Use uint32_t for sample rate
Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index d64ecd4..33078bb 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -87,7 +87,7 @@
static float linearToLog(int volume);
static int logToLinear(float volume);
- static status_t getOutputSamplingRate(int* samplingRate,
+ static status_t getOutputSamplingRate(uint32_t* samplingRate,
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
static status_t getOutputFrameCount(int* frameCount,
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
@@ -95,7 +95,7 @@
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
static status_t getSamplingRate(audio_io_handle_t output,
audio_stream_type_t streamType,
- int* samplingRate);
+ uint32_t* samplingRate);
// returns the number of frames per audio HAL write buffer. Corresponds to
// audio_stream->get_buffer_size()/audio_stream_frame_size()
static status_t getFrameCount(audio_io_handle_t output,
@@ -237,7 +237,7 @@
static const sp<IAudioPolicyService>& get_audio_policy_service();
// helpers for android.media.AudioManager.getProperty(), see description there for meaning
- static int32_t getPrimaryOutputSamplingRate();
+ static uint32_t getPrimaryOutputSamplingRate();
static int32_t getPrimaryOutputFrameCount();
// ----------------------------------------------------------------------------
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index b82f814..99d583d 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -282,7 +282,9 @@
/* Set sample rate for this track in Hz, mostly used for games' sound effects
*/
- status_t setSampleRate(int sampleRate);
+ status_t setSampleRate(uint32_t sampleRate);
+
+ /* Return current sample rate in Hz, or 0 if unknown */
uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 0aa48c6..5fd5044 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -192,7 +192,7 @@
// helpers for android.media.AudioManager.getProperty(), see description there for meaning
// FIXME move these APIs to AudioPolicy to permit a more accurate implementation
// that looks on primary device for a stream with fast flag, primary flag, or first one.
- virtual int32_t getPrimaryOutputSamplingRate() = 0;
+ virtual uint32_t getPrimaryOutputSamplingRate() = 0;
virtual int32_t getPrimaryOutputFrameCount() = 0;
};
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 29c8fd9..0529bcd 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -263,7 +263,7 @@
unsigned short mLoopCounter; // Current tone loopback count
- int mSamplingRate; // AudioFlinger Sampling rate
+ uint32_t mSamplingRate; // AudioFlinger Sampling rate
AudioTrack *mpAudioTrack; // Pointer to audio track used for playback
Mutex mLock; // Mutex to control concurent access to ToneGenerator object from audio callback and application API
Mutex mCbkCondLock; // Mutex associated to mWaitCbkCond
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index fc9fb49..d34b6d3 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -406,7 +406,7 @@
}
ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount);
if (mTrack) close();
- int afSampleRate;
+ uint32_t afSampleRate;
int afFrameCount;
int frameCount;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 062f546..8f45a57 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -54,7 +54,7 @@
}
if (size == 0) {
- ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
+ ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
@@ -127,7 +127,7 @@
int sessionId)
{
- ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask,
+ ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %d", sampleRate, channelMask,
frameCount);
AutoMutex lock(mLock);
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 488edac..f3b74a2 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -205,7 +205,7 @@
return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
}
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType)
+status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t streamType)
{
audio_io_handle_t output;
@@ -223,7 +223,7 @@
status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
audio_stream_type_t streamType,
- int* samplingRate)
+ uint32_t* samplingRate)
{
OutputDescriptor *outputDesc;
@@ -241,7 +241,7 @@
gLock.unlock();
}
- ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output,
+ ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output,
*samplingRate);
return NO_ERROR;
@@ -442,7 +442,7 @@
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %#x frameCount %d "
+ ALOGV("ioConfigChanged() new output samplingRate %u, format %d channels %#x frameCount %d "
"latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channels,
outputDesc->frameCount, outputDesc->latency);
@@ -466,7 +466,7 @@
if (param2 == NULL) break;
desc = (const OutputDescriptor *)param2;
- ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %#x "
+ ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channels %#x "
"frameCount %d latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channels, desc->frameCount, desc->latency);
@@ -740,7 +740,7 @@
return NO_ERROR;
}
-int32_t AudioSystem::getPrimaryOutputSamplingRate()
+uint32_t AudioSystem::getPrimaryOutputSamplingRate()
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return 0;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index daf6d07..7480807 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -65,7 +65,7 @@
// audio_format_t format
// audio_channel_mask_t channelMask
// audio_output_flags_t flags
- int afSampleRate;
+ uint32_t afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
@@ -193,7 +193,7 @@
}
if (sampleRate == 0) {
- int afSampleRate;
+ uint32_t afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
@@ -535,9 +535,9 @@
}
}
-status_t AudioTrack::setSampleRate(int rate)
+status_t AudioTrack::setSampleRate(uint32_t rate)
{
- int afSamplingRate;
+ uint32_t afSamplingRate;
if (mIsTimed) {
return INVALID_OPERATION;
@@ -547,7 +547,7 @@
return NO_INIT;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
- if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
+ if (rate == 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
AutoMutex lock(mLock);
mCblk->sampleRate = rate;
@@ -557,7 +557,7 @@
uint32_t AudioTrack::getSampleRate() const
{
if (mIsTimed) {
- return INVALID_OPERATION;
+ return 0;
}
AutoMutex lock(mLock);
@@ -802,7 +802,7 @@
} else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
- int afSampleRate;
+ uint32_t afSampleRate;
if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
return NO_INIT;
}
@@ -816,7 +816,7 @@
if (minBufCount < 2) minBufCount = 2;
int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
- ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
+ ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
", afLatency=%d",
minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
@@ -1423,7 +1423,7 @@
snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat,
mChannelCount, cblk->frameCount);
result.append(buffer);
- snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n",
+ snprintf(buffer, 255, " sample rate(%u), status(%d), muted(%d)\n",
(cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted);
result.append(buffer);
snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 55658db..0eeb6d9 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -695,7 +695,7 @@
return (audio_module_handle_t) reply.readInt32();
}
- virtual int32_t getPrimaryOutputSamplingRate()
+ virtual uint32_t getPrimaryOutputSamplingRate()
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index abc8899..b321e92 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -569,7 +569,7 @@
// initialize track
int afFrameCount;
- int afSampleRate;
+ uint32_t afSampleRate;
audio_stream_type_t streamType = mSoundPool->streamType();
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
afFrameCount = kDefaultFrameCount;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 9bedff1..769b322 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1387,7 +1387,7 @@
}
ALOGV("open(%u, %d, 0x%x, %d, %d, %d)", sampleRate, channelCount, channelMask,
format, bufferCount, mSessionId);
- int afSampleRate;
+ uint32_t afSampleRate;
int afFrameCount;
uint32_t frameCount;
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();
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 54cf239..8816929 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -207,7 +207,7 @@
virtual audio_module_handle_t loadHwModule(const char *name);
- virtual int32_t getPrimaryOutputSamplingRate();
+ virtual uint32_t getPrimaryOutputSamplingRate();
virtual int32_t getPrimaryOutputFrameCount();
virtual status_t onTransact(
@@ -423,7 +423,7 @@
audio_channel_mask_t channelMask() const { return mChannelMask; }
- int sampleRate() const; // FIXME inline after cblk sr moved
+ uint32_t sampleRate() const; // FIXME inline after cblk sr moved
// Return a pointer to the start of a contiguous slice of the track buffer.
// Parameter 'offset' is the requested start position, expressed in