chromium/third_party/webrtc/test/mock_audio_encoder.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 "test/mock_audio_encoder.h"

namespace webrtc {

MockAudioEncoder::MockAudioEncoder() = default;
MockAudioEncoder::~MockAudioEncoder() = default;

MockAudioEncoder::FakeEncoding::FakeEncoding(
    const AudioEncoder::EncodedInfo& info)
    : … { … }

MockAudioEncoder::FakeEncoding::FakeEncoding(size_t encoded_bytes) { … }

AudioEncoder::EncodedInfo MockAudioEncoder::FakeEncoding::operator()(
    uint32_t timestamp,
    rtc::ArrayView<const int16_t> audio,
    rtc::Buffer* encoded) { … }

MockAudioEncoder::CopyEncoding::~CopyEncoding() = default;

MockAudioEncoder::CopyEncoding::CopyEncoding(
    AudioEncoder::EncodedInfo info,
    rtc::ArrayView<const uint8_t> payload)
    : … { … }

MockAudioEncoder::CopyEncoding::CopyEncoding(
    rtc::ArrayView<const uint8_t> payload)
    : … { … }

AudioEncoder::EncodedInfo MockAudioEncoder::CopyEncoding::operator()(
    uint32_t timestamp,
    rtc::ArrayView<const int16_t> audio,
    rtc::Buffer* encoded) { … }

}  // namespace webrtc