| /* |
| * 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/rtc_base/checks.h" |
| #include "webrtc/rtc_base/safe_conversions.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator, |
| std::unique_ptr<AudioEncoder> encoder, |
| int64_t input_duration_ms) |
| : generator_(std::move(generator)), |
| 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); |
| } |
| |
| 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(), 0); |
| |
| // 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); |
| |
| info = encoder_->Encode(rtp_timestamp_, generator_->Generate(num_samples), |
| &packet_data_->payload); |
| |
| rtp_timestamp_ += rtc::dchecked_cast<uint32_t>( |
| num_samples * encoder_->RtpTimestampRateHz() / |
| encoder_->SampleRateHz()); |
| ++num_blocks; |
| } |
| packet_data_->header.timestamp = info.encoded_timestamp; |
| packet_data_->header.payloadType = info.payload_type; |
| packet_data_->header.sequenceNumber = sequence_number_++; |
| packet_data_->time_ms = next_packet_time_ms_; |
| next_packet_time_ms_ += num_blocks * kOutputPeriodMs; |
| } |
| |
| } // namespace test |
| } // namespace webrtc |