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-rw-r--r--media/libmedia/AudioRecord.cpp22
-rw-r--r--media/libmedia/AudioSystem.cpp32
-rw-r--r--media/libmedia/AudioTrack.cpp276
-rw-r--r--media/libmedia/AudioTrackShared.cpp12
-rw-r--r--media/libmedia/CharacterEncodingDetector.cpp10
-rw-r--r--media/libmedia/IAudioPolicyService.cpp57
-rw-r--r--media/libmedia/IMediaMetadataRetriever.cpp6
-rw-r--r--media/libmedia/IMediaRecorder.cpp8
-rw-r--r--media/libmedia/MediaProfiles.cpp6
-rw-r--r--media/libmedia/SoundPool.cpp10
-rw-r--r--media/libmedia/mediametadataretriever.cpp6
-rw-r--r--media/libmedia/mediaplayer.cpp14
-rw-r--r--media/libmedia/mediarecorder.cpp5
-rw-r--r--media/libmediaplayerservice/MediaPlayerService.cpp14
-rw-r--r--media/libmediaplayerservice/MetadataRetrieverClient.cpp4
-rw-r--r--media/libmediaplayerservice/MidiFile.cpp2
-rw-r--r--media/libnbaio/MonoPipe.cpp6
-rw-r--r--media/libnbaio/NBAIO.cpp2
18 files changed, 390 insertions, 102 deletions
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 50b444a..f865d38 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -18,7 +18,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioRecord"
+#include <inttypes.h>
#include <sys/resource.h>
+
#include <binder/IPCThreadState.h>
#include <media/AudioRecord.h>
#include <utils/Log.h>
@@ -468,7 +470,7 @@ status_t AudioRecord::openRecord_l(size_t epoch)
if (frameCount == 0) {
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
- ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
+ ALOGE("frameCount %zu < minFrameCount %zu", frameCount, minFrameCount);
return BAD_VALUE;
}
@@ -555,17 +557,17 @@ status_t AudioRecord::openRecord_l(size_t epoch)
mCblk = cblk;
// 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);
+ ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
}
frameCount = temp;
mAwaitBoost = false;
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
- ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", frameCount);
+ ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu", frameCount);
mAwaitBoost = true;
} else {
- ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
+ ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
// once denied, do not request again if IAudioRecord is re-created
mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
}
@@ -740,7 +742,7 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize)
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// sanity-check. user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
- ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize);
+ ALOGE("AudioRecord::read(buffer=%p, size=%zu (%zu)", buffer, userSize, userSize);
return BAD_VALUE;
}
@@ -921,10 +923,10 @@ nsecs_t AudioRecord::processAudioBuffer()
size_t nonContig;
status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
- "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+ "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
requested = &ClientProxy::kNonBlocking;
size_t avail = audioBuffer.frameCount + nonContig;
- ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+ ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
@@ -952,8 +954,8 @@ nsecs_t AudioRecord::processAudioBuffer()
// Sanity check on returned size
if (ssize_t(readSize) < 0 || readSize > reqSize) {
- ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
- reqSize, (int) readSize);
+ ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
+ reqSize, ssize_t(readSize));
return NS_NEVER;
}
@@ -1092,7 +1094,7 @@ bool AudioRecord::AudioRecordThread::threadLoop()
ns = 1000000000LL;
// fall through
default:
- LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+ LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
pauseInternal(ns);
return true;
}
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 15b32ff..a47d45c 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -245,6 +245,19 @@ status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream
return getSamplingRate(output, samplingRate);
}
+status_t AudioSystem::getOutputSamplingRateForAttr(uint32_t* samplingRate,
+ const audio_attributes_t *attr)
+{
+ if (attr == NULL) {
+ return BAD_VALUE;
+ }
+ audio_io_handle_t output = getOutputForAttr(attr);
+ if (output == 0) {
+ return PERMISSION_DENIED;
+ }
+ return getSamplingRate(output, samplingRate);
+}
+
status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
uint32_t* samplingRate)
{
@@ -310,7 +323,7 @@ status_t AudioSystem::getFrameCount(audio_io_handle_t output,
return BAD_VALUE;
}
- ALOGV("getFrameCount() output %d, frameCount %d", output, *frameCount);
+ ALOGV("getFrameCount() output %d, frameCount %zu", output, *frameCount);
return NO_ERROR;
}
@@ -476,7 +489,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %u "
+ ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %zu "
"latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
outputDesc->frameCount, outputDesc->latency);
@@ -501,7 +514,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
desc = (const OutputDescriptor *)param2;
ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
- "frameCount %d latency %d",
+ "frameCount %zu latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channelMask, desc->frameCount, desc->latency);
OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -633,6 +646,19 @@ audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
return aps->getOutput(stream, samplingRate, format, channelMask, flags, offloadInfo);
}
+audio_io_handle_t AudioSystem::getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ if (attr == NULL) return 0;
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return 0;
+ return aps->getOutputForAttr(attr, samplingRate, format, channelMask, flags, offloadInfo);
+}
+
status_t AudioSystem::startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session)
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index e6827ee..898d58d 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -15,12 +15,13 @@
** limitations under the License.
*/
-
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioTrack"
+#include <inttypes.h>
#include <math.h>
#include <sys/resource.h>
+
#include <audio_utils/primitives.h>
#include <binder/IPCThreadState.h>
#include <media/AudioTrack.h>
@@ -89,7 +90,7 @@ status_t AudioTrack::getMinFrameCount(
streamType, sampleRate);
return BAD_VALUE;
}
- ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
+ ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d",
*frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
return NO_ERROR;
}
@@ -103,6 +104,10 @@ AudioTrack::AudioTrack()
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0)
{
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
+ mAttributes.usage = AUDIO_USAGE_UNKNOWN;
+ mAttributes.flags = 0x0;
+ strcpy(mAttributes.tags, "");
}
AudioTrack::AudioTrack(
@@ -129,7 +134,7 @@ AudioTrack::AudioTrack(
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
- offloadInfo, uid, pid);
+ offloadInfo, uid, pid, NULL /*no audio attributes*/);
}
AudioTrack::AudioTrack(
@@ -156,7 +161,7 @@ AudioTrack::AudioTrack(
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
- uid, pid);
+ uid, pid, NULL /*no audio attributes*/);
}
AudioTrack::~AudioTrack()
@@ -199,7 +204,8 @@ status_t AudioTrack::set(
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
- pid_t pid)
+ pid_t pid,
+ audio_attributes_t* pAttributes)
{
ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"flags #%x, notificationFrames %u, sessionId %d, transferType %d",
@@ -245,7 +251,7 @@ status_t AudioTrack::set(
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
sharedBuffer->size());
- ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
+ ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags);
AutoMutex lock(mLock);
@@ -259,18 +265,33 @@ status_t AudioTrack::set(
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;
+
+ if (pAttributes == NULL) {
+ if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+ ALOGE("Invalid stream type %d", streamType);
+ return BAD_VALUE;
+ }
+ setAttributesFromStreamType(streamType);
+ mStreamType = streamType;
+ } else {
+ if (!isValidAttributes(pAttributes)) {
+ ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+ pAttributes->usage, pAttributes->content_type, pAttributes->flags,
+ pAttributes->tags);
+ }
+ // stream type shouldn't be looked at, this track has audio attributes
+ memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
+ setStreamTypeFromAttributes(mAttributes);
+ ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+ mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
}
- mStreamType = streamType;
status_t status;
if (sampleRate == 0) {
- status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType);
+ status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes);
if (status != NO_ERROR) {
ALOGE("Could not get output sample rate for stream type %d; status %d",
- streamType, status);
+ mStreamType, status);
return status;
}
}
@@ -314,7 +335,7 @@ status_t AudioTrack::set(
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// only allow deep buffering for music stream type
- if (streamType != AUDIO_STREAM_MUSIC) {
+ if (mStreamType != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
@@ -615,12 +636,12 @@ void AudioTrack::getAuxEffectSendLevel(float* level) const
status_t AudioTrack::setSampleRate(uint32_t rate)
{
- if (mIsTimed || isOffloaded()) {
+ if (mIsTimed || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
uint32_t afSamplingRate;
- if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
+ if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) {
return NO_INIT;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
@@ -646,7 +667,7 @@ uint32_t AudioTrack::getSampleRate() const
// sample rate can be updated during playback by the offloaded decoder so we need to
// query the HAL and update if needed.
// FIXME use Proxy return channel to update the rate from server and avoid polling here
- if (isOffloaded_l()) {
+ if (isOffloadedOrDirect_l()) {
if (mOutput != AUDIO_IO_HANDLE_NONE) {
uint32_t sampleRate = 0;
status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
@@ -660,7 +681,7 @@ uint32_t AudioTrack::getSampleRate() const
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
- if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+ if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
@@ -694,7 +715,7 @@ void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
// The only purpose of setting marker position is to get a callback
- if (mCbf == NULL || isOffloaded()) {
+ if (mCbf == NULL || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
@@ -707,7 +728,7 @@ status_t AudioTrack::setMarkerPosition(uint32_t marker)
status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
{
- if (isOffloaded()) {
+ if (isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (marker == NULL) {
@@ -723,7 +744,7 @@ status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
// The only purpose of setting position update period is to get a callback
- if (mCbf == NULL || isOffloaded()) {
+ if (mCbf == NULL || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
@@ -736,7 +757,7 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
- if (isOffloaded()) {
+ if (isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (updatePeriod == NULL) {
@@ -751,7 +772,7 @@ status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
status_t AudioTrack::setPosition(uint32_t position)
{
- if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+ if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (position > mFrameCount) {
@@ -784,10 +805,10 @@ status_t AudioTrack::getPosition(uint32_t *position) const
}
AutoMutex lock(mLock);
- if (isOffloaded_l()) {
+ if (isOffloadedOrDirect_l()) {
uint32_t dspFrames = 0;
- if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) {
+ if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition);
*position = mPausedPosition;
return NO_ERROR;
@@ -822,7 +843,7 @@ status_t AudioTrack::getBufferPosition(uint32_t *position)
status_t AudioTrack::reload()
{
- if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+ if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
@@ -867,12 +888,12 @@ status_t AudioTrack::createTrack_l(size_t epoch)
return NO_INIT;
}
- audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat,
+ audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat,
mChannelMask, mFlags, mOffloadInfo);
if (output == AUDIO_IO_HANDLE_NONE) {
- ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, "
- "channel mask %#x, flags %#x",
- mStreamType, mSampleRate, mFormat, mChannelMask, mFlags);
+ ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x,"
+ " channel mask %#x, flags %#x",
+ mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
return BAD_VALUE;
}
{
@@ -973,14 +994,14 @@ status_t AudioTrack::createTrack_l(size_t epoch)
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
- ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",
+ ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d",
afFrameCount, minBufCount, afSampleRate, afLatency);
if (minBufCount <= nBuffering) {
minBufCount = nBuffering;
}
size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
- ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
+ ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
", afLatency=%d",
minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
@@ -988,7 +1009,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
// not ALOGW because it happens all the time when playing key clicks over A2DP
- ALOGV("Minimum buffer size corrected from %d to %d",
+ ALOGV("Minimum buffer size corrected from %zu to %zu",
frameCount, minFrameCount);
frameCount = minFrameCount;
}
@@ -1018,6 +1039,10 @@ status_t AudioTrack::createTrack_l(size_t epoch)
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
+ if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ trackFlags |= IAudioFlinger::TRACK_DIRECT;
+ }
+
size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
// but we will still need the original value also
sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType,
@@ -1071,14 +1096,14 @@ status_t AudioTrack::createTrack_l(size_t epoch)
// In current design, AudioTrack client checks and ensures frame count validity before
// passing it to AudioFlinger so AudioFlinger should not return a different value except
// for fast track as it uses a special method of assigning frame count.
- ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+ ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
}
frameCount = temp;
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
- ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
+ ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount);
mAwaitBoost = true;
if (mSharedBuffer == 0) {
// Theoretically double-buffering is not required for fast tracks,
@@ -1089,7 +1114,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
}
}
} else {
- ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
+ ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
// once denied, do not request again if IAudioTrack is re-created
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
if (mSharedBuffer == 0) {
@@ -1109,6 +1134,16 @@ status_t AudioTrack::createTrack_l(size_t epoch)
//return NO_INIT;
}
}
+ if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (trackFlags & IAudioFlinger::TRACK_DIRECT) {
+ ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful");
+ } else {
+ ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server");
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+ // FIXME This is a warning, not an error, so don't return error status
+ //return NO_INIT;
+ }
+ }
// We retain a copy of the I/O handle, but don't own the reference
mOutput = output;
@@ -1304,6 +1339,16 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
return INVALID_OPERATION;
}
+ if (isDirect()) {
+ AutoMutex lock(mLock);
+ int32_t flags = android_atomic_and(
+ ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
+ &mCblk->mFlags);
+ if (flags & CBLK_INVALID) {
+ return DEAD_OBJECT;
+ }
+ }
+
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// Sanity-check: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
@@ -1452,7 +1497,7 @@ nsecs_t AudioTrack::processAudioBuffer()
// for offloaded tracks restoreTrack_l() will just update the sequence and clear
// AudioSystem cache. We should not exit here but after calling the callback so
// that the upper layers can recreate the track
- if (!isOffloaded_l() || (mSequence == mObservedSequence)) {
+ if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
status_t status = restoreTrack_l("processAudioBuffer");
mLock.unlock();
// Run again immediately, but with a new IAudioTrack
@@ -1578,7 +1623,7 @@ nsecs_t AudioTrack::processAudioBuffer()
mObservedSequence = sequence;
mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
// for offloaded tracks, just wait for the upper layers to recreate the track
- if (isOffloaded()) {
+ if (isOffloadedOrDirect()) {
return NS_INACTIVE;
}
}
@@ -1636,10 +1681,10 @@ nsecs_t AudioTrack::processAudioBuffer()
size_t nonContig;
status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
- "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+ "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
requested = &ClientProxy::kNonBlocking;
size_t avail = audioBuffer.frameCount + nonContig;
- ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+ ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
@@ -1674,8 +1719,8 @@ nsecs_t AudioTrack::processAudioBuffer()
// Sanity check on returned size
if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
- ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
- reqSize, (int) writtenSize);
+ ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
+ reqSize, ssize_t(writtenSize));
return NS_NEVER;
}
@@ -1736,7 +1781,7 @@ nsecs_t AudioTrack::processAudioBuffer()
status_t AudioTrack::restoreTrack_l(const char *from)
{
ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
- isOffloaded_l() ? "Offloaded" : "PCM", from);
+ isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
++mSequence;
status_t result;
@@ -1744,7 +1789,7 @@ status_t AudioTrack::restoreTrack_l(const char *from)
// output parameters in createTrack_l()
AudioSystem::clearAudioConfigCache();
- if (isOffloaded_l()) {
+ if (isOffloadedOrDirect_l()) {
// FIXME re-creation of offloaded tracks is not yet implemented
return DEAD_OBJECT;
}
@@ -1830,6 +1875,19 @@ bool AudioTrack::isOffloaded() const
return isOffloaded_l();
}
+bool AudioTrack::isDirect() const
+{
+ AutoMutex lock(mLock);
+ return isDirect_l();
+}
+
+bool AudioTrack::isOffloadedOrDirect() const
+{
+ AutoMutex lock(mLock);
+ return isOffloadedOrDirect_l();
+}
+
+
status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
{
@@ -1858,6 +1916,136 @@ uint32_t AudioTrack::getUnderrunFrames() const
return mProxy->getUnderrunFrames();
}
+void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) {
+ mAttributes.flags = 0x0;
+
+ switch(streamType) {
+ case AUDIO_STREAM_DEFAULT:
+ case AUDIO_STREAM_MUSIC:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+ mAttributes.usage = AUDIO_USAGE_MEDIA;
+ break;
+ case AUDIO_STREAM_VOICE_CALL:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+ break;
+ case AUDIO_STREAM_ENFORCED_AUDIBLE:
+ mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED;
+ // intended fall through, attributes in common with STREAM_SYSTEM
+ case AUDIO_STREAM_SYSTEM:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
+ break;
+ case AUDIO_STREAM_RING:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
+ break;
+ case AUDIO_STREAM_ALARM:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_ALARM;
+ break;
+ case AUDIO_STREAM_NOTIFICATION:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_NOTIFICATION;
+ break;
+ case AUDIO_STREAM_BLUETOOTH_SCO:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+ mAttributes.flags |= AUDIO_FLAG_SCO;
+ break;
+ case AUDIO_STREAM_DTMF:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
+ break;
+ case AUDIO_STREAM_TTS:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
+ break;
+ default:
+ ALOGE("invalid stream type %d when converting to attributes", streamType);
+ }
+}
+
+void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) {
+ // flags to stream type mapping
+ if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE;
+ return;
+ }
+ if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+ mStreamType = AUDIO_STREAM_BLUETOOTH_SCO;
+ return;
+ }
+
+ // usage to stream type mapping
+ switch (aa.usage) {
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ mStreamType = AUDIO_STREAM_MUSIC;
+ return;
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ mStreamType = AUDIO_STREAM_SYSTEM;
+ return;
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ mStreamType = AUDIO_STREAM_VOICE_CALL;
+ return;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ mStreamType = AUDIO_STREAM_DTMF;
+ return;
+
+ case AUDIO_USAGE_ALARM:
+ mStreamType = AUDIO_STREAM_ALARM;
+ return;
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ mStreamType = AUDIO_STREAM_RING;
+ return;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ mStreamType = AUDIO_STREAM_NOTIFICATION;
+ return;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ mStreamType = AUDIO_STREAM_MUSIC;
+ }
+}
+
+bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) {
+ // has flags that map to a strategy?
+ if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) {
+ return true;
+ }
+
+ // has known usage?
+ switch (paa->usage) {
+ case AUDIO_USAGE_UNKNOWN:
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ case AUDIO_USAGE_GAME:
+ break;
+ default:
+ return false;
+ }
+ return true;
+}
// =========================================================================
void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
@@ -1918,7 +2106,7 @@ bool AudioTrack::AudioTrackThread::threadLoop()
ns = 1000000000LL;
// fall through
default:
- LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+ LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
pauseInternal(ns);
return true;
}
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 0dbfa62..eec025e 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -135,7 +135,7 @@ status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *reques
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
if (mIsOut) {
- ALOGE("Shared memory control block is corrupt (filled=%d, mFrameCount=%u); "
+ ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); "
"shutting down", filled, mFrameCount);
mIsShutdown = true;
status = NO_INIT;
@@ -338,7 +338,7 @@ size_t ClientProxy::getFramesFilled() {
ssize_t filled = rear - front;
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
return 0;
}
return (size_t)filled;
@@ -555,7 +555,7 @@ status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
ssize_t filled = rear - front;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
mIsShutdown = true;
}
if (mIsShutdown) {
@@ -642,7 +642,7 @@ void ServerProxy::releaseBuffer(Buffer* buffer)
}
// FIXME AudioRecord wakeup needs to be optimized; it currently wakes up client every time
if (!mIsOut || (mAvailToClient + stepCount >= minimum)) {
- ALOGV("mAvailToClient=%u stepCount=%u minimum=%u", mAvailToClient, stepCount, minimum);
+ ALOGV("mAvailToClient=%zu stepCount=%zu minimum=%zu", mAvailToClient, stepCount, minimum);
int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
(void) syscall(__NR_futex, &cblk->mFutex,
@@ -675,7 +675,7 @@ size_t AudioTrackServerProxy::framesReady()
ssize_t filled = rear - cblk->u.mStreaming.mFront;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
mIsShutdown = true;
return 0;
}
@@ -834,7 +834,7 @@ void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
size_t newPosition = position + stepCount;
int32_t setFlags = 0;
if (!(position <= newPosition && newPosition <= mFrameCount)) {
- ALOGW("%s newPosition %u outside [%u, %u]", __func__, newPosition, position, mFrameCount);
+ ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, mFrameCount);
newPosition = mFrameCount;
} else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) {
diff --git a/media/libmedia/CharacterEncodingDetector.cpp b/media/libmedia/CharacterEncodingDetector.cpp
index 4992798..7d1ddfd 100644
--- a/media/libmedia/CharacterEncodingDetector.cpp
+++ b/media/libmedia/CharacterEncodingDetector.cpp
@@ -112,7 +112,7 @@ void CharacterEncodingDetector::detectAndConvert() {
if (allprintable) {
// since 'buf' is empty, ICU would return a UTF-8 matcher with low confidence, so
// no need to even call it
- ALOGV("all tags are printable, assuming ascii (%d)", strlen(buf));
+ ALOGV("all tags are printable, assuming ascii (%zu)", strlen(buf));
} else {
ucsdet_setText(csd, buf, strlen(buf), &status);
int32_t matches;
@@ -267,11 +267,11 @@ const UCharsetMatch *CharacterEncodingDetector::getPreferred(
Vector<const UCharsetMatch*> matches;
UErrorCode status = U_ZERO_ERROR;
- ALOGV("%d matches", nummatches);
+ ALOGV("%zu matches", nummatches);
for (size_t i = 0; i < nummatches; i++) {
const char *encname = ucsdet_getName(ucma[i], &status);
int confidence = ucsdet_getConfidence(ucma[i], &status);
- ALOGV("%d: %s %d", i, encname, confidence);
+ ALOGV("%zu: %s %d", i, encname, confidence);
matches.push_back(ucma[i]);
}
@@ -287,7 +287,7 @@ const UCharsetMatch *CharacterEncodingDetector::getPreferred(
return matches[0];
}
- ALOGV("considering %d matches", num);
+ ALOGV("considering %zu matches", num);
// keep track of how many "special" characters result when converting the input using each
// encoding
@@ -315,7 +315,7 @@ const UCharsetMatch *CharacterEncodingDetector::getPreferred(
freqcoverage = frequent_ja_coverage;
}
- ALOGV("%d: %s %d", i, encname, confidence);
+ ALOGV("%zu: %s %d", i, encname, confidence);
UConverter *conv = ucnv_open(encname, &status);
const char *source = input;
const char *sourceLimit = input + len;
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 77d131b..41a9065 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -64,7 +64,8 @@ enum {
RELEASE_AUDIO_PATCH,
LIST_AUDIO_PATCHES,
SET_AUDIO_PORT_CONFIG,
- REGISTER_CLIENT
+ REGISTER_CLIENT,
+ GET_OUTPUT_FOR_ATTR
};
class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
@@ -155,6 +156,36 @@ public:
return static_cast <audio_io_handle_t> (reply.readInt32());
}
+ virtual audio_io_handle_t getOutputForAttr(
+ const audio_attributes_t *attr,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ if (attr == NULL) {
+ ALOGE("Writing NULL audio attributes - shouldn't happen");
+ return (audio_io_handle_t) 0;
+ }
+ data.write(attr, sizeof(audio_attributes_t));
+ data.writeInt32(samplingRate);
+ data.writeInt32(static_cast <uint32_t>(format));
+ data.writeInt32(channelMask);
+ data.writeInt32(static_cast <uint32_t>(flags));
+ // hasOffloadInfo
+ if (offloadInfo == NULL) {
+ data.writeInt32(0);
+ } else {
+ data.writeInt32(1);
+ data.write(offloadInfo, sizeof(audio_offload_info_t));
+ }
+ remote()->transact(GET_OUTPUT_FOR_ATTR, data, &reply);
+ return static_cast <audio_io_handle_t> (reply.readInt32());
+ }
+
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session)
@@ -614,6 +645,30 @@ status_t BnAudioPolicyService::onTransact(
return NO_ERROR;
} break;
+ case GET_OUTPUT_FOR_ATTR: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ audio_attributes_t *attr = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
+ data.read(attr, sizeof(audio_attributes_t));
+ uint32_t samplingRate = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
+ audio_output_flags_t flags =
+ static_cast <audio_output_flags_t>(data.readInt32());
+ bool hasOffloadInfo = data.readInt32() != 0;
+ audio_offload_info_t offloadInfo;
+ if (hasOffloadInfo) {
+ data.read(&offloadInfo, sizeof(audio_offload_info_t));
+ }
+ audio_io_handle_t output = getOutputForAttr(attr,
+ samplingRate,
+ format,
+ channelMask,
+ flags,
+ hasOffloadInfo ? &offloadInfo : NULL);
+ reply->writeInt32(static_cast <int>(output));
+ return NO_ERROR;
+ } break;
+
case START_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index 432d890..38f717c 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -15,8 +15,10 @@
** limitations under the License.
*/
+#include <inttypes.h>
#include <stdint.h>
#include <sys/types.h>
+
#include <binder/Parcel.h>
#include <media/IMediaHTTPService.h>
#include <media/IMediaMetadataRetriever.h>
@@ -125,7 +127,7 @@ public:
sp<IMemory> getFrameAtTime(int64_t timeUs, int option)
{
- ALOGV("getTimeAtTime: time(%lld us) and option(%d)", timeUs, option);
+ ALOGV("getTimeAtTime: time(%" PRId64 " us) and option(%d)", timeUs, option);
Parcel data, reply;
data.writeInterfaceToken(IMediaMetadataRetriever::getInterfaceDescriptor());
data.writeInt64(timeUs);
@@ -237,7 +239,7 @@ status_t BnMediaMetadataRetriever::onTransact(
CHECK_INTERFACE(IMediaMetadataRetriever, data, reply);
int64_t timeUs = data.readInt64();
int option = data.readInt32();
- ALOGV("getTimeAtTime: time(%lld us) and option(%d)", timeUs, option);
+ ALOGV("getTimeAtTime: time(%" PRId64 " us) and option(%d)", timeUs, option);
#ifndef DISABLE_GROUP_SCHEDULE_HACK
setSchedPolicy(data);
#endif
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index 8e58162..95af006 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -17,6 +17,10 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "IMediaRecorder"
+
+#include <inttypes.h>
+#include <unistd.h>
+
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <camera/ICamera.h>
@@ -24,8 +28,6 @@
#include <media/IMediaRecorder.h>
#include <gui/Surface.h>
#include <gui/IGraphicBufferProducer.h>
-#include <unistd.h>
-
namespace android {
@@ -167,7 +169,7 @@ public:
}
status_t setOutputFile(int fd, int64_t offset, int64_t length) {
- ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
Parcel data, reply;
data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor());
data.writeFileDescriptor(fd);
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index 28238c4..e9e453b 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -475,7 +475,7 @@ static bool isTimelapseProfile(camcorder_quality quality) {
}
void MediaProfiles::initRequiredProfileRefs(const Vector<int>& cameraIds) {
- ALOGV("Number of camera ids: %d", cameraIds.size());
+ ALOGV("Number of camera ids: %zu", cameraIds.size());
CHECK(cameraIds.size() > 0);
mRequiredProfileRefs = new RequiredProfiles[cameraIds.size()];
for (size_t i = 0, n = cameraIds.size(); i < n; ++i) {
@@ -602,14 +602,14 @@ void MediaProfiles::checkAndAddRequiredProfilesIfNecessary() {
int index = getCamcorderProfileIndex(cameraId, profile->mQuality);
if (index != -1) {
- ALOGV("Profile quality %d for camera %d already exists",
+ ALOGV("Profile quality %d for camera %zu already exists",
profile->mQuality, cameraId);
CHECK(index == refIndex);
continue;
}
// Insert the new profile
- ALOGV("Add a profile: quality %d=>%d for camera %d",
+ ALOGV("Add a profile: quality %d=>%d for camera %zu",
mCamcorderProfiles[info->mRefProfileIndex]->mQuality,
profile->mQuality, cameraId);
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index a55e09c..2aa0592 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -16,6 +16,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "SoundPool"
+
+#include <inttypes.h>
+
#include <utils/Log.h>
#define USE_SHARED_MEM_BUFFER
@@ -212,7 +215,7 @@ int SoundPool::load(const char* path, int priority __unused)
int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused)
{
- ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
+ ALOGV("load: fd=%d, offset=%" PRId64 ", length=%" PRId64 ", priority=%d",
fd, offset, length, priority);
Mutex::Autolock lock(&mLock);
sp<Sample> sample = new Sample(++mNextSampleID, fd, offset, length);
@@ -462,7 +465,8 @@ Sample::Sample(int sampleID, int fd, int64_t offset, int64_t length)
mFd = dup(fd);
mOffset = offset;
mLength = length;
- ALOGV("create sampleID=%d, fd=%d, offset=%lld, length=%lld", mSampleID, mFd, mLength, mOffset);
+ ALOGV("create sampleID=%d, fd=%d, offset=%" PRId64 " length=%" PRId64,
+ mSampleID, mFd, mLength, mOffset);
}
void Sample::init()
@@ -516,7 +520,7 @@ status_t Sample::doLoad()
ALOGE("Unable to load sample: %s", mUrl);
goto error;
}
- ALOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d",
+ ALOGV("pointer = %p, size = %zu, sampleRate = %u, numChannels = %d",
mHeap->getBase(), mSize, sampleRate, numChannels);
if (sampleRate > kMaxSampleRate) {
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 1d6bb6f..39a239d 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -18,6 +18,8 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaMetadataRetriever"
+#include <inttypes.h>
+
#include <binder/IServiceManager.h>
#include <binder/IPCThreadState.h>
#include <media/mediametadataretriever.h>
@@ -114,7 +116,7 @@ status_t MediaMetadataRetriever::setDataSource(
status_t MediaMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)
{
- ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setDataSource(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
Mutex::Autolock _l(mLock);
if (mRetriever == 0) {
ALOGE("retriever is not initialized");
@@ -129,7 +131,7 @@ status_t MediaMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t l
sp<IMemory> MediaMetadataRetriever::getFrameAtTime(int64_t timeUs, int option)
{
- ALOGV("getFrameAtTime: time(%lld us) option(%d)", timeUs, option);
+ ALOGV("getFrameAtTime: time(%" PRId64 " us) option(%d)", timeUs, option);
Mutex::Autolock _l(mLock);
if (mRetriever == 0) {
ALOGE("retriever is not initialized");
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 0be01a9..406f9f2 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -17,12 +17,14 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaPlayer"
-#include <utils/Log.h>
-#include <sys/types.h>
+#include <fcntl.h>
+#include <inttypes.h>
#include <sys/stat.h>
+#include <sys/types.h>
#include <unistd.h>
-#include <fcntl.h>
+
+#include <utils/Log.h>
#include <binder/IServiceManager.h>
#include <binder/IPCThreadState.h>
@@ -157,7 +159,7 @@ status_t MediaPlayer::setDataSource(
status_t MediaPlayer::setDataSource(int fd, int64_t offset, int64_t length)
{
- ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setDataSource(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
status_t err = UNKNOWN_ERROR;
const sp<IMediaPlayerService>& service(getMediaPlayerService());
if (service != 0) {
@@ -194,7 +196,7 @@ status_t MediaPlayer::invoke(const Parcel& request, Parcel *reply)
(mCurrentState != MEDIA_PLAYER_STATE_ERROR) &&
((mCurrentState & MEDIA_PLAYER_IDLE) != MEDIA_PLAYER_IDLE);
if ((mPlayer != NULL) && hasBeenInitialized) {
- ALOGV("invoke %d", request.dataSize());
+ ALOGV("invoke %zu", request.dataSize());
return mPlayer->invoke(request, reply);
}
ALOGE("invoke failed: wrong state %X", mCurrentState);
@@ -818,7 +820,7 @@ void MediaPlayer::died()
audio_format_t* pFormat,
const sp<IMemoryHeap>& heap, size_t *pSize)
{
- ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("decode(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
status_t status;
const sp<IMediaPlayerService>& service = getMediaPlayerService();
if (service != 0) {
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 3710e46..c8192e9 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -17,6 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaRecorder"
+
+#include <inttypes.h>
+
#include <utils/Log.h>
#include <media/mediarecorder.h>
#include <binder/IServiceManager.h>
@@ -286,7 +289,7 @@ status_t MediaRecorder::setOutputFile(const char* path)
status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length)
{
- ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 778eb9a..76632a7 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -307,7 +307,7 @@ sp<IRemoteDisplay> MediaPlayerService::listenForRemoteDisplay(
return new RemoteDisplay(client, iface.string());
}
-status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& args) const
+status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& /*args*/) const
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -673,8 +673,8 @@ status_t MediaPlayerService::Client::setDataSource(int fd, int64_t offset, int64
ALOGV("st_dev = %llu", sb.st_dev);
ALOGV("st_mode = %u", sb.st_mode);
- ALOGV("st_uid = %lu", sb.st_uid);
- ALOGV("st_gid = %lu", sb.st_gid);
+ ALOGV("st_uid = %lu", static_cast<unsigned long>(sb.st_uid));
+ ALOGV("st_gid = %lu", static_cast<unsigned long>(sb.st_gid));
ALOGV("st_size = %llu", sb.st_size);
if (offset >= sb.st_size) {
@@ -803,7 +803,7 @@ status_t MediaPlayerService::Client::setMetadataFilter(const Parcel& filter)
}
status_t MediaPlayerService::Client::getMetadata(
- bool update_only, bool apply_filter, Parcel *reply)
+ bool update_only, bool /*apply_filter*/, Parcel *reply)
{
sp<MediaPlayerBase> player = getPlayer();
if (player == 0) return UNKNOWN_ERROR;
@@ -1926,8 +1926,8 @@ bool CallbackThread::threadLoop() {
status_t MediaPlayerService::AudioCache::open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
- AudioCallback cb, void *cookie, audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo)
+ AudioCallback cb, void *cookie, audio_output_flags_t /*flags*/,
+ const audio_offload_info_t* /*offloadInfo*/)
{
ALOGV("open(%u, %d, 0x%x, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount);
if (mHeap->getHeapID() < 0) {
@@ -1994,7 +1994,7 @@ status_t MediaPlayerService::AudioCache::wait()
}
void MediaPlayerService::AudioCache::notify(
- void* cookie, int msg, int ext1, int ext2, const Parcel *obj)
+ void* cookie, int msg, int ext1, int ext2, const Parcel* /*obj*/)
{
ALOGV("notify(%p, %d, %d, %d)", cookie, msg, ext1, ext2);
AudioCache* p = static_cast<AudioCache*>(cookie);
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 80c7e0a..a91b0e5 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -147,8 +147,8 @@ status_t MetadataRetrieverClient::setDataSource(int fd, int64_t offset, int64_t
}
ALOGV("st_dev = %llu", sb.st_dev);
ALOGV("st_mode = %u", sb.st_mode);
- ALOGV("st_uid = %lu", sb.st_uid);
- ALOGV("st_gid = %lu", sb.st_gid);
+ ALOGV("st_uid = %lu", static_cast<unsigned long>(sb.st_uid));
+ ALOGV("st_gid = %lu", static_cast<unsigned long>(sb.st_gid));
ALOGV("st_size = %llu", sb.st_size);
if (offset >= sb.st_size) {
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index deeddd1..749ef96 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -114,7 +114,7 @@ MidiFile::~MidiFile() {
}
status_t MidiFile::setDataSource(
- const sp<IMediaHTTPService> &httpService,
+ const sp<IMediaHTTPService> & /*httpService*/,
const char* path,
const KeyedVector<String8, String8> *) {
ALOGV("MidiFile::setDataSource url=%s", path);
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index 4adf018..0b65861 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -14,6 +14,8 @@
* limitations under the License.
*/
+#include <inttypes.h>
+
#define LOG_TAG "MonoPipe"
//#define LOG_NDEBUG 0
@@ -87,7 +89,7 @@ MonoPipe::MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBl
static const uint64_t kUnsignedHiBitsMask = ~(0xFFFFFFFFull);
if ((N & kSignedHiBitsMask) || (D & kUnsignedHiBitsMask)) {
ALOGE("Cannot reduce sample rate to local clock frequency ratio to fit"
- " in a 32/32 bit rational. (max reduction is 0x%016llx/0x%016llx"
+ " in a 32/32 bit rational. (max reduction is 0x%016" PRIx64 "/0x%016" PRIx64
"). getNextWriteTimestamp calls will be non-functional", N, D);
return;
}
@@ -308,7 +310,7 @@ int64_t MonoPipe::offsetTimestampByAudioFrames(int64_t ts, size_t audFrames)
// error, but then zero out the ratio in the linear transform so
// that we don't try to do any conversions from now on. This
// MonoPipe's getNextWriteTimestamp is now broken for good.
- ALOGE("Overflow when attempting to convert %d audio frames to"
+ ALOGE("Overflow when attempting to convert %zu audio frames to"
" duration in local time. getNextWriteTimestamp will fail from"
" now on.", audFrames);
mSamplesToLocalTime.a_to_b_numer = 0;
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index ff3284c..d641e74 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -137,7 +137,7 @@ ssize_t NBAIO_Source::readVia(readVia_t via, size_t total, void *user,
ssize_t NBAIO_Port::negotiate(const NBAIO_Format offers[], size_t numOffers,
NBAIO_Format counterOffers[], size_t& numCounterOffers)
{
- ALOGV("negotiate offers=%p numOffers=%u countersOffers=%p numCounterOffers=%u",
+ ALOGV("negotiate offers=%p numOffers=%zu countersOffers=%p numCounterOffers=%zu",
offers, numOffers, counterOffers, numCounterOffers);
if (Format_isValid(mFormat)) {
for (size_t i = 0; i < numOffers; ++i) {
generated by cgit v1.1 at 2024-05-15 18:40:49 +0000