Issue 3058745: fix audio HAL downsampler.

Latest audio HAL downsampler implementation had a problem that
caused one sample to be dropped if the input buffer size was odd
while downsampling by a factor of 2.
This explains why record timing was correct for 22 and 16kHz but
not for 11 or 8kHz: in the later case, the second stage of resampling
(22 to 11) receives 503 frames for each buffer.

Change-Id: Ib8fcba3ddfbbab50a908e6b0a6cdc2b398acd862

1 files changed, 9 insertions, 6 deletions

diff --git a/libaudio/AudioHardwareALSA.cpp b/libaudio/AudioHardwareALSA.cpp
index 6c8a610..4df3c42 100755
--- a/libaudio/AudioHardwareALSA.cpp
+++ b/libaudio/AudioHardwareALSA.cpp
@@ -2305,13 +2305,16 @@ void resample_2_1(int16_t* input, int16_t* output, int* num_samples_in, int* num
         return;
     }
 
-        for (int i = 0; i < *num_samples_in - (int)OVERLAP_22KHZ; i += 2) {
-                output[i / 2] = clip(fir_convolve(input + i, filter_22khz_coeff, NUM_COEFF_22KHZ));
-        }
+    int odd_smp = *num_samples_in & 0x1;
+    int num_samples = *num_samples_in - odd_smp - OVERLAP_22KHZ;
+
+    for (int i = 0; i < num_samples; i += 2) {
+            output[i / 2] = clip(fir_convolve(input + i, filter_22khz_coeff, NUM_COEFF_22KHZ));
+    }
 
-    memmove(input, input + *num_samples_in - OVERLAP_22KHZ, OVERLAP_22KHZ * sizeof(*input));
-    *num_samples_out = (*num_samples_in - OVERLAP_22KHZ) / 2;
-    *num_samples_in = OVERLAP_22KHZ;
+    memmove(input, input + num_samples, (OVERLAP_22KHZ + odd_smp) * sizeof(*input));
+    *num_samples_out = num_samples / 2;
+    *num_samples_in = OVERLAP_22KHZ + odd_smp;
 }
 
 /*