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-rw-r--r--media/libmedia/AudioRecord.cpp4
-rw-r--r--media/libmedia/AudioSystem.cpp12
-rw-r--r--media/libmedia/AudioTrack.cpp18
-rw-r--r--media/libmedia/IAudioFlinger.cpp2
-rw-r--r--media/libmedia/SoundPool.cpp2
5 files changed, 19 insertions, 19 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;
generated by cgit v1.1 at 2025-01-11 06:48:52 +0000