1451 lines
43 KiB
C++
1451 lines
43 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 <utils/IServiceManager.h>
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#include <utils/Log.h>
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#include <utils/Parcel.h>
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#include <utils/IPCThreadState.h>
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#include <utils/String16.h>
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#include <utils/threads.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/AudioHardwareInterface.h>
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#include "AudioMixer.h"
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#include "AudioFlinger.h"
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namespace android {
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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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#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(), Thread(false),
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mMasterVolume(0), mMasterMute(true),
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mAudioMixer(0), mAudioHardware(0), mOutput(0), mAudioRecordThread(0),
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mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0),
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mMixBuffer(0), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0),
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mStandby(false), mInWrite(false)
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{
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mHardwareStatus = AUDIO_HW_IDLE;
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mAudioHardware = AudioHardwareInterface::create();
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mHardwareStatus = AUDIO_HW_INIT;
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if (mAudioHardware->initCheck() == NO_ERROR) {
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// open 16-bit output stream for s/w mixer
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mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
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mOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT);
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mHardwareStatus = AUDIO_HW_IDLE;
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if (mOutput) {
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mSampleRate = mOutput->sampleRate();
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mChannelCount = mOutput->channelCount();
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mFormat = mOutput->format();
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mMixBufferSize = mOutput->bufferSize();
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mFrameCount = mMixBufferSize / mChannelCount / sizeof(int16_t);
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mMixBuffer = new int16_t[mFrameCount * mChannelCount];
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memset(mMixBuffer, 0, mMixBufferSize);
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mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
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// FIXME - this should come from settings
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setMasterVolume(1.0f);
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setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
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setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
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setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
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setMode(AudioSystem::MODE_NORMAL);
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mMasterMute = false;
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} else {
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LOGE("Failed to initialize output stream");
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}
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} else {
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LOGE("Couldn't even initialize the stubbed audio hardware!");
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}
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}
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AudioFlinger::~AudioFlinger()
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{
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delete mOutput;
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delete mAudioHardware;
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delete [] mMixBuffer;
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delete mAudioMixer;
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mAudioRecordThread.clear();
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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::dumpTracks(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("Tracks:\n");
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result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
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for (size_t i = 0; i < mTracks.size(); ++i) {
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wp<Track> wTrack = mTracks[i];
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if (wTrack != 0) {
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sp<Track> track = wTrack.promote();
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if (track != 0) {
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track->dump(buffer, SIZE);
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result.append(buffer);
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}
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}
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}
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result.append("Active Tracks:\n");
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result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
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for (size_t i = 0; i < mActiveTracks.size(); ++i) {
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wp<Track> wTrack = mTracks[i];
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if (wTrack != 0) {
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sp<Track> track = wTrack.promote();
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if (track != 0) {
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track->dump(buffer, SIZE);
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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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snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", audioMixer().trackNames());
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result.append(buffer);
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snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
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result.append(buffer);
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snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
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result.append(buffer);
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snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
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result.append(buffer);
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snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
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result.append(buffer);
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snprintf(buffer, SIZE, "standby: %d\n", mStandby);
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result.append(buffer);
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snprintf(buffer, SIZE, "Hardware status: %d\n", mHardwareStatus);
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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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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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AutoMutex lock(&mLock);
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dumpClients(fd, args);
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dumpTracks(fd, args);
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dumpInternals(fd, args);
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if (mAudioHardware) {
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mAudioHardware->dumpState(fd, args);
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}
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}
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return NO_ERROR;
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}
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// Thread virtuals
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bool AudioFlinger::threadLoop()
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{
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nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
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unsigned long sleepTime = kBufferRecoveryInUsecs;
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const size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
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int16_t* curBuf = mMixBuffer;
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Vector< sp<Track> > tracksToRemove;
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size_t enabledTracks;
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nsecs_t standbyTime = systemTime();
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do {
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enabledTracks = 0;
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{ // scope for the lock
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Mutex::Autolock _l(mLock);
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const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
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// put audio hardware into standby after short delay
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if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
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// wait until we have something to do...
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LOGV("Audio hardware entering standby\n");
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mHardwareStatus = AUDIO_HW_STANDBY;
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if (!mStandby) {
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mAudioHardware->standby();
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mStandby = true;
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}
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mHardwareStatus = AUDIO_HW_IDLE;
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// we're about to wait, flush the binder command buffer
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IPCThreadState::self()->flushCommands();
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mWaitWorkCV.wait(mLock);
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LOGV("Audio hardware exiting standby\n");
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standbyTime = systemTime() + kStandbyTimeInNsecs;
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continue;
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}
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// find out which tracks need to be processed
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size_t count = activeTracks.size();
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for (size_t i=0 ; i<count ; i++) {
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sp<Track> t = activeTracks[i].promote();
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if (t == 0) continue;
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Track* const track = t.get();
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audio_track_cblk_t* cblk = track->cblk();
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uint32_t u = cblk->user;
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uint32_t s = cblk->server;
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// The first time a track is added we wait
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// for all its buffers to be filled before processing it
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audioMixer().setActiveTrack(track->name());
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if ((u > s) && (track->isReady(u, s) || track->isStopped()) &&
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!track->isPaused())
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{
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//LOGD("u=%08x, s=%08x [OK]", u, s);
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// compute volume for this track
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int16_t left, right;
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if (track->isMuted() || mMasterMute || track->isPausing()) {
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left = right = 0;
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if (track->isPausing()) {
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LOGV("paused(%d)", track->name());
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track->setPaused();
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}
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} else {
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float typeVolume = mStreamTypes[track->type()].volume;
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float v = mMasterVolume * typeVolume;
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float v_clamped = v * cblk->volume[0];
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if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
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left = int16_t(v_clamped);
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v_clamped = v * cblk->volume[1];
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if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
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right = int16_t(v_clamped);
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}
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// XXX: these things DON'T need to be done each time
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AudioMixer& mixer(audioMixer());
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mixer.setBufferProvider(track);
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mixer.enable(AudioMixer::MIXING);
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int param;
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if ( track->mFillingUpStatus == Track::FS_FILLED) {
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// no ramp for the first volume setting
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track->mFillingUpStatus = Track::FS_ACTIVE;
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if (track->mState == TrackBase::RESUMING) {
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track->mState = TrackBase::ACTIVE;
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param = AudioMixer::RAMP_VOLUME;
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} else {
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param = AudioMixer::VOLUME;
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}
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} else {
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param = AudioMixer::RAMP_VOLUME;
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}
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mixer.setParameter(param, AudioMixer::VOLUME0, left);
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mixer.setParameter(param, AudioMixer::VOLUME1, right);
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mixer.setParameter(
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AudioMixer::TRACK,
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AudioMixer::FORMAT, track->format());
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mixer.setParameter(
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AudioMixer::TRACK,
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AudioMixer::CHANNEL_COUNT, track->channelCount());
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mixer.setParameter(
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AudioMixer::RESAMPLE,
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AudioMixer::SAMPLE_RATE,
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int(cblk->sampleRate));
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// reset retry count
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track->mRetryCount = kMaxTrackRetries;
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enabledTracks++;
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} else {
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//LOGD("u=%08x, s=%08x [NOT READY]", u, s);
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if (track->isStopped()) {
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track->mFillingUpStatus = Track::FS_FILLING;
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track->mFlags = 0;
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}
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if (track->isTerminated() || track->isStopped() || track->isPaused()) {
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// We have consumed all the buffers of this track.
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// Remove it from the list of active tracks.
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LOGV("remove(%d) from active list", track->name());
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tracksToRemove.add(track);
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} else {
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// No buffers for this track. Give it a few chances to
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// fill a buffer, then remove it from active list.
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if (--(track->mRetryCount) <= 0) {
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LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
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tracksToRemove.add(track);
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}
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}
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// LOGV("disable(%d)", track->name());
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audioMixer().disable(AudioMixer::MIXING);
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}
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}
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// remove all the tracks that need to be...
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count = tracksToRemove.size();
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if (UNLIKELY(count)) {
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for (size_t i=0 ; i<count ; i++) {
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const sp<Track>& track = tracksToRemove[i];
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mActiveTracks.remove(track);
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if (track->isTerminated()) {
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mTracks.remove(track);
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audioMixer().deleteTrackName(track->mName);
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}
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}
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}
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}
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if (LIKELY(enabledTracks)) {
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// mix buffers...
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audioMixer().process(curBuf);
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// output audio to hardware
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mLastWriteTime = systemTime();
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mInWrite = true;
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mOutput->write(curBuf, mixBufferSize);
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mNumWrites++;
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mInWrite = false;
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mStandby = false;
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nsecs_t temp = systemTime();
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standbyTime = temp + kStandbyTimeInNsecs;
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nsecs_t delta = temp - mLastWriteTime;
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if (delta > maxPeriod) {
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LOGW("write blocked for %llu msecs", ns2ms(delta));
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mNumDelayedWrites++;
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}
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sleepTime = kBufferRecoveryInUsecs;
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} else {
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// There was nothing to mix this round, which means all
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// active tracks were late. Sleep a little bit to give
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// them another chance. If we're too late, the audio
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// hardware will zero-fill for us.
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LOGV("no buffers - usleep(%lu)", sleepTime);
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usleep(sleepTime);
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if (sleepTime < kMaxBufferRecoveryInUsecs) {
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sleepTime += kBufferRecoveryInUsecs;
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}
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}
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// finally let go of all our tracks, without the lock held
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// since we can't guarantee the destructors won't acquire that
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// same lock.
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tracksToRemove.clear();
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} while (true);
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return false;
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}
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status_t AudioFlinger::readyToRun()
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{
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if (mSampleRate == 0) {
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LOGE("No working audio driver found.");
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return NO_INIT;
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}
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LOGI("AudioFlinger's main thread ready to run.");
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return NO_ERROR;
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}
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void AudioFlinger::onFirstRef()
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{
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run("AudioFlinger", ANDROID_PRIORITY_URGENT_AUDIO);
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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 bufferCount,
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uint32_t flags)
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{
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if (streamType >= AudioTrack::NUM_STREAM_TYPES) {
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LOGE("invalid stream type");
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return NULL;
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}
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if (sampleRate > MAX_SAMPLE_RATE) {
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LOGE("Sample rate out of range: %d", sampleRate);
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return NULL;
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}
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sp<Track> track;
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sp<TrackHandle> trackHandle;
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Mutex::Autolock _l(mLock);
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if (mSampleRate == 0) {
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LOGE("Audio driver not initialized.");
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return trackHandle;
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}
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sp<Client> client;
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wp<Client> 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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// FIXME: Buffer size should be based on sample rate for consistent latency
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track = new Track(this, client, streamType, sampleRate, format,
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channelCount, bufferCount, channelCount == 1 ? mMixBufferSize>>1 : mMixBufferSize);
|
|
mTracks.add(track);
|
|
trackHandle = new TrackHandle(track);
|
|
return trackHandle;
|
|
}
|
|
|
|
uint32_t AudioFlinger::sampleRate() const
|
|
{
|
|
return mSampleRate;
|
|
}
|
|
|
|
int AudioFlinger::channelCount() const
|
|
{
|
|
return mChannelCount;
|
|
}
|
|
|
|
int AudioFlinger::format() const
|
|
{
|
|
return mFormat;
|
|
}
|
|
|
|
size_t AudioFlinger::frameCount() const
|
|
{
|
|
return mFrameCount;
|
|
}
|
|
|
|
status_t AudioFlinger::setMasterVolume(float value)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
// when hw supports master volume, don't scale in sw mixer
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
|
|
if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
|
|
mMasterVolume = 1.0f;
|
|
}
|
|
else {
|
|
mMasterVolume = value;
|
|
}
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
|
|
LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_HW_GET_ROUTING;
|
|
uint32_t r;
|
|
uint32_t err = mAudioHardware->getRouting(mode, &r);
|
|
if (err == NO_ERROR) {
|
|
r = (r & ~mask) | (routes & mask);
|
|
mHardwareStatus = AUDIO_HW_SET_ROUTING;
|
|
err = mAudioHardware->setRouting(mode, r);
|
|
}
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return err;
|
|
}
|
|
|
|
uint32_t AudioFlinger::getRouting(int mode) const
|
|
{
|
|
uint32_t routes = 0;
|
|
if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
|
|
mHardwareStatus = AUDIO_HW_GET_ROUTING;
|
|
mAudioHardware->getRouting(mode, &routes);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
} else {
|
|
LOGW("Illegal value: getRouting(%d)", mode);
|
|
}
|
|
return routes;
|
|
}
|
|
|
|
status_t AudioFlinger::setMode(int mode)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
|
|
LOGW("Illegal value: setMode(%d)", mode);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_HW_SET_MODE;
|
|
status_t ret = mAudioHardware->setMode(mode);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return ret;
|
|
}
|
|
|
|
int AudioFlinger::getMode() const
|
|
{
|
|
int mode = AudioSystem::MODE_INVALID;
|
|
mHardwareStatus = AUDIO_HW_SET_MODE;
|
|
mAudioHardware->getMode(&mode);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return mode;
|
|
}
|
|
|
|
status_t AudioFlinger::setMicMute(bool state)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
|
|
status_t ret = mAudioHardware->setMicMute(state);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return ret;
|
|
}
|
|
|
|
bool AudioFlinger::getMicMute() const
|
|
{
|
|
bool state = AudioSystem::MODE_INVALID;
|
|
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
|
|
mAudioHardware->getMicMute(&state);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return state;
|
|
}
|
|
|
|
status_t AudioFlinger::setMasterMute(bool muted)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
mMasterMute = muted;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::masterVolume() const
|
|
{
|
|
return mMasterVolume;
|
|
}
|
|
|
|
bool AudioFlinger::masterMute() const
|
|
{
|
|
return mMasterMute;
|
|
}
|
|
|
|
status_t AudioFlinger::setStreamVolume(int stream, float value)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
mStreamTypes[stream].volume = value;
|
|
status_t ret = NO_ERROR;
|
|
if (stream == AudioTrack::VOICE_CALL) {
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
|
|
ret = mAudioHardware->setVoiceVolume(value);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
status_t AudioFlinger::setStreamMute(int stream, bool muted)
|
|
{
|
|
// check calling permissions
|
|
if (!settingsAllowed()) {
|
|
return PERMISSION_DENIED;
|
|
}
|
|
|
|
if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
|
|
return BAD_VALUE;
|
|
}
|
|
mStreamTypes[stream].mute = muted;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
float AudioFlinger::streamVolume(int stream) const
|
|
{
|
|
if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
|
|
return 0.0f;
|
|
}
|
|
return mStreamTypes[stream].volume;
|
|
}
|
|
|
|
bool AudioFlinger::streamMute(int stream) const
|
|
{
|
|
if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) {
|
|
return true;
|
|
}
|
|
return mStreamTypes[stream].mute;
|
|
}
|
|
|
|
bool AudioFlinger::isMusicActive() const
|
|
{
|
|
size_t count = mActiveTracks.size();
|
|
for (size_t i = 0 ; i < count ; ++i) {
|
|
sp<Track> t = mActiveTracks[i].promote();
|
|
if (t == 0) continue;
|
|
Track* const track = t.get();
|
|
if (t->mStreamType == AudioTrack::MUSIC)
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::setParameter(const char* key, const char* value)
|
|
{
|
|
status_t result;
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_SET_PARAMETER;
|
|
result = mAudioHardware->setParameter(key, value);
|
|
mHardwareStatus = AUDIO_HW_IDLE;
|
|
return result;
|
|
}
|
|
|
|
void AudioFlinger::removeClient(pid_t pid)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
mClients.removeItem(pid);
|
|
}
|
|
|
|
status_t AudioFlinger::addTrack(const sp<Track>& track)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
// here the track could be either new, or restarted
|
|
// in both cases "unstop" the track
|
|
if (track->isPaused()) {
|
|
track->mState = TrackBase::RESUMING;
|
|
LOGV("PAUSED => RESUMING (%d)", track->name());
|
|
} else {
|
|
track->mState = TrackBase::ACTIVE;
|
|
LOGV("? => ACTIVE (%d)", track->name());
|
|
}
|
|
// set retry count for buffer fill
|
|
track->mRetryCount = kMaxTrackStartupRetries;
|
|
LOGV("mWaitWorkCV.broadcast");
|
|
mWaitWorkCV.broadcast();
|
|
|
|
if (mActiveTracks.indexOf(track) < 0) {
|
|
// the track is newly added, make sure it fills up all its
|
|
// buffers before playing. This is to ensure the client will
|
|
// effectively get the latency it requested.
|
|
track->mFillingUpStatus = Track::FS_FILLING;
|
|
mActiveTracks.add(track);
|
|
return NO_ERROR;
|
|
}
|
|
return ALREADY_EXISTS;
|
|
}
|
|
|
|
void AudioFlinger::removeTrack(wp<Track> track, int name)
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
sp<Track> t = track.promote();
|
|
if (t!=NULL && (t->mState <= TrackBase::STOPPED)) {
|
|
remove_track_l(track, name);
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::remove_track_l(wp<Track> track, int name)
|
|
{
|
|
sp<Track> t = track.promote();
|
|
if (t!=NULL) {
|
|
t->reset();
|
|
}
|
|
audioMixer().deleteTrackName(name);
|
|
mActiveTracks.remove(track);
|
|
mWaitWorkCV.broadcast();
|
|
}
|
|
|
|
void AudioFlinger::destroyTrack(const sp<Track>& track)
|
|
{
|
|
// NOTE: We're acquiring a strong reference on the track before
|
|
// acquiring the lock, this is to make sure removing it from
|
|
// mTracks won't cause the destructor to be called while the lock is
|
|
// held (note that technically, 'track' could be a reference to an item
|
|
// in mTracks, which is why we need to do this).
|
|
sp<Track> keep(track);
|
|
Mutex::Autolock _l(mLock);
|
|
track->mState = TrackBase::TERMINATED;
|
|
if (mActiveTracks.indexOf(track) < 0) {
|
|
LOGV("remove track (%d) and delete from mixer", track->name());
|
|
mTracks.remove(track);
|
|
audioMixer().deleteTrackName(keep->name());
|
|
}
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
|
|
: RefBase(),
|
|
mAudioFlinger(audioFlinger),
|
|
mMemoryDealer(new MemoryDealer(1024*1024)),
|
|
mPid(pid)
|
|
{
|
|
// 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
|
|
}
|
|
|
|
AudioFlinger::Client::~Client()
|
|
{
|
|
mAudioFlinger->removeClient(mPid);
|
|
}
|
|
|
|
const sp<MemoryDealer>& AudioFlinger::Client::heap() const
|
|
{
|
|
return mMemoryDealer;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::TrackBase::TrackBase(
|
|
const sp<AudioFlinger>& audioFlinger,
|
|
const sp<Client>& client,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int bufferCount,
|
|
int bufferSize)
|
|
: RefBase(),
|
|
mAudioFlinger(audioFlinger),
|
|
mClient(client),
|
|
mStreamType(streamType),
|
|
mFormat(format),
|
|
mChannelCount(channelCount),
|
|
mBufferCount(bufferCount),
|
|
mFlags(0),
|
|
mBufferSize(bufferSize),
|
|
mState(IDLE),
|
|
mClientTid(-1)
|
|
{
|
|
mName = audioFlinger->audioMixer().getTrackName();
|
|
if (mName < 0) {
|
|
LOGE("no more track names availlable");
|
|
return;
|
|
}
|
|
|
|
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
|
|
size_t size = sizeof(audio_track_cblk_t) + bufferCount * bufferSize;
|
|
mCblkMemory = client->heap()->allocate(size);
|
|
if (mCblkMemory != 0) {
|
|
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
|
|
if (mCblk) { // construct the shared structure in-place.
|
|
new(mCblk) audio_track_cblk_t();
|
|
// clear all buffers
|
|
mCblk->size = bufferSize;
|
|
mCblk->sampleRate = sampleRate;
|
|
mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t);
|
|
memset(mBuffers, 0, bufferCount * bufferSize);
|
|
}
|
|
} else {
|
|
LOGE("not enough memory for AudioTrack size=%u", size);
|
|
client->heap()->dump("AudioTrack");
|
|
return;
|
|
}
|
|
}
|
|
|
|
AudioFlinger::TrackBase::~TrackBase()
|
|
{
|
|
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
|
|
mCblkMemory.clear(); // and free the shared memory
|
|
mClient.clear();
|
|
}
|
|
|
|
void AudioFlinger::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
buffer->raw = 0;
|
|
buffer->frameCount = 0;
|
|
step();
|
|
}
|
|
|
|
bool AudioFlinger::TrackBase::step() {
|
|
bool result;
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
|
|
result = cblk->stepServer(bufferCount());
|
|
if (!result) {
|
|
LOGV("stepServer failed acquiring cblk mutex");
|
|
mFlags |= STEPSERVER_FAILED;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
void AudioFlinger::TrackBase::reset() {
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
|
|
cblk->user = 0;
|
|
cblk->server = 0;
|
|
mFlags = 0;
|
|
}
|
|
|
|
sp<IMemory> AudioFlinger::TrackBase::getCblk() const
|
|
{
|
|
return mCblkMemory;
|
|
}
|
|
|
|
int AudioFlinger::TrackBase::sampleRate() const {
|
|
return mCblk->sampleRate;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::Track::Track(
|
|
const sp<AudioFlinger>& audioFlinger,
|
|
const sp<Client>& client,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int bufferCount,
|
|
int bufferSize)
|
|
: TrackBase(audioFlinger, client, streamType, sampleRate, format, channelCount, bufferCount, bufferSize)
|
|
{
|
|
mVolume[0] = 1.0f;
|
|
mVolume[1] = 1.0f;
|
|
mMute = false;
|
|
}
|
|
|
|
AudioFlinger::Track::~Track()
|
|
{
|
|
wp<Track> weak(this); // never create a strong ref from the dtor
|
|
mState = TERMINATED;
|
|
mAudioFlinger->removeTrack(weak, mName);
|
|
}
|
|
|
|
void AudioFlinger::Track::destroy()
|
|
{
|
|
mAudioFlinger->destroyTrack(this);
|
|
}
|
|
|
|
void AudioFlinger::Track::dump(char* buffer, size_t size)
|
|
{
|
|
snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
|
|
mName - AudioMixer::TRACK0,
|
|
mClient->pid(),
|
|
mStreamType,
|
|
mFormat,
|
|
mChannelCount,
|
|
mBufferCount,
|
|
mState,
|
|
mMute,
|
|
mFillingUpStatus,
|
|
mCblk->sampleRate,
|
|
mCblk->volume[0],
|
|
mCblk->volume[1],
|
|
mCblk->server,
|
|
mCblk->user);
|
|
}
|
|
|
|
status_t AudioFlinger::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
uint32_t u = cblk->user;
|
|
uint32_t s = cblk->server;
|
|
|
|
// Check if last stepServer failed, try to step now
|
|
if (mFlags & TrackBase::STEPSERVER_FAILED) {
|
|
if (!step()) goto getNextBuffer_exit;
|
|
LOGV("stepServer recovered");
|
|
mFlags &= ~TrackBase::STEPSERVER_FAILED;
|
|
}
|
|
|
|
if (LIKELY(u > s)) {
|
|
int index = s & audio_track_cblk_t::BUFFER_MASK;
|
|
buffer->raw = getBuffer(index);
|
|
buffer->frameCount = mAudioFlinger->frameCount();
|
|
return NO_ERROR;
|
|
}
|
|
getNextBuffer_exit:
|
|
buffer->raw = 0;
|
|
buffer->frameCount = 0;
|
|
return NOT_ENOUGH_DATA;
|
|
}
|
|
|
|
bool AudioFlinger::Track::isReady(uint32_t u, int32_t s) const {
|
|
if (mFillingUpStatus != FS_FILLING) return true;
|
|
const uint32_t u_seq = u & audio_track_cblk_t::SEQUENCE_MASK;
|
|
const uint32_t u_buf = u & audio_track_cblk_t::BUFFER_MASK;
|
|
const uint32_t s_seq = s & audio_track_cblk_t::SEQUENCE_MASK;
|
|
const uint32_t s_buf = s & audio_track_cblk_t::BUFFER_MASK;
|
|
if (u_seq > s_seq && u_buf == s_buf) {
|
|
mFillingUpStatus = FS_FILLED;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::Track::start()
|
|
{
|
|
LOGV("start(%d)", mName);
|
|
mAudioFlinger->addTrack(this);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioFlinger::Track::stop()
|
|
{
|
|
LOGV("stop(%d)", mName);
|
|
Mutex::Autolock _l(mAudioFlinger->mLock);
|
|
if (mState > STOPPED) {
|
|
mState = STOPPED;
|
|
// If the track is not active (PAUSED and buffers full), flush buffers
|
|
if (mAudioFlinger->mActiveTracks.indexOf(this) < 0) {
|
|
reset();
|
|
}
|
|
LOGV("(> STOPPED) => STOPPED (%d)", mName);
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::Track::pause()
|
|
{
|
|
LOGV("pause(%d)", mName);
|
|
Mutex::Autolock _l(mAudioFlinger->mLock);
|
|
if (mState == ACTIVE || mState == RESUMING) {
|
|
mState = PAUSING;
|
|
LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::Track::flush()
|
|
{
|
|
LOGV("flush(%d)", mName);
|
|
Mutex::Autolock _l(mAudioFlinger->mLock);
|
|
if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
|
|
return;
|
|
}
|
|
// No point remaining in PAUSED state after a flush => go to
|
|
// STOPPED state
|
|
mState = STOPPED;
|
|
|
|
// NOTE: reset() will reset cblk->user and cblk->server with
|
|
// the risk that at the same time, the AudioMixer is trying to read
|
|
// data. In this case, getNextBuffer() would return a NULL pointer
|
|
// as audio buffer => the AudioMixer code MUST always test that pointer
|
|
// returned by getNextBuffer() is not NULL!
|
|
reset();
|
|
}
|
|
|
|
void AudioFlinger::Track::reset()
|
|
{
|
|
TrackBase::reset();
|
|
mFillingUpStatus = FS_FILLING;
|
|
}
|
|
|
|
void AudioFlinger::Track::mute(bool muted)
|
|
{
|
|
mMute = muted;
|
|
}
|
|
|
|
void AudioFlinger::Track::setVolume(float left, float right)
|
|
{
|
|
mVolume[0] = left;
|
|
mVolume[1] = right;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::Track>& track)
|
|
: BnAudioTrack(),
|
|
mTrack(track)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::TrackHandle::~TrackHandle() {
|
|
// just stop the track on deletion, associated resources
|
|
// will be freed from the main thread once all pending buffers have
|
|
// been played. Unless it's not in the active track list, in which
|
|
// case we free everything now...
|
|
mTrack->destroy();
|
|
}
|
|
|
|
status_t AudioFlinger::TrackHandle::start() {
|
|
return mTrack->start();
|
|
}
|
|
|
|
void AudioFlinger::TrackHandle::stop() {
|
|
mTrack->stop();
|
|
}
|
|
|
|
void AudioFlinger::TrackHandle::flush() {
|
|
mTrack->flush();
|
|
}
|
|
|
|
void AudioFlinger::TrackHandle::mute(bool e) {
|
|
mTrack->mute(e);
|
|
}
|
|
|
|
void AudioFlinger::TrackHandle::pause() {
|
|
mTrack->pause();
|
|
}
|
|
|
|
void AudioFlinger::TrackHandle::setVolume(float left, float right) {
|
|
mTrack->setVolume(left, right);
|
|
}
|
|
|
|
sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
|
|
return mTrack->getCblk();
|
|
}
|
|
|
|
status_t AudioFlinger::TrackHandle::onTransact(
|
|
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
|
|
{
|
|
return BnAudioTrack::onTransact(code, data, reply, flags);
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
sp<AudioFlinger::AudioRecordThread> AudioFlinger::audioRecordThread()
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
return mAudioRecordThread;
|
|
}
|
|
|
|
void AudioFlinger::endRecord()
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
mAudioRecordThread.clear();
|
|
}
|
|
|
|
sp<IAudioRecord> AudioFlinger::openRecord(
|
|
pid_t pid,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int bufferCount,
|
|
uint32_t flags)
|
|
{
|
|
sp<AudioRecordThread> thread;
|
|
sp<RecordTrack> recordTrack;
|
|
sp<RecordHandle> recordHandle;
|
|
sp<Client> client;
|
|
wp<Client> wclient;
|
|
AudioStreamIn* input = 0;
|
|
|
|
// check calling permissions
|
|
if (!recordingAllowed()) {
|
|
goto Exit;
|
|
}
|
|
|
|
if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
|
|
LOGE("invalid stream type");
|
|
goto Exit;
|
|
}
|
|
|
|
if (sampleRate > MAX_SAMPLE_RATE) {
|
|
LOGE("Sample rate out of range");
|
|
goto Exit;
|
|
}
|
|
|
|
if (mSampleRate == 0) {
|
|
LOGE("Audio driver not initialized");
|
|
goto Exit;
|
|
}
|
|
|
|
// Create audio thread - take mutex to prevent race condition
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
if (mAudioRecordThread != 0) {
|
|
LOGE("Record channel already open");
|
|
goto Exit;
|
|
}
|
|
thread = new AudioRecordThread(this);
|
|
mAudioRecordThread = thread;
|
|
}
|
|
// It's safe to release the mutex here since the client doesn't get a
|
|
// handle until we return from this call
|
|
|
|
// open driver, initialize h/w
|
|
input = mAudioHardware->openInputStream(
|
|
AudioSystem::PCM_16_BIT, channelCount, sampleRate);
|
|
if (!input) {
|
|
LOGE("Error opening input stream");
|
|
mAudioRecordThread.clear();
|
|
goto Exit;
|
|
}
|
|
|
|
// add client to list
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
wclient = mClients.valueFor(pid);
|
|
if (wclient != NULL) {
|
|
client = wclient.promote();
|
|
} else {
|
|
client = new Client(this, pid);
|
|
mClients.add(pid, client);
|
|
}
|
|
}
|
|
|
|
// create new record track and pass to record thread
|
|
recordTrack = new RecordTrack(this, client, streamType, sampleRate,
|
|
format, channelCount, bufferCount, input->bufferSize());
|
|
|
|
// spin up record thread
|
|
thread->open(recordTrack, input);
|
|
thread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
|
|
|
|
// return to handle to client
|
|
recordHandle = new RecordHandle(recordTrack);
|
|
|
|
Exit:
|
|
return recordHandle;
|
|
}
|
|
|
|
status_t AudioFlinger::startRecord() {
|
|
sp<AudioRecordThread> t = audioRecordThread();
|
|
if (t == 0) return NO_INIT;
|
|
return t->start();
|
|
}
|
|
|
|
void AudioFlinger::stopRecord() {
|
|
sp<AudioRecordThread> t = audioRecordThread();
|
|
if (t != 0) t->stop();
|
|
}
|
|
|
|
void AudioFlinger::exitRecord()
|
|
{
|
|
sp<AudioRecordThread> t = audioRecordThread();
|
|
if (t != 0) t->exit();
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::RecordTrack::RecordTrack(
|
|
const sp<AudioFlinger>& audioFlinger,
|
|
const sp<Client>& client,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int bufferCount,
|
|
int bufferSize)
|
|
: TrackBase(audioFlinger, client, streamType, sampleRate, format,
|
|
channelCount, bufferCount, bufferSize),
|
|
mOverflow(false)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::RecordTrack::~RecordTrack()
|
|
{
|
|
mAudioFlinger->audioMixer().deleteTrackName(mName);
|
|
mAudioFlinger->exitRecord();
|
|
}
|
|
|
|
status_t AudioFlinger::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
|
|
{
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
const uint32_t u_seq = cblk->user & audio_track_cblk_t::SEQUENCE_MASK;
|
|
const uint32_t u_buf = cblk->user & audio_track_cblk_t::BUFFER_MASK;
|
|
const uint32_t s_seq = cblk->server & audio_track_cblk_t::SEQUENCE_MASK;
|
|
const uint32_t s_buf = cblk->server & audio_track_cblk_t::BUFFER_MASK;
|
|
|
|
// Check if last stepServer failed, try to step now
|
|
if (mFlags & TrackBase::STEPSERVER_FAILED) {
|
|
if (!step()) goto getNextBuffer_exit;
|
|
LOGV("stepServer recovered");
|
|
mFlags &= ~TrackBase::STEPSERVER_FAILED;
|
|
}
|
|
|
|
if (LIKELY(s_seq == u_seq || s_buf != u_buf)) {
|
|
buffer->raw = getBuffer(s_buf);
|
|
buffer->frameCount = mAudioFlinger->frameCount();
|
|
return NO_ERROR;
|
|
}
|
|
|
|
getNextBuffer_exit:
|
|
buffer->raw = 0;
|
|
buffer->frameCount = 0;
|
|
return NOT_ENOUGH_DATA;
|
|
}
|
|
|
|
status_t AudioFlinger::RecordTrack::start()
|
|
{
|
|
return mAudioFlinger->startRecord();
|
|
}
|
|
|
|
void AudioFlinger::RecordTrack::stop()
|
|
{
|
|
mAudioFlinger->stopRecord();
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordTrack>& recordTrack)
|
|
: BnAudioRecord(),
|
|
mRecordTrack(recordTrack)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::RecordHandle::~RecordHandle() {}
|
|
|
|
status_t AudioFlinger::RecordHandle::start() {
|
|
LOGV("RecordHandle::start()");
|
|
return mRecordTrack->start();
|
|
}
|
|
|
|
void AudioFlinger::RecordHandle::stop() {
|
|
LOGV("RecordHandle::stop()");
|
|
mRecordTrack->stop();
|
|
}
|
|
|
|
sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
|
|
return mRecordTrack->getCblk();
|
|
}
|
|
|
|
status_t AudioFlinger::RecordHandle::onTransact(
|
|
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
|
|
{
|
|
return BnAudioRecord::onTransact(code, data, reply, flags);
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::AudioRecordThread::AudioRecordThread(const sp<AudioFlinger>& audioFlinger) :
|
|
mAudioFlinger(audioFlinger),
|
|
mRecordTrack(0),
|
|
mInput(0),
|
|
mActive(false)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::AudioRecordThread::~AudioRecordThread()
|
|
{
|
|
}
|
|
|
|
bool AudioFlinger::AudioRecordThread::threadLoop()
|
|
{
|
|
LOGV("AudioRecordThread: start record loop");
|
|
|
|
// start recording
|
|
while (!exitPending()) {
|
|
if (!mActive) {
|
|
mLock.lock();
|
|
if (!mActive && !exitPending()) {
|
|
LOGV("AudioRecordThread: loop stopping");
|
|
mWaitWorkCV.wait(mLock);
|
|
LOGV("AudioRecordThread: loop starting");
|
|
}
|
|
mLock.unlock();
|
|
} else {
|
|
// promote strong ref so track isn't deleted while we access it
|
|
sp<RecordTrack> t = mRecordTrack.promote();
|
|
|
|
// if we lose the weak reference, client is gone.
|
|
if (t == 0) {
|
|
LOGV("AudioRecordThread: client deleted track");
|
|
break;
|
|
}
|
|
|
|
if (LIKELY(t->getNextBuffer(&mBuffer) == NO_ERROR)) {
|
|
if (mInput->read(mBuffer.raw, t->mBufferSize) < 0) {
|
|
LOGE("Error reading audio input");
|
|
sleep(1);
|
|
}
|
|
t->releaseBuffer(&mBuffer);
|
|
}
|
|
|
|
// client isn't retrieving buffers fast enough
|
|
else {
|
|
if (!t->setOverflow())
|
|
LOGW("AudioRecordThread: buffer overflow");
|
|
}
|
|
}
|
|
};
|
|
|
|
// close hardware
|
|
close();
|
|
|
|
// delete this object - no more data references after this call
|
|
mAudioFlinger->endRecord();
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::AudioRecordThread::open(const sp<RecordTrack>& recordTrack, AudioStreamIn *input) {
|
|
LOGV("AudioRecordThread::open");
|
|
// check for record channel already open
|
|
AutoMutex lock(&mLock);
|
|
if (mRecordTrack != NULL) {
|
|
LOGE("Record channel already open");
|
|
return ALREADY_EXISTS;
|
|
}
|
|
mRecordTrack = recordTrack;
|
|
mInput = input;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::AudioRecordThread::start()
|
|
{
|
|
LOGV("AudioRecordThread::start");
|
|
AutoMutex lock(&mLock);
|
|
if (mActive) return -EBUSY;
|
|
|
|
sp<RecordTrack> t = mRecordTrack.promote();
|
|
if (t == 0) return UNKNOWN_ERROR;
|
|
|
|
// signal thread to start
|
|
LOGV("Signal record thread");
|
|
mActive = true;
|
|
mWaitWorkCV.signal();
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioFlinger::AudioRecordThread::stop() {
|
|
LOGV("AudioRecordThread::stop");
|
|
AutoMutex lock(&mLock);
|
|
if (mActive) {
|
|
mActive = false;
|
|
mWaitWorkCV.signal();
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::AudioRecordThread::exit()
|
|
{
|
|
LOGV("AudioRecordThread::exit");
|
|
AutoMutex lock(&mLock);
|
|
requestExit();
|
|
mWaitWorkCV.signal();
|
|
}
|
|
|
|
|
|
status_t AudioFlinger::AudioRecordThread::close()
|
|
{
|
|
LOGV("AudioRecordThread::close");
|
|
AutoMutex lock(&mLock);
|
|
if (!mInput) return NO_INIT;
|
|
delete mInput;
|
|
mInput = 0;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioFlinger::onTransact(
|
|
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
|
|
{
|
|
return BnAudioFlinger::onTransact(code, data, reply, flags);
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
void AudioFlinger::instantiate() {
|
|
defaultServiceManager()->addService(
|
|
String16("media.audio_flinger"), new AudioFlinger());
|
|
}
|
|
|
|
}; // namespace android
|