Setting up an RTP input fuzzer for NetEq

webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc

Index: webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
new file mode 100644
index 0000000000000000000000000000000000000000..54682166aa7a7dd1588c53cf9f8119f96fa2e91a
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -0,0 +1,89 @@
+/*
+ *  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/encode_neteq_input.h"
+
+#include <utility>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<InputAudioFile> input,
+                                   std::unique_ptr<AudioEncoder> encoder,
+                                   int64_t input_duration_ms)
+    : input_(std::move(input)),
+      encoder_(std::move(encoder)),
+      input_duration_ms_(input_duration_ms) {
+  CreatePacket();
+}
+
+rtc::Optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
+  RTC_DCHECK(packet_data_);
+  return rtc::Optional<int64_t>(static_cast<int64_t>(packet_data_->time_ms));
+}
+
+rtc::Optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
+  return rtc::Optional<int64_t>(next_output_event_ms_);
+}
+
+std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() {
+  RTC_DCHECK(packet_data_);
+  // Grab the packet to return...
+  std::unique_ptr<PacketData> packet_to_return = std::move(packet_data_);
+  // ... and line up the next packet for future use.
+  CreatePacket();
+
+  return packet_to_return;
+}
+
+void EncodeNetEqInput::AdvanceOutputEvent() {
+  next_output_event_ms_ += kOutputPeriodMs;
+}
+
+rtc::Optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
+  RTC_DCHECK(packet_data_);
+  return rtc::Optional<RTPHeader>(packet_data_->header.header);
+}
+
+void EncodeNetEqInput::CreatePacket() {
+  // Create a new PacketData object.
+  RTC_DCHECK(!packet_data_);
+  packet_data_.reset(new NetEqInput::PacketData);
+  RTC_DCHECK_EQ(packet_data_->payload.size(), 0u);
+
+  // Loop until we get a packet.
+  AudioEncoder::EncodedInfo info;
+  RTC_DCHECK(!info.send_even_if_empty);
+  int num_blocks = 0;
+  while (packet_data_->payload.size() == 0 && !info.send_even_if_empty) {
+    const size_t num_samples = rtc::CheckedDivExact(
+        static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
+    std::unique_ptr<int16_t[]> audio(new int16_t[num_samples]);
+    RTC_CHECK(input_->Read(num_samples, audio.get()));
+
+    info = encoder_->Encode(
+        rtp_timestamp_, rtc::ArrayView<const int16_t>(audio.get(), num_samples),
+        &packet_data_->payload);
+
+    rtp_timestamp_ +=
+        num_samples * encoder_->RtpTimestampRateHz() / encoder_->SampleRateHz();
+    ++num_blocks;
+  }
+  packet_data_->header.header.timestamp = info.encoded_timestamp;
+  packet_data_->header.header.payloadType = info.payload_type;
+  packet_data_->header.header.sequenceNumber = sequence_number_++;
+  packet_data_->time_ms = next_packet_time_ms_;
+  next_packet_time_ms_ += num_blocks * kOutputPeriodMs;
+}
+
+}  // namespace test
+}  // namespace webrtc