Index: webrtc/modules/audio_processing/level_controller/down_sampler.cc |
diff --git a/webrtc/modules/audio_processing/level_controller/down_sampler.cc b/webrtc/modules/audio_processing/level_controller/down_sampler.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..e1be7edabc5c796d35a8305d1a48d3bfdfb7b0e7 |
--- /dev/null |
+++ b/webrtc/modules/audio_processing/level_controller/down_sampler.cc |
@@ -0,0 +1,101 @@ |
+/* |
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
+ * |
+ * Use of this source code is governed by a BSD-style license |
+ * that can be found in the LICENSE file in the root of the source |
+ * tree. An additional intellectual property rights grant can be found |
+ * in the file PATENTS. All contributing project authors may |
+ * be found in the AUTHORS file in the root of the source tree. |
+ */ |
+ |
+#include "webrtc/modules/audio_processing/level_controller/down_sampler.h" |
+ |
+#include <string.h> |
+#include <algorithm> |
+#include <vector> |
+ |
+#include "webrtc/base/checks.h" |
+#include "webrtc/modules/audio_processing/include/audio_processing.h" |
+#include "webrtc/modules/audio_processing/level_controller/biquad_filter.h" |
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" |
+ |
+namespace webrtc { |
+namespace { |
+ |
+// Bandlimiter coefficients computed based on that only |
+// the first 40 bins of the spectrum for the downsampled |
+// signal are used. |
+// [B,A] = butter(2,(41/64*4000)/8000) |
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_16kHz = { |
+ {0.1455f, 0.2911f, 0.1455f}, |
+ {-0.6698f, 0.2520f}}; |
+ |
+// [B,A] = butter(2,(41/64*4000)/16000) |
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_32kHz = { |
+ {0.0462f, 0.0924f, 0.0462f}, |
+ {-1.3066f, 0.4915f}}; |
+ |
+// [B,A] = butter(2,(41/64*4000)/24000) |
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_48kHz = { |
+ {0.0226f, 0.0452f, 0.0226f}, |
+ {-1.5320f, 0.6224f}}; |
+ |
+} // namespace |
+ |
+DownSampler::DownSampler(ApmDataDumper* data_dumper) |
+ : data_dumper_(data_dumper) { |
+ Initialize(48000); |
+} |
+void DownSampler::Initialize(int sample_rate_hz) { |
+ RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz || |
+ sample_rate_hz == AudioProcessing::kSampleRate16kHz || |
+ sample_rate_hz == AudioProcessing::kSampleRate32kHz || |
+ sample_rate_hz == AudioProcessing::kSampleRate48kHz); |
+ |
+ sample_rate_hz_ = sample_rate_hz; |
+ down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000); |
+ |
+ /// Note that the down sampling filter is not used if the sample rate is 8 |
+ /// kHz. |
+ if (sample_rate_hz_ == AudioProcessing::kSampleRate16kHz) { |
+ low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz); |
+ } else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) { |
+ low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz); |
+ } else if (sample_rate_hz_ == AudioProcessing::kSampleRate48kHz) { |
+ low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz); |
+ } |
+} |
+ |
+void DownSampler::DownSample(rtc::ArrayView<const float> in, |
+ rtc::ArrayView<float> out) { |
+ data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1); |
+ RTC_DCHECK_EQ(static_cast<size_t>(sample_rate_hz_ * |
+ AudioProcessing::kChunkSizeMs / 1000), |
+ in.size()); |
+ RTC_DCHECK_EQ(static_cast<size_t>(AudioProcessing::kSampleRate8kHz * |
+ AudioProcessing::kChunkSizeMs / 1000), |
+ out.size()); |
+ const size_t kMaxNumFrames = |
+ AudioProcessing::kSampleRate48kHz * AudioProcessing::kChunkSizeMs / 1000; |
+ float x[kMaxNumFrames]; |
+ |
+ // Band-limit the signal to 4 kHz. |
+ if (sample_rate_hz_ != AudioProcessing::kSampleRate8kHz) { |
+ low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size())); |
+ |
+ // Downsample the signal. |
+ size_t k = 0; |
+ for (size_t j = 0; j < out.size(); ++j) { |
+ RTC_DCHECK_GT(kMaxNumFrames, k); |
+ out[j] = x[k]; |
+ k += down_sampling_factor_; |
+ } |
+ } else { |
+ std::copy(in.data(), in.data() + in.size(), out.data()); |
+ } |
+ |
+ data_dumper_->DumpWav("lc_down_sampler_output", out, |
+ AudioProcessing::kSampleRate8kHz, 1); |
+} |
+ |
+} // namespace webrtc |