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| author | bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-06-05 08:42:53 +0000 |
|---|---|---|
| committer | bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-06-05 08:42:53 +0000 |
| commit | edbe886a0ba31422b130feee430e82a4051c2d3d (patch) | |
| tree | 980fce0125d5f816841f35983813c78e6087affe /webrtc | |
| parent | ef92755780253c6a7940c89598a206e58e05b810 (diff) | |
| download | webrtc-edbe886a0ba31422b130feee430e82a4051c2d3d.tar.gz | |
common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND
This macro was only used at two places in fixed point iSAC, where it has been replaced with the operation.
BUG=3348,3353
TESTED=trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6336 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'webrtc')
3 files changed, 4 insertions, 5 deletions
diff --git a/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/webrtc/common_audio/signal_processing/include/signal_processing_library.h index 3db793b00c..ce84ec85fb 100644 --- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h +++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h @@ -99,8 +99,6 @@ #define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \ ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t) \ (((int32_t)1) << ((c) - 1)))) >> (c)) -#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b) \ - ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t) (1 << 14))) >> 15) // C + the 32 most significant bits of A * B #define WEBRTC_SPL_SCALEDIFF32(A, B, C) \ diff --git a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc index 7b1f7fd5e5..90e116f360 100644 --- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc +++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc @@ -68,7 +68,6 @@ TEST_F(SplTest, MacroTest) { EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2)); EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2)); - EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b)); EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b)); EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B)); diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c index 0dc8174399..deba0d5e29 100644 --- a/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c +++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c @@ -834,13 +834,15 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0, /* bandwidth expansion */ for (n = 1; n <= ORDERLO; n++) { - a_LOQ11[n] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecLo[n-1], a_LOQ11[n]); + a_LOQ11[n] = (int16_t) ((WEBRTC_SPL_MUL_16_16( + kPolyVecLo[n-1], a_LOQ11[n]) + ((int32_t) (1 << 14))) >> 15); } polyHI[0] = a_HIQ12[0]; for (n = 1; n <= ORDERHI; n++) { - a_HIQ12[n] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecHi[n-1], a_HIQ12[n]); + a_HIQ12[n] = (int16_t) ((WEBRTC_SPL_MUL_16_16( + kPolyVecHi[n-1], a_HIQ12[n]) + ((int32_t) (1 << 14))) >> 15); polyHI[n] = a_HIQ12[n]; } |