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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
73 lines
2.6 KiB
C++
73 lines
2.6 KiB
C++
/*
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* Copyright (c) 2014 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 COMMON_AUDIO_AUDIO_CONVERTER_H_
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#define COMMON_AUDIO_AUDIO_CONVERTER_H_
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#include <stddef.h>
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#include <memory>
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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// Format conversion (remixing and resampling) for audio. Only simple remixing
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// conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
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// upmix from mono (i.e. |src_channels == 1|).
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//
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// The source and destination chunks have the same duration in time; specifying
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// the number of frames is equivalent to specifying the sample rates.
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class AudioConverter {
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public:
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// Returns a new AudioConverter, which will use the supplied format for its
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// lifetime. Caller is responsible for the memory.
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static std::unique_ptr<AudioConverter> Create(size_t src_channels,
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size_t src_frames,
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size_t dst_channels,
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size_t dst_frames);
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virtual ~AudioConverter() {}
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// Convert |src|, containing |src_size| samples, to |dst|, having a sample
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// capacity of |dst_capacity|. Both point to a series of buffers containing
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// the samples for each channel. The sizes must correspond to the format
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// passed to Create().
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virtual void Convert(const float* const* src,
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size_t src_size,
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float* const* dst,
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size_t dst_capacity) = 0;
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size_t src_channels() const { return src_channels_; }
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size_t src_frames() const { return src_frames_; }
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size_t dst_channels() const { return dst_channels_; }
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size_t dst_frames() const { return dst_frames_; }
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protected:
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AudioConverter();
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AudioConverter(size_t src_channels,
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size_t src_frames,
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size_t dst_channels,
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size_t dst_frames);
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// Helper to RTC_CHECK that inputs are correctly sized.
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void CheckSizes(size_t src_size, size_t dst_capacity) const;
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private:
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const size_t src_channels_;
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const size_t src_frames_;
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const size_t dst_channels_;
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const size_t dst_frames_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_AUDIO_CONVERTER_H_
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