Use uint32_t for sample rate

Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80

6 files changed, 20 insertions, 20 deletions

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;