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Implement PacketDuration() for FakeDecoderFromFile.

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Bug: None
Change-Id: Ie4ab1ce737608706f12f298f793f76571805ca91
Reviewed-on: https://webrtc-review.googlesource.com/86160
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23780}
This commit is contained in:
Minyue Li 2018-06-28 23:06:45 +02:00 committed by Commit Bot
parent c54f706993
commit a91decab4f
2 changed files with 27 additions and 17 deletions
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42
modules/audio_coding/neteq/tools/fake_decode_from_file.cc
View file

@ -22,6 +22,7 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, SampleRateHz());
if (encoded_len == 0) {
// Decoder is asked to produce codec-internal comfort noise.
RTC_DCHECK(!encoded); // NetEq always sends nullptr in this case.
@ -35,23 +36,6 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
RTC_CHECK_GE(encoded_len, 12);
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 ||
samples_to_decode % rtc::CheckedDivExact(sample_rate_hz, 100) != 0) {
// Number of samples being zero or non-multiple of 10ms is considered
// erroneous.
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::dchecked_cast<int>(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_) {
@ -61,6 +45,7 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
RTC_CHECK(input_->Seek(jump));
}
int samples_to_decode = PacketDuration(encoded, encoded_len);
next_timestamp_from_input_ = timestamp_to_decode + samples_to_decode;
uint32_t original_payload_size_bytes =
@ -88,6 +73,29 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
return rtc::dchecked_cast<int>(last_decoded_length_);
}
int FakeDecodeFromFile::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
uint32_t samples_to_decode =
encoded_len == 0 ? 0
: ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
if (samples_to_decode == 0 ||
samples_to_decode % rtc::CheckedDivExact(SampleRateHz(), 100) != 0) {
// Number of samples being zero or non-multiple of 10ms is considered
// erroneous.
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::dchecked_cast<int>(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(SampleRateHz(), 100);
}
}
return samples_to_decode;
}
void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
size_t samples,
size_t original_payload_size_bytes,

2
modules/audio_coding/neteq/tools/fake_decode_from_file.h
View file

@ -50,6 +50,8 @@ class FakeDecodeFromFile : public AudioDecoder {
int16_t* decoded,
SpeechType* speech_type) override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
// Helper method. Writes |timestamp|, |samples| and
// |original_payload_size_bytes| to |encoded| in a format that the
// FakeDecodeFromFile decoder will understand. |encoded| must be at least 12