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| 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |
| 12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |
| 13 |
| 14 #include <memory> |
| 15 |
| 16 #include "webrtc/base/array_view.h" |
| 17 #include "webrtc/base/optional.h" |
| 18 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
| 19 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" |
| 20 |
| 21 namespace webrtc { |
| 22 namespace test { |
| 23 |
| 24 // Provides an AudioDecoder implementation that delivers audio data from a file. |
| 25 // The "encoded" input should contain information about what RTP timestamp the |
| 26 // encoding represents, and how many samples the decoder should produce for that |
| 27 // encoding. A helper method PrepareEncoded is provided to prepare such |
| 28 // encodings. If packets are missing, as determined from the timestamps, the |
| 29 // file reading will skip forward to match the loss. |
| 30 class FakeDecodeFromFile : public AudioDecoder { |
| 31 public: |
| 32 FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input, |
| 33 int sample_rate_hz, |
| 34 bool stereo) |
| 35 : input_(std::move(input)), |
| 36 sample_rate_hz_(sample_rate_hz), |
| 37 stereo_(stereo) {} |
| 38 |
| 39 ~FakeDecodeFromFile() = default; |
| 40 |
| 41 void Reset() override {} |
| 42 |
| 43 int SampleRateHz() const override { return sample_rate_hz_; } |
| 44 |
| 45 size_t Channels() const override { return stereo_ ? 2 : 1; } |
| 46 |
| 47 int DecodeInternal(const uint8_t* encoded, |
| 48 size_t encoded_len, |
| 49 int sample_rate_hz, |
| 50 int16_t* decoded, |
| 51 SpeechType* speech_type) override; |
| 52 |
| 53 // Helper method. Writes |timestamp| and |samples| to |encoded| in a format |
| 54 // that the FakeDecpdeFromFile decoder will understand. |encoded| must be at |
| 55 // least 8 bytes long. |
| 56 static void PrepareEncoded(uint32_t timestamp, |
| 57 size_t samples, |
| 58 rtc::ArrayView<uint8_t> encoded); |
| 59 |
| 60 private: |
| 61 std::unique_ptr<InputAudioFile> input_; |
| 62 rtc::Optional<uint32_t> next_timestamp_from_input_; |
| 63 const int sample_rate_hz_; |
| 64 const bool stereo_; |
| 65 }; |
| 66 |
| 67 } // namespace test |
| 68 } // namespace webrtc |
| 69 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ |
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