| Index: webrtc/common_audio/audio_ring_buffer.cc
|
| diff --git a/webrtc/common_audio/audio_ring_buffer.cc b/webrtc/common_audio/audio_ring_buffer.cc
|
| deleted file mode 100644
|
| index a29e53a61c626d7134302caefcc2354e5e0d0adf..0000000000000000000000000000000000000000
|
| --- a/webrtc/common_audio/audio_ring_buffer.cc
|
| +++ /dev/null
|
| @@ -1,75 +0,0 @@
|
| -/*
|
| - * 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.
|
| - */
|
| -
|
| -#include "webrtc/common_audio/audio_ring_buffer.h"
|
| -
|
| -#include "webrtc/base/checks.h"
|
| -#include "webrtc/common_audio/ring_buffer.h"
|
| -
|
| -// This is a simple multi-channel wrapper over the ring_buffer.h C interface.
|
| -
|
| -namespace webrtc {
|
| -
|
| -AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) {
|
| - buffers_.reserve(channels);
|
| - for (size_t i = 0; i < channels; ++i)
|
| - buffers_.push_back(WebRtc_CreateBuffer(max_frames, sizeof(float)));
|
| -}
|
| -
|
| -AudioRingBuffer::~AudioRingBuffer() {
|
| - for (auto buf : buffers_)
|
| - WebRtc_FreeBuffer(buf);
|
| -}
|
| -
|
| -void AudioRingBuffer::Write(const float* const* data, size_t channels,
|
| - size_t frames) {
|
| - RTC_DCHECK_EQ(buffers_.size(), channels);
|
| - for (size_t i = 0; i < channels; ++i) {
|
| - const size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames);
|
| - RTC_CHECK_EQ(written, frames);
|
| - }
|
| -}
|
| -
|
| -void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) {
|
| - RTC_DCHECK_EQ(buffers_.size(), channels);
|
| - for (size_t i = 0; i < channels; ++i) {
|
| - const size_t read =
|
| - WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames);
|
| - RTC_CHECK_EQ(read, frames);
|
| - }
|
| -}
|
| -
|
| -size_t AudioRingBuffer::ReadFramesAvailable() const {
|
| - // All buffers have the same amount available.
|
| - return WebRtc_available_read(buffers_[0]);
|
| -}
|
| -
|
| -size_t AudioRingBuffer::WriteFramesAvailable() const {
|
| - // All buffers have the same amount available.
|
| - return WebRtc_available_write(buffers_[0]);
|
| -}
|
| -
|
| -void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
|
| - for (auto buf : buffers_) {
|
| - const size_t moved =
|
| - static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames)));
|
| - RTC_CHECK_EQ(moved, frames);
|
| - }
|
| -}
|
| -
|
| -void AudioRingBuffer::MoveReadPositionBackward(size_t frames) {
|
| - for (auto buf : buffers_) {
|
| - const size_t moved = static_cast<size_t>(
|
| - -WebRtc_MoveReadPtr(buf, -static_cast<int>(frames)));
|
| - RTC_CHECK_EQ(moved, frames);
|
| - }
|
| -}
|
| -
|
| -} // namespace webrtc
|
|
|