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More or less bit-exact, only difference is that we don't seek in the input file before returning silence for DTX packets. Bug: webrtc:13322 Change-Id: I147b70d4a0f2c78719c9673b55df6617e064bd61 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301104 Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39851}
122 lines
4.4 KiB
C++
122 lines
4.4 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/numerics/safe_conversions.h"
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namespace webrtc {
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namespace test {
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namespace {
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class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame {
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public:
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FakeEncodedFrame(FakeDecodeFromFile* decoder,
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uint32_t timestamp,
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size_t duration,
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bool is_dtx)
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: decoder_(decoder),
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timestamp_(timestamp),
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duration_(duration),
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is_dtx_(is_dtx) {}
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size_t Duration() const override { return duration_; }
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absl::optional<DecodeResult> Decode(
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rtc::ArrayView<int16_t> decoded) const override {
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if (is_dtx_) {
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std::fill_n(decoded.data(), duration_, 0);
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return DecodeResult{duration_, AudioDecoder::kComfortNoise};
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}
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decoder_->ReadFromFile(timestamp_, duration_, decoded.data());
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return DecodeResult{Duration(), AudioDecoder::kSpeech};
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}
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bool IsDtxPacket() const override { return is_dtx_; }
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private:
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FakeDecodeFromFile* const decoder_;
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const uint32_t timestamp_;
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const size_t duration_;
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const bool is_dtx_;
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};
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} // namespace
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void FakeDecodeFromFile::ReadFromFile(uint32_t timestamp,
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size_t samples,
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int16_t* destination) {
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if (next_timestamp_from_input_ && timestamp != *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 - *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_ = timestamp + samples;
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RTC_CHECK(input_->Read(static_cast<size_t>(samples), destination));
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if (stereo_) {
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InputAudioFile::DuplicateInterleaved(destination, samples, 2, destination);
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}
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}
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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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// This call is only used to produce codec-internal comfort noise.
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RTC_DCHECK_EQ(sample_rate_hz, SampleRateHz());
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RTC_DCHECK_EQ(encoded_len, 0);
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RTC_DCHECK(!encoded); // NetEq always sends nullptr in this case.
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const int samples_to_decode = rtc::CheckedDivExact(SampleRateHz(), 100);
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const int total_samples_to_decode = samples_to_decode * (stereo_ ? 2 : 1);
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std::fill_n(decoded, total_samples_to_decode, 0);
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*speech_type = kComfortNoise;
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return rtc::dchecked_cast<int>(total_samples_to_decode);
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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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std::vector<AudioDecoder::ParseResult> FakeDecodeFromFile::ParsePayload(
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rtc::Buffer&& payload,
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uint32_t timestamp) {
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RTC_CHECK_GE(payload.size(), 12);
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// Parse payload encoded in PrepareEncoded.
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RTC_CHECK_EQ(timestamp, ByteReader<uint32_t>::ReadLittleEndian(&payload[0]));
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size_t samples = ByteReader<uint32_t>::ReadLittleEndian(&payload[4]);
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size_t original_payload_size_bytes =
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ByteReader<uint32_t>::ReadLittleEndian(&payload[8]);
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bool opus_dtx = original_payload_size_bytes <= 2;
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std::vector<ParseResult> results;
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results.emplace_back(
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timestamp, 0,
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std::make_unique<FakeEncodedFrame>(this, timestamp, samples, opus_dtx));
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return results;
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}
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} // namespace test
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} // namespace webrtc
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