chromium/third_party/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc

/*
 *  Copyright (c) 2016 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/neteq/tools/fake_decode_from_file.h"

#include "modules/rtp_rtcp/source/byte_io.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"

namespace webrtc {
namespace test {

namespace {

class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame { … };

}  // namespace

void FakeDecodeFromFile::ReadFromFile(uint32_t timestamp,
                                      size_t samples,
                                      int16_t* destination) { … }

int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
                                       size_t encoded_len,
                                       int sample_rate_hz,
                                       int16_t* decoded,
                                       SpeechType* speech_type) { … }

void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
                                        size_t samples,
                                        size_t original_payload_size_bytes,
                                        rtc::ArrayView<uint8_t> encoded) { … }

std::vector<AudioDecoder::ParseResult> FakeDecodeFromFile::ParsePayload(
    rtc::Buffer&& payload,
    uint32_t timestamp) { … }

}  // namespace test
}  // namespace webrtc