Revert of Setting up an RTP input fuzzer for NetEq (patchset #2 id:20001 of https://codereview.webrtc.org/2315633002/ )
Reason for revert: Broke all Chromium libFuzzer builds https://bugs.chromium.org/p/chromium/issues/detail?id=645069 Original issue's description: > Setting up an RTP input fuzzer for NetEq > > This CL introduces a new fuzzer target neteq_rtp_fuzzer that > manipulates the RTP header fields before inserting the packets into > NetEq. A few helper classes are also introduced. > > BUG=webrtc:5447 > NOTRY=True > > Committed: https://crrev.com/2d273f1e97cd5030ed1686f27ce1118291b66395 > Cr-Commit-Position: refs/heads/master@{#14103} TBR=ivoc@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5447 Review-Url: https://codereview.webrtc.org/2328483002 Cr-Commit-Position: refs/heads/master@{#14131}
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henrik.lundin
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@ -231,8 +231,6 @@
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'tools/audio_sink.cc',
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'tools/constant_pcm_packet_source.cc',
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'tools/constant_pcm_packet_source.h',
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'tools/encode_neteq_input.cc',
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'tools/encode_neteq_input.h',
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'tools/fake_decode_from_file.cc',
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'tools/fake_decode_from_file.h',
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'tools/input_audio_file.cc',
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@ -1,89 +0,0 @@
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/*
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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 "webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h"
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#include <utility>
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#include "webrtc/base/checks.h"
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namespace webrtc {
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namespace test {
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EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<InputAudioFile> input,
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std::unique_ptr<AudioEncoder> encoder,
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int64_t input_duration_ms)
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: input_(std::move(input)),
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encoder_(std::move(encoder)),
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input_duration_ms_(input_duration_ms) {
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CreatePacket();
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}
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rtc::Optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
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RTC_DCHECK(packet_data_);
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return rtc::Optional<int64_t>(static_cast<int64_t>(packet_data_->time_ms));
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}
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rtc::Optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
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return rtc::Optional<int64_t>(next_output_event_ms_);
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}
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std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() {
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RTC_DCHECK(packet_data_);
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// Grab the packet to return...
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std::unique_ptr<PacketData> packet_to_return = std::move(packet_data_);
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// ... and line up the next packet for future use.
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CreatePacket();
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return packet_to_return;
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}
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void EncodeNetEqInput::AdvanceOutputEvent() {
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next_output_event_ms_ += kOutputPeriodMs;
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}
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rtc::Optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
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RTC_DCHECK(packet_data_);
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return rtc::Optional<RTPHeader>(packet_data_->header.header);
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}
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void EncodeNetEqInput::CreatePacket() {
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// Create a new PacketData object.
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RTC_DCHECK(!packet_data_);
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packet_data_.reset(new NetEqInput::PacketData);
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RTC_DCHECK_EQ(packet_data_->payload.size(), 0u);
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// Loop until we get a packet.
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AudioEncoder::EncodedInfo info;
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RTC_DCHECK(!info.send_even_if_empty);
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int num_blocks = 0;
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while (packet_data_->payload.size() == 0 && !info.send_even_if_empty) {
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const size_t num_samples = rtc::CheckedDivExact(
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static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
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std::unique_ptr<int16_t[]> audio(new int16_t[num_samples]);
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RTC_CHECK(input_->Read(num_samples, audio.get()));
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info = encoder_->Encode(
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rtp_timestamp_, rtc::ArrayView<const int16_t>(audio.get(), num_samples),
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&packet_data_->payload);
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rtp_timestamp_ +=
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num_samples * encoder_->RtpTimestampRateHz() / encoder_->SampleRateHz();
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++num_blocks;
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}
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packet_data_->header.header.timestamp = info.encoded_timestamp;
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packet_data_->header.header.payloadType = info.payload_type;
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packet_data_->header.header.sequenceNumber = sequence_number_++;
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packet_data_->time_ms = next_packet_time_ms_;
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next_packet_time_ms_ += num_blocks * kOutputPeriodMs;
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}
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} // namespace test
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} // namespace webrtc
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@ -1,64 +0,0 @@
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/*
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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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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
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#include <memory>
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#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
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#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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namespace test {
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// This class provides a NetEqInput that takes audio from an input file and
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// encodes it using a given audio encoder.
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class EncodeNetEqInput : public NetEqInput {
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public:
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// The source will end after the given input duration.
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EncodeNetEqInput(std::unique_ptr<InputAudioFile> input,
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std::unique_ptr<AudioEncoder> encoder,
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int64_t input_duration_ms);
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rtc::Optional<int64_t> NextPacketTime() const override;
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rtc::Optional<int64_t> NextOutputEventTime() const override;
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std::unique_ptr<PacketData> PopPacket() override;
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void AdvanceOutputEvent() override;
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bool ended() const override {
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return next_output_event_ms_ <= input_duration_ms_;
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}
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rtc::Optional<RTPHeader> NextHeader() const override;
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private:
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static constexpr int64_t kOutputPeriodMs = 10;
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void CreatePacket();
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std::unique_ptr<InputAudioFile> input_;
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std::unique_ptr<AudioEncoder> encoder_;
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std::unique_ptr<PacketData> packet_data_;
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int32_t rtp_timestamp_ = 0;
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int16_t sequence_number_ = 0;
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int64_t next_packet_time_ms_ = 0;
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int64_t next_output_event_ms_ = 0;
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const int64_t input_duration_ms_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
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@ -65,9 +65,7 @@ class NetEqInput {
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// time).
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virtual void AdvanceOutputEvent() = 0;
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// Returns true if the source has come to an end. An implementation must
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// eventually return true from this method, or the test will end up in an
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// infinite loop.
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// Returns true if the source has come to an end.
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virtual bool ended() const = 0;
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// Returns the RTP header for the next packet, i.e., the packet that will be
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