3522c80819
Add a parameter to ToneGenerator.startTone() allowing the caller to specify the tone duration. This is used by the phone application to have a precise control on the DTMF tone duration which was not possible with the use of delayed messaged. Also modified AudioFlinger output threads so that 0s are written to the audio output stream when no more tracks are ready to mix instead of just sleeping. This avoids an issue where the end of a previous DTMF tone could stay in audio hardware buffers and be played just before the beginning of the next DTMF tone.
3582 lines
116 KiB
C++
3582 lines
116 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 kDumpLockRetries = 50;
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static const int kDumpLockSleep = 20000;
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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), mMasterVolume(1.0f), mMasterMute(false), mNextThreadId(0)
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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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setMode(AudioSystem::MODE_NORMAL);
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setMasterVolume(1.0f);
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setMasterMute(false);
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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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while (!mRecordThreads.isEmpty()) {
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// closeInput() will remove first entry from mRecordThreads
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closeInput(mRecordThreads.keyAt(0));
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}
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while (!mPlaybackThreads.isEmpty()) {
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// closeOutput() will remove first entry from mPlaybackThreads
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closeOutput(mPlaybackThreads.keyAt(0));
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}
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if (mAudioHardware) {
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delete mAudioHardware;
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}
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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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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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// dump playback threads
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for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
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mPlaybackThreads.valueAt(i)->dump(fd, args);
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}
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// dump record threads
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for (size_t i = 0; i < mRecordThreads.size(); i++) {
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mRecordThreads.valueAt(i)->dump(fd, args);
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}
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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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int output,
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status_t *status)
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{
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sp<PlaybackThread::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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PlaybackThread *thread = checkPlaybackThread_l(output);
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if (thread == NULL) {
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LOGE("unknown output thread");
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lStatus = BAD_VALUE;
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goto Exit;
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}
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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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track = thread->createTrack_l(client, streamType, sampleRate, format,
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channelCount, frameCount, sharedBuffer, &lStatus);
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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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Mutex::Autolock _l(mLock);
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PlaybackThread *thread = checkPlaybackThread_l(output);
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if (thread == NULL) {
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LOGW("sampleRate() unknown thread %d", output);
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return 0;
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}
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return thread->sampleRate();
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}
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int AudioFlinger::channelCount(int output) const
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{
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Mutex::Autolock _l(mLock);
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PlaybackThread *thread = checkPlaybackThread_l(output);
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if (thread == NULL) {
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LOGW("channelCount() unknown thread %d", output);
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return 0;
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}
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return thread->channelCount();
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}
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int AudioFlinger::format(int output) const
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{
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Mutex::Autolock _l(mLock);
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PlaybackThread *thread = checkPlaybackThread_l(output);
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if (thread == NULL) {
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LOGW("format() unknown thread %d", output);
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return 0;
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}
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return thread->format();
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}
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size_t AudioFlinger::frameCount(int output) const
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{
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Mutex::Autolock _l(mLock);
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PlaybackThread *thread = checkPlaybackThread_l(output);
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if (thread == NULL) {
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LOGW("frameCount() unknown thread %d", output);
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return 0;
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}
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return thread->frameCount();
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}
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uint32_t AudioFlinger::latency(int output) const
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{
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Mutex::Autolock _l(mLock);
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PlaybackThread *thread = checkPlaybackThread_l(output);
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if (thread == NULL) {
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LOGW("latency() unknown thread %d", output);
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return 0;
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}
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return thread->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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mMasterVolume = value;
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for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
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mPlaybackThreads.valueAt(i)->setMasterVolume(value);
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return NO_ERROR;
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}
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status_t AudioFlinger::setMode(int mode)
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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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if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
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LOGW("Illegal value: setMode(%d)", mode);
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return BAD_VALUE;
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}
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|
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AutoMutex lock(mHardwareLock);
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mHardwareStatus = AUDIO_HW_SET_MODE;
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status_t ret = mAudioHardware->setMode(mode);
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mHardwareStatus = AUDIO_HW_IDLE;
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return ret;
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}
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status_t AudioFlinger::setMicMute(bool state)
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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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AutoMutex lock(mHardwareLock);
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mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
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status_t ret = mAudioHardware->setMicMute(state);
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mHardwareStatus = AUDIO_HW_IDLE;
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return ret;
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}
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bool AudioFlinger::getMicMute() const
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{
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bool state = AudioSystem::MODE_INVALID;
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mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
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mAudioHardware->getMicMute(&state);
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mHardwareStatus = AUDIO_HW_IDLE;
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return state;
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}
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status_t AudioFlinger::setMasterMute(bool muted)
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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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|
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mMasterMute = muted;
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for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
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mPlaybackThreads.valueAt(i)->setMasterMute(muted);
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|
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return NO_ERROR;
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}
|
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|
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float AudioFlinger::masterVolume() const
|
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{
|
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return mMasterVolume;
|
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}
|
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|
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bool AudioFlinger::masterMute() const
|
|
{
|
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return mMasterMute;
|
|
}
|
|
|
|
status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
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return PERMISSION_DENIED;
|
|
}
|
|
|
|
if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
|
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return BAD_VALUE;
|
|
}
|
|
|
|
AutoMutex lock(mLock);
|
|
PlaybackThread *thread = NULL;
|
|
if (output) {
|
|
thread = checkPlaybackThread_l(output);
|
|
if (thread == NULL) {
|
|
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)
|
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hwValue = 1.0f;
|
|
}
|
|
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
|
|
ret = mAudioHardware->setVoiceVolume(hwValue);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
|
|
}
|
|
|
|
mStreamTypes[stream].volume = value;
|
|
|
|
if (thread == NULL) {
|
|
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
|
|
mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
|
|
|
|
} else {
|
|
thread->setStreamVolume(stream, value);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
status_t AudioFlinger::setStreamMute(int stream, bool muted)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
|
|
uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
mStreamTypes[stream].mute = muted;
|
|
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
|
|
mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::streamVolume(int stream, int output) const
|
|
{
|
|
if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
|
|
return 0.0f;
|
|
}
|
|
|
|
AutoMutex lock(mLock);
|
|
float volume;
|
|
if (output) {
|
|
PlaybackThread *thread = checkPlaybackThread_l(output);
|
|
if (thread == NULL) {
|
|
return 0.0f;
|
|
}
|
|
volume = thread->streamVolume(stream);
|
|
} else {
|
|
volume = mStreamTypes[stream].volume;
|
|
}
|
|
|
|
// 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 (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) {
|
|
return true;
|
|
}
|
|
|
|
return mStreamTypes[stream].mute;
|
|
}
|
|
|
|
bool AudioFlinger::isMusicActive() const
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
|
|
if (mPlaybackThreads.valueAt(i)->isMusicActive()) {
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
|
|
{
|
|
status_t result;
|
|
|
|
LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
|
|
ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
// ioHandle == 0 means the parameters are global to the audio hardware interface
|
|
if (ioHandle == 0) {
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_SET_PARAMETER;
|
|
result = mAudioHardware->setParameters(keyValuePairs);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return result;
|
|
}
|
|
|
|
// Check if parameters are for an output
|
|
PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
|
|
if (playbackThread != NULL) {
|
|
return playbackThread->setParameters(keyValuePairs);
|
|
}
|
|
|
|
// Check if parameters are for an input
|
|
RecordThread *recordThread = checkRecordThread_l(ioHandle);
|
|
if (recordThread != NULL) {
|
|
return recordThread->setParameters(keyValuePairs);
|
|
}
|
|
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
|
|
{
|
|
// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
|
|
// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
|
|
|
|
if (ioHandle == 0) {
|
|
return mAudioHardware->getParameters(keys);
|
|
}
|
|
PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
|
|
if (playbackThread != NULL) {
|
|
return playbackThread->getParameters(keys);
|
|
}
|
|
RecordThread *recordThread = checkRecordThread_l(ioHandle);
|
|
if (recordThread != NULL) {
|
|
return recordThread->getParameters(keys);
|
|
}
|
|
return String8("");
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
// the config change is always sent from playback or record threads to avoid deadlock
|
|
// with AudioSystem::gLock
|
|
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
|
|
mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
|
|
}
|
|
|
|
for (size_t i = 0; i < mRecordThreads.size(); i++) {
|
|
mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
|
|
}
|
|
}
|
|
|
|
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::audioConfigChanged(int event, const sp<ThreadBase>& thread, void *param2) {
|
|
Mutex::Autolock _l(mLock);
|
|
int ioHandle = 0;
|
|
|
|
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
|
|
if (mPlaybackThreads.valueAt(i) == thread) {
|
|
ioHandle = mPlaybackThreads.keyAt(i);
|
|
break;
|
|
}
|
|
}
|
|
if (ioHandle == 0) {
|
|
for (size_t i = 0; i < mRecordThreads.size(); i++) {
|
|
if (mRecordThreads.valueAt(i) == thread) {
|
|
ioHandle = mRecordThreads.keyAt(i);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (ioHandle != 0) {
|
|
size_t size = mNotificationClients.size();
|
|
for (size_t i = 0; i < size; i++) {
|
|
sp<IBinder> binder = mNotificationClients.itemAt(i);
|
|
LOGV("audioConfigChanged() Notifying change to client %p", binder.get());
|
|
sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
|
|
client->ioConfigChanged(event, ioHandle, param2);
|
|
}
|
|
}
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger)
|
|
: Thread(false),
|
|
mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
|
|
mFormat(0), mFrameSize(1), mStandby(false)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::ThreadBase::~ThreadBase()
|
|
{
|
|
mParamCond.broadcast();
|
|
mNewParameters.clear();
|
|
}
|
|
|
|
void AudioFlinger::ThreadBase::exit()
|
|
{
|
|
// keep a strong ref on ourself so that we want get
|
|
// destroyed in the middle of requestExitAndWait()
|
|
sp <ThreadBase> strongMe = this;
|
|
|
|
LOGV("ThreadBase::exit");
|
|
{
|
|
AutoMutex lock(&mLock);
|
|
requestExit();
|
|
mWaitWorkCV.signal();
|
|
}
|
|
requestExitAndWait();
|
|
}
|
|
|
|
uint32_t AudioFlinger::ThreadBase::sampleRate() const
|
|
{
|
|
return mSampleRate;
|
|
}
|
|
|
|
int AudioFlinger::ThreadBase::channelCount() const
|
|
{
|
|
return mChannelCount;
|
|
}
|
|
|
|
int AudioFlinger::ThreadBase::format() const
|
|
{
|
|
return mFormat;
|
|
}
|
|
|
|
size_t AudioFlinger::ThreadBase::frameCount() const
|
|
{
|
|
return mFrameCount;
|
|
}
|
|
|
|
status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
|
|
{
|
|
status_t status;
|
|
|
|
LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
mNewParameters.add(keyValuePairs);
|
|
mWaitWorkCV.signal();
|
|
mParamCond.wait(mLock);
|
|
status = mParamStatus;
|
|
mWaitWorkCV.signal();
|
|
return status;
|
|
}
|
|
|
|
void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
sendConfigEvent_l(event, param);
|
|
}
|
|
|
|
// sendConfigEvent_l() must be called with ThreadBase::mLock held
|
|
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
|
|
{
|
|
ConfigEvent *configEvent = new ConfigEvent();
|
|
configEvent->mEvent = event;
|
|
configEvent->mParam = param;
|
|
mConfigEvents.add(configEvent);
|
|
LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
|
|
mWaitWorkCV.signal();
|
|
}
|
|
|
|
void AudioFlinger::ThreadBase::processConfigEvents()
|
|
{
|
|
mLock.lock();
|
|
while(!mConfigEvents.isEmpty()) {
|
|
LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
|
|
ConfigEvent *configEvent = mConfigEvents[0];
|
|
mConfigEvents.removeAt(0);
|
|
// release mLock because audioConfigChanged() will call
|
|
// Audioflinger::audioConfigChanged() which locks AudioFlinger mLock thus creating
|
|
// potential cross deadlock between AudioFlinger::mLock and mLock
|
|
mLock.unlock();
|
|
audioConfigChanged(configEvent->mEvent, configEvent->mParam);
|
|
delete configEvent;
|
|
mLock.lock();
|
|
}
|
|
mLock.unlock();
|
|
}
|
|
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
|
|
: ThreadBase(audioFlinger),
|
|
mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
|
|
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
|
|
{
|
|
readOutputParameters();
|
|
|
|
mMasterVolume = mAudioFlinger->masterVolume();
|
|
mMasterMute = mAudioFlinger->masterMute();
|
|
|
|
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
|
|
mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
|
|
mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
|
|
}
|
|
// notify client processes that a new input has been opened
|
|
sendConfigEvent(AudioSystem::OUTPUT_OPENED);
|
|
}
|
|
|
|
AudioFlinger::PlaybackThread::~PlaybackThread()
|
|
{
|
|
delete [] mMixBuffer;
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
|
|
{
|
|
dumpInternals(fd, args);
|
|
dumpTracks(fd, args);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
|
|
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 thread %p active tracks\n", this);
|
|
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::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, "Output thread %p internals\n", this);
|
|
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
|
|
status_t AudioFlinger::PlaybackThread::readyToRun()
|
|
{
|
|
if (mSampleRate == 0) {
|
|
LOGE("No working audio driver found.");
|
|
return NO_INIT;
|
|
}
|
|
LOGI("AudioFlinger's thread %p ready to run", this);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::onFirstRef()
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
|
|
snprintf(buffer, SIZE, "Playback Thread %p", this);
|
|
|
|
run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
|
|
}
|
|
|
|
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
|
|
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::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;
|
|
|
|
if (mType == DIRECT) {
|
|
if (sampleRate != mSampleRate || format != mFormat || channelCount != mChannelCount) {
|
|
LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
|
|
sampleRate, format, channelCount, mOutput);
|
|
lStatus = BAD_VALUE;
|
|
goto Exit;
|
|
}
|
|
} else {
|
|
// 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 (mOutput == 0) {
|
|
LOGE("Audio driver not initialized.");
|
|
lStatus = NO_INIT;
|
|
goto Exit;
|
|
}
|
|
|
|
{ // scope for mLock
|
|
Mutex::Autolock _l(mLock);
|
|
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;
|
|
}
|
|
|
|
uint32_t AudioFlinger::PlaybackThread::latency() const
|
|
{
|
|
if (mOutput) {
|
|
return mOutput->latency();
|
|
}
|
|
else {
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
|
|
{
|
|
mMasterVolume = value;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
|
|
{
|
|
mMasterMute = muted;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::PlaybackThread::masterVolume() const
|
|
{
|
|
return mMasterVolume;
|
|
}
|
|
|
|
bool AudioFlinger::PlaybackThread::masterMute() const
|
|
{
|
|
return mMasterMute;
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
|
|
{
|
|
mStreamTypes[stream].volume = value;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
|
|
{
|
|
mStreamTypes[stream].mute = muted;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::PlaybackThread::streamVolume(int stream) const
|
|
{
|
|
return mStreamTypes[stream].volume;
|
|
}
|
|
|
|
bool AudioFlinger::PlaybackThread::streamMute(int stream) const
|
|
{
|
|
return mStreamTypes[stream].mute;
|
|
}
|
|
|
|
bool AudioFlinger::PlaybackThread::isMusicActive() const
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
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->type() == AudioSystem::MUSIC)
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
// addTrack_l() must be called with ThreadBase::mLock held
|
|
status_t AudioFlinger::PlaybackThread::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;
|
|
mActiveTracks.add(track);
|
|
status = NO_ERROR;
|
|
}
|
|
|
|
LOGV("mWaitWorkCV.broadcast");
|
|
mWaitWorkCV.broadcast();
|
|
|
|
return status;
|
|
}
|
|
|
|
// destroyTrack_l() must be called with ThreadBase::mLock held
|
|
void AudioFlinger::PlaybackThread::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());
|
|
}
|
|
}
|
|
|
|
String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
|
|
{
|
|
return mOutput->getParameters(keys);
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
|
|
AudioSystem::OutputDescriptor desc;
|
|
void *param2 = 0;
|
|
|
|
LOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, param);
|
|
|
|
switch (event) {
|
|
case AudioSystem::OUTPUT_OPENED:
|
|
case AudioSystem::OUTPUT_CONFIG_CHANGED:
|
|
desc.channels = mChannelCount;
|
|
desc.samplingRate = mSampleRate;
|
|
desc.format = mFormat;
|
|
desc.frameCount = mFrameCount;
|
|
desc.latency = latency();
|
|
param2 = &desc;
|
|
break;
|
|
|
|
case AudioSystem::STREAM_CONFIG_CHANGED:
|
|
param2 = ¶m;
|
|
case AudioSystem::OUTPUT_CLOSED:
|
|
default:
|
|
break;
|
|
}
|
|
mAudioFlinger->audioConfigChanged(event, this, param2);
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::readOutputParameters()
|
|
{
|
|
mSampleRate = mOutput->sampleRate();
|
|
mChannelCount = AudioSystem::popCount(mOutput->channels());
|
|
|
|
mFormat = mOutput->format();
|
|
mFrameSize = mOutput->frameSize();
|
|
mFrameCount = mOutput->bufferSize() / mFrameSize;
|
|
|
|
mMinBytesToWrite = (mOutput->latency() * mSampleRate * mFrameSize) / 1000;
|
|
// FIXME - Current mixer implementation only supports stereo output: Always
|
|
// Allocate a stereo buffer even if HW output is mono.
|
|
if (mMixBuffer != NULL) delete mMixBuffer;
|
|
mMixBuffer = new int16_t[mFrameCount * 2];
|
|
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
|
|
: PlaybackThread(audioFlinger, output),
|
|
mAudioMixer(0)
|
|
{
|
|
mType = PlaybackThread::MIXER;
|
|
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
|
|
|
|
// FIXME - Current mixer implementation only supports stereo output
|
|
if (mChannelCount == 1) {
|
|
LOGE("Invalid audio hardware channel count");
|
|
}
|
|
}
|
|
|
|
AudioFlinger::MixerThread::~MixerThread()
|
|
{
|
|
delete mAudioMixer;
|
|
}
|
|
|
|
bool AudioFlinger::MixerThread::threadLoop()
|
|
{
|
|
unsigned long sleepTime = 0;
|
|
int16_t* curBuf = mMixBuffer;
|
|
Vector< sp<Track> > tracksToRemove;
|
|
size_t enabledTracks = 0;
|
|
nsecs_t standbyTime = systemTime();
|
|
size_t mixBufferSize = mFrameCount * mFrameSize;
|
|
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
|
|
|
|
while (!exitPending())
|
|
{
|
|
processConfigEvents();
|
|
|
|
enabledTracks = 0;
|
|
{ // scope for mLock
|
|
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
if (checkForNewParameters_l()) {
|
|
mixBufferSize = mFrameCount * mFrameSize;
|
|
maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
|
|
}
|
|
|
|
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
|
|
|
|
// put audio hardware into standby after short delay
|
|
if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
|
|
mSuspended) {
|
|
if (!mStandby) {
|
|
LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
|
|
mOutput->standby();
|
|
mStandby = true;
|
|
mBytesWritten = 0;
|
|
}
|
|
|
|
if (!activeTracks.size() && mConfigEvents.isEmpty()) {
|
|
// we're about to wait, flush the binder command buffer
|
|
IPCThreadState::self()->flushCommands();
|
|
|
|
if (exitPending()) break;
|
|
|
|
// wait until we have something to do...
|
|
LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
|
|
mWaitWorkCV.wait(mLock);
|
|
LOGV("MixerThread %p TID %d waking up\n", this, gettid());
|
|
|
|
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;
|
|
sleepTime = 0;
|
|
continue;
|
|
}
|
|
}
|
|
|
|
enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove);
|
|
}
|
|
|
|
|
|
// output audio to hardware
|
|
if (mSuspended) {
|
|
usleep(kMaxBufferRecoveryInUsecs);
|
|
} else {
|
|
if (LIKELY(enabledTracks)) {
|
|
// mix buffers...
|
|
mAudioMixer->process(curBuf);
|
|
sleepTime = 0;
|
|
standbyTime = systemTime() + kStandbyTimeInNsecs;
|
|
} else {
|
|
sleepTime += kBufferRecoveryInUsecs;
|
|
// 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, write 0s to audio
|
|
// hardware to avoid underrun.
|
|
if (sleepTime < kMaxBufferRecoveryInUsecs) {
|
|
usleep(kBufferRecoveryInUsecs);
|
|
} else {
|
|
memset (curBuf, 0, mixBufferSize);
|
|
sleepTime = 0;
|
|
}
|
|
}
|
|
// sleepTime == 0 means PCM data were written to mMixBuffer[]
|
|
if (sleepTime == 0) {
|
|
mLastWriteTime = systemTime();
|
|
mInWrite = true;
|
|
int bytesWritten = (int)mOutput->write(curBuf, mixBufferSize);
|
|
if (bytesWritten > 0) mBytesWritten += bytesWritten;
|
|
mNumWrites++;
|
|
mInWrite = false;
|
|
mStandby = false;
|
|
nsecs_t delta = systemTime() - mLastWriteTime;
|
|
if (delta > maxPeriod) {
|
|
LOGW("write blocked for %llu msecs", ns2ms(delta));
|
|
mNumDelayedWrites++;
|
|
}
|
|
}
|
|
}
|
|
|
|
// 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();
|
|
}
|
|
|
|
if (!mStandby) {
|
|
mOutput->standby();
|
|
}
|
|
sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
|
|
processConfigEvents();
|
|
|
|
LOGV("MixerThread %p exiting", this);
|
|
return false;
|
|
}
|
|
|
|
// prepareTracks_l() must be called with ThreadBase::mLock held
|
|
size_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
|
|
{
|
|
|
|
size_t enabledTracks = 0;
|
|
// 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() ||
|
|
mStreamTypes[track->type()].mute) {
|
|
left = right = 0;
|
|
if (track->isPausing()) {
|
|
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 = AudioMixer::VOLUME;
|
|
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 if (cblk->server != 0) {
|
|
// If the track is stopped before the first frame was mixed,
|
|
// do not apply ramp
|
|
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.
|
|
tracksToRemove->add(track);
|
|
mAudioMixer->disable(AudioMixer::MIXING);
|
|
} 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);
|
|
}
|
|
// For tracks using static shared memory buffer, make sure that we have
|
|
// written enough data to audio hardware before disabling the track
|
|
// NOTE: this condition with arrive before track->mRetryCount <= 0 so we
|
|
// don't care about code removing track from active list above.
|
|
if ((track->mSharedBuffer == 0) || (mBytesWritten >= mMinBytesToWrite)) {
|
|
mAudioMixer->disable(AudioMixer::MIXING);
|
|
} else {
|
|
enabledTracks++;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// 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->itemAt(i);
|
|
mActiveTracks.remove(track);
|
|
if (track->isTerminated()) {
|
|
mTracks.remove(track);
|
|
deleteTrackName_l(track->mName);
|
|
}
|
|
}
|
|
}
|
|
|
|
return enabledTracks;
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::getTracks(
|
|
SortedVector < sp<Track> >& tracks,
|
|
SortedVector < wp<Track> >& activeTracks,
|
|
int streamType)
|
|
{
|
|
LOGV ("MixerThread::getTracks() mixer %p, mTracks.size %d, mActiveTracks.size %d", this, mTracks.size(), mActiveTracks.size());
|
|
Mutex::Autolock _l(mLock);
|
|
size_t size = mTracks.size();
|
|
for (size_t i = 0; i < size; i++) {
|
|
sp<Track> t = mTracks[i];
|
|
if (t->type() == streamType) {
|
|
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++) {
|
|
mActiveTracks.remove(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());
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::MixerThread::putTracks(
|
|
SortedVector < sp<Track> >& tracks,
|
|
SortedVector < wp<Track> >& activeTracks)
|
|
{
|
|
LOGV ("MixerThread::putTracks() mixer %p, tracks.size %d, activeTracks.size %d", this, tracks.size(), activeTracks.size());
|
|
Mutex::Autolock _l(mLock);
|
|
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->mThread = this;
|
|
mTracks.add(t);
|
|
|
|
int j = activeTracks.indexOf(t);
|
|
if (j >= 0) {
|
|
mActiveTracks.add(t);
|
|
// force buffer refilling and no ramp volume when the track is mixed for the first time
|
|
t->mFillingUpStatus = Track::FS_FILLING;
|
|
}
|
|
}
|
|
}
|
|
|
|
// getTrackName_l() must be called with ThreadBase::mLock held
|
|
int AudioFlinger::MixerThread::getTrackName_l()
|
|
{
|
|
return mAudioMixer->getTrackName();
|
|
}
|
|
|
|
// deleteTrackName_l() must be called with ThreadBase::mLock held
|
|
void AudioFlinger::MixerThread::deleteTrackName_l(int name)
|
|
{
|
|
mAudioMixer->deleteTrackName(name);
|
|
}
|
|
|
|
// checkForNewParameters_l() must be called with ThreadBase::mLock held
|
|
bool AudioFlinger::MixerThread::checkForNewParameters_l()
|
|
{
|
|
bool reconfig = false;
|
|
|
|
while (!mNewParameters.isEmpty()) {
|
|
status_t status = NO_ERROR;
|
|
String8 keyValuePair = mNewParameters[0];
|
|
AudioParameter param = AudioParameter(keyValuePair);
|
|
int value;
|
|
|
|
mNewParameters.removeAt(0);
|
|
|
|
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
|
|
reconfig = true;
|
|
}
|
|
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
|
|
if (value != AudioSystem::PCM_16_BIT) {
|
|
status = BAD_VALUE;
|
|
} else {
|
|
reconfig = true;
|
|
}
|
|
}
|
|
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
|
|
if (value != AudioSystem::CHANNEL_OUT_STEREO) {
|
|
status = BAD_VALUE;
|
|
} else {
|
|
reconfig = true;
|
|
}
|
|
}
|
|
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
|
|
// do not accept frame count changes if tracks are open as the track buffer
|
|
// size depends on frame count and correct behavior would not be garantied
|
|
// if frame count is changed after track creation
|
|
if (!mTracks.isEmpty()) {
|
|
status = INVALID_OPERATION;
|
|
} else {
|
|
reconfig = true;
|
|
}
|
|
}
|
|
if (status == NO_ERROR) {
|
|
status = mOutput->setParameters(keyValuePair);
|
|
if (!mStandby && status == INVALID_OPERATION) {
|
|
mOutput->standby();
|
|
mStandby = true;
|
|
mBytesWritten = 0;
|
|
status = mOutput->setParameters(keyValuePair);
|
|
}
|
|
if (status == NO_ERROR && reconfig) {
|
|
delete mAudioMixer;
|
|
readOutputParameters();
|
|
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
|
|
for (size_t i = 0; i < mTracks.size() ; i++) {
|
|
int name = getTrackName_l();
|
|
if (name < 0) break;
|
|
mTracks[i]->mName = name;
|
|
// limit track sample rate to 2 x new output sample rate
|
|
if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
|
|
mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
|
|
}
|
|
}
|
|
sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
|
|
}
|
|
}
|
|
mParamStatus = status;
|
|
mParamCond.signal();
|
|
mWaitWorkCV.wait(mLock);
|
|
}
|
|
return reconfig;
|
|
}
|
|
|
|
status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
PlaybackThread::dumpInternals(fd, args);
|
|
|
|
snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
|
|
result.append(buffer);
|
|
write(fd, result.string(), result.size());
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
|
|
: PlaybackThread(audioFlinger, output),
|
|
mLeftVolume (1.0), mRightVolume(1.0)
|
|
{
|
|
mType = PlaybackThread::DIRECT;
|
|
}
|
|
|
|
AudioFlinger::DirectOutputThread::~DirectOutputThread()
|
|
{
|
|
}
|
|
|
|
|
|
bool AudioFlinger::DirectOutputThread::threadLoop()
|
|
{
|
|
unsigned long sleepTime = 0;
|
|
sp<Track> trackToRemove;
|
|
sp<Track> activeTrack;
|
|
nsecs_t standbyTime = systemTime();
|
|
int8_t *curBuf;
|
|
size_t mixBufferSize = mFrameCount*mFrameSize;
|
|
|
|
while (!exitPending())
|
|
{
|
|
processConfigEvents();
|
|
|
|
{ // scope for the mLock
|
|
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
if (checkForNewParameters_l()) {
|
|
mixBufferSize = mFrameCount*mFrameSize;
|
|
}
|
|
|
|
// put audio hardware into standby after short delay
|
|
if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
|
|
mSuspended) {
|
|
// wait until we have something to do...
|
|
if (!mStandby) {
|
|
LOGV("Audio hardware entering standby, mixer %p\n", this);
|
|
mOutput->standby();
|
|
mStandby = true;
|
|
mBytesWritten = 0;
|
|
}
|
|
|
|
if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
|
|
// we're about to wait, flush the binder command buffer
|
|
IPCThreadState::self()->flushCommands();
|
|
|
|
if (exitPending()) break;
|
|
|
|
LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
|
|
mWaitWorkCV.wait(mLock);
|
|
LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
|
|
|
|
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;
|
|
sleepTime = 0;
|
|
continue;
|
|
}
|
|
}
|
|
|
|
// find out which tracks need to be processed
|
|
if (mActiveTracks.size() != 0) {
|
|
sp<Track> t = mActiveTracks[0].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
|
|
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
|
|
float left, right;
|
|
if (track->isMuted() || mMasterMute || track->isPausing() ||
|
|
mStreamTypes[track->type()].mute) {
|
|
left = right = 0;
|
|
if (track->isPausing()) {
|
|
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 = v_clamped/MAX_GAIN;
|
|
v_clamped = v * cblk->volume[1];
|
|
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
|
|
right = v_clamped/MAX_GAIN;
|
|
}
|
|
|
|
if (left != mLeftVolume || right != mRightVolume) {
|
|
mOutput->setVolume(left, right);
|
|
left = mLeftVolume;
|
|
right = mRightVolume;
|
|
}
|
|
|
|
if (track->mFillingUpStatus == Track::FS_FILLED) {
|
|
track->mFillingUpStatus = Track::FS_ACTIVE;
|
|
if (track->mState == TrackBase::RESUMING) {
|
|
track->mState = TrackBase::ACTIVE;
|
|
}
|
|
}
|
|
|
|
// reset retry count
|
|
track->mRetryCount = kMaxTrackRetries;
|
|
activeTrack = t;
|
|
} 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.
|
|
trackToRemove = 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());
|
|
trackToRemove = track;
|
|
}
|
|
|
|
// For tracks using static shared memry buffer, make sure that we have
|
|
// written enough data to audio hardware before disabling the track
|
|
// NOTE: this condition with arrive before track->mRetryCount <= 0 so we
|
|
// don't care about code removing track from active list above.
|
|
if ((track->mSharedBuffer != 0) && (mBytesWritten < mMinBytesToWrite)) {
|
|
activeTrack = t;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// remove all the tracks that need to be...
|
|
if (UNLIKELY(trackToRemove != 0)) {
|
|
mActiveTracks.remove(trackToRemove);
|
|
if (trackToRemove->isTerminated()) {
|
|
mTracks.remove(trackToRemove);
|
|
deleteTrackName_l(trackToRemove->mName);
|
|
}
|
|
}
|
|
}
|
|
|
|
// output audio to hardware
|
|
if (mSuspended) {
|
|
usleep(kMaxBufferRecoveryInUsecs);
|
|
} else {
|
|
if (activeTrack != 0) {
|
|
AudioBufferProvider::Buffer buffer;
|
|
size_t frameCount = mFrameCount;
|
|
curBuf = (int8_t *)mMixBuffer;
|
|
// output audio to hardware
|
|
while(frameCount) {
|
|
buffer.frameCount = frameCount;
|
|
activeTrack->getNextBuffer(&buffer);
|
|
if (UNLIKELY(buffer.raw == 0)) {
|
|
memset(curBuf, 0, frameCount * mFrameSize);
|
|
break;
|
|
}
|
|
memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
|
|
frameCount -= buffer.frameCount;
|
|
curBuf += buffer.frameCount * mFrameSize;
|
|
activeTrack->releaseBuffer(&buffer);
|
|
}
|
|
sleepTime = 0;
|
|
standbyTime = systemTime() + kStandbyTimeInNsecs;
|
|
} else {
|
|
sleepTime += kBufferRecoveryInUsecs;
|
|
if (sleepTime < kMaxBufferRecoveryInUsecs) {
|
|
usleep(kBufferRecoveryInUsecs);
|
|
} else {
|
|
memset (mMixBuffer, 0, mFrameCount * mFrameSize);
|
|
sleepTime = 0;
|
|
}
|
|
}
|
|
|
|
// sleepTime == 0 means PCM data were written to mMixBuffer[]
|
|
if (sleepTime == 0) {
|
|
mLastWriteTime = systemTime();
|
|
mInWrite = true;
|
|
int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
|
|
if (bytesWritten) mBytesWritten += bytesWritten;
|
|
mNumWrites++;
|
|
mInWrite = false;
|
|
mStandby = false;
|
|
}
|
|
}
|
|
|
|
// finally let go of removed track, without the lock held
|
|
// since we can't guarantee the destructors won't acquire that
|
|
// same lock.
|
|
trackToRemove.clear();
|
|
activeTrack.clear();
|
|
}
|
|
|
|
if (!mStandby) {
|
|
mOutput->standby();
|
|
}
|
|
sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
|
|
processConfigEvents();
|
|
|
|
LOGV("DirectOutputThread %p exiting", this);
|
|
return false;
|
|
}
|
|
|
|
// getTrackName_l() must be called with ThreadBase::mLock held
|
|
int AudioFlinger::DirectOutputThread::getTrackName_l()
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
// deleteTrackName_l() must be called with ThreadBase::mLock held
|
|
void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
|
|
{
|
|
}
|
|
|
|
// checkForNewParameters_l() must be called with ThreadBase::mLock held
|
|
bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
|
|
{
|
|
bool reconfig = false;
|
|
|
|
while (!mNewParameters.isEmpty()) {
|
|
status_t status = NO_ERROR;
|
|
String8 keyValuePair = mNewParameters[0];
|
|
AudioParameter param = AudioParameter(keyValuePair);
|
|
int value;
|
|
|
|
mNewParameters.removeAt(0);
|
|
|
|
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
|
|
// do not accept frame count changes if tracks are open as the track buffer
|
|
// size depends on frame count and correct behavior would not be garantied
|
|
// if frame count is changed after track creation
|
|
if (!mTracks.isEmpty()) {
|
|
status = INVALID_OPERATION;
|
|
} else {
|
|
reconfig = true;
|
|
}
|
|
}
|
|
if (status == NO_ERROR) {
|
|
status = mOutput->setParameters(keyValuePair);
|
|
if (!mStandby && status == INVALID_OPERATION) {
|
|
mOutput->standby();
|
|
mStandby = true;
|
|
mBytesWritten = 0;
|
|
status = mOutput->setParameters(keyValuePair);
|
|
}
|
|
if (status == NO_ERROR && reconfig) {
|
|
readOutputParameters();
|
|
sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
|
|
}
|
|
}
|
|
mParamStatus = status;
|
|
mParamCond.signal();
|
|
mWaitWorkCV.wait(mLock);
|
|
}
|
|
return reconfig;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread)
|
|
: MixerThread(audioFlinger, mainThread->getOutput())
|
|
{
|
|
mType = PlaybackThread::DUPLICATING;
|
|
addOutputTrack(mainThread);
|
|
}
|
|
|
|
AudioFlinger::DuplicatingThread::~DuplicatingThread()
|
|
{
|
|
mOutputTracks.clear();
|
|
}
|
|
|
|
bool AudioFlinger::DuplicatingThread::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*mFrameSize;
|
|
SortedVector< sp<OutputTrack> > outputTracks;
|
|
|
|
while (!exitPending())
|
|
{
|
|
processConfigEvents();
|
|
|
|
enabledTracks = 0;
|
|
{ // scope for the mLock
|
|
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
if (checkForNewParameters_l()) {
|
|
mixBufferSize = mFrameCount*mFrameSize;
|
|
}
|
|
|
|
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
|
|
|
|
for (size_t i = 0; i < mOutputTracks.size(); i++) {
|
|
outputTracks.add(mOutputTracks[i]);
|
|
}
|
|
|
|
// put audio hardware into standby after short delay
|
|
if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
|
|
mSuspended) {
|
|
if (!mStandby) {
|
|
for (size_t i = 0; i < outputTracks.size(); i++) {
|
|
outputTracks[i]->stop();
|
|
}
|
|
mStandby = true;
|
|
mBytesWritten = 0;
|
|
}
|
|
|
|
if (!activeTracks.size() && mConfigEvents.isEmpty()) {
|
|
// we're about to wait, flush the binder command buffer
|
|
IPCThreadState::self()->flushCommands();
|
|
outputTracks.clear();
|
|
|
|
if (exitPending()) break;
|
|
|
|
LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
|
|
mWaitWorkCV.wait(mLock);
|
|
LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
|
|
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;
|
|
sleepTime = kBufferRecoveryInUsecs;
|
|
continue;
|
|
}
|
|
}
|
|
|
|
enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove);
|
|
}
|
|
|
|
bool mustSleep = true;
|
|
if (LIKELY(enabledTracks)) {
|
|
// mix buffers...
|
|
mAudioMixer->process(curBuf);
|
|
if (!mSuspended) {
|
|
for (size_t i = 0; i < outputTracks.size(); i++) {
|
|
outputTracks[i]->write(curBuf, mFrameCount);
|
|
mustSleep = false;
|
|
}
|
|
mStandby = false;
|
|
mBytesWritten += mixBufferSize;
|
|
}
|
|
} else {
|
|
// flush remaining overflow buffers in output tracks
|
|
for (size_t i = 0; i < outputTracks.size(); i++) {
|
|
if (outputTracks[i]->isActive()) {
|
|
outputTracks[i]->write(curBuf, 0);
|
|
standbyTime = systemTime() + kStandbyTimeInNsecs;
|
|
mustSleep = false;
|
|
}
|
|
}
|
|
}
|
|
if (mustSleep) {
|
|
// LOGV("threadLoop() sleeping %d", sleepTime);
|
|
usleep(sleepTime);
|
|
if (sleepTime < kMaxBufferRecoveryInUsecs) {
|
|
sleepTime += kBufferRecoveryInUsecs;
|
|
}
|
|
} else {
|
|
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();
|
|
outputTracks.clear();
|
|
}
|
|
|
|
{ // scope for the mLock
|
|
|
|
Mutex::Autolock _l(mLock);
|
|
if (!mStandby) {
|
|
LOGV("DuplicatingThread() exiting out of standby");
|
|
for (size_t i = 0; i < mOutputTracks.size(); i++) {
|
|
mOutputTracks[i]->destroy();
|
|
}
|
|
}
|
|
}
|
|
|
|
sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
|
|
processConfigEvents();
|
|
|
|
return false;
|
|
}
|
|
|
|
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
|
|
{
|
|
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
|
|
OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
|
|
mSampleRate,
|
|
mFormat,
|
|
mChannelCount,
|
|
frameCount);
|
|
if (outputTrack->cblk() != NULL) {
|
|
thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
|
|
mOutputTracks.add(outputTrack);
|
|
LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
for (size_t i = 0; i < mOutputTracks.size(); i++) {
|
|
if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
|
|
mOutputTracks[i]->destroy();
|
|
mOutputTracks.removeAt(i);
|
|
return;
|
|
}
|
|
}
|
|
LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
|
|
}
|
|
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
// TrackBase constructor must be called with AudioFlinger::mLock held
|
|
AudioFlinger::ThreadBase::TrackBase::TrackBase(
|
|
const wp<ThreadBase>& thread,
|
|
const sp<Client>& client,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags,
|
|
const sp<IMemory>& sharedBuffer)
|
|
: RefBase(),
|
|
mThread(thread),
|
|
mClient(client),
|
|
mFrameCount(0),
|
|
mState(IDLE),
|
|
mClientTid(-1),
|
|
mFormat(format),
|
|
mFlags(flags & ~SYSTEM_FLAGS_MASK)
|
|
{
|
|
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::PlaybackThread::TrackBase::~TrackBase()
|
|
{
|
|
if (mCblk) {
|
|
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
|
|
if (mClient == NULL) {
|
|
delete mCblk;
|
|
}
|
|
}
|
|
mCblkMemory.clear(); // and free the shared memory
|
|
mClient.clear();
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
buffer->raw = 0;
|
|
mFrameCount = buffer->frameCount;
|
|
step();
|
|
buffer->frameCount = 0;
|
|
}
|
|
|
|
bool AudioFlinger::PlaybackThread::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::PlaybackThread::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::PlaybackThread::TrackBase::getCblk() const
|
|
{
|
|
return mCblkMemory;
|
|
}
|
|
|
|
int AudioFlinger::PlaybackThread::TrackBase::sampleRate() const {
|
|
return (int)mCblk->sampleRate;
|
|
}
|
|
|
|
int AudioFlinger::PlaybackThread::TrackBase::channelCount() const {
|
|
return (int)mCblk->channels;
|
|
}
|
|
|
|
void* AudioFlinger::PlaybackThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
|
|
int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
|
|
|
|
// Check validity of returned pointer in case the track control block would have been corrupted.
|
|
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
|
|
((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
|
|
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 and ThreadBase::mLock held
|
|
AudioFlinger::PlaybackThread::Track::Track(
|
|
const wp<ThreadBase>& thread,
|
|
const sp<Client>& client,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
const sp<IMemory>& sharedBuffer)
|
|
: TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer),
|
|
mMute(false), mSharedBuffer(sharedBuffer), mName(-1)
|
|
{
|
|
sp<ThreadBase> baseThread = thread.promote();
|
|
if (baseThread != 0) {
|
|
PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
|
|
mName = playbackThread->getTrackName_l();
|
|
}
|
|
LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
|
|
if (mName < 0) {
|
|
LOGE("no more track names available");
|
|
}
|
|
mVolume[0] = 1.0f;
|
|
mVolume[1] = 1.0f;
|
|
mStreamType = streamType;
|
|
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
|
|
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
|
|
mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
|
|
}
|
|
|
|
AudioFlinger::PlaybackThread::Track::~Track()
|
|
{
|
|
LOGV("PlaybackThread::Track destructor");
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
Mutex::Autolock _l(thread->mLock);
|
|
mState = TERMINATED;
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::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 mLock,
|
|
// we must acquire a strong reference on this Track before locking 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 mLock
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
Mutex::Autolock _l(thread->mLock);
|
|
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
|
|
playbackThread->destroyTrack_l(this);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::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::PlaybackThread::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;
|
|
LOGV("getNextBuffer() no more data");
|
|
return NOT_ENOUGH_DATA;
|
|
}
|
|
|
|
bool AudioFlinger::PlaybackThread::Track::isReady() const {
|
|
if (mFillingUpStatus != FS_FILLING) return true;
|
|
|
|
if (mCblk->framesReady() >= mCblk->frameCount ||
|
|
mCblk->forceReady) {
|
|
mFillingUpStatus = FS_FILLED;
|
|
mCblk->forceReady = 0;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::Track::start()
|
|
{
|
|
LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
Mutex::Autolock _l(thread->mLock);
|
|
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
|
|
playbackThread->addTrack_l(this);
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::Track::stop()
|
|
{
|
|
LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
Mutex::Autolock _l(thread->mLock);
|
|
if (mState > STOPPED) {
|
|
mState = STOPPED;
|
|
// If the track is not active (PAUSED and buffers full), flush buffers
|
|
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
|
|
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
|
|
reset();
|
|
}
|
|
LOGV("(> STOPPED) => STOPPED (%d)", mName);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::Track::pause()
|
|
{
|
|
LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
Mutex::Autolock _l(thread->mLock);
|
|
if (mState == ACTIVE || mState == RESUMING) {
|
|
mState = PAUSING;
|
|
LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::Track::flush()
|
|
{
|
|
LOGV("flush(%d)", mName);
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
Mutex::Autolock _l(thread->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::PlaybackThread::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::PlaybackThread::Track::mute(bool muted)
|
|
{
|
|
mMute = muted;
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
|
|
{
|
|
mVolume[0] = left;
|
|
mVolume[1] = right;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
// RecordTrack constructor must be called with AudioFlinger::mLock held
|
|
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
|
|
const wp<ThreadBase>& thread,
|
|
const sp<Client>& client,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags)
|
|
: TrackBase(thread, client, sampleRate, format,
|
|
channelCount, frameCount, flags, 0),
|
|
mOverflow(false)
|
|
{
|
|
LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
|
|
if (format == AudioSystem::PCM_16_BIT) {
|
|
mCblk->frameSize = channelCount * sizeof(int16_t);
|
|
} else if (format == AudioSystem::PCM_8_BIT) {
|
|
mCblk->frameSize = channelCount * sizeof(int8_t);
|
|
} else {
|
|
mCblk->frameSize = sizeof(int8_t);
|
|
}
|
|
}
|
|
|
|
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
|
|
{
|
|
}
|
|
|
|
status_t AudioFlinger::RecordThread::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::RecordThread::RecordTrack::start()
|
|
{
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
RecordThread *recordThread = (RecordThread *)thread.get();
|
|
return recordThread->start(this);
|
|
}
|
|
return NO_INIT;
|
|
}
|
|
|
|
void AudioFlinger::RecordThread::RecordTrack::stop()
|
|
{
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
RecordThread *recordThread = (RecordThread *)thread.get();
|
|
recordThread->stop(this);
|
|
TrackBase::reset();
|
|
// Force overerrun condition to avoid false overrun callback until first data is
|
|
// read from buffer
|
|
mCblk->flowControlFlag = 1;
|
|
}
|
|
}
|
|
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
|
|
const wp<ThreadBase>& thread,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount)
|
|
: Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL),
|
|
mActive(false)
|
|
{
|
|
|
|
PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
|
|
if (mCblk != NULL) {
|
|
mCblk->out = 1;
|
|
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
|
|
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
|
|
mOutBuffer.frameCount = 0;
|
|
mWaitTimeMs = (playbackThread->frameCount() * 2 * 1000) / playbackThread->sampleRate();
|
|
playbackThread->mTracks.add(this);
|
|
LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p mWaitTimeMs %d",
|
|
mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd, mWaitTimeMs);
|
|
} else {
|
|
LOGW("Error creating output track on thread %p", playbackThread);
|
|
}
|
|
}
|
|
|
|
AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
|
|
{
|
|
clearBufferQueue();
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::OutputTrack::start()
|
|
{
|
|
status_t status = Track::start();
|
|
if (status != NO_ERROR) {
|
|
return status;
|
|
}
|
|
|
|
mActive = true;
|
|
mRetryCount = 127;
|
|
return status;
|
|
}
|
|
|
|
void AudioFlinger::PlaybackThread::OutputTrack::stop()
|
|
{
|
|
Track::stop();
|
|
clearBufferQueue();
|
|
mOutBuffer.frameCount = 0;
|
|
mActive = false;
|
|
}
|
|
|
|
bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
|
|
{
|
|
Buffer *pInBuffer;
|
|
Buffer inBuffer;
|
|
uint32_t channels = mCblk->channels;
|
|
bool outputBufferFull = false;
|
|
inBuffer.frameCount = frames;
|
|
inBuffer.i16 = data;
|
|
|
|
uint32_t waitTimeLeftMs = mWaitTimeMs;
|
|
|
|
if (!mActive) {
|
|
start();
|
|
sp<ThreadBase> thread = mThread.promote();
|
|
if (thread != 0) {
|
|
MixerThread *mixerThread = (MixerThread *)thread.get();
|
|
if (mCblk->frameCount > frames){
|
|
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
|
|
uint32_t startFrames = (mCblk->frameCount - frames);
|
|
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() %p no more buffers in queue", this);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
while (waitTimeLeftMs) {
|
|
// 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;
|
|
nsecs_t startTime = systemTime();
|
|
if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
|
|
LOGV ("OutputTrack::write() %p no more output buffers", this);
|
|
outputBufferFull = true;
|
|
break;
|
|
}
|
|
uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
|
|
// LOGV("OutputTrack::write() waitTimeMs %d waitTimeLeftMs %d", waitTimeMs, waitTimeLeftMs)
|
|
if (waitTimeLeftMs >= waitTimeMs) {
|
|
waitTimeLeftMs -= waitTimeMs;
|
|
} else {
|
|
waitTimeLeftMs = 0;
|
|
}
|
|
}
|
|
|
|
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;
|
|
LOGV("OutputTrack::write() %p released overflow buffer %d", this, mBufferQueue.size());
|
|
} else {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
// If we could not write all frames, allocate a buffer and queue it for next time.
|
|
if (inBuffer.frameCount) {
|
|
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
|
|
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);
|
|
LOGV("OutputTrack::write() %p adding overflow buffer %d", this, mBufferQueue.size());
|
|
} else {
|
|
LOGW("OutputTrack::write() %p no more overflow buffers", this);
|
|
}
|
|
}
|
|
|
|
// 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) {
|
|
if (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);
|
|
} else {
|
|
stop();
|
|
}
|
|
}
|
|
|
|
return outputBufferFull;
|
|
}
|
|
|
|
status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
|
|
{
|
|
int active;
|
|
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) {
|
|
Mutex::Autolock _l(cblk->lock);
|
|
goto start_loop_here;
|
|
while (framesAvail == 0) {
|
|
active = mActive;
|
|
if (UNLIKELY(!active)) {
|
|
LOGV("Not active and NO_MORE_BUFFERS");
|
|
return AudioTrack::NO_MORE_BUFFERS;
|
|
}
|
|
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
|
|
if (result != NO_ERROR) {
|
|
return AudioTrack::NO_MORE_BUFFERS;
|
|
}
|
|
// read the server count again
|
|
start_loop_here:
|
|
framesAvail = cblk->framesAvailable_l();
|
|
}
|
|
}
|
|
|
|
// if (framesAvail < framesReq) {
|
|
// 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::PlaybackThread::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::PlaybackThread::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 input,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags,
|
|
status_t *status)
|
|
{
|
|
sp<RecordThread::RecordTrack> recordTrack;
|
|
sp<RecordHandle> recordHandle;
|
|
sp<Client> client;
|
|
wp<Client> wclient;
|
|
status_t lStatus;
|
|
RecordThread *thread;
|
|
size_t inFrameCount;
|
|
|
|
// check calling permissions
|
|
if (!recordingAllowed()) {
|
|
lStatus = PERMISSION_DENIED;
|
|
goto Exit;
|
|
}
|
|
|
|
// add client to list
|
|
{ // scope for mLock
|
|
Mutex::Autolock _l(mLock);
|
|
thread = checkRecordThread_l(input);
|
|
if (thread == NULL) {
|
|
lStatus = BAD_VALUE;
|
|
goto Exit;
|
|
}
|
|
|
|
wclient = mClients.valueFor(pid);
|
|
if (wclient != NULL) {
|
|
client = wclient.promote();
|
|
} else {
|
|
client = new Client(this, pid);
|
|
mClients.add(pid, client);
|
|
}
|
|
|
|
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
|
|
recordTrack = new RecordThread::RecordTrack(thread, client, 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;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::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::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels) :
|
|
ThreadBase(audioFlinger),
|
|
mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
|
|
{
|
|
mReqChannelCount = AudioSystem::popCount(channels);
|
|
mReqSampleRate = sampleRate;
|
|
readInputParameters();
|
|
sendConfigEvent(AudioSystem::INPUT_OPENED);
|
|
}
|
|
|
|
|
|
AudioFlinger::RecordThread::~RecordThread()
|
|
{
|
|
delete[] mRsmpInBuffer;
|
|
if (mResampler != 0) {
|
|
delete mResampler;
|
|
delete[] mRsmpOutBuffer;
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::RecordThread::onFirstRef()
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
|
|
snprintf(buffer, SIZE, "Record Thread %p", this);
|
|
|
|
run(buffer, PRIORITY_URGENT_AUDIO);
|
|
}
|
|
bool AudioFlinger::RecordThread::threadLoop()
|
|
{
|
|
AudioBufferProvider::Buffer buffer;
|
|
sp<RecordTrack> activeTrack;
|
|
|
|
// start recording
|
|
while (!exitPending()) {
|
|
|
|
processConfigEvents();
|
|
|
|
{ // scope for mLock
|
|
Mutex::Autolock _l(mLock);
|
|
checkForNewParameters_l();
|
|
if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
|
|
if (!mStandby) {
|
|
mInput->standby();
|
|
mStandby = true;
|
|
}
|
|
|
|
if (exitPending()) break;
|
|
|
|
LOGV("RecordThread: loop stopping");
|
|
// go to sleep
|
|
mWaitWorkCV.wait(mLock);
|
|
LOGV("RecordThread: loop starting");
|
|
continue;
|
|
}
|
|
if (mActiveTrack != 0) {
|
|
if (mActiveTrack->mState == TrackBase::PAUSING) {
|
|
mActiveTrack.clear();
|
|
mStartStopCond.broadcast();
|
|
} else if (mActiveTrack->mState == TrackBase::RESUMING) {
|
|
mRsmpInIndex = mFrameCount;
|
|
if (mReqChannelCount != mActiveTrack->channelCount()) {
|
|
mActiveTrack.clear();
|
|
} else {
|
|
mActiveTrack->mState = TrackBase::ACTIVE;
|
|
}
|
|
mStartStopCond.broadcast();
|
|
}
|
|
mStandby = false;
|
|
}
|
|
}
|
|
|
|
if (mActiveTrack != 0) {
|
|
buffer.frameCount = mFrameCount;
|
|
if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
|
|
size_t framesOut = buffer.frameCount;
|
|
if (mResampler == 0) {
|
|
// no resampling
|
|
while (framesOut) {
|
|
size_t framesIn = mFrameCount - mRsmpInIndex;
|
|
if (framesIn) {
|
|
int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
|
|
int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
|
|
if (framesIn > framesOut)
|
|
framesIn = framesOut;
|
|
mRsmpInIndex += framesIn;
|
|
framesOut -= framesIn;
|
|
if (mChannelCount == mReqChannelCount ||
|
|
mFormat != AudioSystem::PCM_16_BIT) {
|
|
memcpy(dst, src, framesIn * mFrameSize);
|
|
} else {
|
|
int16_t *src16 = (int16_t *)src;
|
|
int16_t *dst16 = (int16_t *)dst;
|
|
if (mChannelCount == 1) {
|
|
while (framesIn--) {
|
|
*dst16++ = *src16;
|
|
*dst16++ = *src16++;
|
|
}
|
|
} else {
|
|
while (framesIn--) {
|
|
*dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
|
|
src16 += 2;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
if (framesOut && mFrameCount == mRsmpInIndex) {
|
|
ssize_t bytesRead;
|
|
if (framesOut == mFrameCount &&
|
|
(mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {
|
|
bytesRead = mInput->read(buffer.raw, mInputBytes);
|
|
framesOut = 0;
|
|
} else {
|
|
bytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
|
|
mRsmpInIndex = 0;
|
|
}
|
|
if (bytesRead < 0) {
|
|
LOGE("Error reading audio input");
|
|
sleep(1);
|
|
mRsmpInIndex = mFrameCount;
|
|
framesOut = 0;
|
|
buffer.frameCount = 0;
|
|
}
|
|
}
|
|
}
|
|
} else {
|
|
// resampling
|
|
|
|
memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
|
|
// alter output frame count as if we were expecting stereo samples
|
|
if (mChannelCount == 1 && mReqChannelCount == 1) {
|
|
framesOut >>= 1;
|
|
}
|
|
mResampler->resample(mRsmpOutBuffer, framesOut, this);
|
|
// ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
|
|
// are 32 bit aligned which should be always true.
|
|
if (mChannelCount == 2 && mReqChannelCount == 1) {
|
|
AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
|
|
// the resampler always outputs stereo samples: do post stereo to mono conversion
|
|
int16_t *src = (int16_t *)mRsmpOutBuffer;
|
|
int16_t *dst = buffer.i16;
|
|
while (framesOut--) {
|
|
*dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
|
|
src += 2;
|
|
}
|
|
} else {
|
|
AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
|
|
}
|
|
|
|
}
|
|
mActiveTrack->releaseBuffer(&buffer);
|
|
mActiveTrack->overflow();
|
|
}
|
|
// client isn't retrieving buffers fast enough
|
|
else {
|
|
if (!mActiveTrack->setOverflow())
|
|
LOGW("RecordThread: 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 (!mStandby) {
|
|
mInput->standby();
|
|
}
|
|
mActiveTrack.clear();
|
|
|
|
sendConfigEvent(AudioSystem::INPUT_CLOSED);
|
|
processConfigEvents();
|
|
|
|
LOGV("RecordThread %p exiting", this);
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
|
|
{
|
|
LOGV("RecordThread::start");
|
|
AutoMutex lock(&mLock);
|
|
|
|
if (mActiveTrack != 0) {
|
|
if (recordTrack != mActiveTrack.get()) return -EBUSY;
|
|
|
|
if (mActiveTrack->mState == TrackBase::PAUSING) mActiveTrack->mState = TrackBase::RESUMING;
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
mActiveTrack = recordTrack;
|
|
mActiveTrack->mState = TrackBase::RESUMING;
|
|
// signal thread to start
|
|
LOGV("Signal record thread");
|
|
mWaitWorkCV.signal();
|
|
mStartStopCond.wait(mLock);
|
|
if (mActiveTrack != 0) {
|
|
LOGV("Record started OK");
|
|
return NO_ERROR;
|
|
} else {
|
|
LOGV("Record failed to start");
|
|
return BAD_VALUE;
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
|
|
LOGV("RecordThread::stop");
|
|
AutoMutex lock(&mLock);
|
|
if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
|
|
mActiveTrack->mState = TrackBase::PAUSING;
|
|
mStartStopCond.wait(mLock);
|
|
}
|
|
}
|
|
|
|
status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
pid_t pid = 0;
|
|
|
|
if (mActiveTrack != 0 && mActiveTrack->mClient != 0) {
|
|
snprintf(buffer, SIZE, "Record client pid: %d\n", mActiveTrack->mClient->pid());
|
|
result.append(buffer);
|
|
} else {
|
|
result.append("No record client\n");
|
|
}
|
|
write(fd, result.string(), result.size());
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
size_t framesReq = buffer->frameCount;
|
|
size_t framesReady = mFrameCount - mRsmpInIndex;
|
|
int channelCount;
|
|
|
|
if (framesReady == 0) {
|
|
ssize_t bytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
|
|
if (bytesRead < 0) {
|
|
LOGE("RecordThread::getNextBuffer() Error reading audio input");
|
|
sleep(1);
|
|
buffer->raw = 0;
|
|
buffer->frameCount = 0;
|
|
return NOT_ENOUGH_DATA;
|
|
}
|
|
mRsmpInIndex = 0;
|
|
framesReady = mFrameCount;
|
|
}
|
|
|
|
if (framesReq > framesReady) {
|
|
framesReq = framesReady;
|
|
}
|
|
|
|
if (mChannelCount == 1 && mReqChannelCount == 2) {
|
|
channelCount = 1;
|
|
} else {
|
|
channelCount = 2;
|
|
}
|
|
buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
|
|
buffer->frameCount = framesReq;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
mRsmpInIndex += buffer->frameCount;
|
|
buffer->frameCount = 0;
|
|
}
|
|
|
|
bool AudioFlinger::RecordThread::checkForNewParameters_l()
|
|
{
|
|
bool reconfig = false;
|
|
|
|
while (!mNewParameters.isEmpty()) {
|
|
status_t status = NO_ERROR;
|
|
String8 keyValuePair = mNewParameters[0];
|
|
AudioParameter param = AudioParameter(keyValuePair);
|
|
int value;
|
|
int reqFormat = mFormat;
|
|
int reqSamplingRate = mReqSampleRate;
|
|
int reqChannelCount = mReqChannelCount;
|
|
|
|
mNewParameters.removeAt(0);
|
|
|
|
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
|
|
reqSamplingRate = value;
|
|
reconfig = true;
|
|
}
|
|
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
|
|
reqFormat = value;
|
|
reconfig = true;
|
|
}
|
|
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
|
|
reqChannelCount = AudioSystem::popCount(value);
|
|
reconfig = true;
|
|
}
|
|
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
|
|
// do not accept frame count changes if tracks are open as the track buffer
|
|
// size depends on frame count and correct behavior would not be garantied
|
|
// if frame count is changed after track creation
|
|
if (mActiveTrack != 0) {
|
|
status = INVALID_OPERATION;
|
|
} else {
|
|
reconfig = true;
|
|
}
|
|
}
|
|
if (status == NO_ERROR) {
|
|
status = mInput->setParameters(keyValuePair);
|
|
if (status == INVALID_OPERATION) {
|
|
mInput->standby();
|
|
status = mInput->setParameters(keyValuePair);
|
|
}
|
|
if (reconfig) {
|
|
if (status == BAD_VALUE &&
|
|
reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT &&
|
|
((int)mInput->sampleRate() <= 2 * reqSamplingRate) &&
|
|
(AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
|
|
status = NO_ERROR;
|
|
}
|
|
if (status == NO_ERROR) {
|
|
readInputParameters();
|
|
sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
|
|
}
|
|
}
|
|
}
|
|
mParamStatus = status;
|
|
mParamCond.signal();
|
|
mWaitWorkCV.wait(mLock);
|
|
}
|
|
return reconfig;
|
|
}
|
|
|
|
String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
|
|
{
|
|
return mInput->getParameters(keys);
|
|
}
|
|
|
|
void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) {
|
|
AudioSystem::OutputDescriptor desc;
|
|
void *param2 = 0;
|
|
|
|
switch (event) {
|
|
case AudioSystem::INPUT_OPENED:
|
|
case AudioSystem::INPUT_CONFIG_CHANGED:
|
|
desc.channels = mChannelCount;
|
|
desc.samplingRate = mSampleRate;
|
|
desc.format = mFormat;
|
|
desc.frameCount = mFrameCount;
|
|
desc.latency = 0;
|
|
param2 = &desc;
|
|
break;
|
|
|
|
case AudioSystem::INPUT_CLOSED:
|
|
default:
|
|
break;
|
|
}
|
|
mAudioFlinger->audioConfigChanged(event, this, param2);
|
|
}
|
|
|
|
void AudioFlinger::RecordThread::readInputParameters()
|
|
{
|
|
if (mRsmpInBuffer) delete mRsmpInBuffer;
|
|
if (mRsmpOutBuffer) delete mRsmpOutBuffer;
|
|
if (mResampler) delete mResampler;
|
|
mResampler = 0;
|
|
|
|
mSampleRate = mInput->sampleRate();
|
|
mChannelCount = AudioSystem::popCount(mInput->channels());
|
|
mFormat = mInput->format();
|
|
mFrameSize = mInput->frameSize();
|
|
mInputBytes = mInput->bufferSize();
|
|
mFrameCount = mInputBytes / mFrameSize;
|
|
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
|
|
|
|
if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
|
|
{
|
|
int channelCount;
|
|
// optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
|
|
// stereo to mono post process as the resampler always outputs stereo.
|
|
if (mChannelCount == 1 && mReqChannelCount == 2) {
|
|
channelCount = 1;
|
|
} else {
|
|
channelCount = 2;
|
|
}
|
|
mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
|
|
mResampler->setSampleRate(mSampleRate);
|
|
mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
|
|
mRsmpOutBuffer = new int32_t[mFrameCount * 2];
|
|
|
|
// optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
|
|
if (mChannelCount == 1 && mReqChannelCount == 1) {
|
|
mFrameCount >>= 1;
|
|
}
|
|
|
|
}
|
|
mRsmpInIndex = mFrameCount;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
int AudioFlinger::openOutput(uint32_t *pDevices,
|
|
uint32_t *pSamplingRate,
|
|
uint32_t *pFormat,
|
|
uint32_t *pChannels,
|
|
uint32_t *pLatencyMs,
|
|
uint32_t flags)
|
|
{
|
|
status_t status;
|
|
PlaybackThread *thread = NULL;
|
|
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
|
|
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
|
|
uint32_t format = pFormat ? *pFormat : 0;
|
|
uint32_t channels = pChannels ? *pChannels : 0;
|
|
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
|
|
|
|
LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
|
|
pDevices ? *pDevices : 0,
|
|
samplingRate,
|
|
format,
|
|
channels,
|
|
flags);
|
|
|
|
if (pDevices == NULL || *pDevices == 0) {
|
|
return 0;
|
|
}
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
|
|
(int *)&format,
|
|
&channels,
|
|
&samplingRate,
|
|
&status);
|
|
LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
|
|
output,
|
|
samplingRate,
|
|
format,
|
|
channels,
|
|
status);
|
|
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
if (output != 0) {
|
|
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
|
|
(format != AudioSystem::PCM_16_BIT) ||
|
|
(channels != AudioSystem::CHANNEL_OUT_STEREO)) {
|
|
thread = new DirectOutputThread(this, output);
|
|
LOGV("openOutput() created direct output: ID %d thread %p", (mNextThreadId + 1), thread);
|
|
} else {
|
|
thread = new MixerThread(this, output);
|
|
LOGV("openOutput() created mixer output: ID %d thread %p", (mNextThreadId + 1), thread);
|
|
}
|
|
mPlaybackThreads.add(++mNextThreadId, thread);
|
|
|
|
if (pSamplingRate) *pSamplingRate = samplingRate;
|
|
if (pFormat) *pFormat = format;
|
|
if (pChannels) *pChannels = channels;
|
|
if (pLatencyMs) *pLatencyMs = thread->latency();
|
|
}
|
|
|
|
return mNextThreadId;
|
|
}
|
|
|
|
int AudioFlinger::openDuplicateOutput(int output1, int output2)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
MixerThread *thread1 = checkMixerThread_l(output1);
|
|
MixerThread *thread2 = checkMixerThread_l(output2);
|
|
|
|
if (thread1 == NULL || thread2 == NULL) {
|
|
LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
|
|
return 0;
|
|
}
|
|
|
|
|
|
DuplicatingThread *thread = new DuplicatingThread(this, thread1);
|
|
thread->addOutputTrack(thread2);
|
|
mPlaybackThreads.add(++mNextThreadId, thread);
|
|
return mNextThreadId;
|
|
}
|
|
|
|
status_t AudioFlinger::closeOutput(int output)
|
|
{
|
|
// keep strong reference on the playback thread so that
|
|
// it is not destroyed while exit() is executed
|
|
sp <PlaybackThread> thread;
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
thread = checkPlaybackThread_l(output);
|
|
if (thread == NULL) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
LOGV("closeOutput() %d", output);
|
|
|
|
if (thread->type() == PlaybackThread::MIXER) {
|
|
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
|
|
if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) {
|
|
DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
|
|
dupThread->removeOutputTrack((MixerThread *)thread.get());
|
|
}
|
|
}
|
|
}
|
|
mPlaybackThreads.removeItem(output);
|
|
}
|
|
thread->exit();
|
|
|
|
if (thread->type() != PlaybackThread::DUPLICATING) {
|
|
mAudioHardware->closeOutputStream(thread->getOutput());
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::suspendOutput(int output)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
PlaybackThread *thread = checkPlaybackThread_l(output);
|
|
|
|
if (thread == NULL) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
LOGV("suspendOutput() %d", output);
|
|
thread->suspend();
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::restoreOutput(int output)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
PlaybackThread *thread = checkPlaybackThread_l(output);
|
|
|
|
if (thread == NULL) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
LOGV("restoreOutput() %d", output);
|
|
|
|
thread->restore();
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
int AudioFlinger::openInput(uint32_t *pDevices,
|
|
uint32_t *pSamplingRate,
|
|
uint32_t *pFormat,
|
|
uint32_t *pChannels,
|
|
uint32_t acoustics)
|
|
{
|
|
status_t status;
|
|
RecordThread *thread = NULL;
|
|
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
|
|
uint32_t format = pFormat ? *pFormat : 0;
|
|
uint32_t channels = pChannels ? *pChannels : 0;
|
|
uint32_t reqSamplingRate = samplingRate;
|
|
uint32_t reqFormat = format;
|
|
uint32_t reqChannels = channels;
|
|
|
|
if (pDevices == NULL || *pDevices == 0) {
|
|
return 0;
|
|
}
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices,
|
|
(int *)&format,
|
|
&channels,
|
|
&samplingRate,
|
|
&status,
|
|
(AudioSystem::audio_in_acoustics)acoustics);
|
|
LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
|
|
input,
|
|
samplingRate,
|
|
format,
|
|
channels,
|
|
acoustics,
|
|
status);
|
|
|
|
// If the input could not be opened with the requested parameters and we can handle the conversion internally,
|
|
// try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
|
|
// or stereo to mono conversions on 16 bit PCM inputs.
|
|
if (input == 0 && status == BAD_VALUE &&
|
|
reqFormat == format && format == AudioSystem::PCM_16_BIT &&
|
|
(samplingRate <= 2 * reqSamplingRate) &&
|
|
(AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) {
|
|
LOGV("openInput() reopening with proposed sampling rate and channels");
|
|
input = mAudioHardware->openInputStream(*pDevices,
|
|
(int *)&format,
|
|
&channels,
|
|
&samplingRate,
|
|
&status,
|
|
(AudioSystem::audio_in_acoustics)acoustics);
|
|
}
|
|
|
|
if (input != 0) {
|
|
// Start record thread
|
|
thread = new RecordThread(this, input, reqSamplingRate, reqChannels);
|
|
mRecordThreads.add(++mNextThreadId, thread);
|
|
LOGV("openInput() created record thread: ID %d thread %p", mNextThreadId, thread);
|
|
if (pSamplingRate) *pSamplingRate = reqSamplingRate;
|
|
if (pFormat) *pFormat = format;
|
|
if (pChannels) *pChannels = reqChannels;
|
|
|
|
input->standby();
|
|
}
|
|
|
|
return mNextThreadId;
|
|
}
|
|
|
|
status_t AudioFlinger::closeInput(int input)
|
|
{
|
|
// keep strong reference on the record thread so that
|
|
// it is not destroyed while exit() is executed
|
|
sp <RecordThread> thread;
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
thread = checkRecordThread_l(input);
|
|
if (thread == NULL) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
LOGV("closeInput() %d", input);
|
|
mRecordThreads.removeItem(input);
|
|
}
|
|
thread->exit();
|
|
|
|
mAudioHardware->closeInputStream(thread->getInput());
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
MixerThread *dstThread = checkMixerThread_l(output);
|
|
if (dstThread == NULL) {
|
|
LOGW("setStreamOutput() bad output id %d", output);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
LOGV("setStreamOutput() stream %d to output %d", stream, output);
|
|
|
|
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
|
|
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
|
|
if (thread != dstThread &&
|
|
thread->type() != PlaybackThread::DIRECT) {
|
|
MixerThread *srcThread = (MixerThread *)thread;
|
|
SortedVector < sp<MixerThread::Track> > tracks;
|
|
SortedVector < wp<MixerThread::Track> > activeTracks;
|
|
srcThread->getTracks(tracks, activeTracks, stream);
|
|
if (tracks.size()) {
|
|
dstThread->putTracks(tracks, activeTracks);
|
|
}
|
|
}
|
|
}
|
|
|
|
dstThread->sendConfigEvent(AudioSystem::STREAM_CONFIG_CHANGED, stream);
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
|
|
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
|
|
{
|
|
PlaybackThread *thread = NULL;
|
|
if (mPlaybackThreads.indexOfKey(output) >= 0) {
|
|
thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
|
|
}
|
|
return thread;
|
|
}
|
|
|
|
// checkMixerThread_l() must be called with AudioFlinger::mLock held
|
|
AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
|
|
{
|
|
PlaybackThread *thread = checkPlaybackThread_l(output);
|
|
if (thread != NULL) {
|
|
if (thread->type() == PlaybackThread::DIRECT) {
|
|
thread = NULL;
|
|
}
|
|
}
|
|
return (MixerThread *)thread;
|
|
}
|
|
|
|
// checkRecordThread_l() must be called with AudioFlinger::mLock held
|
|
AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
|
|
{
|
|
RecordThread *thread = NULL;
|
|
if (mRecordThreads.indexOfKey(input) >= 0) {
|
|
thread = (RecordThread *)mRecordThreads.valueFor(input).get();
|
|
}
|
|
return thread;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
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
|