Add a wrapper around PushSincResampler and the old Resampler.

The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.

Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.

BUG=webrtc:1395
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/common_audio/resampler/push_resampler.cc

@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2013 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/common_audio/resampler/include/push_resampler.h"
+
+#include <cstring>
+
+#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/common_audio/resampler/include/resampler.h"
+#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
+
+namespace webrtc {
+
+PushResampler::PushResampler()
+      // Requires valid values at construction, so give it something arbitrary.
+    : resampler_(new Resampler(48000, 48000, kResamplerSynchronous)),
+      sinc_resampler_(NULL),
+      sinc_resampler_right_(NULL),
+      src_sample_rate_hz_(0),
+      dst_sample_rate_hz_(0),
+      num_channels_(0),
+      use_sinc_resampler_(false),
+      src_left_(NULL),
+      src_right_(NULL),
+      dst_left_(NULL),
+      dst_right_(NULL) {
+}
+
+PushResampler::~PushResampler() {
+}
+
+int PushResampler::InitializeIfNeeded(int src_sample_rate_hz,
+                                      int dst_sample_rate_hz,
+                                      int num_channels) {
+  if (src_sample_rate_hz == src_sample_rate_hz_ &&
+      dst_sample_rate_hz == dst_sample_rate_hz_ &&
+      num_channels == num_channels_) {
+    // No-op if settings haven't changed.
+    return 0;
+  }
+
+  if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
+      num_channels <= 0 || num_channels > 2) {
+    return -1;
+  }
+
+  src_sample_rate_hz_ = src_sample_rate_hz;
+  dst_sample_rate_hz_ = dst_sample_rate_hz;
+  num_channels_ = num_channels;
+
+  const ResamplerType resampler_type =
+      num_channels == 1 ? kResamplerSynchronous : kResamplerSynchronousStereo;
+  if (resampler_->Reset(src_sample_rate_hz, dst_sample_rate_hz,
+                        resampler_type) == 0) {
+    // The resampler supports these rates.
+    use_sinc_resampler_ = false;
+    return 0;
+  }
+
+  use_sinc_resampler_ = true;
+  const int src_size_10ms_mono = src_sample_rate_hz / 100;
+  const int dst_size_10ms_mono = dst_sample_rate_hz / 100;
+  sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
+                                              dst_size_10ms_mono));
+  if (num_channels_ == 2) {
+    src_left_.reset(new int16_t[src_size_10ms_mono]);
+    src_right_.reset(new int16_t[src_size_10ms_mono]);
+    dst_left_.reset(new int16_t[dst_size_10ms_mono]);
+    dst_right_.reset(new int16_t[dst_size_10ms_mono]);
+    sinc_resampler_right_.reset(new PushSincResampler(src_size_10ms_mono,
+                                                      dst_size_10ms_mono));
+  }
+
+  return 0;
+}
+
+int PushResampler::Resample(const int16_t* src, int src_length,
+                            int16_t* dst, int dst_capacity) {
+  const int src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
+  const int dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
+  if (src_length != src_size_10ms || dst_capacity < dst_size_10ms) {
+    return -1;
+  }
+
+  if (use_sinc_resampler_) {
+    return ResampleSinc(src, src_length, dst, dst_capacity);
+  }
+
+  int resulting_length = 0;
+  if (resampler_->Push(src, src_length, dst, dst_capacity,
+                       resulting_length) != 0) {
+    return -1;
+  }
+  return resulting_length;
+}
+
+int PushResampler::ResampleSinc(const int16_t* src, int src_length,
+                                int16_t* dst, int dst_capacity) {
+  if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
+    // The old resampler provides this memcpy facility in the case of matching
+    // sample rates, so reproduce it here for the sinc resampler.
+    memcpy(dst, src, src_length * sizeof(int16_t));
+    return src_length;
+  }
+  if (num_channels_ == 2) {
+    const int src_length_mono = src_length / num_channels_;
+    const int dst_capacity_mono = dst_capacity / num_channels_;
+    int16_t* deinterleaved[] = {src_left_.get(), src_right_.get()};
+    Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
+
+    int dst_length_mono =
+        sinc_resampler_->Resample(src_left_.get(), src_length_mono,
+                                  dst_left_.get(), dst_capacity_mono);
+    sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
+                                    dst_right_.get(), dst_capacity_mono);
+
+    deinterleaved[0] = dst_left_.get();
+    deinterleaved[1] = dst_right_.get();
+    Interleave(deinterleaved, dst_length_mono, num_channels_, dst);
+    return dst_length_mono * num_channels_;
+  } else {
+    return sinc_resampler_->Resample(src, src_length, dst, dst_capacity);
+  }
+}
+
+}  // namespace webrtc