Use uint32_t for sample rate

Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80

13 files changed, 44 insertions, 42 deletions

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 @@ public:
     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 @@ public:
             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 @@ public:
     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 @@ public:
 
     /* 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 @@ public:
     // 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 @@ private:
 
     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 @@ status_t VideoEditorPlayer::VeAudioOutput::open(
     }
     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 @@ status_t AudioRecord::getMinFrameCount(
     }
 
     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 @@ status_t AudioRecord::set(
         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 @@ int AudioSystem::logToLinear(float volume)
     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::getOutputSamplingRate(int* samplingRate, audio_stream_type
 
 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 @@ status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
         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 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
 
         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 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
         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 @@ status_t AudioSystem::isSourceActive(audio_source_t stream, bool* state)
     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 @@ status_t AudioTrack::getMinFrameCount(
     //          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 @@ status_t AudioTrack::set(
     }
 
     if (sampleRate == 0) {
-        int afSampleRate;
+        uint32_t afSampleRate;
         if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
             return NO_INIT;
         }
@@ -535,9 +535,9 @@ void AudioTrack::getAuxEffectSendLevel(float* level) const
     }
 }
 
-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 @@ status_t AudioTrack::setSampleRate(int rate)
         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 @@ status_t AudioTrack::setSampleRate(int rate)
 uint32_t AudioTrack::getSampleRate() const
 {
     if (mIsTimed) {
-        return INVALID_OPERATION;
+        return 0;
     }
 
     AutoMutex lock(mLock);
@@ -802,7 +802,7 @@ status_t AudioTrack::createTrack_l(
     } 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 @@ status_t AudioTrack::createTrack_l(
         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 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
     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 @@ public:
         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 @@ void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftV
 
         // 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 @@ status_t MediaPlayerService::AudioOutput::open(
     }
     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 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
     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 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
                 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 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
     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 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
     } 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 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud
         // 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 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
         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 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
             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 @@ void AudioFlinger::ThreadBase::TrackBase::reset() {
     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 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
         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 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a
         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 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
         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 @@ audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
 
 // ----------------------------------------------------------------------------
 
-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 @@ public:
 
     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 @@ private:
 
             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