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authorandrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-09-06 21:15:55 +0000
committerandrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-09-06 21:15:55 +0000
commit11a8868e8e4803e6375eb1bd2002351786f55ebc (patch)
tree499c8b0126d25f66b646b54fb2d295c1a6d8d631
parentca20f3d4c2d6823908e3b7e23119aeebb2ba7de1 (diff)
downloadwebrtc-11a8868e8e4803e6375eb1bd2002351786f55ebc.tar.gz
Reduce cost of PushSincResampler::Resample().
Ideally, PushSincResampler would have very little overhead on SincResampler. This gets closer to that ideal. Replace std::min/max and floor with inline functions. Add a benchmark test to verify the improvement. On a MacBook Retina, this results in PushSincResampler::Resample() accounting for ~1% of CPU usage on voe_cmd_test vs the earlier ~2% (with ISAC16 and 48 kHz audio devices). Using the new benchmark, this results in a performance improvement of: 16 -> 44.1 : 1.7x 16 -> 48 : 1.9x 32 -> 44.1 : 1.6x 32 -> 48 : 1.7x 44.1 -> 16 : 1.5x 44.1 -> 32 : 1.7x 44.1 -> 48 : 1.7x 48 -> 16 : 1.5x 48 -> 32 : 1.5x 48 -> 44.1 : 1.8x R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2157005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4695 4adac7df-926f-26a2-2b94-8c16560cd09d
-rw-r--r--common_audio/audio_util_unittest.cc14
-rw-r--r--common_audio/include/audio_util.h14
-rw-r--r--common_audio/resampler/push_sinc_resampler.cc10
-rw-r--r--common_audio/resampler/push_sinc_resampler_unittest.cc54
4 files changed, 85 insertions, 7 deletions
diff --git a/common_audio/audio_util_unittest.cc b/common_audio/audio_util_unittest.cc
index 9ffed73b..c724e378 100644
--- a/common_audio/audio_util_unittest.cc
+++ b/common_audio/audio_util_unittest.cc
@@ -20,6 +20,20 @@ void ExpectArraysEq(const int16_t* ref, const int16_t* test, int length) {
}
}
+TEST(AudioUtilTest, Clamp) {
+ EXPECT_EQ(1000.f, ClampInt16(1000.f));
+ EXPECT_EQ(32767.f, ClampInt16(32767.5f));
+ EXPECT_EQ(-32768.f, ClampInt16(-32768.5f));
+}
+
+TEST(AudioUtilTest, Round) {
+ EXPECT_EQ(0, RoundToInt16(0.f));
+ EXPECT_EQ(0, RoundToInt16(0.4f));
+ EXPECT_EQ(1, RoundToInt16(0.5f));
+ EXPECT_EQ(0, RoundToInt16(-0.4f));
+ EXPECT_EQ(-1, RoundToInt16(-0.5f));
+}
+
TEST(AudioUtilTest, InterleavingStereo) {
const int16_t kInterleaved[] = {2, 3, 4, 9, 8, 27, 16, 81};
const int kSamplesPerChannel = 4;
diff --git a/common_audio/include/audio_util.h b/common_audio/include/audio_util.h
index 2196fc34..5e86c1f0 100644
--- a/common_audio/include/audio_util.h
+++ b/common_audio/include/audio_util.h
@@ -15,6 +15,20 @@
namespace webrtc {
+// Clamp the floating |value| to the range representable by an int16_t.
+static inline float ClampInt16(float value) {
+ const float kMaxInt16 = 32767.f;
+ const float kMinInt16 = -32768.f;
+ return value < kMinInt16 ? kMinInt16 :
+ (value > kMaxInt16 ? kMaxInt16 : value);
+}
+
+// Return a rounded int16_t of the floating |value|. Doesn't handle overflow;
+// use ClampInt16 if necessary.
+static inline int16_t RoundToInt16(float value) {
+ return static_cast<int16_t>(value < 0.f ? value - 0.5f : value + 0.5f);
+}
+
// Deinterleave audio from |interleaved| to the channel buffers pointed to
// by |deinterleaved|. There must be sufficient space allocated in the
// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
diff --git a/common_audio/resampler/push_sinc_resampler.cc b/common_audio/resampler/push_sinc_resampler.cc
index 97febbd4..f4f8e131 100644
--- a/common_audio/resampler/push_sinc_resampler.cc
+++ b/common_audio/resampler/push_sinc_resampler.cc
@@ -8,13 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
-#include <math.h>
#include <string.h>
-#include <algorithm>
-
namespace webrtc {
PushSincResampler::PushSincResampler(int source_frames,
@@ -61,10 +59,8 @@ int PushSincResampler::Resample(const int16_t* source,
resampler_->Resample(resampler_->ChunkSize(), float_buffer_.get());
resampler_->Resample(destination_frames_, float_buffer_.get());
- for (int i = 0; i < destination_frames_; ++i) {
- float clipped = std::max(std::min(float_buffer_[i], 32767.0f), -32768.0f);
- destination[i] = static_cast<int16_t>(floor(clipped + 0.5));
- }
+ for (int i = 0; i < destination_frames_; ++i)
+ destination[i] = RoundToInt16(ClampInt16(float_buffer_[i]));
source_ptr_ = NULL;
return destination_frames_;
}
diff --git a/common_audio/resampler/push_sinc_resampler_unittest.cc b/common_audio/resampler/push_sinc_resampler_unittest.cc
index 28b26ac1..d5005c6d 100644
--- a/common_audio/resampler/push_sinc_resampler_unittest.cc
+++ b/common_audio/resampler/push_sinc_resampler_unittest.cc
@@ -15,6 +15,7 @@
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -39,6 +40,59 @@ class PushSincResamplerTest
double low_freq_error_;
};
+class ZeroSource : public SincResamplerCallback {
+ public:
+ void Run(int frames, float* destination) {
+ memset(destination, 0, sizeof(float) * frames);
+ }
+};
+
+// Disabled because it takes too long to run routinely. Use for performance
+// benchmarking when needed.
+TEST_P(PushSincResamplerTest, DISABLED_ResampleBenchmark) {
+ const int input_samples = input_rate_ / 100;
+ const int output_samples = output_rate_ / 100;
+ const int kResampleIterations = 200000;
+
+ // Source for data to be resampled.
+ ZeroSource resampler_source;
+
+ scoped_array<float> resampled_destination(new float[output_samples]);
+ scoped_array<float> source(new float[input_samples]);
+ scoped_array<int16_t> source_int(new int16_t[input_samples]);
+ scoped_array<int16_t> destination_int(new int16_t[output_samples]);
+
+ resampler_source.Run(input_samples, source.get());
+ for (int i = 0; i < input_samples; ++i) {
+ source_int[i] = static_cast<int16_t>(floor(32767 * source[i] + 0.5));
+ }
+
+ printf("Benchmarking %d iterations of %d Hz -> %d Hz:\n",
+ kResampleIterations, input_rate_, output_rate_);
+ const double io_ratio = input_rate_ / static_cast<double>(output_rate_);
+ SincResampler sinc_resampler(io_ratio, SincResampler::kDefaultRequestSize,
+ &resampler_source);
+ TickTime start = TickTime::Now();
+ for (int i = 0; i < kResampleIterations; ++i) {
+ sinc_resampler.Resample(output_samples, resampled_destination.get());
+ }
+ double total_time_sinc_us = (TickTime::Now() - start).Microseconds();
+ printf("SincResampler took %.2f us per frame.\n",
+ total_time_sinc_us / kResampleIterations);
+
+ PushSincResampler resampler(input_samples, output_samples);
+ start = TickTime::Now();
+ for (int i = 0; i < kResampleIterations; ++i) {
+ EXPECT_EQ(output_samples,
+ resampler.Resample(source_int.get(), input_samples,
+ destination_int.get(), output_samples));
+ }
+ double total_time_us = (TickTime::Now() - start).Microseconds();
+ printf("PushSincResampler took %.2f us per frame; which is a %.1f%% overhead "
+ "on SincResampler.\n\n", total_time_us / kResampleIterations,
+ (total_time_us - total_time_sinc_us) / total_time_sinc_us * 100);
+}
+
// Tests resampling using a given input and output sample rate.
TEST_P(PushSincResamplerTest, Resample) {
// Make comparisons using one second of data.