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Diffstat (limited to 'media/libmedia/AudioRecord.cpp')
-rw-r--r--media/libmedia/AudioRecord.cpp90
1 files changed, 57 insertions, 33 deletions
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 0a6f4f7..5e35564 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -28,6 +28,7 @@
#include <media/AudioSystem.h>
#include <media/AudioRecord.h>
+#include <media/mediarecorder.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
@@ -45,7 +46,7 @@ namespace android {
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord()
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT), mInput(0)
{
}
@@ -53,15 +54,15 @@ AudioRecord::AudioRecord(
int inputSource,
uint32_t sampleRate,
int format,
- int channelCount,
+ uint32_t channels,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT), mInput(0)
{
- mStatus = set(inputSource, sampleRate, format, channelCount,
+ mStatus = set(inputSource, sampleRate, format, channels,
frameCount, flags, cbf, user, notificationFrames);
}
@@ -78,6 +79,7 @@ AudioRecord::~AudioRecord()
}
mAudioRecord.clear();
IPCThreadState::self()->flushCommands();
+ AudioSystem::releaseInput(mInput);
}
}
@@ -85,7 +87,7 @@ status_t AudioRecord::set(
int inputSource,
uint32_t sampleRate,
int format,
- int channelCount,
+ uint32_t channels,
int frameCount,
uint32_t flags,
callback_t cbf,
@@ -94,7 +96,7 @@ status_t AudioRecord::set(
bool threadCanCallJava)
{
- LOGV("set(): sampleRate %d, channelCount %d, frameCount %d",sampleRate, channelCount, frameCount);
+ LOGV("set(): sampleRate %d, channels %d, frameCount %d",sampleRate, channels, frameCount);
if (mAudioRecord != 0) {
return INVALID_OPERATION;
}
@@ -104,8 +106,8 @@ status_t AudioRecord::set(
return NO_INIT;
}
- if (inputSource == DEFAULT_INPUT) {
- inputSource = MIC_INPUT;
+ if (inputSource == AUDIO_SOURCE_DEFAULT) {
+ inputSource = AUDIO_SOURCE_MIC;
}
if (sampleRate == 0) {
@@ -115,15 +117,21 @@ status_t AudioRecord::set(
if (format == 0) {
format = AudioSystem::PCM_16_BIT;
}
- if (channelCount == 0) {
- channelCount = 1;
+ // validate parameters
+ if (!AudioSystem::isValidFormat(format)) {
+ LOGE("Invalid format");
+ return BAD_VALUE;
}
- // validate parameters
- if (format != AudioSystem::PCM_16_BIT) {
+ if (!AudioSystem::isInputChannel(channels)) {
return BAD_VALUE;
}
- if (channelCount != 1 && channelCount != 2) {
+ int channelCount = AudioSystem::popCount(channels);
+
+ mInput = AudioSystem::getInput(inputSource,
+ sampleRate, format, channels, (AudioSystem::audio_in_acoustics)flags);
+ if (mInput == 0) {
+ LOGE("Could not get audio output for stream type %d", inputSource);
return BAD_VALUE;
}
@@ -132,14 +140,22 @@ status_t AudioRecord::set(
if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &inputBuffSizeInBytes)
!= NO_ERROR) {
LOGE("AudioSystem could not query the input buffer size.");
- return NO_INIT;
+ 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;
@@ -157,11 +173,11 @@ status_t AudioRecord::set(
// open record channel
status_t status;
- sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), inputSource,
+ sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), mInput,
sampleRate, format,
channelCount,
frameCount,
- ((uint16_t)flags) << 16,
+ ((uint16_t)flags) << 16,
&status);
if (record == 0) {
LOGE("AudioFlinger could not create record track, status: %d", status);
@@ -188,7 +204,7 @@ status_t AudioRecord::set(
mFormat = format;
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount;
- mChannelCount = channelCount;
+ mChannelCount = (uint8_t)channelCount;
mActive = 0;
mCbf = cbf;
mNotificationFrames = notificationFrames;
@@ -234,7 +250,11 @@ uint32_t AudioRecord::frameCount() const
int AudioRecord::frameSize() const
{
- return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+ 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
@@ -262,15 +282,18 @@ status_t AudioRecord::start()
}
if (android_atomic_or(1, &mActive) == 0) {
- 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);
+ ret = AudioSystem::startInput(mInput);
+ 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);
+ }
+ ret = mAudioRecord->start();
}
- ret = mAudioRecord->start();
}
if (t != 0) {
@@ -301,6 +324,7 @@ status_t AudioRecord::stop()
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
}
+ AudioSystem::stopInput(mInput);
}
if (t != 0) {
@@ -421,7 +445,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
"this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
cblk->waitTimeMs = 0;
-
+
if (framesReq > framesReady) {
framesReq = framesReady;
}
@@ -437,7 +461,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
audioBuffer->channelCount= mChannelCount;
audioBuffer->format = mFormat;
audioBuffer->frameCount = framesReq;
- audioBuffer->size = framesReq*mChannelCount*sizeof(int16_t);
+ audioBuffer->size = framesReq*cblk->frameSize;
audioBuffer->raw = (int8_t*)cblk->buffer(u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
@@ -468,7 +492,7 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize)
do {
- audioBuffer.frameCount = userSize/mChannelCount/sizeof(int16_t);
+ audioBuffer.frameCount = userSize/frameSize();
// Calling obtainBuffer() with a negative wait count causes
// an (almost) infinite wait time.
@@ -519,8 +543,8 @@ bool AudioRecord::processAudioBuffer(const sp<ClientRecordThread>& thread)
do {
audioBuffer.frameCount = frames;
- // Calling obtainBuffer() with a wait count of 1
- // limits wait time to WAIT_PERIOD_MS. This prevents from being
+ // 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) {
@@ -548,14 +572,14 @@ bool AudioRecord::processAudioBuffer(const sp<ClientRecordThread>& thread)
if (readSize > reqSize) readSize = reqSize;
audioBuffer.size = readSize;
- audioBuffer.frameCount = readSize/mChannelCount/sizeof(int16_t);
+ 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);
generated by cgit v1.1 at 2024-05-15 23:37:09 +0000