Use uint32_t for sample rate
Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 9353e70..6406b6c 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1291,7 +1291,7 @@
result.append(buffer);
snprintf(buffer, SIZE, "standby: %d\n", mStandby);
result.append(buffer);
- snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
+ snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
result.append(buffer);
snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
result.append(buffer);
@@ -1776,7 +1776,7 @@
frameCount, mFrameCount);
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
- "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d "
+ "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
"hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
audio_is_linear_pcm(format),
@@ -1801,7 +1801,7 @@
if (mType == DIRECT) {
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
- ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x "
+ ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
"for output %p with format %d",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
@@ -1811,7 +1811,7 @@
} else {
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (sampleRate > mSampleRate*2) {
- ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
+ ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -2280,7 +2280,7 @@
// mNormalSink below
{
ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
- ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
+ ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
"mFrameCount=%d, mNormalFrameCount=%d",
mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
mNormalFrameCount);
@@ -3126,7 +3126,7 @@
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
- if (t->sampleRate() == (int)mSampleRate) {
+ if (t->sampleRate() == mSampleRate) {
minFrames = mNormalFrameCount;
} else {
// +1 for rounding and +1 for additional sample needed for interpolation
@@ -3624,7 +3624,7 @@
NBAIO_Format format = teeSource->format();
unsigned channelCount = Format_channelCount(format);
ALOG_ASSERT(channelCount <= FCC_2);
- unsigned sampleRate = Format_sampleRate(format);
+ uint32_t sampleRate = Format_sampleRate(format);
wavHeader[22] = channelCount; // number of channels
wavHeader[24] = sampleRate; // sample rate
wavHeader[25] = sampleRate >> 8;
@@ -4306,8 +4306,8 @@
ALOGV("TrackBase::reset");
}
-int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
- return (int)mCblk->sampleRate;
+uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
+ return mCblk->sampleRate;
}
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
@@ -5541,7 +5541,7 @@
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
- "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
+ "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p",
mCblk, mBuffer, mCblk->buffers,
mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
} else {
@@ -6558,7 +6558,7 @@
result.append(buffer);
snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
result.append(buffer);
- snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
+ snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
result.append(buffer);
} else {
result.append("No active record client\n");
@@ -6653,7 +6653,7 @@
AudioParameter param = AudioParameter(keyValuePair);
int value;
audio_format_t reqFormat = mFormat;
- int reqSamplingRate = mReqSampleRate;
+ uint32_t reqSamplingRate = mReqSampleRate;
int reqChannelCount = mReqChannelCount;
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
@@ -6987,7 +6987,7 @@
// ----------------------------------------------------------------------------
-int32_t AudioFlinger::getPrimaryOutputSamplingRate()
+uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = primaryPlaybackThread_l();