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
diff --git a/webrtc/common_audio/resampler/include/push_resampler.h b/webrtc/common_audio/resampler/include/push_resampler.h
new file mode 100644
index 0000000..0183f91
--- /dev/null
+++ b/webrtc/common_audio/resampler/include/push_resampler.h
@@ -0,0 +1,61 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
+#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
+
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Resampler;
+class PushSincResampler;
+
+// Wraps the old resampler and new arbitrary rate conversion resampler. The
+// old resampler will be used whenever it supports the requested rates, and
+// otherwise the sinc resampler will be enabled.
+class PushResampler {
+ public:
+ PushResampler();
+ virtual ~PushResampler();
+
+ // Must be called whenever the parameters change. Free to be called at any
+ // time as it is a no-op if parameters have not changed since the last call.
+ int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
+ int num_channels);
+
+ // Returns the total number of samples provided in destination (e.g. 32 kHz,
+ // 2 channel audio gives 640 samples).
+ int Resample(const int16_t* src, int src_length, int16_t* dst,
+ int dst_capacity);
+
+ bool use_sinc_resampler() const { return use_sinc_resampler_; }
+
+ private:
+ int ResampleSinc(const int16_t* src, int src_length, int16_t* dst,
+ int dst_capacity);
+
+ scoped_ptr<Resampler> resampler_;
+ scoped_ptr<PushSincResampler> sinc_resampler_;
+ scoped_ptr<PushSincResampler> sinc_resampler_right_;
+ int src_sample_rate_hz_;
+ int dst_sample_rate_hz_;
+ int num_channels_;
+ bool use_sinc_resampler_;
+ scoped_array<int16_t> src_left_;
+ scoped_array<int16_t> src_right_;
+ scoped_array<int16_t> dst_left_;
+ scoped_array<int16_t> dst_right_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_