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-rw-r--r--include/media/AudioTrack.h58
-rw-r--r--include/media/stagefright/BufferProducerWrapper.h1
-rw-r--r--media/libmedia/AudioTrack.cpp11
-rw-r--r--media/libmediaplayerservice/MediaPlayerService.cpp3
-rw-r--r--media/libmediaplayerservice/tests/Android.mk11
-rw-r--r--media/libmediaplayerservice/tests/DrmSessionManager_test.cpp2
-rw-r--r--media/libstagefright/codecs/aacdec/SoftAAC2.cpp2
-rw-r--r--media/libstagefright/mpeg2ts/AnotherPacketSource.cpp12
-rw-r--r--services/audioflinger/Threads.cpp22
9 files changed, 75 insertions, 47 deletions
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 2e1ed6c..3de0774 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -63,7 +63,7 @@ public:
// See AudioTimestamp for the information included with event.
};
- /* Client should declare Buffer on the stack and pass address to obtainBuffer()
+ /* Client should declare a Buffer and pass the address to obtainBuffer()
* and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
*/
@@ -72,14 +72,20 @@ public:
public:
// FIXME use m prefix
size_t frameCount; // number of sample frames corresponding to size;
- // on input it is the number of frames desired,
- // on output is the number of frames actually filled
- // (currently ignored, but will make the primary field in future)
+ // on input to obtainBuffer() it is the number of frames desired,
+ // on output from obtainBuffer() it is the number of available
+ // [empty slots for] frames to be filled
+ // on input to releaseBuffer() it is currently ignored
size_t size; // input/output in bytes == frameCount * frameSize
- // on input it is unused
- // on output is the number of bytes actually filled
- // FIXME this is redundant with respect to frameCount.
+ // on input to obtainBuffer() it is ignored
+ // on output from obtainBuffer() it is the number of available
+ // [empty slots for] bytes to be filled,
+ // which is frameCount * frameSize
+ // on input to releaseBuffer() it is the number of bytes to
+ // release
+ // FIXME This is redundant with respect to frameCount. Consider
+ // removing size and making frameCount the primary field.
union {
void* raw;
@@ -484,10 +490,18 @@ public:
*/
status_t attachAuxEffect(int effectId);
- /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
+ /* Public API for TRANSFER_OBTAIN mode.
+ * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
* After filling these slots with data, the caller should release them with releaseBuffer().
* If the track buffer is not full, obtainBuffer() returns as many contiguous
* [empty slots for] frames as are available immediately.
+ *
+ * If nonContig is non-NULL, it is an output parameter that will be set to the number of
+ * additional non-contiguous frames that are predicted to be available immediately,
+ * if the client were to release the first frames and then call obtainBuffer() again.
+ * This value is only a prediction, and needs to be confirmed.
+ * It will be set to zero for an error return.
+ *
* If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK
* regardless of the value of waitCount.
* If the track buffer is full and track is not stopped, obtainBuffer() blocks with a
@@ -496,7 +510,6 @@ public:
* is exhausted, at which point obtainBuffer() will either block
* or return WOULD_BLOCK depending on the value of the "waitCount"
* parameter.
- * Each sample is 16-bit signed PCM.
*
* obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
* which should use write() or callback EVENT_MORE_DATA instead.
@@ -508,24 +521,29 @@ public:
*
* Buffer fields
* On entry:
- * frameCount number of frames requested
+ * frameCount number of [empty slots for] frames requested
+ * size ignored
+ * raw ignored
* After error return:
* frameCount 0
* size 0
* raw undefined
* After successful return:
- * frameCount actual number of frames available, <= number requested
+ * frameCount actual number of [empty slots for] frames available, <= number requested
* size actual number of bytes available
* raw pointer to the buffer
*/
-
/* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
- status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
+ status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount,
+ size_t *nonContig = NULL)
__attribute__((__deprecated__));
private:
/* If nonContig is non-NULL, it is an output parameter that will be set to the number of
- * additional non-contiguous frames that are available immediately.
+ * additional non-contiguous frames that are predicted to be available immediately,
+ * if the client were to release the first frames and then call obtainBuffer() again.
+ * This value is only a prediction, and needs to be confirmed.
+ * It will be set to zero for an error return.
* FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(),
* in case the requested amount of frames is in two or more non-contiguous regions.
* FIXME requested and elapsed are both relative times. Consider changing to absolute time.
@@ -534,9 +552,17 @@ private:
struct timespec *elapsed = NULL, size_t *nonContig = NULL);
public:
- /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
+ /* Public API for TRANSFER_OBTAIN mode.
+ * Release a filled buffer of frames for AudioFlinger to process.
+ *
+ * Buffer fields:
+ * frameCount currently ignored but recommend to set to actual number of frames filled
+ * size actual number of bytes filled, must be multiple of frameSize
+ * raw ignored
+ *
+ */
// FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
- void releaseBuffer(Buffer* audioBuffer);
+ void releaseBuffer(const Buffer* audioBuffer);
/* As a convenience we provide a write() interface to the audio buffer.
* Input parameter 'size' is in byte units.
diff --git a/include/media/stagefright/BufferProducerWrapper.h b/include/media/stagefright/BufferProducerWrapper.h
index d8acf30..4caa2c6 100644
--- a/include/media/stagefright/BufferProducerWrapper.h
+++ b/include/media/stagefright/BufferProducerWrapper.h
@@ -19,6 +19,7 @@
#define BUFFER_PRODUCER_WRAPPER_H_
#include <gui/IGraphicBufferProducer.h>
+#include <media/stagefright/foundation/ABase.h>
namespace android {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 1d5fc95..0ad9cc0 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1002,7 +1002,9 @@ status_t AudioTrack::createTrack_l()
// use case 1: shared buffer
(mSharedBuffer != 0) ||
// use case 2: callback transfer mode
- (mTransfer == TRANSFER_CALLBACK)) &&
+ (mTransfer == TRANSFER_CALLBACK) ||
+ // use case 3: obtain/release mode
+ (mTransfer == TRANSFER_OBTAIN)) &&
// matching sample rate
(mSampleRate == afSampleRate))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
@@ -1236,7 +1238,7 @@ release:
return status;
}
-status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
+status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
{
if (audioBuffer == NULL) {
return BAD_VALUE;
@@ -1263,7 +1265,7 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
ALOGE("%s invalid waitCount %d", __func__, waitCount);
requested = NULL;
}
- return obtainBuffer(audioBuffer, requested);
+ return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
@@ -1338,8 +1340,9 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *re
return status;
}
-void AudioTrack::releaseBuffer(Buffer* audioBuffer)
+void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
{
+ // FIXME add error checking on mode, by adding an internal version
if (mTransfer == TRANSFER_SHARED) {
return;
}
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 5e5d099..3a399af 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -290,8 +290,9 @@ MediaPlayerService::MediaPlayerService()
const sp<IServiceManager> sm(defaultServiceManager());
if (sm != NULL) {
const String16 name("batterystats");
+ // use checkService() to avoid blocking if service is not up yet
sp<IBatteryStats> batteryStats =
- interface_cast<IBatteryStats>(sm->getService(name));
+ interface_cast<IBatteryStats>(sm->checkService(name));
if (batteryStats != NULL) {
batteryStats->noteResetVideo();
batteryStats->noteResetAudio();
diff --git a/media/libmediaplayerservice/tests/Android.mk b/media/libmediaplayerservice/tests/Android.mk
index 69d4ad1..7bc78ff 100644
--- a/media/libmediaplayerservice/tests/Android.mk
+++ b/media/libmediaplayerservice/tests/Android.mk
@@ -1,7 +1,6 @@
# Build the unit tests.
LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
-LOCAL_ADDITIONAL_DEPENDENCIES := $(LOCAL_PATH)/Android.mk
LOCAL_MODULE := DrmSessionManager_test
@@ -19,13 +18,7 @@ LOCAL_C_INCLUDES := \
frameworks/av/include \
frameworks/av/media/libmediaplayerservice \
-include $(BUILD_NATIVE_TEST)
+LOCAL_32_BIT_ONLY := true
-# Include subdirectory makefiles
-# ============================================================
+include $(BUILD_NATIVE_TEST)
-# If we're building with ONE_SHOT_MAKEFILE (mm, mmm), then what the framework
-# team really wants is to build the stuff defined by this makefile.
-ifeq (,$(ONE_SHOT_MAKEFILE))
-include $(call first-makefiles-under,$(LOCAL_PATH))
-endif
diff --git a/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp b/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp
index 782c1a5..27b482b 100644
--- a/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp
+++ b/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp
@@ -227,7 +227,7 @@ TEST_F(DrmSessionManagerTest, reclaimSession) {
// add a session from a higher priority process.
sp<FakeDrm> drm = new FakeDrm;
- const uint8_t ids[] = {456, 7890, 123};
+ const uint8_t ids[] = {1, 3, 5};
Vector<uint8_t> sessionId;
GetSessionId(ids, ARRAY_SIZE(ids), &sessionId);
mDrmSessionManager->addSession(15, drm, sessionId);
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 1505f08..10937ec 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -975,6 +975,7 @@ void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
mBufferSizes.clear();
mDecodedSizes.clear();
mLastInHeader = NULL;
+ mEndOfInput = false;
} else {
int avail;
while ((avail = outputDelayRingBufferSamplesAvailable()) > 0) {
@@ -989,6 +990,7 @@ void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
mOutputBufferCount++;
}
mOutputDelayRingBufferReadPos = mOutputDelayRingBufferWritePos;
+ mEndOfOutput = false;
}
}
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index f266fe7..bb05417 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -91,13 +91,11 @@ sp<MetaData> AnotherPacketSource::getFormat() {
while (it != mBuffers.end()) {
sp<ABuffer> buffer = *it;
int32_t discontinuity;
- if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
- break;
- }
-
- sp<RefBase> object;
- if (buffer->meta()->findObject("format", &object)) {
- return mFormat = static_cast<MetaData*>(object.get());
+ if (!buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+ sp<RefBase> object;
+ if (buffer->meta()->findObject("format", &object)) {
+ return mFormat = static_cast<MetaData*>(object.get());
+ }
}
++it;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 3474f24..c1da6bc 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -741,6 +741,7 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u
dprintf(fd, "thread %p may be deadlocked\n", this);
}
+ dprintf(fd, " Thread name: %s\n", mThreadName);
dprintf(fd, " I/O handle: %d\n", mId);
dprintf(fd, " TID: %d\n", getTid());
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
@@ -764,6 +765,9 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u
} else {
dprintf(fd, " none\n");
}
+ dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string());
+ dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string());
+ dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource));
if (locked) {
mLock.unlock();
@@ -1479,6 +1483,9 @@ void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& ar
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type()));
+
+ dumpBase(fd, args);
+
dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
dprintf(fd, " Total writes: %d\n", mNumWrites);
@@ -1493,8 +1500,6 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>&
audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
String8 flagsAsString = outputFlagsToString(flags);
dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string());
-
- dumpBase(fd, args);
}
// Thread virtuals
@@ -1545,9 +1550,10 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
(
(sharedBuffer != 0)
) ||
- // use case 2: callback handler and frame count is default or at least as large as HAL
+ // use case 2: frame count is default or at least as large as HAL
(
- (tid != -1) &&
+ // we formerly checked for a callback handler (non-0 tid),
+ // but that is no longer required for TRANSFER_OBTAIN mode
((frameCount == 0) ||
(frameCount >= mFrameCount))
)
@@ -6151,15 +6157,13 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a
{
dprintf(fd, "\nInput thread %p:\n", this);
- if (mActiveTracks.size() > 0) {
- dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
- } else {
+ dumpBase(fd, args);
+
+ if (mActiveTracks.size() == 0) {
dprintf(fd, " No active record clients\n");
}
dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
-
- dumpBase(fd, args);
}
void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
generated by cgit v1.1 at 2024-06-11 09:27:52 +0000