| /* |
| * 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. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |
| |
| #include <memory> |
| |
| #include "webrtc/base/array_view.h" |
| #include "webrtc/base/optional.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| // Provides an AudioDecoder implementation that delivers audio data from a file. |
| // The "encoded" input should contain information about what RTP timestamp the |
| // encoding represents, and how many samples the decoder should produce for that |
| // encoding. A helper method PrepareEncoded is provided to prepare such |
| // encodings. If packets are missing, as determined from the timestamps, the |
| // file reading will skip forward to match the loss. |
| class FakeDecodeFromFile : public AudioDecoder { |
| public: |
| FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input, |
| int sample_rate_hz, |
| bool stereo) |
| : input_(std::move(input)), |
| sample_rate_hz_(sample_rate_hz), |
| stereo_(stereo) {} |
| |
| ~FakeDecodeFromFile() = default; |
| |
| void Reset() override {} |
| |
| int SampleRateHz() const override { return sample_rate_hz_; } |
| |
| size_t Channels() const override { return stereo_ ? 2 : 1; } |
| |
| int DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) override; |
| |
| // Helper method. Writes |timestamp|, |samples| and |
| // |original_payload_size_bytes| to |encoded| in a format that the |
| // FakeDecodeFromFile decoder will understand. |encoded| must be at least 12 |
| // bytes long. |
| static void PrepareEncoded(uint32_t timestamp, |
| size_t samples, |
| size_t original_payload_size_bytes, |
| rtc::ArrayView<uint8_t> encoded); |
| |
| private: |
| std::unique_ptr<InputAudioFile> input_; |
| rtc::Optional<uint32_t> next_timestamp_from_input_; |
| const int sample_rate_hz_; |
| const bool stereo_; |
| size_t last_decoded_length_ = 0; |
| bool cng_mode_ = false; |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |