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author | andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-09-06 21:15:55 +0000 |
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committer | andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-09-06 21:15:55 +0000 |
commit | 11a8868e8e4803e6375eb1bd2002351786f55ebc (patch) | |
tree | 499c8b0126d25f66b646b54fb2d295c1a6d8d631 | |
parent | ca20f3d4c2d6823908e3b7e23119aeebb2ba7de1 (diff) | |
download | webrtc-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.cc | 14 | ||||
-rw-r--r-- | common_audio/include/audio_util.h | 14 | ||||
-rw-r--r-- | common_audio/resampler/push_sinc_resampler.cc | 10 | ||||
-rw-r--r-- | common_audio/resampler/push_sinc_resampler_unittest.cc | 54 |
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. |