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author | Glenn Kasten <gkasten@google.com> | 2012-11-14 08:44:39 -0800 |
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committer | Glenn Kasten <gkasten@google.com> | 2012-11-14 16:19:23 -0800 |
commit | 3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17 (patch) | |
tree | 56e1f373606202e5d5277f9645158d91b1a0a80c /media/libmedia/AudioSystem.cpp | |
parent | b4a17e834b718eff1ba2eac4232de6e73a4bf9f5 (diff) | |
download | frameworks_av-3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17.zip frameworks_av-3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17.tar.gz frameworks_av-3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17.tar.bz2 |
Use uint32_t for sample rate
Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80
Diffstat (limited to 'media/libmedia/AudioSystem.cpp')
-rw-r--r-- | media/libmedia/AudioSystem.cpp | 12 |
1 files changed, 6 insertions, 6 deletions
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; |