Merge "Use uint32_t sequence numbers for IAudioRecord"
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index f4ced52..45134c4 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -39,8 +39,12 @@
* Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
*/
enum event_type {
- EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer.
- EVENT_OVERRUN = 1, // PCM buffer overrun occurred.
+ EVENT_MORE_DATA = 0, // Request to read available data from buffer.
+ // If this event is delivered but the callback handler
+ // does not want to read the available data, the handler must
+ // explicitly
+ // ignore the event by setting frameCount to zero.
+ EVENT_OVERRUN = 1, // Buffer overrun occurred.
EVENT_MARKER = 2, // Record head is at the specified marker position
// (See setMarkerPosition()).
EVENT_NEW_POS = 3, // Record head is at a new position
@@ -63,6 +67,7 @@
// (currently ignored but will make the primary field in future)
size_t size; // input/output in bytes == frameCount * frameSize
+ // on output is the number of bytes actually drained
// FIXME this is redundant with respect to frameCount,
// and TRANSFER_OBTAIN mode is broken for 8-bit data
// since we don't define the frame format
@@ -76,7 +81,7 @@
/* As a convenience, if a callback is supplied, a handler thread
* is automatically created with the appropriate priority. This thread
- * invokes the callback when a new buffer becomes ready or various conditions occur.
+ * invokes the callback when a new buffer becomes available or various conditions occur.
* Parameters:
*
* event: type of event notified (see enum AudioRecord::event_type).
@@ -99,6 +104,8 @@
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
* - BAD_VALUE: unsupported configuration
+ * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+ * and is undefined otherwise.
*/
static status_t getMinFrameCount(size_t* frameCount,
@@ -109,7 +116,7 @@
/* How data is transferred from AudioRecord
*/
enum transfer_type {
- TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters
+ TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters
TRANSFER_CALLBACK, // callback EVENT_MORE_DATA
TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer()
TRANSFER_SYNC, // synchronous read()
@@ -137,7 +144,7 @@
* be larger if the requested size is not compatible with current audio HAL
* latency. Zero means to use a default value.
* cbf: Callback function. If not null, this function is called periodically
- * to consume new PCM data and inform of marker, position updates, etc.
+ * to consume new data and inform of marker, position updates, etc.
* user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames are ready in record track output buffer.
@@ -171,9 +178,10 @@
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful intialization
* - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
- * - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
+ * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
+ * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
*
* Parameters not listed in the AudioRecord constructors above:
*
@@ -192,7 +200,7 @@
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
- /* Result of constructing the AudioRecord. This must be checked
+ /* Result of constructing the AudioRecord. This must be checked for successful initialization
* before using any AudioRecord API (except for set()), because using
* an uninitialized AudioRecord produces undefined results.
* See set() method above for possible return codes.
@@ -221,7 +229,7 @@
status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
- /* Stop a track. If set, the callback will cease being called. Note that obtainBuffer() still
+ /* Stop a track. The callback will cease being called. Note that obtainBuffer() still
* works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
*/
void stop();
@@ -236,7 +244,7 @@
* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
* with marker == 0 cancels marker notification callback.
* To set a marker at a position which would compute as 0,
- * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+ * a workaround is to set the marker at a nearby position such as ~0 or 1.
* If the AudioRecord has been opened with no callback function associated,
* the operation will fail.
*
@@ -378,8 +386,10 @@
* returning the current value by this function call. Such loss typically occurs when the
* user space process is blocked longer than the capacity of audio driver buffers.
* Units: the number of input audio frames.
+ * FIXME The side-effect of resetting the counter may be incompatible with multi-client.
+ * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects.
*/
- unsigned int getInputFramesLost() const;
+ uint32_t getInputFramesLost() const;
private:
/* copying audio record objects is not allowed */
@@ -426,6 +436,7 @@
nsecs_t processAudioBuffer();
// caller must hold lock on mLock for all _l methods
+
status_t openRecord_l(size_t epoch);
// FIXME enum is faster than strcmp() for parameter 'from'
@@ -477,7 +488,7 @@
audio_io_handle_t mInput; // returned by AudioSystem::getInput()
- // may be changed if IAudioRecord object is re-created
+ // Next 3 fields may be changed if IAudioRecord is re-created, but always != 0
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index ca9aaf7..5abab8a 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -138,7 +138,7 @@
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
// return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
- static size_t getInputFramesLost(audio_io_handle_t ioHandle);
+ static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
static int newAudioSessionId();
static void acquireAudioSessionId(int audioSession);
@@ -200,12 +200,16 @@
static status_t setPhoneState(audio_mode_t state);
static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+
+ // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
+ // or release it with releaseOutput().
static audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
const audio_offload_info_t *offloadInfo = NULL);
+
static status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session);
@@ -213,11 +217,15 @@
audio_stream_type_t stream,
int session);
static void releaseOutput(audio_io_handle_t output);
+
+ // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
+ // or release it with releaseInput().
static audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int sessionId);
+
static status_t startInput(audio_io_handle_t input);
static status_t stopInput(audio_io_handle_t input);
static void releaseInput(audio_io_handle_t input);
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index c86a67b..644e55c 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -160,7 +160,7 @@
* sampleRate: Data source sampling rate in Hz.
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
- * channelMask: Channel mask.
+ * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true.
* frameCount: Minimum size of track PCM buffer in frames. This defines the
* application's contribution to the
* latency of the track. The actual size selected by the AudioTrack could be
@@ -338,7 +338,7 @@
*/
status_t setSampleRate(uint32_t sampleRate);
- /* Return current source sample rate in Hz, or 0 if unknown */
+ /* Return current source sample rate in Hz */
uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
@@ -363,7 +363,7 @@
/* Sets marker position. When playback reaches the number of frames specified, a callback with
* event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
* notification callback. To set a marker at a position which would compute as 0,
- * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+ * a workaround is to set the marker at a nearby position such as ~0 or 1.
* If the AudioTrack has been opened with no callback function associated, the operation will
* fail.
*
@@ -661,6 +661,7 @@
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
sp<AudioTrackThread> mAudioTrackThread;
+
float mVolume[2];
float mSendLevel;
mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it.
@@ -674,7 +675,10 @@
audio_stream_type_t mStreamType;
uint32_t mChannelCount;
audio_channel_mask_t mChannelMask;
+ sp<IMemory> mSharedBuffer;
transfer_type mTransfer;
+ audio_offload_info_t mOffloadInfoCopy;
+ const audio_offload_info_t* mOffloadInfo;
// mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's
// twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
@@ -715,8 +719,8 @@
bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer()
uint32_t mObservedSequence; // last observed value of mSequence
- sp<IMemory> mSharedBuffer;
uint32_t mLoopPeriod; // in frames, zero means looping is disabled
+
uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
uint32_t mNewPosition; // in frames
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 899d79f..a794e87 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -67,6 +67,9 @@
size_t frameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
+ // On successful return, AudioFlinger takes over the handle
+ // reference and will release it when the track is destroyed.
+ // However on failure, the client is responsible for release.
audio_io_handle_t output,
pid_t tid, // -1 means unused, otherwise must be valid non-0
int *sessionId,
@@ -78,6 +81,9 @@
status_t *status) = 0;
virtual sp<IAudioRecord> openRecord(
+ // On successful return, AudioFlinger takes over the handle
+ // reference and will release it when the track is destroyed.
+ // However on failure, the client is responsible for release.
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
@@ -170,7 +176,7 @@
virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
audio_io_handle_t output) const = 0;
- virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
+ virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
virtual int newAudioSessionId() = 0;
@@ -188,6 +194,7 @@
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
+ // AudioFlinger doesn't take over handle reference from client
audio_io_handle_t output,
int sessionId,
status_t *status,
diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index 3a87474..db8216b 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -215,6 +215,8 @@
bool findData(uint32_t key, uint32_t *type,
const void **data, size_t *size) const;
+ bool hasData(uint32_t key) const;
+
void dumpToLog() const;
protected:
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 076404b..6565a16 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -289,6 +289,9 @@
// reset current position as seen by client to 0
mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
+ // force refresh of remaining frames by processAudioBuffer() as last
+ // read before stop could be partial.
+ mRefreshRemaining = true;
mNewPosition = mProxy->getPosition() + mUpdatePeriod;
int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
@@ -352,6 +355,7 @@
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
+ // The only purpose of setting marker position is to get a callback
if (mCbf == NULL) {
return INVALID_OPERATION;
}
@@ -377,6 +381,7 @@
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
+ // The only purpose of setting position update period is to get a callback
if (mCbf == NULL) {
return INVALID_OPERATION;
}
@@ -412,7 +417,7 @@
return NO_ERROR;
}
-unsigned int AudioRecord::getInputFramesLost() const
+uint32_t AudioRecord::getInputFramesLost() const
{
// no need to check mActive, because if inactive this will return 0, which is what we want
return AudioSystem::getInputFramesLost(getInput());
@@ -591,6 +596,9 @@
if (newSequence == oldSequence) {
status = restoreRecord_l("obtainBuffer");
if (status != NO_ERROR) {
+ buffer.mFrameCount = 0;
+ buffer.mRaw = NULL;
+ buffer.mNonContig = 0;
break;
}
}
@@ -770,7 +778,7 @@
uint32_t sequence = mSequence;
// These fields don't need to be cached, because they are assigned only by set():
- // mTransfer, mCbf, mUserData, mSampleRate
+ // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
mLock.unlock();
@@ -844,8 +852,8 @@
"obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
requested = &ClientProxy::kNonBlocking;
size_t avail = audioBuffer.frameCount + nonContig;
- ALOGV("obtainBuffer(%u) returned %u = %u + %u",
- mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
+ ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+ mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
break;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 69d9273..dcb72f8 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -413,9 +413,9 @@
return af->getRenderPosition(halFrames, dspFrames, output);
}
-size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+uint32_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- unsigned int result = 0;
+ uint32_t result = 0;
if (af == 0) return result;
if (ioHandle == 0) return result;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index d5e3e67..72be5ca 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -54,16 +54,22 @@
status_t status;
status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
if (status != NO_ERROR) {
+ ALOGE("Unable to query output sample rate for stream type %d; status %d",
+ streamType, status);
return status;
}
size_t afFrameCount;
status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
if (status != NO_ERROR) {
+ ALOGE("Unable to query output frame count for stream type %d; status %d",
+ streamType, status);
return status;
}
uint32_t afLatency;
status = AudioSystem::getOutputLatency(&afLatency, streamType);
if (status != NO_ERROR) {
+ ALOGE("Unable to query output latency for stream type %d; status %d",
+ streamType, status);
return status;
}
@@ -216,6 +222,7 @@
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
+ mSharedBuffer = sharedBuffer;
mTransfer = transferType;
// FIXME "int" here is legacy and will be replaced by size_t later
@@ -244,6 +251,11 @@
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
+ if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+ ALOGE("Invalid stream type %d", streamType);
+ return BAD_VALUE;
+ }
+ mStreamType = streamType;
status_t status;
if (sampleRate == 0) {
@@ -266,6 +278,7 @@
ALOGE("Invalid format %d", format);
return BAD_VALUE;
}
+ mFormat = format;
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
@@ -317,6 +330,16 @@
return BAD_VALUE;
}
+ // Make copy of input parameter offloadInfo so that in the future:
+ // (a) createTrack_l doesn't need it as an input parameter
+ // (b) we can support re-creation of offloaded tracks
+ if (offloadInfo != NULL) {
+ mOffloadInfoCopy = *offloadInfo;
+ mOffloadInfo = &mOffloadInfoCopy;
+ } else {
+ mOffloadInfo = NULL;
+ }
+
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0.0f;
@@ -355,7 +378,7 @@
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
- //Use of direct and offloaded output streams is ref counted by audio policy manager.
+ // Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
AudioSystem::releaseOutput(output);
@@ -363,9 +386,6 @@
}
mStatus = NO_ERROR;
- mStreamType = streamType;
- mFormat = format;
- mSharedBuffer = sharedBuffer;
mState = STATE_STOPPED;
mUserData = user;
mLoopPeriod = 0;
@@ -698,6 +718,7 @@
AutoMutex lock(mLock);
mNewPosition = mProxy->getPosition() + updatePeriod;
mUpdatePeriod = updatePeriod;
+
return NO_ERROR;
}
@@ -1004,6 +1025,11 @@
ALOGE("Could not get control block");
return NO_INIT;
}
+ void *iMemPointer = iMem->pointer();
+ if (iMemPointer == NULL) {
+ ALOGE("Could not get control block pointer");
+ return NO_INIT;
+ }
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
@@ -1011,7 +1037,7 @@
}
mAudioTrack = track;
mCblkMemory = iMem;
- audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
+ audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
size_t temp = cblk->frameCount_;
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
@@ -1610,7 +1636,6 @@
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
size_t writtenSize = audioBuffer.size;
- size_t writtenFrames = writtenSize / mFrameSize;
// Sanity check on returned size
if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
@@ -1735,7 +1760,7 @@
}
}
if (result != NO_ERROR) {
- //Use of direct and offloaded output streams is ref counted by audio policy manager.
+ // Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
AudioSystem::releaseOutput(output);
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index c9c8d58..86a4d74 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -138,6 +138,17 @@
name = reply.readString8();
lStatus = reply.readInt32();
track = interface_cast<IAudioTrack>(reply.readStrongBinder());
+ if (lStatus == NO_ERROR) {
+ if (track == 0) {
+ ALOGE("createTrack should have returned an IAudioTrack");
+ lStatus = UNKNOWN_ERROR;
+ }
+ } else {
+ if (track != 0) {
+ ALOGE("createTrack returned an IAudioTrack but with status %d", lStatus);
+ track.clear();
+ }
+ }
}
if (status != NULL) {
*status = lStatus;
@@ -564,13 +575,16 @@
return status;
}
- virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const
+ virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32((int32_t) ioHandle);
- remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
- return reply.readInt32();
+ status_t status = remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
+ if (status != NO_ERROR) {
+ return 0;
+ }
+ return (uint32_t) reply.readInt32();
}
virtual int newAudioSessionId()
@@ -795,6 +809,7 @@
(audio_stream_type_t) streamType, sampleRate, format,
channelMask, frameCount, &flags, buffer, output, tid,
&sessionId, name, clientUid, &status);
+ LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
}
reply->writeInt32(flags);
reply->writeInt32(sessionId);
@@ -1044,7 +1059,7 @@
case GET_INPUT_FRAMES_LOST: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
- reply->writeInt32(getInputFramesLost(ioHandle));
+ reply->writeInt32((int32_t) getInputFramesLost(ioHandle));
return NO_ERROR;
} break;
case NEW_AUDIO_SESSION_ID: {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 92b9a92..bf5271e 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -422,7 +422,9 @@
ALOGV("video late by %lld us (%.2f secs)",
mVideoLateByUs, mVideoLateByUs / 1E6);
} else {
- ALOGV("rendering video at media time %.2f secs", mediaTimeUs / 1E6);
+ ALOGV("rendering video at media time %.2f secs",
+ (mFlags & FLAG_REAL_TIME ? realTimeUs :
+ (realTimeUs + mAnchorTimeMediaUs - mAnchorTimeRealUs)) / 1E6);
if (mSoftRenderer != NULL) {
mSoftRenderer->render(entry->mBuffer->data(), entry->mBuffer->size(), NULL);
}
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index d7223d9..cadadc8 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -278,7 +278,7 @@
// Drop retrieved and previously lost audio data.
if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
- mRecord->getInputFramesLost();
+ (void) mRecord->getInputFramesLost();
ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs);
return OK;
}
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 9c89e82..dfb5c04 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -39,6 +39,7 @@
#include <utils/String8.h>
#include <byteswap.h>
+#include "include/ID3.h"
namespace android {
@@ -1787,6 +1788,18 @@
break;
}
+ case FOURCC('I', 'D', '3', '2'):
+ {
+ if (chunk_data_size < 6) {
+ return ERROR_MALFORMED;
+ }
+
+ parseID3v2MetaData(data_offset + 6);
+
+ *offset += chunk_size;
+ break;
+ }
+
case FOURCC('-', '-', '-', '-'):
{
mLastCommentMean.clear();
@@ -2167,7 +2180,7 @@
break;
}
- if (size >= 8 && metadataKey) {
+ if (size >= 8 && metadataKey && !mFileMetaData->hasData(metadataKey)) {
if (metadataKey == kKeyAlbumArt) {
mFileMetaData->setData(
kKeyAlbumArt, MetaData::TYPE_NONE,
@@ -2316,6 +2329,62 @@
return OK;
}
+void MPEG4Extractor::parseID3v2MetaData(off64_t offset) {
+ ID3 id3(mDataSource, true /* ignorev1 */, offset);
+
+ if (id3.isValid()) {
+ struct Map {
+ int key;
+ const char *tag1;
+ const char *tag2;
+ };
+ static const Map kMap[] = {
+ { kKeyAlbum, "TALB", "TAL" },
+ { kKeyArtist, "TPE1", "TP1" },
+ { kKeyAlbumArtist, "TPE2", "TP2" },
+ { kKeyComposer, "TCOM", "TCM" },
+ { kKeyGenre, "TCON", "TCO" },
+ { kKeyTitle, "TIT2", "TT2" },
+ { kKeyYear, "TYE", "TYER" },
+ { kKeyAuthor, "TXT", "TEXT" },
+ { kKeyCDTrackNumber, "TRK", "TRCK" },
+ { kKeyDiscNumber, "TPA", "TPOS" },
+ { kKeyCompilation, "TCP", "TCMP" },
+ };
+ static const size_t kNumMapEntries = sizeof(kMap) / sizeof(kMap[0]);
+
+ for (size_t i = 0; i < kNumMapEntries; ++i) {
+ if (!mFileMetaData->hasData(kMap[i].key)) {
+ ID3::Iterator *it = new ID3::Iterator(id3, kMap[i].tag1);
+ if (it->done()) {
+ delete it;
+ it = new ID3::Iterator(id3, kMap[i].tag2);
+ }
+
+ if (it->done()) {
+ delete it;
+ continue;
+ }
+
+ String8 s;
+ it->getString(&s);
+ delete it;
+
+ mFileMetaData->setCString(kMap[i].key, s);
+ }
+ }
+
+ size_t dataSize;
+ String8 mime;
+ const void *data = id3.getAlbumArt(&dataSize, &mime);
+
+ if (data) {
+ mFileMetaData->setData(kKeyAlbumArt, MetaData::TYPE_NONE, data, dataSize);
+ mFileMetaData->setCString(kKeyAlbumArtMIME, mime.string());
+ }
+ }
+}
+
sp<MediaSource> MPEG4Extractor::getTrack(size_t index) {
status_t err;
if ((err = readMetaData()) != OK) {
diff --git a/media/libstagefright/MetaData.cpp b/media/libstagefright/MetaData.cpp
index 7b60afc..1daead7 100644
--- a/media/libstagefright/MetaData.cpp
+++ b/media/libstagefright/MetaData.cpp
@@ -221,6 +221,16 @@
return true;
}
+bool MetaData::hasData(uint32_t key) const {
+ ssize_t i = mItems.indexOfKey(key);
+
+ if (i < 0) {
+ return false;
+ }
+
+ return true;
+}
+
MetaData::typed_data::typed_data()
: mType(0),
mSize(0) {
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index a486522..f0f203c 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -56,14 +56,14 @@
DISALLOW_EVIL_CONSTRUCTORS(MemorySource);
};
-ID3::ID3(const sp<DataSource> &source, bool ignoreV1)
+ID3::ID3(const sp<DataSource> &source, bool ignoreV1, off64_t offset)
: mIsValid(false),
mData(NULL),
mSize(0),
mFirstFrameOffset(0),
mVersion(ID3_UNKNOWN),
mRawSize(0) {
- mIsValid = parseV2(source);
+ mIsValid = parseV2(source, offset);
if (!mIsValid && !ignoreV1) {
mIsValid = parseV1(source);
@@ -79,7 +79,7 @@
mRawSize(0) {
sp<MemorySource> source = new MemorySource(data, size);
- mIsValid = parseV2(source);
+ mIsValid = parseV2(source, 0);
if (!mIsValid && !ignoreV1) {
mIsValid = parseV1(source);
@@ -115,7 +115,7 @@
return true;
}
-bool ID3::parseV2(const sp<DataSource> &source) {
+bool ID3::parseV2(const sp<DataSource> &source, off64_t offset) {
struct id3_header {
char id[3];
uint8_t version_major;
@@ -126,7 +126,7 @@
id3_header header;
if (source->readAt(
- 0, &header, sizeof(header)) != (ssize_t)sizeof(header)) {
+ offset, &header, sizeof(header)) != (ssize_t)sizeof(header)) {
return false;
}
@@ -185,7 +185,7 @@
mSize = size;
mRawSize = mSize + sizeof(header);
- if (source->readAt(sizeof(header), mData, mSize) != (ssize_t)mSize) {
+ if (source->readAt(offset + sizeof(header), mData, mSize) != (ssize_t)mSize) {
free(mData);
mData = NULL;
diff --git a/media/libstagefright/include/ID3.h b/media/libstagefright/include/ID3.h
index cca83ab..e83f3ef 100644
--- a/media/libstagefright/include/ID3.h
+++ b/media/libstagefright/include/ID3.h
@@ -35,7 +35,7 @@
ID3_V2_4,
};
- ID3(const sp<DataSource> &source, bool ignoreV1 = false);
+ ID3(const sp<DataSource> &source, bool ignoreV1 = false, off64_t offset = 0);
ID3(const uint8_t *data, size_t size, bool ignoreV1 = false);
~ID3();
@@ -86,7 +86,7 @@
size_t mRawSize;
bool parseV1(const sp<DataSource> &source);
- bool parseV2(const sp<DataSource> &source);
+ bool parseV2(const sp<DataSource> &source, off64_t offset);
void removeUnsynchronization();
bool removeUnsynchronizationV2_4(bool iTunesHack);
diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index bd5e4b9..7b4bc6d 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -97,6 +97,7 @@
status_t parseChunk(off64_t *offset, int depth);
status_t parseITunesMetaData(off64_t offset, size_t size);
status_t parse3GPPMetaData(off64_t offset, size_t size, int depth);
+ void parseID3v2MetaData(off64_t offset);
status_t updateAudioTrackInfoFromESDS_MPEG4Audio(
const void *esds_data, size_t esds_size);
diff --git a/media/mtp/MtpDataPacket.cpp b/media/mtp/MtpDataPacket.cpp
index 930f0b0..c4f87a0 100644
--- a/media/mtp/MtpDataPacket.cpp
+++ b/media/mtp/MtpDataPacket.cpp
@@ -331,7 +331,7 @@
void MtpDataPacket::putString(const uint16_t* string) {
int count = 0;
- for (int i = 0; i < 256; i++) {
+ for (int i = 0; i <= MTP_STRING_MAX_CHARACTER_NUMBER; i++) {
if (string[i])
count++;
else
diff --git a/media/mtp/MtpStringBuffer.cpp b/media/mtp/MtpStringBuffer.cpp
index fe8cf04..f3420a4 100644
--- a/media/mtp/MtpStringBuffer.cpp
+++ b/media/mtp/MtpStringBuffer.cpp
@@ -56,42 +56,47 @@
}
void MtpStringBuffer::set(const char* src) {
- int length = strlen(src);
- if (length >= sizeof(mBuffer))
- length = sizeof(mBuffer) - 1;
- memcpy(mBuffer, src, length);
-
// count the characters
int count = 0;
char ch;
- while ((ch = *src++) != 0) {
+ char* dest = (char*)mBuffer;
+
+ while ((ch = *src++) != 0 && count < MTP_STRING_MAX_CHARACTER_NUMBER) {
if ((ch & 0x80) == 0) {
// single byte character
+ *dest++ = ch;
} else if ((ch & 0xE0) == 0xC0) {
// two byte character
- if (! *src++) {
+ char ch1 = *src++;
+ if (! ch1) {
// last character was truncated, so ignore last byte
- length--;
break;
}
+
+ *dest++ = ch;
+ *dest++ = ch1;
} else if ((ch & 0xF0) == 0xE0) {
// 3 byte char
- if (! *src++) {
+ char ch1 = *src++;
+ if (! ch1) {
// last character was truncated, so ignore last byte
- length--;
break;
}
- if (! *src++) {
- // last character was truncated, so ignore last two bytes
- length -= 2;
+ char ch2 = *src++;
+ if (! ch2) {
+ // last character was truncated, so ignore last byte
break;
}
+
+ *dest++ = ch;
+ *dest++ = ch1;
+ *dest++ = ch2;
}
count++;
}
- mByteCount = length + 1;
- mBuffer[length] = 0;
+ *dest++ = 0;
+ mByteCount = dest - (char*)mBuffer;
mCharCount = count;
}
@@ -100,7 +105,7 @@
uint16_t ch;
uint8_t* dest = mBuffer;
- while ((ch = *src++) != 0 && count < 255) {
+ while ((ch = *src++) != 0 && count < MTP_STRING_MAX_CHARACTER_NUMBER) {
if (ch >= 0x0800) {
*dest++ = (uint8_t)(0xE0 | (ch >> 12));
*dest++ = (uint8_t)(0x80 | ((ch >> 6) & 0x3F));
diff --git a/media/mtp/MtpStringBuffer.h b/media/mtp/MtpStringBuffer.h
index cbc8307..e5150df 100644
--- a/media/mtp/MtpStringBuffer.h
+++ b/media/mtp/MtpStringBuffer.h
@@ -19,6 +19,9 @@
#include <stdint.h>
+// Max Character number of a MTP String
+#define MTP_STRING_MAX_CHARACTER_NUMBER 255
+
namespace android {
class MtpDataPacket;
@@ -29,7 +32,7 @@
private:
// mBuffer contains string in UTF8 format
// maximum 3 bytes/character, with 1 extra for zero termination
- uint8_t mBuffer[255 * 3 + 1];
+ uint8_t mBuffer[MTP_STRING_MAX_CHARACTER_NUMBER * 3 + 1];
int mCharCount;
int mByteCount;
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 34811a7..1257161 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -559,6 +559,7 @@
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
+ LOG_ALWAYS_FATAL_IF((track != 0) != (lStatus == NO_ERROR));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
// move effect chain to this output thread if an effect on same session was waiting
@@ -1058,7 +1059,7 @@
return size;
}
-unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
+uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 066d5d5..0ab43e0 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -189,7 +189,7 @@
virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
audio_io_handle_t output) const;
- virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
+ virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
virtual int newAudioSessionId();
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 01b90a8..d5a0e21 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1339,6 +1339,7 @@
// but TimedTrack::create() is a factory that could fail by returning NULL
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
+ ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
track.clear();
goto Exit;
}
@@ -5371,7 +5372,7 @@
mRsmpInIndex = mFrameCount;
}
-unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+uint32_t AudioFlinger::RecordThread::getInputFramesLost()
{
Mutex::Autolock _l(mLock);
if (initCheck() != NO_ERROR) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 6b81c38..2b749fa 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -907,7 +907,7 @@
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
void readInputParameters();
- virtual unsigned int getInputFramesLost();
+ virtual uint32_t getInputFramesLost();
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);