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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
103 lines
3.9 KiB
C++
103 lines
3.9 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
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#include "modules/rtp_rtcp/source/byte_io.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/safe_conversions.h"
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namespace webrtc {
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namespace test {
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int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) {
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if (encoded_len == 0) {
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// Decoder is asked to produce codec-internal comfort noise.
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RTC_DCHECK(!encoded); // NetEq always sends nullptr in this case.
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RTC_DCHECK(cng_mode_);
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RTC_DCHECK_GT(last_decoded_length_, 0);
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std::fill_n(decoded, last_decoded_length_, 0);
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*speech_type = kComfortNoise;
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return rtc::dchecked_cast<int>(last_decoded_length_);
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}
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RTC_CHECK_GE(encoded_len, 12);
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uint32_t timestamp_to_decode =
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ByteReader<uint32_t>::ReadLittleEndian(encoded);
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uint32_t samples_to_decode =
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ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
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if (samples_to_decode == 0) {
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// Number of samples in packet is unknown.
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if (last_decoded_length_ > 0) {
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// Use length of last decoded packet, but since this is the total for all
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// channels, we have to divide by 2 in the stereo case.
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samples_to_decode = rtc::dchecked_cast<int>(rtc::CheckedDivExact(
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last_decoded_length_, static_cast<size_t>(stereo_ ? 2uL : 1uL)));
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} else {
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// This is the first packet to decode, and we do not know the length of
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// it. Set it to 10 ms.
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samples_to_decode = rtc::CheckedDivExact(sample_rate_hz, 100);
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}
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}
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if (next_timestamp_from_input_ &&
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timestamp_to_decode != *next_timestamp_from_input_) {
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// A gap in the timestamp sequence is detected. Skip the same number of
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// samples from the file.
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uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_;
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RTC_CHECK(input_->Seek(jump));
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}
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next_timestamp_from_input_ =
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rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode);
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uint32_t original_payload_size_bytes =
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ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
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if (original_payload_size_bytes == 1) {
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// This is a comfort noise payload.
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RTC_DCHECK_GT(last_decoded_length_, 0);
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std::fill_n(decoded, last_decoded_length_, 0);
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*speech_type = kComfortNoise;
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cng_mode_ = true;
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return rtc::dchecked_cast<int>(last_decoded_length_);
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}
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cng_mode_ = false;
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RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
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if (stereo_) {
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InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2,
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decoded);
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samples_to_decode *= 2;
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}
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*speech_type = kSpeech;
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last_decoded_length_ = samples_to_decode;
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return rtc::dchecked_cast<int>(last_decoded_length_);
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}
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void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
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size_t samples,
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size_t original_payload_size_bytes,
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rtc::ArrayView<uint8_t> encoded) {
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RTC_CHECK_GE(encoded.size(), 12);
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ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
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ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
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rtc::checked_cast<uint32_t>(samples));
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ByteWriter<uint32_t>::WriteLittleEndian(
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&encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes));
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}
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} // namespace test
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} // namespace webrtc
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