Chromium Code Reviews| Index: webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc |
| diff --git a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc |
| index 9ca5fb810bd64f21298680634793cc1b53a55ae5..9f6c745349026fbbd02f3c6ca6e6c3d029e7959b 100644 |
| --- a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc |
| +++ b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc |
| @@ -19,14 +19,39 @@ namespace test { |
| int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| - int /*sample_rate_hz*/, |
| + int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| - RTC_CHECK_GE(encoded_len, 8u); |
| + 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_, (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); |
| + } |
| + } |
| + uint32_t original_payload_size_bytes = |
| + ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]); |
|
ivoc
2016/08/24 14:55:28
I think it makes sense to move this below to just
hlundin-webrtc
2016/08/24 15:27:36
Done.
|
| if (next_timestamp_from_input_ && |
| timestamp_to_decode != *next_timestamp_from_input_) { |
| @@ -36,10 +61,21 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, |
| RTC_CHECK(input_->Seek(jump)); |
| } |
| - RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded)); |
| next_timestamp_from_input_ = |
| rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode); |
| + 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); |
| @@ -47,16 +83,19 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, |
| } |
| *speech_type = kSpeech; |
| - return samples_to_decode; |
| + 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(), 8u); |
| + 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 |