| /* | 
 |  *  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" | 
 |  | 
 | #include <cstddef> | 
 | #include <cstdint> | 
 |  | 
 | #include "api/array_view.h" | 
 | #include "api/audio_codecs/audio_encoder.h" | 
 | #include "rtc_base/buffer.h" | 
 | #include "rtc_base/checks.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | MockAudioEncoder::MockAudioEncoder() = default; | 
 | MockAudioEncoder::~MockAudioEncoder() = default; | 
 |  | 
 | MockAudioEncoder::FakeEncoding::FakeEncoding( | 
 |     const AudioEncoder::EncodedInfo& info) | 
 |     : info_(info) {} | 
 |  | 
 | MockAudioEncoder::FakeEncoding::FakeEncoding(size_t encoded_bytes) { | 
 |   info_.encoded_bytes = encoded_bytes; | 
 | } | 
 |  | 
 | AudioEncoder::EncodedInfo MockAudioEncoder::FakeEncoding::operator()( | 
 |     uint32_t timestamp, | 
 |     ArrayView<const int16_t> audio, | 
 |     Buffer* encoded) { | 
 |   encoded->SetSize(encoded->size() + info_.encoded_bytes); | 
 |   return info_; | 
 | } | 
 |  | 
 | MockAudioEncoder::CopyEncoding::~CopyEncoding() = default; | 
 |  | 
 | MockAudioEncoder::CopyEncoding::CopyEncoding(AudioEncoder::EncodedInfo info, | 
 |                                              ArrayView<const uint8_t> payload) | 
 |     : info_(info), payload_(payload) {} | 
 |  | 
 | MockAudioEncoder::CopyEncoding::CopyEncoding(ArrayView<const uint8_t> payload) | 
 |     : payload_(payload) { | 
 |   info_.encoded_bytes = payload_.size(); | 
 | } | 
 |  | 
 | AudioEncoder::EncodedInfo MockAudioEncoder::CopyEncoding::operator()( | 
 |     uint32_t timestamp, | 
 |     ArrayView<const int16_t> audio, | 
 |     Buffer* encoded) { | 
 |   RTC_CHECK(encoded); | 
 |   RTC_CHECK_LE(info_.encoded_bytes, payload_.size()); | 
 |   encoded->AppendData(payload_.data(), info_.encoded_bytes); | 
 |   return info_; | 
 | } | 
 |  | 
 | }  // namespace webrtc |