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Side by Side Diff: webrtc/common_audio/resampler/include/push_resampler.h

Issue 1316523002: Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Fix compile Created 4 years, 11 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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22 // TODO(ajm): add support for an arbitrary number of channels. 22 // TODO(ajm): add support for an arbitrary number of channels.
23 template <typename T> 23 template <typename T>
24 class PushResampler { 24 class PushResampler {
25 public: 25 public:
26 PushResampler(); 26 PushResampler();
27 virtual ~PushResampler(); 27 virtual ~PushResampler();
28 28
29 // Must be called whenever the parameters change. Free to be called at any 29 // Must be called whenever the parameters change. Free to be called at any
30 // time as it is a no-op if parameters have not changed since the last call. 30 // time as it is a no-op if parameters have not changed since the last call.
31 int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz, 31 int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
32 int num_channels); 32 size_t num_channels);
33 33
34 // Returns the total number of samples provided in destination (e.g. 32 kHz, 34 // Returns the total number of samples provided in destination (e.g. 32 kHz,
35 // 2 channel audio gives 640 samples). 35 // 2 channel audio gives 640 samples).
36 int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity); 36 int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
37 37
38 private: 38 private:
39 rtc::scoped_ptr<PushSincResampler> sinc_resampler_; 39 rtc::scoped_ptr<PushSincResampler> sinc_resampler_;
40 rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_; 40 rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_;
41 int src_sample_rate_hz_; 41 int src_sample_rate_hz_;
42 int dst_sample_rate_hz_; 42 int dst_sample_rate_hz_;
43 int num_channels_; 43 size_t num_channels_;
44 rtc::scoped_ptr<T[]> src_left_; 44 rtc::scoped_ptr<T[]> src_left_;
45 rtc::scoped_ptr<T[]> src_right_; 45 rtc::scoped_ptr<T[]> src_right_;
46 rtc::scoped_ptr<T[]> dst_left_; 46 rtc::scoped_ptr<T[]> dst_left_;
47 rtc::scoped_ptr<T[]> dst_right_; 47 rtc::scoped_ptr<T[]> dst_right_;
48 }; 48 };
49 49
50 } // namespace webrtc 50 } // namespace webrtc
51 51
52 #endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ 52 #endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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