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Unified Diff: webrtc/common_audio/audio_ring_buffer.cc

Issue 1846903004: Moved ring-buffer related files from common_audio to audio_processing (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 8 months ago
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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
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