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d4161a3c9d
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}
76 lines
2.4 KiB
C++
76 lines
2.4 KiB
C++
/*
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* Copyright (c) 2015 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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#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
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#include "common_audio/ring_buffer.h"
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#include "rtc_base/checks.h"
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// This is a simple multi-channel wrapper over the ring_buffer.h C interface.
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namespace webrtc {
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AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) {
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buffers_.reserve(channels);
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for (size_t i = 0; i < channels; ++i)
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buffers_.push_back(WebRtc_CreateBuffer(max_frames, sizeof(float)));
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}
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AudioRingBuffer::~AudioRingBuffer() {
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for (auto* buf : buffers_)
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WebRtc_FreeBuffer(buf);
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}
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void AudioRingBuffer::Write(const float* const* data,
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size_t channels,
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size_t frames) {
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RTC_DCHECK_EQ(buffers_.size(), channels);
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for (size_t i = 0; i < channels; ++i) {
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const size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames);
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RTC_CHECK_EQ(written, frames);
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}
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}
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void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) {
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RTC_DCHECK_EQ(buffers_.size(), channels);
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for (size_t i = 0; i < channels; ++i) {
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const size_t read =
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WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames);
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RTC_CHECK_EQ(read, frames);
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}
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}
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size_t AudioRingBuffer::ReadFramesAvailable() const {
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// All buffers have the same amount available.
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return WebRtc_available_read(buffers_[0]);
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}
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size_t AudioRingBuffer::WriteFramesAvailable() const {
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// All buffers have the same amount available.
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return WebRtc_available_write(buffers_[0]);
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}
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void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
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for (auto* buf : buffers_) {
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const size_t moved =
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static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames)));
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RTC_CHECK_EQ(moved, frames);
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}
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}
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void AudioRingBuffer::MoveReadPositionBackward(size_t frames) {
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for (auto* buf : buffers_) {
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const size_t moved = static_cast<size_t>(
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-WebRtc_MoveReadPtr(buf, -static_cast<int>(frames)));
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RTC_CHECK_EQ(moved, frames);
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}
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}
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} // namespace webrtc
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