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-rw-r--r--media/libmedia/Android.mk1
-rw-r--r--media/libmedia/AudioRecord.cpp80
-rw-r--r--media/libmedia/AudioTrack.cpp19
3 files changed, 56 insertions, 44 deletions
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index e0acae6..f3770e4 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -72,7 +72,6 @@ LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper
LOCAL_MODULE:= libmedia
LOCAL_C_INCLUDES := \
- $(call include-path-for, graphics corecg) \
$(TOP)/frameworks/native/include/media/openmax \
external/icu4c/common \
external/icu4c/i18n \
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 700718d..4438dfd 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -41,30 +41,22 @@ status_t AudioRecord::getMinFrameCount(
return BAD_VALUE;
}
- // default to 0 in case of error
- *frameCount = 0;
-
- size_t size = 0;
+ size_t size;
status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
if (status != NO_ERROR) {
- ALOGE("AudioSystem could not query the input buffer size; status %d", status);
- return NO_INIT;
+ ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, "
+ "channelMask %#x; status %d", sampleRate, format, channelMask, status);
+ return status;
}
- if (size == 0) {
+ // We double the size of input buffer for ping pong use of record buffer.
+ // Assumes audio_is_linear_pcm(format)
+ if ((*frameCount = (size * 2) / (popcount(channelMask) * audio_bytes_per_sample(format))) == 0) {
ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
- // We double the size of input buffer for ping pong use of record buffer.
- size <<= 1;
-
- // Assumes audio_is_linear_pcm(format)
- uint32_t channelCount = popcount(channelMask);
- size /= channelCount * audio_bytes_per_sample(format);
-
- *frameCount = size;
return NO_ERROR;
}
@@ -110,10 +102,8 @@ AudioRecord::~AudioRecord()
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
- if (mAudioRecord != 0) {
- mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
- mAudioRecord.clear();
- }
+ mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
+ mAudioRecord.clear();
IPCThreadState::self()->flushCommands();
AudioSystem::releaseAudioSessionId(mSessionId, -1);
}
@@ -133,6 +123,11 @@ status_t AudioRecord::set(
transfer_type transferType,
audio_input_flags_t flags)
{
+ ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %d, "
+ "notificationFrames %d, sessionId %d, transferType %d, flags %#x",
+ inputSource, sampleRate, format, channelMask, frameCountInt, notificationFrames,
+ sessionId, transferType, flags);
+
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
@@ -163,16 +158,15 @@ status_t AudioRecord::set(
}
size_t frameCount = frameCountInt;
- ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
- frameCount);
-
AutoMutex lock(mLock);
+ // invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
+ // handle default values first.
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
@@ -209,15 +203,19 @@ status_t AudioRecord::set(
uint32_t channelCount = popcount(channelMask);
mChannelCount = channelCount;
- // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t)
- mFrameSize = channelCount * audio_bytes_per_sample(format);
+ if (audio_is_linear_pcm(format)) {
+ mFrameSize = channelCount * audio_bytes_per_sample(format);
+ } else {
+ mFrameSize = sizeof(uint8_t);
+ }
// validate framecount
- size_t minFrameCount = 0;
+ size_t minFrameCount;
status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
sampleRate, format, channelMask);
if (status != NO_ERROR) {
- ALOGE("getMinFrameCount() failed; status %d", status);
+ ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d",
+ sampleRate, format, channelMask, status);
return status;
}
ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
@@ -258,7 +256,6 @@ status_t AudioRecord::set(
mActive = false;
mCbf = cbf;
- mRefreshRemaining = true;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
@@ -485,10 +482,12 @@ status_t AudioRecord::openRecord_l(size_t epoch)
ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
"session ID changed from %d to %d", originalSessionId, mSessionId);
- if (record == 0 || status != NO_ERROR) {
+ if (status != NO_ERROR) {
ALOGE("AudioFlinger could not create record track, status: %d", status);
goto release;
}
+ ALOG_ASSERT(record != 0);
+
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
@@ -502,27 +501,21 @@ status_t AudioRecord::openRecord_l(size_t epoch)
ALOGE("Could not get control block pointer");
return NO_INIT;
}
+ // invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
-
- // We retain a copy of the I/O handle, but don't own the reference
- mInput = input;
mAudioRecord = record;
+
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
- // note that temp is the (possibly revised) value of mFrameCount
+ // note that temp is the (possibly revised) value of frameCount
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
}
frameCount = temp;
- // If IAudioRecord is re-created, don't let the requested frameCount
- // decrease. This can confuse clients that cache frameCount().
- if (frameCount > mReqFrameCount) {
- mReqFrameCount = frameCount;
- }
// FIXME missing fast track frameCount logic
mAwaitBoost = false;
@@ -544,10 +537,21 @@ status_t AudioRecord::openRecord_l(size_t epoch)
}
}
- // starting address of buffers in shared memory
+ // We retain a copy of the I/O handle, but don't own the reference
+ mInput = input;
+ mRefreshRemaining = true;
+
+ // Starting address of buffers in shared memory, immediately after the control block. This
+ // address is for the mapping within client address space. AudioFlinger::TrackBase::mBuffer
+ // is for the server address space.
void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
mFrameCount = frameCount;
+ // If IAudioRecord is re-created, don't let the requested frameCount
+ // decrease. This can confuse clients that cache frameCount().
+ if (frameCount > mReqFrameCount) {
+ mReqFrameCount = frameCount;
+ }
// update proxy
mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 5c62260..adf3847 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -195,6 +195,11 @@ status_t AudioTrack::set(
int uid,
pid_t pid)
{
+ ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %d, "
+ "flags #%x, notificationFrames %d, sessionId %d, transferType %d",
+ streamType, sampleRate, format, channelMask, frameCountInt, flags, notificationFrames,
+ sessionId, transferType);
+
switch (transferType) {
case TRANSFER_DEFAULT:
if (sharedBuffer != 0) {
@@ -288,6 +293,9 @@ status_t AudioTrack::set(
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
+ mChannelMask = channelMask;
+ uint32_t channelCount = popcount(channelMask);
+ mChannelCount = channelCount;
// AudioFlinger does not currently support 8-bit data in shared memory
if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
@@ -311,10 +319,6 @@ status_t AudioTrack::set(
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
- mChannelMask = channelMask;
- uint32_t channelCount = popcount(channelMask);
- mChannelCount = channelCount;
-
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
mFrameSizeAF = channelCount * sizeof(int16_t);
@@ -1012,10 +1016,12 @@ status_t AudioTrack::createTrack_l(size_t epoch)
mClientUid,
&status);
- if (track == 0) {
+ if (status != NO_ERROR) {
ALOGE("AudioFlinger could not create track, status: %d", status);
goto release;
}
+ ALOG_ASSERT(track != 0);
+
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
@@ -1035,6 +1041,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
mDeathNotifier.clear();
}
mAudioTrack = track;
+
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
@@ -1046,6 +1053,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
}
frameCount = temp;
+
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
@@ -1099,6 +1107,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME don't believe this lie
mLatency = afLatency + (1000*frameCount) / mSampleRate;
+
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
generated by cgit v1.1 at 2024-08-02 10:06:47 +0000