blob: 29beed5644f31402bda6d74d75fde25b676d4ab3 [file] [log] [blame]
/*
* 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 "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
namespace webrtc {
namespace test {
int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
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_) {
// A gap in the timestamp sequence is detected. Skip the same number of
// samples from the file.
uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_;
RTC_CHECK(input_->Seek(jump));
}
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);
samples_to_decode *= 2;
}
*speech_type = kSpeech;
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(), 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
} // namespace webrtc