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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h" | 11 #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h" |
12 | 12 |
13 #include "webrtc/base/checks.h" | 13 #include "webrtc/base/checks.h" |
14 #include "webrtc/base/safe_conversions.h" | 14 #include "webrtc/base/safe_conversions.h" |
15 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 15 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
16 | 16 |
17 namespace webrtc { | 17 namespace webrtc { |
18 namespace test { | 18 namespace test { |
19 | 19 |
20 int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, | 20 int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, |
21 size_t encoded_len, | 21 size_t encoded_len, |
22 int /*sample_rate_hz*/, | 22 int sample_rate_hz, |
23 int16_t* decoded, | 23 int16_t* decoded, |
24 SpeechType* speech_type) { | 24 SpeechType* speech_type) { |
25 RTC_CHECK_GE(encoded_len, 8u); | 25 if (encoded_len == 0) { |
| 26 // Decoder is asked to produce codec-internal comfort noise. |
| 27 RTC_DCHECK(!encoded); // NetEq always sends nullptr in this case. |
| 28 RTC_DCHECK(cng_mode_); |
| 29 RTC_DCHECK_GT(last_decoded_length_, 0u); |
| 30 std::fill_n(decoded, last_decoded_length_, 0); |
| 31 *speech_type = kComfortNoise; |
| 32 return last_decoded_length_; |
| 33 } |
| 34 |
| 35 RTC_CHECK_GE(encoded_len, 12u); |
26 uint32_t timestamp_to_decode = | 36 uint32_t timestamp_to_decode = |
27 ByteReader<uint32_t>::ReadLittleEndian(encoded); | 37 ByteReader<uint32_t>::ReadLittleEndian(encoded); |
28 uint32_t samples_to_decode = | 38 uint32_t samples_to_decode = |
29 ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]); | 39 ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]); |
| 40 if (samples_to_decode == 0) { |
| 41 // Number of samples in packet is unknown. |
| 42 if (last_decoded_length_ > 0) { |
| 43 // Use length of last decoded packet, but since this is the total for all |
| 44 // channels, we have to divide by 2 in the stereo case. |
| 45 samples_to_decode = rtc::CheckedDivExact( |
| 46 last_decoded_length_, static_cast<size_t>(stereo_ ? 2uL : 1uL)); |
| 47 } else { |
| 48 // This is the first packet to decode, and we do not know the length of |
| 49 // it. Set it to 10 ms. |
| 50 samples_to_decode = rtc::CheckedDivExact(sample_rate_hz, 100); |
| 51 } |
| 52 } |
30 | 53 |
31 if (next_timestamp_from_input_ && | 54 if (next_timestamp_from_input_ && |
32 timestamp_to_decode != *next_timestamp_from_input_) { | 55 timestamp_to_decode != *next_timestamp_from_input_) { |
33 // A gap in the timestamp sequence is detected. Skip the same number of | 56 // A gap in the timestamp sequence is detected. Skip the same number of |
34 // samples from the file. | 57 // samples from the file. |
35 uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_; | 58 uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_; |
36 RTC_CHECK(input_->Seek(jump)); | 59 RTC_CHECK(input_->Seek(jump)); |
37 } | 60 } |
38 | 61 |
39 RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded)); | |
40 next_timestamp_from_input_ = | 62 next_timestamp_from_input_ = |
41 rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode); | 63 rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode); |
42 | 64 |
| 65 uint32_t original_payload_size_bytes = |
| 66 ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]); |
| 67 if (original_payload_size_bytes == 1) { |
| 68 // This is a comfort noise payload. |
| 69 RTC_DCHECK_GT(last_decoded_length_, 0u); |
| 70 std::fill_n(decoded, last_decoded_length_, 0); |
| 71 *speech_type = kComfortNoise; |
| 72 cng_mode_ = true; |
| 73 return last_decoded_length_; |
| 74 } |
| 75 |
| 76 cng_mode_ = false; |
| 77 RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded)); |
| 78 |
43 if (stereo_) { | 79 if (stereo_) { |
44 InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2, | 80 InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2, |
45 decoded); | 81 decoded); |
46 samples_to_decode *= 2; | 82 samples_to_decode *= 2; |
47 } | 83 } |
48 | 84 |
49 *speech_type = kSpeech; | 85 *speech_type = kSpeech; |
50 return samples_to_decode; | 86 return last_decoded_length_ = samples_to_decode; |
51 } | 87 } |
52 | 88 |
53 void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp, | 89 void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp, |
54 size_t samples, | 90 size_t samples, |
| 91 size_t original_payload_size_bytes, |
55 rtc::ArrayView<uint8_t> encoded) { | 92 rtc::ArrayView<uint8_t> encoded) { |
56 RTC_CHECK_GE(encoded.size(), 8u); | 93 RTC_CHECK_GE(encoded.size(), 12u); |
57 ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp); | 94 ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp); |
58 ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4], | 95 ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4], |
59 rtc::checked_cast<uint32_t>(samples)); | 96 rtc::checked_cast<uint32_t>(samples)); |
| 97 ByteWriter<uint32_t>::WriteLittleEndian( |
| 98 &encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes)); |
60 } | 99 } |
61 | 100 |
62 } // namespace test | 101 } // namespace test |
63 } // namespace webrtc | 102 } // namespace webrtc |
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