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 MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ 12 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ 13 14 #include <memory> 15 16 #include "absl/types/optional.h" 17 #include "api/array_view.h" 18 #include "api/audio_codecs/audio_decoder.h" 19 #include "modules/audio_coding/neteq/tools/input_audio_file.h" 20 21 namespace webrtc { 22 namespace test { 23 // Provides an AudioDecoder implementation that delivers audio data from a file. 24 // The "encoded" input should contain information about what RTP timestamp the 25 // encoding represents, and how many samples the decoder should produce for that 26 // encoding. A helper method PrepareEncoded is provided to prepare such 27 // encodings. If packets are missing, as determined from the timestamps, the 28 // file reading will skip forward to match the loss. 29 class FakeDecodeFromFile : public AudioDecoder { 30 public: FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input,int sample_rate_hz,bool stereo)31 FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input, 32 int sample_rate_hz, 33 bool stereo) 34 : input_(std::move(input)), 35 sample_rate_hz_(sample_rate_hz), 36 stereo_(stereo) {} 37 38 ~FakeDecodeFromFile() = default; 39 40 std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, 41 uint32_t timestamp) override; 42 Reset()43 void Reset() override {} 44 SampleRateHz()45 int SampleRateHz() const override { return sample_rate_hz_; } 46 Channels()47 size_t Channels() const override { return stereo_ ? 2 : 1; } 48 49 int DecodeInternal(const uint8_t* encoded, 50 size_t encoded_len, 51 int sample_rate_hz, 52 int16_t* decoded, 53 SpeechType* speech_type) override; 54 55 int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override; 56 57 // Helper method. Writes `timestamp`, `samples` and 58 // `original_payload_size_bytes` to `encoded` in a format that the 59 // FakeDecodeFromFile decoder will understand. `encoded` must be at least 12 60 // bytes long. 61 static void PrepareEncoded(uint32_t timestamp, 62 size_t samples, 63 size_t original_payload_size_bytes, 64 rtc::ArrayView<uint8_t> encoded); 65 66 private: 67 std::unique_ptr<InputAudioFile> input_; 68 absl::optional<uint32_t> next_timestamp_from_input_; 69 const int sample_rate_hz_; 70 const bool stereo_; 71 size_t last_decoded_length_ = 0; 72 bool cng_mode_ = false; 73 }; 74 75 } // namespace test 76 } // namespace webrtc 77 #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ 78