| /* |
| * 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/neteq_replacement_input.h" |
| |
| #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h" |
| #include "webrtc/rtc_base/checks.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| NetEqReplacementInput::NetEqReplacementInput( |
| std::unique_ptr<NetEqInput> source, |
| uint8_t replacement_payload_type, |
| const std::set<uint8_t>& comfort_noise_types, |
| const std::set<uint8_t>& forbidden_types) |
| : source_(std::move(source)), |
| replacement_payload_type_(replacement_payload_type), |
| comfort_noise_types_(comfort_noise_types), |
| forbidden_types_(forbidden_types) { |
| RTC_CHECK(source_); |
| packet_ = source_->PopPacket(); |
| ReplacePacket(); |
| RTC_CHECK(packet_); |
| } |
| |
| rtc::Optional<int64_t> NetEqReplacementInput::NextPacketTime() const { |
| return packet_ |
| ? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms)) |
| : rtc::Optional<int64_t>(); |
| } |
| |
| rtc::Optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const { |
| return source_->NextOutputEventTime(); |
| } |
| |
| std::unique_ptr<NetEqInput::PacketData> NetEqReplacementInput::PopPacket() { |
| std::unique_ptr<PacketData> to_return = std::move(packet_); |
| packet_ = source_->PopPacket(); |
| ReplacePacket(); |
| return to_return; |
| } |
| |
| void NetEqReplacementInput::AdvanceOutputEvent() { |
| source_->AdvanceOutputEvent(); |
| } |
| |
| bool NetEqReplacementInput::ended() const { |
| return source_->ended(); |
| } |
| |
| rtc::Optional<RTPHeader> NetEqReplacementInput::NextHeader() const { |
| return source_->NextHeader(); |
| } |
| |
| void NetEqReplacementInput::ReplacePacket() { |
| if (!source_->NextPacketTime()) { |
| // End of input. Cannot do proper replacement on the very last packet, so we |
| // delete it instead. |
| packet_.reset(); |
| return; |
| } |
| |
| RTC_DCHECK(packet_); |
| |
| RTC_CHECK_EQ(forbidden_types_.count(packet_->header.payloadType), 0) |
| << "Payload type " << static_cast<int>(packet_->header.payloadType) |
| << " is forbidden."; |
| |
| // Check if this packet is comfort noise. |
| if (comfort_noise_types_.count(packet_->header.payloadType) != 0) { |
| // If CNG, simply insert a zero-energy one-byte payload. |
| uint8_t cng_payload[1] = {127}; // Max attenuation of CNG. |
| packet_->payload.SetData(cng_payload); |
| return; |
| } |
| |
| rtc::Optional<RTPHeader> next_hdr = source_->NextHeader(); |
| RTC_DCHECK(next_hdr); |
| uint8_t payload[12]; |
| RTC_DCHECK_LE(last_frame_size_timestamps_, 120 * 48); |
| uint32_t input_frame_size_timestamps = last_frame_size_timestamps_; |
| const uint32_t timestamp_diff = |
| next_hdr->timestamp - packet_->header.timestamp; |
| if (next_hdr->sequenceNumber == packet_->header.sequenceNumber + 1 && |
| timestamp_diff <= 120 * 48) { |
| // Packets are in order and the timestamp diff is less than 5760 samples. |
| // Accept the timestamp diff as a valid frame size. |
| input_frame_size_timestamps = timestamp_diff; |
| last_frame_size_timestamps_ = input_frame_size_timestamps; |
| } |
| RTC_DCHECK_LE(input_frame_size_timestamps, 120 * 48); |
| FakeDecodeFromFile::PrepareEncoded(packet_->header.timestamp, |
| input_frame_size_timestamps, |
| packet_->payload.size(), payload); |
| packet_->payload.SetData(payload); |
| packet_->header.payloadType = replacement_payload_type_; |
| return; |
| } |
| |
| } // namespace test |
| } // namespace webrtc |