Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #include "webrtc/modules/audio_processing/level_controller/down_sampler.h"
      12             : 
      13             : #include <string.h>
      14             : #include <algorithm>
      15             : #include <vector>
      16             : 
      17             : #include "webrtc/base/checks.h"
      18             : #include "webrtc/modules/audio_processing/include/audio_processing.h"
      19             : #include "webrtc/modules/audio_processing/level_controller/biquad_filter.h"
      20             : #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
      21             : 
      22             : namespace webrtc {
      23             : namespace {
      24             : 
      25             : // Bandlimiter coefficients computed based on that only
      26             : // the first 40 bins of the spectrum for the downsampled
      27             : // signal are used.
      28             : // [B,A] = butter(2,(41/64*4000)/8000)
      29             : const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_16kHz = {
      30             :     {0.1455f, 0.2911f, 0.1455f},
      31             :     {-0.6698f, 0.2520f}};
      32             : 
      33             : // [B,A] = butter(2,(41/64*4000)/16000)
      34             : const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_32kHz = {
      35             :     {0.0462f, 0.0924f, 0.0462f},
      36             :     {-1.3066f, 0.4915f}};
      37             : 
      38             : // [B,A] = butter(2,(41/64*4000)/24000)
      39             : const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_48kHz = {
      40             :     {0.0226f, 0.0452f, 0.0226f},
      41             :     {-1.5320f, 0.6224f}};
      42             : 
      43             : }  // namespace
      44             : 
      45           0 : DownSampler::DownSampler(ApmDataDumper* data_dumper)
      46           0 :     : data_dumper_(data_dumper) {
      47           0 :   Initialize(48000);
      48           0 : }
      49           0 : void DownSampler::Initialize(int sample_rate_hz) {
      50           0 :   RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
      51             :              sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
      52             :              sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
      53           0 :              sample_rate_hz == AudioProcessing::kSampleRate48kHz);
      54             : 
      55           0 :   sample_rate_hz_ = sample_rate_hz;
      56           0 :   down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000);
      57             : 
      58             :   /// Note that the down sampling filter is not used if the sample rate is 8
      59             :   /// kHz.
      60           0 :   if (sample_rate_hz_ == AudioProcessing::kSampleRate16kHz) {
      61           0 :     low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
      62           0 :   } else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) {
      63           0 :     low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
      64           0 :   } else if (sample_rate_hz_ == AudioProcessing::kSampleRate48kHz) {
      65           0 :     low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz);
      66             :   }
      67           0 : }
      68             : 
      69           0 : void DownSampler::DownSample(rtc::ArrayView<const float> in,
      70             :                              rtc::ArrayView<float> out) {
      71           0 :   data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1);
      72           0 :   RTC_DCHECK_EQ(sample_rate_hz_ * AudioProcessing::kChunkSizeMs / 1000,
      73           0 :                 in.size());
      74           0 :   RTC_DCHECK_EQ(
      75             :       AudioProcessing::kSampleRate8kHz * AudioProcessing::kChunkSizeMs / 1000,
      76           0 :       out.size());
      77             :   const size_t kMaxNumFrames =
      78           0 :       AudioProcessing::kSampleRate48kHz * AudioProcessing::kChunkSizeMs / 1000;
      79             :   float x[kMaxNumFrames];
      80             : 
      81             :   // Band-limit the signal to 4 kHz.
      82           0 :   if (sample_rate_hz_ != AudioProcessing::kSampleRate8kHz) {
      83           0 :     low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size()));
      84             : 
      85             :     // Downsample the signal.
      86           0 :     size_t k = 0;
      87           0 :     for (size_t j = 0; j < out.size(); ++j) {
      88           0 :       RTC_DCHECK_GT(kMaxNumFrames, k);
      89           0 :       out[j] = x[k];
      90           0 :       k += down_sampling_factor_;
      91             :     }
      92             :   } else {
      93           0 :     std::copy(in.data(), in.data() + in.size(), out.data());
      94             :   }
      95             : 
      96           0 :   data_dumper_->DumpWav("lc_down_sampler_output", out,
      97           0 :                         AudioProcessing::kSampleRate8kHz, 1);
      98           0 : }
      99             : 
     100             : }  // namespace webrtc