatus);
|
|
return mStartStatus;
|
|
}
|
|
|
|
void AudioFlinger::AudioRecordThread::stop(MixerThread::RecordTrack* recordTrack) {
|
|
LOGV("AudioRecordThread::stop");
|
|
AutoMutex lock(&mLock);
|
|
if (mActive && (recordTrack == mRecordTrack.get())) {
|
|
mActive = false;
|
|
mStopped.wait(mLock);
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::AudioRecordThread::exit()
|
|
{
|
|
LOGV("AudioRecordThread::exit");
|
|
{
|
|
AutoMutex lock(&mLock);
|
|
requestExit();
|
|
mWaitWorkCV.signal();
|
|
}
|
|
requestExitAndWait();
|
|
}
|
|
|
|
status_t AudioFlinger::AudioRecordThread::dump(int fd, const Vector<String16>& args)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
pid_t pid = 0;
|
|
|
|
if (mRecordTrack != 0 && mRecordTrack->mClient != 0) {
|
|
snprintf(buffer, SIZE, "Record client pid: %d\n", mRecordTrack->mClient->pid());
|
|
result.append(buffer);
|
|
} else {
|
|
result.append("No record client\n");
|
|
}
|
|
write(fd, result.string(), result.size());
|
|
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
|