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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc | 79 |
1 files changed, 79 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc b/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc new file mode 100644 index 0000000000..75574961b0 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc @@ -0,0 +1,79 @@ +/* + * 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_processing/test/protobuf_utils.h" + +#include <memory> + +#include "rtc_base/system/arch.h" + +namespace { +// Allocates new memory in the memory owned by the unique_ptr to fit the raw +// message and returns the number of bytes read when having a string stream as +// input. +size_t ReadMessageBytesFromString(std::stringstream* input, + std::unique_ptr<uint8_t[]>* bytes) { + int32_t size = 0; + input->read(reinterpret_cast<char*>(&size), sizeof(int32_t)); + int32_t size_read = input->gcount(); + if (size_read != sizeof(int32_t)) + return 0; + if (size <= 0) + return 0; + + *bytes = std::make_unique<uint8_t[]>(size); + input->read(reinterpret_cast<char*>(bytes->get()), + size * sizeof((*bytes)[0])); + size_read = input->gcount(); + return size_read == size ? size : 0; +} +} // namespace + +namespace webrtc { + +size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes) { +// The "wire format" for the size is little-endian. Assume we're running on +// a little-endian machine. +#ifndef WEBRTC_ARCH_LITTLE_ENDIAN +#error "Need to convert messsage from little-endian." +#endif + int32_t size = 0; + if (fread(&size, sizeof(size), 1, file) != 1) + return 0; + if (size <= 0) + return 0; + + *bytes = std::make_unique<uint8_t[]>(size); + return fread(bytes->get(), sizeof((*bytes)[0]), size, file); +} + +// Returns true on success, false on error or end-of-file. +bool ReadMessageFromFile(FILE* file, MessageLite* msg) { + std::unique_ptr<uint8_t[]> bytes; + size_t size = ReadMessageBytesFromFile(file, &bytes); + if (!size) + return false; + + msg->Clear(); + return msg->ParseFromArray(bytes.get(), size); +} + +// Returns true on success, false on error or end of string stream. +bool ReadMessageFromString(std::stringstream* input, MessageLite* msg) { + std::unique_ptr<uint8_t[]> bytes; + size_t size = ReadMessageBytesFromString(input, &bytes); + if (!size) + return false; + + msg->Clear(); + return msg->ParseFromArray(bytes.get(), size); +} + +} // namespace webrtc |