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/*
* 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 "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
namespace test {
namespace {
class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame {
public:
FakeEncodedFrame(FakeDecodeFromFile* decoder,
uint32_t timestamp,
size_t duration,
bool is_dtx)
: decoder_(decoder),
timestamp_(timestamp),
duration_(duration),
is_dtx_(is_dtx) {}
size_t Duration() const override { return duration_; }
absl::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
if (is_dtx_) {
std::fill_n(decoded.data(), duration_, 0);
return DecodeResult{duration_, AudioDecoder::kComfortNoise};
}
decoder_->ReadFromFile(timestamp_, duration_, decoded.data());
return DecodeResult{Duration(), AudioDecoder::kSpeech};
}
bool IsDtxPacket() const override { return is_dtx_; }
private:
FakeDecodeFromFile* const decoder_;
const uint32_t timestamp_;
const size_t duration_;
const bool is_dtx_;
};
} // namespace
void FakeDecodeFromFile::ReadFromFile(uint32_t timestamp,
size_t samples,
int16_t* destination) {
if (next_timestamp_from_input_ && timestamp != *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 - *next_timestamp_from_input_;
RTC_CHECK(input_->Seek(jump));
}
next_timestamp_from_input_ = timestamp + samples;
RTC_CHECK(input_->Read(static_cast<size_t>(samples), destination));
if (stereo_) {
InputAudioFile::DuplicateInterleaved(destination, samples, 2, destination);
}
}
int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
// This call is only used to produce codec-internal comfort noise.
RTC_DCHECK_EQ(sample_rate_hz, SampleRateHz());
RTC_DCHECK_EQ(encoded_len, 0);
RTC_DCHECK(!encoded); // NetEq always sends nullptr in this case.
const int samples_to_decode = rtc::CheckedDivExact(SampleRateHz(), 100);
const int total_samples_to_decode = samples_to_decode * (stereo_ ? 2 : 1);
std::fill_n(decoded, total_samples_to_decode, 0);
*speech_type = kComfortNoise;
return rtc::dchecked_cast<int>(total_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(), 12);
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));
}
std::vector<AudioDecoder::ParseResult> FakeDecodeFromFile::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
RTC_CHECK_GE(payload.size(), 12);
// Parse payload encoded in PrepareEncoded.
RTC_CHECK_EQ(timestamp, ByteReader<uint32_t>::ReadLittleEndian(&payload[0]));
size_t samples = ByteReader<uint32_t>::ReadLittleEndian(&payload[4]);
size_t original_payload_size_bytes =
ByteReader<uint32_t>::ReadLittleEndian(&payload[8]);
bool opus_dtx = original_payload_size_bytes <= 2;
std::vector<ParseResult> results;
results.emplace_back(
timestamp, 0,
std::make_unique<FakeEncodedFrame>(this, timestamp, samples, opus_dtx));
return results;
}
} // namespace test
} // namespace webrtc