webrtc_m130/webrtc/common_audio/audio_ring_buffer.cc
Edward Lemur c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00

76 lines
2.3 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.
*/
#include "webrtc/common_audio/audio_ring_buffer.h"
#include "webrtc/common_audio/ring_buffer.h"
#include "webrtc/rtc_base/checks.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