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authorGlenn Kasten <gkasten@google.com>2013-08-19 18:45:42 -0700
committerGlenn Kasten <gkasten@google.com>2013-11-06 15:49:00 -0800
commitf551e9961a6f66d63bb57f288c4be01b00725289 (patch)
tree1253e7bc4034a35606881fe75e6c22b3a8c602cc /services
parentd21952fe5bc94f7bb6ee729f806c0e641fdb1a0d (diff)
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Channel counts are uint32_t
Change-Id: I5676b6a291b72d7db6326f7eaf6d17588de51b05
Diffstat (limited to 'services')
-rw-r--r--services/audioflinger/AudioMixer.cpp2
-rw-r--r--services/audioflinger/Threads.cpp4
2 files changed, 3 insertions, 3 deletions
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 91aedbb..8bd04c6 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -566,7 +566,7 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
resampler = AudioResampler::create(
format,
// the resampler sees the number of channels after the downmixer, if any
- downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount,
+ (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
devSampleRate, quality);
resampler->setLocalTimeFreq(sLocalTimeFreq);
}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index ef90952..c14d4b4 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5249,7 +5249,7 @@ void AudioFlinger::RecordThread::readInputParameters()
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) {
- int channelCount;
+ uint32_t channelCount;
// optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
// stereo to mono post process as the resampler always outputs stereo.
if (mChannelCount == 1 && mReqChannelCount == 2) {
@@ -5257,7 +5257,7 @@ void AudioFlinger::RecordThread::readInputParameters()
} else {
channelCount = 2;
}
- mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+ mResampler = AudioResampler::create(16, (int) channelCount, mReqSampleRate);
mResampler->setSampleRate(mSampleRate);
mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2];