1721 lines
51 KiB
C++
1721 lines
51 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 <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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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), mHardwareAudioMixer(0), mA2dpAudioMixer(0),
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mAudioMixer(0), mAudioHardware(0), mA2dpAudioInterface(0), mHardwareOutput(0),
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mA2dpOutput(0), mOutput(0), mRequestedOutput(0), mAudioRecordThread(0),
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mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0),
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mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false),
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mInWrite(false), mA2dpDisableCount(0), mA2dpSuppressed(false)
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{
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mHardwareStatus = AUDIO_HW_IDLE;
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mAudioHardware = AudioHardwareInterface::create();
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mHardwareStatus = AUDIO_HW_INIT;
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if (mAudioHardware->initCheck() == NO_ERROR) {
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// open 16-bit output stream for s/w mixer
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mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
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status_t status;
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mHardwareOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
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mHardwareStatus = AUDIO_HW_IDLE;
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if (mHardwareOutput) {
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mHardwareAudioMixer = new AudioMixer(getOutputFrameCount(mHardwareOutput), mHardwareOutput->sampleRate());
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mRequestedOutput = mHardwareOutput;
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doSetOutput(mHardwareOutput);
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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, status: %d", status);
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}
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#ifdef WITH_A2DP
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// Create A2DP interface
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mA2dpAudioInterface = new A2dpAudioInterface();
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mA2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
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mA2dpAudioMixer = new AudioMixer(getOutputFrameCount(mA2dpOutput), mA2dpOutput->sampleRate());
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// create a buffer big enough for both hardware and A2DP audio output.
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size_t hwFrameCount = getOutputFrameCount(mHardwareOutput);
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size_t a2dpFrameCount = getOutputFrameCount(mA2dpOutput);
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size_t frameCount = (hwFrameCount > a2dpFrameCount ? hwFrameCount : a2dpFrameCount);
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#else
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size_t frameCount = getOutputFrameCount(mHardwareOutput);
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#endif
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// FIXME - Current mixer implementation only supports stereo output: Always
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// Allocate a stereo buffer even if HW output is mono.
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mMixBuffer = new int16_t[frameCount * 2];
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memset(mMixBuffer, 0, frameCount * 2 * sizeof(int16_t));
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// Start record thread
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mAudioRecordThread = new AudioRecordThread(mAudioHardware);
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if (mAudioRecordThread != 0) {
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mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
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}
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} else {
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LOGE("Couldn't even initialize the stubbed audio hardware!");
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}
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char value[PROPERTY_VALUE_MAX];
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property_get("ro.audio.silent", value, "0");
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if (atoi(value)) {
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LOGD("Silence is golden");
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mMasterMute = true;
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}
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}
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AudioFlinger::~AudioFlinger()
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{
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if (mAudioRecordThread != 0) {
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mAudioRecordThread->exit();
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mAudioRecordThread.clear();
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}
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delete mAudioHardware;
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// deleting mA2dpAudioInterface also deletes mA2dpOutput;
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delete mA2dpAudioInterface;
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delete [] mMixBuffer;
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delete mHardwareAudioMixer;
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delete mA2dpAudioMixer;
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}
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void AudioFlinger::setOutput(AudioStreamOut* output)
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{
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mRequestedOutput = output;
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}
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void AudioFlinger::doSetOutput(AudioStreamOut* output)
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{
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mSampleRate = output->sampleRate();
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mChannelCount = output->channelCount();
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// FIXME - Current mixer implementation only supports stereo output
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if (mChannelCount == 1) {
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LOGE("Invalid audio hardware channel count");
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}
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mFormat = output->format();
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mFrameCount = getOutputFrameCount(output);
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mAudioMixer = (output == mA2dpOutput ? mA2dpAudioMixer : mHardwareAudioMixer);
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mOutput = output;
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}
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size_t AudioFlinger::getOutputFrameCount(AudioStreamOut* output)
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{
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return output->bufferSize() / output->channelCount() / sizeof(int16_t);
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}
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#ifdef WITH_A2DP
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bool AudioFlinger::streamDisablesA2dp(int streamType)
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{
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return (streamType == AudioTrack::SYSTEM ||
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streamType == AudioTrack::RING ||
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streamType == AudioTrack::ALARM ||
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streamType == AudioTrack::NOTIFICATION);
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}
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void AudioFlinger::setA2dpEnabled(bool enable)
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{
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if (enable) {
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LOGD("set output to A2DP\n");
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setOutput(mA2dpOutput);
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} else {
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LOGD("set output to hardware audio\n");
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setOutput(mHardwareOutput);
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}
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}
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#endif // WITH_A2DP
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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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unsigned long sleepTime = kBufferRecoveryInUsecs;
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int16_t* curBuf = mMixBuffer;
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Vector< sp<Track> > tracksToRemove;
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size_t enabledTracks = 0;
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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 mLock
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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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mOutput->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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// check for change in output
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if (mRequestedOutput != mOutput) {
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// put current output into standby mode
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if (mOutput) mOutput->standby();
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// change output
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doSetOutput(mRequestedOutput);
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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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// 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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mAudioMixer->setActiveTrack(track->name());
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if (cblk->framesReady() && (track->isReady() || 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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mAudioMixer->setBufferProvider(track);
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mAudioMixer->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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mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
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mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
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mAudioMixer->setParameter(
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AudioMixer::TRACK,
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AudioMixer::FORMAT, track->format());
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mAudioMixer->setParameter(
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AudioMixer::TRACK,
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AudioMixer::CHANNEL_COUNT, track->channelCount());
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mAudioMixer->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->reset();
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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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mAudioMixer->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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removeActiveTrack(track);
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if (track->isTerminated()) {
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mTracks.remove(track);
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mAudioMixer->deleteTrackName(track->mName);
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}
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}
|
|
}
|
|
}
|
|
if (LIKELY(enabledTracks)) {
|
|
// mix buffers...
|
|
mAudioMixer->process(curBuf);
|
|
|
|
// output audio to hardware
|
|
mLastWriteTime = systemTime();
|
|
mInWrite = true;
|
|
size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
|
|
mOutput->write(curBuf, mixBufferSize);
|
|
mNumWrites++;
|
|
mInWrite = false;
|
|
mStandby = false;
|
|
nsecs_t temp = systemTime();
|
|
standbyTime = temp + kStandbyTimeInNsecs;
|
|
nsecs_t delta = temp - mLastWriteTime;
|
|
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
|
|
if (delta > maxPeriod) {
|
|
LOGW("write blocked for %llu msecs", ns2ms(delta));
|
|
mNumDelayedWrites++;
|
|
}
|
|
sleepTime = kBufferRecoveryInUsecs;
|
|
} else {
|
|
// There was nothing to mix this round, which means all
|
|
// active tracks were late. Sleep a little bit to give
|
|
// them another chance. If we're too late, the audio
|
|
// hardware will zero-fill for us.
|
|
LOGV("no buffers - usleep(%lu)", sleepTime);
|
|
usleep(sleepTime);
|
|
if (sleepTime < kMaxBufferRecoveryInUsecs) {
|
|
sleepTime += kBufferRecoveryInUsecs;
|
|
}
|
|
}
|
|
|
|
// finally let go of all our tracks, without the lock held
|
|
// since we can't guarantee the destructors won't acquire that
|
|
// same lock.
|
|
tracksToRemove.clear();
|
|
} while (true);
|
|
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::readyToRun()
|
|
{
|
|
if (mSampleRate == 0) {
|
|
LOGE("No working audio driver found.");
|
|
return NO_INIT;
|
|
}
|
|
LOGI("AudioFlinger's main thread ready to run.");
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioFlinger::onFirstRef()
|
|
{
|
|
run("AudioFlinger", ANDROID_PRIORITY_URGENT_AUDIO);
|
|
}
|
|
|
|
// IAudioFlinger interface
|
|
sp<IAudioTrack> AudioFlinger::createTrack(
|
|
pid_t pid,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags,
|
|
const sp<IMemory>& sharedBuffer,
|
|
status_t *status)
|
|
{
|
|
sp<Track> track;
|
|
sp<TrackHandle> trackHandle;
|
|
sp<Client> client;
|
|
wp<Client> wclient;
|
|
status_t lStatus;
|
|
|
|
if (streamType >= AudioTrack::NUM_STREAM_TYPES) {
|
|
LOGE("invalid stream type");
|
|
lStatus = BAD_VALUE;
|
|
goto Exit;
|
|
}
|
|
|
|
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
|
|
if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) {
|
|
LOGE("Sample rate out of range: %d", sampleRate);
|
|
lStatus = BAD_VALUE;
|
|
goto Exit;
|
|
}
|
|
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
|
|
if (mSampleRate == 0) {
|
|
LOGE("Audio driver not initialized.");
|
|
lStatus = NO_INIT;
|
|
goto Exit;
|
|
}
|
|
|
|
wclient = mClients.valueFor(pid);
|
|
|
|
if (wclient != NULL) {
|
|
client = wclient.promote();
|
|
} else {
|
|
client = new Client(this, pid);
|
|
mClients.add(pid, client);
|
|
}
|
|
|
|
track = new Track(this, client, streamType, sampleRate, format,
|
|
channelCount, frameCount, sharedBuffer);
|
|
mTracks.add(track);
|
|
trackHandle = new TrackHandle(track);
|
|
|
|
lStatus = NO_ERROR;
|
|
}
|
|
|
|
Exit:
|
|
if(status) {
|
|
*status = lStatus;
|
|
}
|
|
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;
|
|
}
|
|
|
|
uint32_t AudioFlinger::latency() const
|
|
{
|
|
if (mOutput) {
|
|
return mOutput->latency();
|
|
}
|
|
else {
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
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)
|
|
{
|
|
status_t err = NO_ERROR;
|
|
|
|
// 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;
|
|
}
|
|
|
|
#ifdef WITH_A2DP
|
|
LOGD("setRouting %d %d %d\n", mode, routes, mask);
|
|
if (mode == AudioSystem::MODE_NORMAL &&
|
|
(mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
|
|
AutoMutex lock(&mLock);
|
|
|
|
bool enableA2dp = false;
|
|
if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) {
|
|
if (mA2dpDisableCount > 0)
|
|
mA2dpSuppressed = true;
|
|
else
|
|
enableA2dp = true;
|
|
}
|
|
setA2dpEnabled(enableA2dp);
|
|
LOGD("setOutput done\n");
|
|
}
|
|
#endif
|
|
|
|
// do nothing if only A2DP routing is affected
|
|
mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP;
|
|
if (mask) {
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_HW_GET_ROUTING;
|
|
uint32_t r;
|
|
err = mAudioHardware->getRouting(mode, &r);
|
|
if (err == NO_ERROR) {
|
|
r = (r & ~mask) | (routes & mask);
|
|
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, result2;
|
|
AutoMutex lock(mHardwareLock);
|
|
mHardwareStatus = AUDIO_SET_PARAMETER;
|
|
result = mAudioHardware->setParameter(key, value);
|
|
if (mA2dpAudioInterface) {
|
|
result2 = mA2dpAudioInterface->setParameter(key, value);
|
|
if (result2)
|
|
result = result2;
|
|
}
|
|
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;
|
|
track->mResetDone = false;
|
|
addActiveTrack(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);
|
|
removeActiveTrack(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());
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::addActiveTrack(const wp<Track>& t)
|
|
{
|
|
mActiveTracks.add(t);
|
|
|
|
#ifdef WITH_A2DP
|
|
// disable A2DP for certain stream types
|
|
sp<Track> track = t.promote();
|
|
if (streamDisablesA2dp(track->type())) {
|
|
if (mA2dpDisableCount++ == 0 && isA2dpEnabled()) {
|
|
setA2dpEnabled(false);
|
|
mA2dpSuppressed = true;
|
|
LOGD("mA2dpSuppressed = true\n");
|
|
}
|
|
LOGD("mA2dpDisableCount incremented to %d\n", mA2dpDisableCount);
|
|
}
|
|
#endif
|
|
}
|
|
|
|
void AudioFlinger::removeActiveTrack(const wp<Track>& t)
|
|
{
|
|
mActiveTracks.remove(t);
|
|
#ifdef WITH_A2DP
|
|
// disable A2DP for certain stream types
|
|
sp<Track> track = t.promote();
|
|
if (streamDisablesA2dp(track->type())) {
|
|
if (mA2dpDisableCount > 0) {
|
|
mA2dpDisableCount--;
|
|
if (mA2dpDisableCount == 0 && mA2dpSuppressed) {
|
|
setA2dpEnabled(true);
|
|
mA2dpSuppressed = false;
|
|
}
|
|
LOGD("mA2dpDisableCount decremented to %d\n", mA2dpDisableCount);
|
|
} else
|
|
LOGE("mA2dpDisableCount is already zero");
|
|
}
|
|
#endif
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
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 frameCount,
|
|
const sp<IMemory>& sharedBuffer)
|
|
: RefBase(),
|
|
mAudioFlinger(audioFlinger),
|
|
mClient(client),
|
|
mStreamType(streamType),
|
|
mFrameCount(0),
|
|
mState(IDLE),
|
|
mClientTid(-1),
|
|
mFormat(format),
|
|
mFlags(0)
|
|
{
|
|
mName = audioFlinger->audioMixer()->getTrackName();
|
|
if (mName < 0) {
|
|
LOGE("no more track names availlable");
|
|
return;
|
|
}
|
|
|
|
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
|
|
|
|
|
|
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
|
|
size_t size = sizeof(audio_track_cblk_t);
|
|
size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
|
|
if (sharedBuffer == 0) {
|
|
size += bufferSize;
|
|
}
|
|
|
|
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 = 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;
|
|
}
|
|
}
|
|
|
|
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;
|
|
mFrameCount = buffer->frameCount;
|
|
step();
|
|
buffer->frameCount = 0;
|
|
}
|
|
|
|
bool AudioFlinger::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::TrackBase::reset() {
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
|
|
cblk->user = 0;
|
|
cblk->server = 0;
|
|
cblk->userBase = 0;
|
|
cblk->serverBase = 0;
|
|
mFlags = 0;
|
|
LOGV("TrackBase::reset");
|
|
}
|
|
|
|
sp<IMemory> AudioFlinger::TrackBase::getCblk() const
|
|
{
|
|
return mCblkMemory;
|
|
}
|
|
|
|
int AudioFlinger::TrackBase::sampleRate() const {
|
|
return mCblk->sampleRate;
|
|
}
|
|
|
|
int AudioFlinger::TrackBase::channelCount() const {
|
|
return mCblk->channels;
|
|
}
|
|
|
|
void* AudioFlinger::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
|
|
audio_track_cblk_t* cblk = this->cblk();
|
|
int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels;
|
|
int16_t *bufferEnd = bufferStart + frames * cblk->channels;
|
|
|
|
// Check validity of returned pointer in case the track control block would have been corrupted.
|
|
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd) {
|
|
LOGW("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
|
|
server %d, serverBase %d, user %d, userBase %d",
|
|
bufferStart, bufferEnd, mBuffer, mBufferEnd,
|
|
cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
|
|
return 0;
|
|
}
|
|
|
|
return bufferStart;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::Track::Track(
|
|
const sp<AudioFlinger>& audioFlinger,
|
|
const sp<Client>& client,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
const sp<IMemory>& sharedBuffer)
|
|
: TrackBase(audioFlinger, client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer)
|
|
{
|
|
mVolume[0] = 1.0f;
|
|
mVolume[1] = 1.0f;
|
|
mMute = false;
|
|
mSharedBuffer = sharedBuffer;
|
|
}
|
|
|
|
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,
|
|
mCblk->channels,
|
|
mFrameCount,
|
|
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 framesReady;
|
|
uint32_t framesReq = buffer->frameCount;
|
|
|
|
// Check if last stepServer failed, try to step now
|
|
if (mFlags & TrackBase::STEPSERVER_FAILED) {
|
|
if (!step()) goto getNextBuffer_exit;
|
|
LOGV("stepServer recovered");
|
|
mFlags &= ~TrackBase::STEPSERVER_FAILED;
|
|
}
|
|
|
|
framesReady = cblk->framesReady();
|
|
|
|
if (LIKELY(framesReady)) {
|
|
uint32_t s = cblk->server;
|
|
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
|
|
|
|
bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
|
|
if (framesReq > framesReady) {
|
|
framesReq = framesReady;
|
|
}
|
|
if (s + framesReq > bufferEnd) {
|
|
framesReq = bufferEnd - s;
|
|
}
|
|
|
|
buffer->raw = getBuffer(s, framesReq);
|
|
if (buffer->raw == 0) goto getNextBuffer_exit;
|
|
|
|
buffer->frameCount = framesReq;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
getNextBuffer_exit:
|
|
buffer->raw = 0;
|
|
buffer->frameCount = 0;
|
|
return NOT_ENOUGH_DATA;
|
|
}
|
|
|
|
bool AudioFlinger::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::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()
|
|
{
|
|
// 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::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<IAudioRecord> AudioFlinger::openRecord(
|
|
pid_t pid,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount,
|
|
uint32_t flags,
|
|
status_t *status)
|
|
{
|
|
sp<AudioRecordThread> thread;
|
|
sp<RecordTrack> recordTrack;
|
|
sp<RecordHandle> recordHandle;
|
|
sp<Client> client;
|
|
wp<Client> wclient;
|
|
AudioStreamIn* input = 0;
|
|
int inFrameCount;
|
|
size_t inputBufferSize;
|
|
status_t lStatus;
|
|
|
|
// check calling permissions
|
|
if (!recordingAllowed()) {
|
|
lStatus = PERMISSION_DENIED;
|
|
goto Exit;
|
|
}
|
|
|
|
if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
|
|
LOGE("invalid stream type");
|
|
lStatus = BAD_VALUE;
|
|
goto Exit;
|
|
}
|
|
|
|
if (sampleRate > MAX_SAMPLE_RATE) {
|
|
LOGE("Sample rate out of range");
|
|
lStatus = BAD_VALUE;
|
|
goto Exit;
|
|
}
|
|
|
|
if (mSampleRate == 0) {
|
|
LOGE("Audio driver not initialized");
|
|
lStatus = NO_INIT;
|
|
goto Exit;
|
|
}
|
|
|
|
if (mAudioRecordThread == 0) {
|
|
LOGE("Audio record thread not started");
|
|
lStatus = NO_INIT;
|
|
goto Exit;
|
|
}
|
|
|
|
|
|
// Check that audio input stream accepts requested audio parameters
|
|
inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
|
|
if (inputBufferSize == 0) {
|
|
lStatus = BAD_VALUE;
|
|
LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount);
|
|
goto Exit;
|
|
}
|
|
|
|
// add client to list
|
|
{
|
|
Mutex::Autolock _l(mLock);
|
|
wclient = mClients.valueFor(pid);
|
|
if (wclient != NULL) {
|
|
client = wclient.promote();
|
|
} else {
|
|
client = new Client(this, pid);
|
|
mClients.add(pid, client);
|
|
}
|
|
}
|
|
|
|
// frameCount must be a multiple of input buffer size
|
|
inFrameCount = inputBufferSize/channelCount/sizeof(short);
|
|
frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
|
|
|
|
// create new record track and pass to record thread
|
|
recordTrack = new RecordTrack(this, client, streamType, sampleRate,
|
|
format, channelCount, frameCount);
|
|
|
|
// return to handle to client
|
|
recordHandle = new RecordHandle(recordTrack);
|
|
lStatus = NO_ERROR;
|
|
|
|
Exit:
|
|
if (status) {
|
|
*status = lStatus;
|
|
}
|
|
return recordHandle;
|
|
}
|
|
|
|
status_t AudioFlinger::startRecord(RecordTrack* recordTrack) {
|
|
if (mAudioRecordThread != 0) {
|
|
return mAudioRecordThread->start(recordTrack);
|
|
}
|
|
return NO_INIT;
|
|
}
|
|
|
|
void AudioFlinger::stopRecord(RecordTrack* recordTrack) {
|
|
if (mAudioRecordThread != 0) {
|
|
mAudioRecordThread->stop(recordTrack);
|
|
}
|
|
}
|
|
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::RecordTrack::RecordTrack(
|
|
const sp<AudioFlinger>& audioFlinger,
|
|
const sp<Client>& client,
|
|
int streamType,
|
|
uint32_t sampleRate,
|
|
int format,
|
|
int channelCount,
|
|
int frameCount)
|
|
: TrackBase(audioFlinger, client, streamType, sampleRate, format,
|
|
channelCount, frameCount, 0),
|
|
mOverflow(false)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::RecordTrack::~RecordTrack()
|
|
{
|
|
mAudioFlinger->audioMixer()->deleteTrackName(mName);
|
|
}
|
|
|
|
status_t AudioFlinger::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::RecordTrack::start()
|
|
{
|
|
return mAudioFlinger->startRecord(this);
|
|
}
|
|
|
|
void AudioFlinger::RecordTrack::stop()
|
|
{
|
|
mAudioFlinger->stopRecord(this);
|
|
TrackBase::reset();
|
|
// Force overerrun condition to avoid false overrun callback until first data is
|
|
// read from buffer
|
|
mCblk->flowControlFlag = 1;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordTrack>& recordTrack)
|
|
: BnAudioRecord(),
|
|
mRecordTrack(recordTrack)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::RecordHandle::~RecordHandle() {
|
|
stop();
|
|
}
|
|
|
|
status_t AudioFlinger::RecordHandle::start() {
|
|
LOGV("RecordHandle::start()");
|
|
return mRecordTrack->start();
|
|
}
|
|
|
|
void AudioFlinger::RecordHandle::stop() {
|
|
LOGV("RecordHandle::stop()");
|
|
mRecordTrack->stop();
|
|
}
|
|
|
|
sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
|
|
return mRecordTrack->getCblk();
|
|
}
|
|
|
|
status_t AudioFlinger::RecordHandle::onTransact(
|
|
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
|
|
{
|
|
return BnAudioRecord::onTransact(code, data, reply, flags);
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware) :
|
|
mAudioHardware(audioHardware),
|
|
mActive(false)
|
|
{
|
|
}
|
|
|
|
AudioFlinger::AudioRecordThread::~AudioRecordThread()
|
|
{
|
|
}
|
|
|
|
bool AudioFlinger::AudioRecordThread::threadLoop()
|
|
{
|
|
LOGV("AudioRecordThread: start record loop");
|
|
AudioBufferProvider::Buffer buffer;
|
|
int inBufferSize = 0;
|
|
int inFrameCount = 0;
|
|
AudioStreamIn* input = 0;
|
|
|
|
mActive = 0;
|
|
|
|
// start recording
|
|
while (!exitPending()) {
|
|
if (!mActive) {
|
|
mLock.lock();
|
|
if (!mActive && !exitPending()) {
|
|
LOGV("AudioRecordThread: loop stopping");
|
|
if (input) {
|
|
delete input;
|
|
input = 0;
|
|
}
|
|
mRecordTrack.clear();
|
|
|
|
mWaitWorkCV.wait(mLock);
|
|
|
|
LOGV("AudioRecordThread: loop starting");
|
|
if (mRecordTrack != 0) {
|
|
input = mAudioHardware->openInputStream(mRecordTrack->format(),
|
|
mRecordTrack->channelCount(),
|
|
mRecordTrack->sampleRate(),
|
|
&mStartStatus);
|
|
if (input != 0) {
|
|
inBufferSize = input->bufferSize();
|
|
inFrameCount = inBufferSize/input->frameSize();
|
|
}
|
|
} else {
|
|
mStartStatus = NO_INIT;
|
|
}
|
|
if (mStartStatus !=NO_ERROR) {
|
|
LOGW("record start failed, status %d", mStartStatus);
|
|
mActive = false;
|
|
mRecordTrack.clear();
|
|
}
|
|
mWaitWorkCV.signal();
|
|
}
|
|
mLock.unlock();
|
|
} else if (mRecordTrack != 0){
|
|
|
|
buffer.frameCount = inFrameCount;
|
|
if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR)) {
|
|
LOGV("AudioRecordThread read: %d frames", buffer.frameCount);
|
|
if (input->read(buffer.raw, inBufferSize) < 0) {
|
|
LOGE("Error reading audio input");
|
|
sleep(1);
|
|
}
|
|
mRecordTrack->releaseBuffer(&buffer);
|
|
mRecordTrack->overflow();
|
|
}
|
|
|
|
// client isn't retrieving buffers fast enough
|
|
else {
|
|
if (!mRecordTrack->setOverflow())
|
|
LOGW("AudioRecordThread: buffer overflow");
|
|
// Release the processor for a while before asking for a new buffer.
|
|
// This will give the application more chance to read from the buffer and
|
|
// clear the overflow.
|
|
usleep(5000);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
if (input) {
|
|
delete input;
|
|
}
|
|
mRecordTrack.clear();
|
|
|
|
return false;
|
|
}
|
|
|
|
status_t AudioFlinger::AudioRecordThread::start(RecordTrack* recordTrack)
|
|
{
|
|
LOGV("AudioRecordThread::start");
|
|
AutoMutex lock(&mLock);
|
|
mActive = true;
|
|
// If starting the active track, just reset mActive in case a stop
|
|
// was pending and exit
|
|
if (recordTrack == mRecordTrack.get()) return NO_ERROR;
|
|
|
|
if (mRecordTrack != 0) return -EBUSY;
|
|
|
|
mRecordTrack = recordTrack;
|
|
|
|
// signal thread to start
|
|
LOGV("Signal record thread");
|
|
mWaitWorkCV.signal();
|
|
mWaitWorkCV.wait(mLock);
|
|
LOGV("Record started, status %d", mStartStatus);
|
|
return mStartStatus;
|
|
}
|
|
|
|
void AudioFlinger::AudioRecordThread::stop(RecordTrack* recordTrack) {
|
|
LOGV("AudioRecordThread::stop");
|
|
AutoMutex lock(&mLock);
|
|
if (mActive && (recordTrack == mRecordTrack.get())) {
|
|
mActive = false;
|
|
}
|
|
}
|
|
|
|
void AudioFlinger::AudioRecordThread::exit()
|
|
{
|
|
LOGV("AudioRecordThread::exit");
|
|
{
|
|
AutoMutex lock(&mLock);
|
|
requestExit();
|
|
mWaitWorkCV.signal();
|
|
}
|
|
requestExitAndWait();
|
|
}
|
|
|
|
|
|
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
|