/*
**
** Copyright 2008, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioRecord"
#include <stdint.h>
#include <sys/types.h>
#include <sched.h>
#include <sys/resource.h>
#include <private/media/AudioTrackShared.h>
#include <media/AudioSystem.h>
#include <media/AudioRecord.h>
#include <media/mediarecorder.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <binder/IPCThreadState.h>
#include <utils/Timers.h>
#include <cutils/atomic.h>
#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
namespace android {
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord()
: mStatus(NO_INIT)
{
}
AudioRecord::AudioRecord(
int inputSource,
uint32_t sampleRate,
int format,
uint32_t channels,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames)
: mStatus(NO_INIT)
{
mStatus = set(inputSource, sampleRate, format, channels,
frameCount, flags, cbf, user, notificationFrames);
}
AudioRecord::~AudioRecord()
{
if (mStatus == NO_ERROR) {
// Make sure that callback function exits in the case where
// it is looping on buffer empty condition in obtainBuffer().
// Otherwise the callback thread will never exit.
stop();
if (mClientRecordThread != 0) {
mClientRecordThread->requestExitAndWait();
mClientRecordThread.clear();
}
mAudioRecord.clear();
IPCThreadState::self()->flushCommands();
}
}
status_t AudioRecord::set(
int inputSource,
uint32_t sampleRate,
int format,
uint32_t channels,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
bool threadCanCallJava)
{
LOGV("set(): sampleRate %d, channels %d, frameCount %d",sampleRate, channels, frameCount);
if (mAudioRecord != 0) {
return INVALID_OPERATION;
}
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
if (sampleRate == 0) {
sampleRate = DEFAULT_SAMPLE_RATE;
}
// these below should probably come from the audioFlinger too...
if (format == 0) {
format = AudioSystem::PCM_16_BIT;
}
// validate parameters
if (!AudioSystem::isValidFormat(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
if (!AudioSystem::isInputChannel(channels)) {
return BAD_VALUE;
}
int channelCount = AudioSystem::popCount(channels);
audio_io_handle_t input = AudioSystem::getInput(inputSource,
sampleRate, format, channels, (AudioSystem::audio_in_acoustics)flags);
if (input == 0) {
LOGE("Could not get audio input for record source %d", inputSource);
return BAD_VALUE;
}
// validate framecount
size_t inputBuffSizeInBytes = -1;
if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &inputBuffSizeInBytes)
!= NO_ERROR) {
LOGE("AudioSystem could not query the input buffer size.");
return NO_INIT;
}
if (inputBuffSizeInBytes == 0) {
LOGE("Recording parameters are not supported: sampleRate %d, channelCount %d, format %d",
sampleRate, channelCount, format);
return BAD_VALUE;
}
int frameSizeInBytes = channelCount * (format == AudioSystem::PCM_16_BIT ? 2 : 1);
if (AudioSystem::isLinearPCM(format)) {
frameSizeInBytes = channelCount * (format == AudioSystem::PCM_16_BIT ? sizeof(int16_t) : sizeof(int8_t));
} else {
frameSizeInBytes = sizeof(int8_t);
}
// We use 2* size of input buffer for ping pong use of record buffer.
int minFrameCount = 2 * inputBuffSizeInBytes / frameSizeInBytes;
LOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
if (frameCount == 0) {
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
return BAD_VALUE;
}
if (notificationFrames == 0) {
notificationFrames = frameCount/2;
}
// create the IAudioRecord
status_t status = openRecord(sampleRate, format, channelCount,
frameCount, flags, input);
if (status != NO_ERROR) {
return status;
}
if (cbf != 0) {
mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
if (mClientRecordThread == 0) {
return NO_INIT;
}
}
mStatus = NO_ERROR;
mFormat = format;
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount;
mChannelCount = (uint8_t)channelCount;
mChannels = channels;
mActive = 0;
mCbf = cbf;
mNotificationFrames = notificationFrames;
mRemainingFrames = notificationFrames;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mInputSource = (uint8_t)inputSource;
mFlags = flags;
mInput = input;
return NO_ERROR;
}
status_t AudioRecord::initCheck() const
{
return mStatus;
}
// -------------------------------------------------------------------------
uint32_t AudioRecord::latency() const
{
return mLatency;
}
int AudioRecord::format() const
{
return mFormat;
}
int AudioRecord::channelCount() const
{
return mChannelCount;
}
uint32_t AudioRecord::frameCount() const
{
return mFrameCount;
}
int AudioRecord::frameSize() const
{
if (AudioSystem::isLinearPCM(mFormat)) {
return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
} else {
return sizeof(uint8_t);
}
}
int AudioRecord::inputSource() const
{
return (int)mInputSource;
}
// -------------------------------------------------------------------------
status_t AudioRecord::start()
{
status_t ret = NO_ERROR;
sp<ClientRecordThread> t = mClientRecordThread;
LOGV("start");
if (t != 0) {
if (t->exitPending()) {
if (t->requestExitAndWait() == WOULD_BLOCK) {
LOGE("AudioRecord::start called from thread");
return WOULD_BLOCK;
}
}
t->mLock.lock();
}
if (android_atomic_or(1, &mActive) == 0) {
ret = mAudioRecord->start();
if (ret == DEAD_OBJECT) {
LOGV("start() dead IAudioRecord: creating a new one");
ret = openRecord(mCblk->sampleRate, mFormat, mChannelCount,
mFrameCount, mFlags, getInput());
if (ret == NO_ERROR) {
ret = mAudioRecord->start();
}
}
if (ret == NO_ERROR) {
mNewPosition = mCblk->user + mUpdatePeriod;
mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
if (t != 0) {
t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
} else {
setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
}
} else {
LOGV("start() failed");
android_atomic_and(~1, &mActive);
}
}
if (t != 0) {
t->mLock.unlock();
}
return ret;
}
status_t AudioRecord::stop()
{
sp<ClientRecordThread> t = mClientRecordThread;
LOGV("stop");
if (t != 0) {
t->mLock.lock();
}
if (android_atomic_and(~1, &mActive) == 1) {
mCblk->cv.signal();
mAudioRecord->stop();
// the record head position will reset to 0, so if a marker is set, we need
// to activate it again
mMarkerReached = false;
if (t != 0) {
t->requestExit();
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
}
}
if (t != 0) {
t->mLock.unlock();
}
return NO_ERROR;
}
bool AudioRecord::stopped() const
{
return !mActive;
}
uint32_t AudioRecord::getSampleRate()
{
return mCblk->sampleRate;
}
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
if (mCbf == 0) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
return NO_ERROR;
}
status_t AudioRecord::getMarkerPosition(uint32_t *marker)
{
if (marker == 0) return BAD_VALUE;
*marker = mMarkerPosition;
return NO_ERROR;
}
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
if (mCbf == 0) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
mNewPosition = curPosition + updatePeriod;
mUpdatePeriod = updatePeriod;
return NO_ERROR;
}
status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod)
{
if (updatePeriod == 0) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
}
status_t AudioRecord::getPosition(uint32_t *position)
{
if (position == 0) return BAD_VALUE;
*position = mCblk->user;
return NO_ERROR;
}
unsigned int AudioRecord::getInputFramesLost()
{
if (mActive)
return AudioSystem::getInputFramesLost(mInput);
else
return 0;
}
// -------------------------------------------------------------------------
status_t AudioRecord::openRecord(
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
audio_io_handle_t input)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
return NO_INIT;
}
sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), input,
sampleRate, format,
channelCount,
frameCount,
((uint16_t)flags) << 16,
&status);
if (record == 0) {
LOGE("AudioFlinger could not create record track, status: %d", status);
return status;
}
sp<IMemory> cblk = record->getCblk();
if (cblk == 0) {
LOGE("Could not get control block");
return NO_INIT;
}
mAudioRecord.clear();
mAudioRecord = record;
mCblkMemory.clear();
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->out = 0;
mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
return NO_ERROR;
}
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
int active;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
audioBuffer->frameCount = 0;
audioBuffer->size = 0;
uint32_t framesReady = cblk->framesReady();
if (framesReady == 0) {
cblk->lock.lock();
goto start_loop_here;
while (framesReady == 0) {
active = mActive;
if (UNLIKELY(!active)) {
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
if (UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (__builtin_expect(result!=NO_ERROR, false)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
LOGW( "obtainBuffer timed out (is the CPU pegged?) "
"user=%08x, server=%08x", cblk->user, cblk->server);
cblk->lock.unlock();
result = mAudioRecord->start();
if (result == DEAD_OBJECT) {
LOGW("obtainBuffer() dead IAudioRecord: creating a new one");
result = openRecord(cblk->sampleRate, mFormat, mChannelCount,
mFrameCount, mFlags, getInput());
if (result == NO_ERROR) {
cblk = mCblk;
mAudioRecord->start();
}
}
cblk->lock.lock();
cblk->waitTimeMs = 0;
}
if (--waitCount == 0) {
cblk->lock.unlock();
return TIMED_OUT;
}
}
// read the server count again
start_loop_here:
framesReady = cblk->framesReady();
}
cblk->lock.unlock();
}
cblk->waitTimeMs = 0;
if (framesReq > framesReady) {
framesReq = framesReady;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (u + framesReq > bufferEnd) {
framesReq = bufferEnd - u;
}
audioBuffer->flags = 0;
audioBuffer->channelCount= mChannelCount;
audioBuffer->format = mFormat;
audioBuffer->frameCount = framesReq;
audioBuffer->size = framesReq*cblk->frameSize;
audioBuffer->raw = (int8_t*)cblk->buffer(u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
void AudioRecord::releaseBuffer(Buffer* audioBuffer)
{
audio_track_cblk_t* cblk = mCblk;
cblk->stepUser(audioBuffer->frameCount);
}
audio_io_handle_t AudioRecord::getInput()
{
mInput = AudioSystem::getInput(mInputSource,
mCblk->sampleRate,
mFormat, mChannels,
(AudioSystem::audio_in_acoustics)mFlags);
return mInput;
}
// -------------------------------------------------------------------------
ssize_t AudioRecord::read(void* buffer, size_t userSize)
{
ssize_t read = 0;
Buffer audioBuffer;
int8_t *dst = static_cast<int8_t*>(buffer);
if (ssize_t(userSize) < 0) {
// sanity-check. user is most-likely passing an error code.
LOGE("AudioRecord::read(buffer=%p, size=%u (%d)",
buffer, userSize, userSize);
return BAD_VALUE;
}
do {
audioBuffer.frameCount = userSize/frameSize();
// By using a wait count corresponding to twice the timeout period in
// obtainBuffer() we give a chance to recover once for a read timeout
// (if media_server crashed for instance) before returning a length of
// 0 bytes read to the client
status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS));
if (err < 0) {
// out of buffers, return #bytes written
if (err == status_t(NO_MORE_BUFFERS))
break;
if (err == status_t(TIMED_OUT))
err = 0;
return ssize_t(err);
}
size_t bytesRead = audioBuffer.size;
memcpy(dst, audioBuffer.i8, bytesRead);
dst += bytesRead;
userSize -= bytesRead;
read += bytesRead;
releaseBuffer(&audioBuffer);
} while (userSize);
return read;
}
// -------------------------------------------------------------------------
bool AudioRecord::processAudioBuffer(const sp<ClientRecordThread>& thread)
{
Buffer audioBuffer;
uint32_t frames = mRemainingFrames;
size_t readSize;
// Manage marker callback
if (!mMarkerReached && (mMarkerPosition > 0)) {
if (mCblk->user >= mMarkerPosition) {
mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
mMarkerReached = true;
}
}
// Manage new position callback
if (mUpdatePeriod > 0) {
while (mCblk->user >= mNewPosition) {
mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
mNewPosition += mUpdatePeriod;
}
}
do {
audioBuffer.frameCount = frames;
// Calling obtainBuffer() with a wait count of 1
// limits wait time to WAIT_PERIOD_MS. This prevents from being
// stuck here not being able to handle timed events (position, markers).
status_t err = obtainBuffer(&audioBuffer, 1);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {
LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
return false;
}
break;
}
if (err == status_t(STOPPED)) return false;
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
readSize = audioBuffer.size;
// Sanity check on returned size
if (ssize_t(readSize) <= 0) {
// The callback is done filling buffers
// Keep this thread going to handle timed events and
// still try to get more data in intervals of WAIT_PERIOD_MS
// but don't just loop and block the CPU, so wait
usleep(WAIT_PERIOD_MS*1000);
break;
}
if (readSize > reqSize) readSize = reqSize;
audioBuffer.size = readSize;
audioBuffer.frameCount = readSize/frameSize();
frames -= audioBuffer.frameCount;
releaseBuffer(&audioBuffer);
} while (frames);
// Manage overrun callback
if (mActive && (mCblk->framesAvailable_l() == 0)) {
LOGV("Overrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
if (mCblk->flowControlFlag == 0) {
mCbf(EVENT_OVERRUN, mUserData, 0);
mCblk->flowControlFlag = 1;
}
}
if (frames == 0) {
mRemainingFrames = mNotificationFrames;
} else {
mRemainingFrames = frames;
}
return true;
}
// =========================================================================
AudioRecord::ClientRecordThread::ClientRecordThread(AudioRecord& receiver, bool bCanCallJava)
: Thread(bCanCallJava), mReceiver(receiver)
{
}
bool AudioRecord::ClientRecordThread::threadLoop()
{
return mReceiver.processAudioBuffer(this);
}
// -------------------------------------------------------------------------
}; // namespace android