Improve dynamic audio resampler filter generation

Improve dynamic audio resampler filter generation speed by 2x.
The resulting filters should be the same (excepting roundoff).

Also added check for upsampling sample rate changes to share
previously generated filters.

Modify the profiling to improve output format and sampling result
reliability.

Change-Id: I9aa6b914fd552a63f79dd4a95945df2f8275772a
Signed-off-by: Andy Hung <hunga@google.com>

1 files changed, 118 insertions, 21 deletions

diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 3f2ce55..66fcd90 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -57,7 +57,8 @@ static int usage(const char* name) {
 int main(int argc, char* argv[]) {
 
     const char* const progname = argv[0];
-    bool profiling = false;
+    bool profileResample = false;
+    bool profileFilter = false;
     bool writeHeader = false;
     int channels = 1;
     int input_freq = 0;
@@ -65,10 +66,13 @@ int main(int argc, char* argv[]) {
     AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
 
     int ch;
-    while ((ch = getopt(argc, argv, "phvsq:i:o:")) != -1) {
+    while ((ch = getopt(argc, argv, "pfhvsq:i:o:")) != -1) {
         switch (ch) {
         case 'p':
-            profiling = true;
+            profileResample = true;
+            break;
+        case 'f':
+            profileFilter = true;
             break;
         case 'h':
             writeHeader = true;
@@ -230,27 +234,108 @@ int main(int argc, char* argv[]) {
     size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq;
     output_size &= ~7; // always stereo, 32-bits
 
-    void* output_vaddr = malloc(output_size);
-    AudioResampler* resampler = AudioResampler::create(16, channels,
-            output_freq, quality);
-    size_t out_frames = output_size/8;
-    resampler->setSampleRate(input_freq);
-    resampler->setVolume(0x1000, 0x1000);
-
-    if (profiling) {
-        const int looplimit = 100;
+    if (profileFilter) {
+        // Check how fast sample rate changes are that require filter changes.
+        // The delta sample rate changes must indicate a downsampling ratio,
+        // and must be larger than 10% changes.
+        //
+        // On fast devices, filters should be generated between 0.1ms - 1ms.
+        // (single threaded).
+        AudioResampler* resampler = AudioResampler::create(16, channels,
+                8000, quality);
+        int looplimit = 100;
         timespec start, end;
         clock_gettime(CLOCK_MONOTONIC, &start);
         for (int i = 0; i < looplimit; ++i) {
-            resampler->resample((int*) output_vaddr, out_frames, &provider);
-            provider.reset(); //  reset only provider as benchmarking
+            resampler->setSampleRate(9000);
+            resampler->setSampleRate(12000);
+            resampler->setSampleRate(20000);
+            resampler->setSampleRate(30000);
         }
         clock_gettime(CLOCK_MONOTONIC, &end);
         int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
         int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
         int64_t time = end_ns - start_ns;
-        printf("time(ns):%lld  channels:%d  quality:%d\n", time, channels, quality);
-        printf("%f Mspl/s\n", out_frames * looplimit / (time / 1e9) / 1e6);
+        printf("%.2f sample rate changes with filter calculation/sec\n",
+                looplimit * 4 / (time / 1e9));
+
+        // Check how fast sample rate changes are without filter changes.
+        // This should be very fast, probably 0.1us - 1us per sample rate
+        // change.
+        resampler->setSampleRate(1000);
+        looplimit = 1000;
+        clock_gettime(CLOCK_MONOTONIC, &start);
+        for (int i = 0; i < looplimit; ++i) {
+            resampler->setSampleRate(1000+i);
+        }
+        clock_gettime(CLOCK_MONOTONIC, &end);
+        start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+        end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+        time = end_ns - start_ns;
+        printf("%.2f sample rate changes without filter calculation/sec\n",
+                looplimit / (time / 1e9));
+        resampler->reset();
+        delete resampler;
+    }
+
+    void* output_vaddr = malloc(output_size);
+    AudioResampler* resampler = AudioResampler::create(16, channels,
+            output_freq, quality);
+    size_t out_frames = output_size/8;
+
+    /* set volume precision to 12 bits, so the volume scale is 1<<12.
+     * This means the "integer" part fits in the Q19.12 precision
+     * representation of output int32_t.
+     *
+     * Generally 0 < volumePrecision <= 14 (due to the limits of
+     * int16_t values for Volume). volumePrecision cannot be 0 due
+     * to rounding and shifts.
+     */
+    const int volumePrecision = 12; // in bits
+
+    resampler->setSampleRate(input_freq);
+    resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
+
+    if (profileResample) {
+        /*
+         * For profiling on mobile devices, upon experimentation
+         * it is better to run a few trials with a shorter loop limit,
+         * and take the minimum time.
+         *
+         * Long tests can cause CPU temperature to build up and thermal throttling
+         * to reduce CPU frequency.
+         *
+         * For frequency checks (index=0, or 1, etc.):
+         * "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq"
+         *
+         * For temperature checks (index=0, or 1, etc.):
+         * "cat /sys/class/thermal/thermal_zone${index}/temp"
+         *
+         * Another way to avoid thermal throttling is to fix the CPU frequency
+         * at a lower level which prevents excessive temperatures.
+         */
+        const int trials = 4;
+        const int looplimit = 4;
+        timespec start, end;
+        int64_t time;
+
+        for (int n = 0; n < trials; ++n) {
+            clock_gettime(CLOCK_MONOTONIC, &start);
+            for (int i = 0; i < looplimit; ++i) {
+                resampler->resample((int*) output_vaddr, out_frames, &provider);
+                provider.reset(); //  during benchmarking reset only the provider
+            }
+            clock_gettime(CLOCK_MONOTONIC, &end);
+            int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+            int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+            int64_t diff_ns = end_ns - start_ns;
+            if (n == 0 || diff_ns < time) {
+                time = diff_ns;   // save the best out of our trials.
+            }
+        }
+        // Mfrms/s is "Millions of output frames per second".
+        printf("quality: %d  channels: %d  msec: %lld  Mfrms/s: %.2lf\n",
+                quality, channels, time/1000000, out_frames * looplimit / (time / 1e9) / 1e6);
         resampler->reset();
     }
 
@@ -266,19 +351,31 @@ int main(int argc, char* argv[]) {
     if (gVerbose) {
         printf("reset() complete\n");
     }
+    delete resampler;
+    resampler = NULL;
 
     // mono takes left channel only
     // stereo right channel is half amplitude of stereo left channel (due to input creation)
     int32_t* out = (int32_t*) output_vaddr;
     int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t));
 
+    // round to half towards zero and saturate at int16 (non-dithered)
+    const int roundVal = (1<<(volumePrecision-1)) - 1; // volumePrecision > 0
+
     for (size_t i = 0; i < out_frames; i++) {
         for (int j = 0; j < channels; j++) {
-            int32_t s = out[i * 2 + j] >> 12;
-            if (s > 32767)
-                s = 32767;
-            else if (s < -32768)
-                s = -32768;
+            int32_t s = out[i * 2 + j] + roundVal; // add offset here
+            if (s < 0) {
+                s = (s + 1) >> volumePrecision; // round to 0
+                if (s < -32768) {
+                    s = -32768;
+                }
+            } else {
+                s = s >> volumePrecision;
+                if (s > 32767) {
+                    s = 32767;
+                }
+            }
             convert[i * channels + j] = int16_t(s);
         }
     }