| /* |
| * 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 "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h" |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/safe_conversions.h" |
| #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| if (encoded_len == 0) { |
| // Decoder is asked to produce codec-internal comfort noise. |
| RTC_DCHECK(!encoded); // NetEq always sends nullptr in this case. |
| RTC_DCHECK(cng_mode_); |
| RTC_DCHECK_GT(last_decoded_length_, 0u); |
| std::fill_n(decoded, last_decoded_length_, 0); |
| *speech_type = kComfortNoise; |
| return last_decoded_length_; |
| } |
| |
| RTC_CHECK_GE(encoded_len, 12u); |
| uint32_t timestamp_to_decode = |
| ByteReader<uint32_t>::ReadLittleEndian(encoded); |
| uint32_t samples_to_decode = |
| ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]); |
| if (samples_to_decode == 0) { |
| // Number of samples in packet is unknown. |
| if (last_decoded_length_ > 0) { |
| // Use length of last decoded packet, but since this is the total for all |
| // channels, we have to divide by 2 in the stereo case. |
| samples_to_decode = rtc::CheckedDivExact( |
| last_decoded_length_, static_cast<size_t>(stereo_ ? 2uL : 1uL)); |
| } else { |
| // This is the first packet to decode, and we do not know the length of |
| // it. Set it to 10 ms. |
| samples_to_decode = rtc::CheckedDivExact(sample_rate_hz, 100); |
| } |
| } |
| |
| if (next_timestamp_from_input_ && |
| timestamp_to_decode != *next_timestamp_from_input_) { |
| // A gap in the timestamp sequence is detected. Skip the same number of |
| // samples from the file. |
| uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_; |
| RTC_CHECK(input_->Seek(jump)); |
| } |
| |
| next_timestamp_from_input_ = |
| rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode); |
| |
| uint32_t original_payload_size_bytes = |
| ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]); |
| if (original_payload_size_bytes == 1) { |
| // This is a comfort noise payload. |
| RTC_DCHECK_GT(last_decoded_length_, 0u); |
| std::fill_n(decoded, last_decoded_length_, 0); |
| *speech_type = kComfortNoise; |
| cng_mode_ = true; |
| return last_decoded_length_; |
| } |
| |
| cng_mode_ = false; |
| RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded)); |
| |
| if (stereo_) { |
| InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2, |
| decoded); |
| samples_to_decode *= 2; |
| } |
| |
| *speech_type = kSpeech; |
| return last_decoded_length_ = samples_to_decode; |
| } |
| |
| void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp, |
| size_t samples, |
| size_t original_payload_size_bytes, |
| rtc::ArrayView<uint8_t> encoded) { |
| RTC_CHECK_GE(encoded.size(), 12u); |
| ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp); |
| ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4], |
| rtc::checked_cast<uint32_t>(samples)); |
| ByteWriter<uint32_t>::WriteLittleEndian( |
| &encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes)); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |