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..29beed5644f31402bda6d74d75fde25b676d4ab3 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 +19,37 @@ 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_, 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_) { |
@@ -36,10 +59,23 @@ 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); |
+ 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); |
@@ -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 |