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Side by Side Diff: webrtc/modules/audio_processing/audio_processing_impl.h

Issue 1316523002: Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Rebase onto cleanup change Created 4 years, 11 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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94 int ProcessReverseStream(const float* const* src, 94 int ProcessReverseStream(const float* const* src,
95 const StreamConfig& reverse_input_config, 95 const StreamConfig& reverse_input_config,
96 const StreamConfig& reverse_output_config, 96 const StreamConfig& reverse_output_config,
97 float* const* dest) override; 97 float* const* dest) override;
98 98
99 // Methods only accessed from APM submodules or 99 // Methods only accessed from APM submodules or
100 // from AudioProcessing tests in a single-threaded manner. 100 // from AudioProcessing tests in a single-threaded manner.
101 // Hence there is no need for locks in these. 101 // Hence there is no need for locks in these.
102 int proc_sample_rate_hz() const override; 102 int proc_sample_rate_hz() const override;
103 int proc_split_sample_rate_hz() const override; 103 int proc_split_sample_rate_hz() const override;
104 int num_input_channels() const override; 104 size_t num_input_channels() const override;
105 int num_output_channels() const override; 105 size_t num_output_channels() const override;
106 int num_reverse_channels() const override; 106 size_t num_reverse_channels() const override;
107 int stream_delay_ms() const override; 107 int stream_delay_ms() const override;
108 bool was_stream_delay_set() const override 108 bool was_stream_delay_set() const override
109 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); 109 EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
110 110
111 // Methods returning pointers to APM submodules. 111 // Methods returning pointers to APM submodules.
112 // No locks are aquired in those, as those locks 112 // No locks are aquired in those, as those locks
113 // would offer no protection (the submodules are 113 // would offer no protection (the submodules are
114 // created only once in a single-treaded manner 114 // created only once in a single-treaded manner
115 // during APM creation). 115 // during APM creation).
116 EchoCancellation* echo_cancellation() const override; 116 EchoCancellation* echo_cancellation() const override;
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332 332
333 struct ApmRenderState { 333 struct ApmRenderState {
334 rtc::scoped_ptr<AudioConverter> render_converter; 334 rtc::scoped_ptr<AudioConverter> render_converter;
335 rtc::scoped_ptr<AudioBuffer> render_audio; 335 rtc::scoped_ptr<AudioBuffer> render_audio;
336 } render_ GUARDED_BY(crit_render_); 336 } render_ GUARDED_BY(crit_render_);
337 }; 337 };
338 338
339 } // namespace webrtc 339 } // namespace webrtc
340 340
341 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ 341 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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