From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../codecs/opus/test/audio_ring_buffer.cc | 76 ++++++++++++++++++++++ 1 file changed, 76 insertions(+) create mode 100644 third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc') diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc new file mode 100644 index 0000000000..2a71b43d2c --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc @@ -0,0 +1,76 @@ +/* + * 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 "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h" + +#include "common_audio/ring_buffer.h" +#include "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(WebRtc_MoveReadPtr(buf, static_cast(frames))); + RTC_CHECK_EQ(moved, frames); + } +} + +void AudioRingBuffer::MoveReadPositionBackward(size_t frames) { + for (auto* buf : buffers_) { + const size_t moved = static_cast( + -WebRtc_MoveReadPtr(buf, -static_cast(frames))); + RTC_CHECK_EQ(moved, frames); + } +} + +} // namespace webrtc -- cgit v1.2.3