blob: 01dad4dfc89699976efd8b5e6925cf5d332725a0 [file] [log] [blame]
* Copyright (c) 2014 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.
#include <memory>
#include "webrtc/base/constructormagic.h"
namespace webrtc {
// Format conversion (remixing and resampling) for audio. Only simple remixing
// conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
// upmix from mono (i.e. |src_channels == 1|).
// The source and destination chunks have the same duration in time; specifying
// the number of frames is equivalent to specifying the sample rates.
class AudioConverter {
// Returns a new AudioConverter, which will use the supplied format for its
// lifetime. Caller is responsible for the memory.
static std::unique_ptr<AudioConverter> Create(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames);
virtual ~AudioConverter() {};
// Convert |src|, containing |src_size| samples, to |dst|, having a sample
// capacity of |dst_capacity|. Both point to a series of buffers containing
// the samples for each channel. The sizes must correspond to the format
// passed to Create().
virtual void Convert(const float* const* src, size_t src_size,
float* const* dst, size_t dst_capacity) = 0;
size_t src_channels() const { return src_channels_; }
size_t src_frames() const { return src_frames_; }
size_t dst_channels() const { return dst_channels_; }
size_t dst_frames() const { return dst_frames_; }
AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
size_t dst_frames);
// Helper to RTC_CHECK that inputs are correctly sized.
void CheckSizes(size_t src_size, size_t dst_capacity) const;
const size_t src_channels_;
const size_t src_frames_;
const size_t dst_channels_;
const size_t dst_frames_;
} // namespace webrtc