webrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
Alessio Bazzica d4161a3c9d Moving LappedTransform, Blocker and AudioRingBuffer.
LappedTransform is only used in BandwidthAdaptationTest and therefore it
should not be anymore a visible target under common_audio.
This CL moves LappedTransform and other two classes it depends on (and which
are not used elsewhere) to modules/audio_coding/codecs/opus/test.

Bug: webrtc:9577, webrtc:5298
Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a
Reviewed-on: https://webrtc-review.googlesource.com/96440
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24509}
2018-08-31 15:27:50 +00:00

57 lines
2 KiB
C++

/*
* Copyright (c) 2015 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
#include <stddef.h>
#include <memory>
#include <vector>
struct RingBuffer;
namespace webrtc {
// A ring buffer tailored for float deinterleaved audio. Any operation that
// cannot be performed as requested will cause a crash (e.g. insufficient data
// in the buffer to fulfill a read request.)
class AudioRingBuffer final {
public:
// Specify the number of channels and maximum number of frames the buffer will
// contain.
AudioRingBuffer(size_t channels, size_t max_frames);
~AudioRingBuffer();
// Copies |data| to the buffer and advances the write pointer. |channels| must
// be the same as at creation time.
void Write(const float* const* data, size_t channels, size_t frames);
// Copies from the buffer to |data| and advances the read pointer. |channels|
// must be the same as at creation time.
void Read(float* const* data, size_t channels, size_t frames);
size_t ReadFramesAvailable() const;
size_t WriteFramesAvailable() const;
// Moves the read position. The forward version advances the read pointer
// towards the write pointer and the backward verison withdraws the read
// pointer away from the write pointer (i.e. flushing and stuffing the buffer
// respectively.)
void MoveReadPositionForward(size_t frames);
void MoveReadPositionBackward(size_t frames);
private:
// TODO(kwiberg): Use std::vector<std::unique_ptr<RingBuffer>> instead.
std::vector<RingBuffer*> buffers_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_