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Resampler modifications in preparation for arbitrary audioproc rates.

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- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.

This is a prerequisite of:
http://review.webrtc.org/9919004/

BUG=2894
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2014-04-19 00:32:07 +00:00
parent 3d9ec1fed4
commit f5a33f145b
12 changed files with 52 additions and 48 deletions
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12
webrtc/voice_engine/utility_unittest.cc
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@ -39,7 +39,7 @@ class UtilityTest : public ::testing::Test {
int dst_channels, int dst_sample_rate_hz,
FunctionToTest function);
PushResampler resampler_;
PushResampler<int16_t> resampler_;
AudioFrame src_frame_;
AudioFrame dst_frame_;
AudioFrame golden_frame_;
@ -127,11 +127,11 @@ void VerifyFramesAreEqual(const AudioFrame& ref_frame,
}
void UtilityTest::RunResampleTest(int src_channels,
int src_sample_rate_hz,
int dst_channels,
int dst_sample_rate_hz,
FunctionToTest function) {
PushResampler resampler; // Create a new one with every test.
int src_sample_rate_hz,
int dst_channels,
int dst_sample_rate_hz,
FunctionToTest function) {
PushResampler<int16_t> resampler; // Create a new one with every test.
const int16_t kSrcLeft = 30; // Shouldn't overflow for any used sample rate.
const int16_t kSrcRight = 15;
const float resampling_factor = (1.0 * src_sample_rate_hz) /