Merge "camera: Let #connectLegacy calls through with unspecified HAL version"
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 @@
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 @@
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 @@
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 @@
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 @@
// 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 @@
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 f5e7a78..a47d45c 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -323,7 +323,7 @@
return BAD_VALUE;
}
- ALOGV("getFrameCount() output %d, frameCount %d", output, *frameCount);
+ ALOGV("getFrameCount() output %d, frameCount %zu", output, *frameCount);
return NO_ERROR;
}
@@ -489,7 +489,7 @@
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);
@@ -514,7 +514,7 @@
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);
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 5c82cfb..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 @@
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;
}
@@ -250,7 +251,7 @@
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);
@@ -993,14 +994,14 @@
// 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);
@@ -1008,7 +1009,7 @@
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;
}
@@ -1095,14 +1096,14 @@
// 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,
@@ -1113,7 +1114,7 @@
}
}
} 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) {
@@ -1680,10 +1681,10 @@
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 ||
@@ -1718,8 +1719,8 @@
// 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;
}
@@ -2105,7 +2106,7 @@
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 @@
// 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 @@
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 @@
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 @@
}
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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/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 @@
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 @@
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 @@
}
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 @@
}
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 @@
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(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 @@
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 @@
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(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 @@
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 <sys/stat.h>
-#include <unistd.h>
#include <fcntl.h>
+#include <inttypes.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <utils/Log.h>
#include <binder/IServiceManager.h>
#include <binder/IPCThreadState.h>
@@ -157,7 +159,7 @@
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 @@
(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 @@
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(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/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk
index 874f18f..f365637 100644
--- a/services/audioflinger/tests/Android.mk
+++ b/services/audioflinger/tests/Android.mk
@@ -1,5 +1,8 @@
# Build the unit tests for audioflinger
+#
+# resampler unit test
+#
LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
@@ -20,6 +23,7 @@
bionic/libstdc++/include \
external/gtest/include \
external/stlport/stlport \
+ $(call include-path-for, audio-utils) \
frameworks/av/services/audioflinger
LOCAL_SRC_FILES := \
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index 8f9c270..4a67d0b 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -33,200 +33,7 @@
#include <gtest/gtest.h>
#include <media/AudioBufferProvider.h>
#include "AudioResampler.h"
-
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-template<typename T, typename U>
-struct is_same
-{
- static const bool value = false;
-};
-
-template<typename T>
-struct is_same<T, T> // partial specialization
-{
- static const bool value = true;
-};
-
-template<typename T>
-static inline T convertValue(double val)
-{
- if (is_same<T, int16_t>::value) {
- return floor(val * 32767.0 + 0.5);
- } else if (is_same<T, int32_t>::value) {
- return floor(val * (1UL<<31) + 0.5);
- }
- return val; // assume float or double
-}
-
-/* Creates a type-independent audio buffer provider from
- * a buffer base address, size, framesize, and input increment array.
- *
- * No allocation or deallocation of the provided buffer is done.
- */
-class TestProvider : public android::AudioBufferProvider {
-public:
- TestProvider(const void* addr, size_t frames, size_t frameSize,
- const std::vector<size_t>& inputIncr)
- : mAddr(addr),
- mNumFrames(frames),
- mFrameSize(frameSize),
- mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
- {
- }
-
- virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS )
- {
- size_t requestedFrames = buffer->frameCount;
- if (requestedFrames > mNumFrames - mNextFrame) {
- buffer->frameCount = mNumFrames - mNextFrame;
- }
- if (!mInputIncr.empty()) {
- size_t provided = mInputIncr[mNextIdx++];
- ALOGV("getNextBuffer() mValue[%d]=%u not %u",
- mNextIdx-1, provided, buffer->frameCount);
- if (provided < buffer->frameCount) {
- buffer->frameCount = provided;
- }
- if (mNextIdx >= mInputIncr.size()) {
- mNextIdx = 0;
- }
- }
- ALOGV("getNextBuffer() requested %u frames out of %u frames available"
- " and returned %u frames\n",
- requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
- mUnrel = buffer->frameCount;
- if (buffer->frameCount > 0) {
- buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
- return android::NO_ERROR;
- } else {
- buffer->raw = NULL;
- return android::NOT_ENOUGH_DATA;
- }
- }
-
- virtual void releaseBuffer(Buffer* buffer)
- {
- if (buffer->frameCount > mUnrel) {
- ALOGE("releaseBuffer() released %u frames but only %u available "
- "to release\n", buffer->frameCount, mUnrel);
- mNextFrame += mUnrel;
- mUnrel = 0;
- } else {
-
- ALOGV("releaseBuffer() released %u frames out of %u frames available "
- "to release\n", buffer->frameCount, mUnrel);
- mNextFrame += buffer->frameCount;
- mUnrel -= buffer->frameCount;
- }
- buffer->frameCount = 0;
- buffer->raw = NULL;
- }
-
- void reset()
- {
- mNextFrame = 0;
- }
-
- size_t getNumFrames()
- {
- return mNumFrames;
- }
-
- void setIncr(const std::vector<size_t> inputIncr)
- {
- mNextIdx = 0;
- mInputIncr = inputIncr;
- }
-
-protected:
- const void* mAddr; // base address
- size_t mNumFrames; // total frames
- int mFrameSize; // frame size (# channels * bytes per sample)
- size_t mNextFrame; // index of next frame to provide
- size_t mUnrel; // number of frames not yet released
- std::vector<size_t> mInputIncr; // number of frames provided per call
- size_t mNextIdx; // index of next entry in mInputIncr to use
-};
-
-/* Creates a buffer filled with a sine wave.
- *
- * Returns a pair consisting of the sine signal buffer and the number of frames.
- * The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
- */
-template<typename T>
-static std::pair<T*, size_t> createSine(size_t channels,
- double freq, double samplingRate, double time)
-{
- double tscale = 1. / samplingRate;
- size_t frames = static_cast<size_t>(samplingRate * time);
- T* buffer = new T[frames * channels];
- for (size_t i = 0; i < frames; ++i) {
- double t = i * tscale;
- double y = sin(2. * M_PI * freq * t);
- T yt = convertValue<T>(y);
-
- for (size_t j = 0; j < channels; ++j) {
- buffer[i*channels + j] = yt / (j + 1);
- }
- }
- return std::make_pair(buffer, frames);
-}
-
-/* Creates a buffer filled with a chirp signal (a sine wave sweep).
- *
- * Returns a pair consisting of the chirp signal buffer and the number of frames.
- * The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
- *
- * When creating the Chirp, note that the frequency is the true sinusoidal
- * frequency not the sampling rate.
- *
- * http://en.wikipedia.org/wiki/Chirp
- */
-template<typename T>
-static std::pair<T*, size_t> createChirp(size_t channels,
- double minfreq, double maxfreq, double samplingRate, double time)
-{
- double tscale = 1. / samplingRate;
- size_t frames = static_cast<size_t>(samplingRate * time);
- T *buffer = new T[frames * channels];
- // note the chirp constant k has a divide-by-two.
- double k = (maxfreq - minfreq) / (2. * time);
- for (size_t i = 0; i < frames; ++i) {
- double t = i * tscale;
- double y = sin(2. * M_PI * (k * t + minfreq) * t);
- T yt = convertValue<T>(y);
-
- for (size_t j = 0; j < channels; ++j) {
- buffer[i*channels + j] = yt / (j + 1);
- }
- }
- return std::make_pair(buffer, frames);
-}
-
-/* This derived class creates a buffer provider of datatype T,
- * consisting of an input signal, e.g. from createChirp().
- * The number of frames can be obtained from the base class
- * TestProvider::getNumFrames().
- */
-template <typename T>
-class SignalProvider : public TestProvider {
-public:
- SignalProvider(const std::pair<T*, size_t>& bufferInfo, size_t channels,
- const std::vector<size_t>& values)
- : TestProvider(bufferInfo.first, bufferInfo.second, channels * sizeof(T), values),
- mManagedPtr(bufferInfo.first)
- {
- }
-
- virtual ~SignalProvider()
- {
- delete[] mManagedPtr;
- }
-
-protected:
- T* mManagedPtr;
-};
+#include "test_utils.h"
void resample(void *output, size_t outputFrames, const std::vector<size_t> &outputIncr,
android::AudioBufferProvider *provider, android::AudioResampler *resampler)
@@ -261,10 +68,11 @@
enum android::AudioResampler::src_quality quality)
{
// create the provider
- std::vector<size_t> inputIncr;
- SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
- 0., outputFreq/2., outputFreq, outputFreq/2000.),
- channels, inputIncr);
+ std::vector<int> inputIncr;
+ SignalProvider provider;
+ provider.setChirp<int16_t>(channels,
+ 0., outputFreq/2., outputFreq, outputFreq/2000.);
+ provider.setIncr(inputIncr);
// calculate the output size
size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
@@ -339,10 +147,11 @@
enum android::AudioResampler::src_quality quality)
{
// create the provider
- std::vector<size_t> inputIncr;
- SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
- 0., inputFreq/2., inputFreq, inputFreq/2000.),
- channels, inputIncr);
+ std::vector<int> inputIncr;
+ SignalProvider provider;
+ provider.setChirp<int16_t>(channels,
+ 0., inputFreq/2., inputFreq, inputFreq/2000.);
+ provider.setIncr(inputIncr);
// calculate the output size
size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
diff --git a/services/audioflinger/tests/test_utils.h b/services/audioflinger/tests/test_utils.h
new file mode 100644
index 0000000..f954292
--- /dev/null
+++ b/services/audioflinger/tests/test_utils.h
@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_TEST_UTILS_H
+#define ANDROID_AUDIO_TEST_UTILS_H
+
+#include <audio_utils/sndfile.h>
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+#endif
+
+template<typename T, typename U>
+struct is_same
+{
+ static const bool value = false;
+};
+
+template<typename T>
+struct is_same<T, T> // partial specialization
+{
+ static const bool value = true;
+};
+
+template<typename T>
+static inline T convertValue(double val)
+{
+ if (is_same<T, int16_t>::value) {
+ return floor(val * 32767.0 + 0.5);
+ } else if (is_same<T, int32_t>::value) {
+ return floor(val * (1UL<<31) + 0.5);
+ }
+ return val; // assume float or double
+}
+
+// Convert a list of integers in CSV format to a Vector of those values.
+// Returns the number of elements in the list, or -1 on error.
+static inline int parseCSV(const char *string, std::vector<int>& values)
+{
+ // pass 1: count the number of values and do syntax check
+ size_t numValues = 0;
+ bool hadDigit = false;
+ for (const char *p = string; ; ) {
+ switch (*p++) {
+ case '0': case '1': case '2': case '3': case '4':
+ case '5': case '6': case '7': case '8': case '9':
+ hadDigit = true;
+ break;
+ case '\0':
+ if (hadDigit) {
+ // pass 2: allocate and initialize vector of values
+ values.resize(++numValues);
+ values[0] = atoi(p = string);
+ for (size_t i = 1; i < numValues; ) {
+ if (*p++ == ',') {
+ values[i++] = atoi(p);
+ }
+ }
+ return numValues;
+ }
+ // fall through
+ case ',':
+ if (hadDigit) {
+ hadDigit = false;
+ numValues++;
+ break;
+ }
+ // fall through
+ default:
+ return -1;
+ }
+ }
+}
+
+/* Creates a type-independent audio buffer provider from
+ * a buffer base address, size, framesize, and input increment array.
+ *
+ * No allocation or deallocation of the provided buffer is done.
+ */
+class TestProvider : public android::AudioBufferProvider {
+public:
+ TestProvider(void* addr, size_t frames, size_t frameSize,
+ const std::vector<int>& inputIncr)
+ : mAddr(addr),
+ mNumFrames(frames),
+ mFrameSize(frameSize),
+ mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
+ {
+ }
+
+ TestProvider()
+ : mAddr(NULL), mNumFrames(0), mFrameSize(0),
+ mNextFrame(0), mUnrel(0), mNextIdx(0)
+ {
+ }
+
+ void setIncr(const std::vector<int>& inputIncr) {
+ mInputIncr = inputIncr;
+ mNextIdx = 0;
+ }
+
+ virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS)
+ {
+ size_t requestedFrames = buffer->frameCount;
+ if (requestedFrames > mNumFrames - mNextFrame) {
+ buffer->frameCount = mNumFrames - mNextFrame;
+ }
+ if (!mInputIncr.empty()) {
+ size_t provided = mInputIncr[mNextIdx++];
+ ALOGV("getNextBuffer() mValue[%d]=%u not %u",
+ mNextIdx-1, provided, buffer->frameCount);
+ if (provided < buffer->frameCount) {
+ buffer->frameCount = provided;
+ }
+ if (mNextIdx >= mInputIncr.size()) {
+ mNextIdx = 0;
+ }
+ }
+ ALOGV("getNextBuffer() requested %u frames out of %u frames available"
+ " and returned %u frames\n",
+ requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
+ mUnrel = buffer->frameCount;
+ if (buffer->frameCount > 0) {
+ buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
+ return android::NO_ERROR;
+ } else {
+ buffer->raw = NULL;
+ return android::NOT_ENOUGH_DATA;
+ }
+ }
+
+ virtual void releaseBuffer(Buffer* buffer)
+ {
+ if (buffer->frameCount > mUnrel) {
+ ALOGE("releaseBuffer() released %u frames but only %u available "
+ "to release\n", buffer->frameCount, mUnrel);
+ mNextFrame += mUnrel;
+ mUnrel = 0;
+ } else {
+
+ ALOGV("releaseBuffer() released %u frames out of %u frames available "
+ "to release\n", buffer->frameCount, mUnrel);
+ mNextFrame += buffer->frameCount;
+ mUnrel -= buffer->frameCount;
+ }
+ buffer->frameCount = 0;
+ buffer->raw = NULL;
+ }
+
+ void reset()
+ {
+ mNextFrame = 0;
+ }
+
+ size_t getNumFrames()
+ {
+ return mNumFrames;
+ }
+
+
+protected:
+ void* mAddr; // base address
+ size_t mNumFrames; // total frames
+ int mFrameSize; // frame size (# channels * bytes per sample)
+ size_t mNextFrame; // index of next frame to provide
+ size_t mUnrel; // number of frames not yet released
+ std::vector<int> mInputIncr; // number of frames provided per call
+ size_t mNextIdx; // index of next entry in mInputIncr to use
+};
+
+/* Creates a buffer filled with a sine wave.
+ */
+template<typename T>
+static void createSine(void *vbuffer, size_t frames,
+ size_t channels, double sampleRate, double freq)
+{
+ double tscale = 1. / sampleRate;
+ T* buffer = reinterpret_cast<T*>(vbuffer);
+ for (size_t i = 0; i < frames; ++i) {
+ double t = i * tscale;
+ double y = sin(2. * M_PI * freq * t);
+ T yt = convertValue<T>(y);
+
+ for (size_t j = 0; j < channels; ++j) {
+ buffer[i*channels + j] = yt / (j + 1);
+ }
+ }
+}
+
+/* Creates a buffer filled with a chirp signal (a sine wave sweep).
+ *
+ * When creating the Chirp, note that the frequency is the true sinusoidal
+ * frequency not the sampling rate.
+ *
+ * http://en.wikipedia.org/wiki/Chirp
+ */
+template<typename T>
+static void createChirp(void *vbuffer, size_t frames,
+ size_t channels, double sampleRate, double minfreq, double maxfreq)
+{
+ double tscale = 1. / sampleRate;
+ T *buffer = reinterpret_cast<T*>(vbuffer);
+ // note the chirp constant k has a divide-by-two.
+ double k = (maxfreq - minfreq) / (2. * tscale * frames);
+ for (size_t i = 0; i < frames; ++i) {
+ double t = i * tscale;
+ double y = sin(2. * M_PI * (k * t + minfreq) * t);
+ T yt = convertValue<T>(y);
+
+ for (size_t j = 0; j < channels; ++j) {
+ buffer[i*channels + j] = yt / (j + 1);
+ }
+ }
+}
+
+/* This derived class creates a buffer provider of datatype T,
+ * consisting of an input signal, e.g. from createChirp().
+ * The number of frames can be obtained from the base class
+ * TestProvider::getNumFrames().
+ */
+
+class SignalProvider : public TestProvider {
+public:
+ SignalProvider()
+ : mSampleRate(0),
+ mChannels(0)
+ {
+ }
+
+ virtual ~SignalProvider()
+ {
+ free(mAddr);
+ mAddr = NULL;
+ }
+
+ template <typename T>
+ void setChirp(size_t channels, double minfreq, double maxfreq, double sampleRate, double time)
+ {
+ createBufferByFrames<T>(channels, sampleRate, sampleRate*time);
+ createChirp<T>(mAddr, mNumFrames, mChannels, mSampleRate, minfreq, maxfreq);
+ }
+
+ template <typename T>
+ void setSine(size_t channels,
+ double freq, double sampleRate, double time)
+ {
+ createBufferByFrames<T>(channels, sampleRate, sampleRate*time);
+ createSine<T>(mAddr, mNumFrames, mChannels, mSampleRate, freq);
+ }
+
+ template <typename T>
+ void setFile(const char *file_in)
+ {
+ SF_INFO info;
+ info.format = 0;
+ SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
+ if (sf == NULL) {
+ perror(file_in);
+ return;
+ }
+ createBufferByFrames<T>(info.channels, info.samplerate, info.frames);
+ if (is_same<T, float>::value) {
+ (void) sf_readf_float(sf, (float *) mAddr, mNumFrames);
+ } else if (is_same<T, short>::value) {
+ (void) sf_readf_short(sf, (short *) mAddr, mNumFrames);
+ }
+ sf_close(sf);
+ }
+
+ template <typename T>
+ void createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames)
+ {
+ mNumFrames = frames;
+ mChannels = channels;
+ mFrameSize = mChannels * sizeof(T);
+ free(mAddr);
+ mAddr = malloc(mFrameSize * mNumFrames);
+ mSampleRate = sampleRate;
+ }
+
+ uint32_t getSampleRate() const {
+ return mSampleRate;
+ }
+
+ uint32_t getNumChannels() const {
+ return mChannels;
+ }
+
+protected:
+ uint32_t mSampleRate;
+ uint32_t mChannels;
+};
+
+#endif // ANDROID_AUDIO_TEST_UTILS_H