88fbb2d86b
Same as https://webrtc-codereview.appspot.com/19519004. The issue in http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux... is solved by this change http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing... (tested locally). BUG=3380 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17619005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
70 lines
2.6 KiB
C++
70 lines
2.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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#define WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/common_audio/resampler/sinc_resampler.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// A thin wrapper over SincResampler to provide a push-based interface as
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// required by WebRTC.
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class PushSincResampler : public SincResamplerCallback {
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public:
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// Provide the size of the source and destination blocks in samples. These
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// must correspond to the same time duration (typically 10 ms) as the sample
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// ratio is inferred from them.
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PushSincResampler(int source_frames, int destination_frames);
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virtual ~PushSincResampler();
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// Perform the resampling. |source_frames| must always equal the
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// |source_frames| provided at construction. |destination_capacity| must be
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// at least as large as |destination_frames|. Returns the number of samples
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// provided in destination (for convenience, since this will always be equal
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// to |destination_frames|).
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int Resample(const int16_t* source, int source_frames,
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int16_t* destination, int destination_capacity);
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int Resample(const float* source,
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int source_frames,
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float* destination,
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int destination_capacity);
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// Implements SincResamplerCallback.
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virtual void Run(int frames, float* destination) OVERRIDE;
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SincResampler* get_resampler_for_testing() { return resampler_.get(); }
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static float AlgorithmicDelaySeconds(int source_rate_hz) {
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return 1.f / source_rate_hz * SincResampler::kKernelSize / 2;
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}
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private:
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scoped_ptr<SincResampler> resampler_;
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scoped_ptr<float[]> float_buffer_;
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const float* source_ptr_;
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const int16_t* source_ptr_int_;
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const int destination_frames_;
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// True on the first call to Resample(), to prime the SincResampler buffer.
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bool first_pass_;
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// Used to assert we are only requested for as much data as is available.
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int source_available_;
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DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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