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/*
* 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 "modules/audio_coding/neteq/tools/encode_neteq_input.h"
#include <cstddef>
#include <cstdint>
#include <memory>
#include <optional>
#include <utility>
#include "api/audio_codecs/audio_encoder.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/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();
}
EncodeNetEqInput::~EncodeNetEqInput() = default;
std::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
RTC_DCHECK(packet_data_);
return packet_data_->arrival_time().ms();
}
std::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
return next_output_event_ms_;
}
std::unique_ptr<RtpPacketReceived> EncodeNetEqInput::PopPacket() {
RTC_DCHECK(packet_data_);
// Grab the packet to return...
std::unique_ptr<RtpPacketReceived> 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;
}
bool EncodeNetEqInput::ended() const {
return next_output_event_ms_ > input_duration_ms_;
}
const RtpPacketReceived* EncodeNetEqInput::NextPacket() const {
RTC_DCHECK(packet_data_);
return packet_data_.get();
}
void EncodeNetEqInput::CreatePacket() {
// Create a new PacketData object.
RTC_DCHECK(!packet_data_);
packet_data_ = std::make_unique<RtpPacketReceived>();
// Loop until we get a packet.
AudioEncoder::EncodedInfo info;
RTC_DCHECK(!info.send_even_if_empty);
int num_blocks = 0;
Buffer payload;
while (payload.empty() && !info.send_even_if_empty) {
const size_t num_samples = CheckedDivExact(
static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
info = encoder_->Encode(rtp_timestamp_, generator_->Generate(num_samples),
&payload);
rtp_timestamp_ +=
dchecked_cast<uint32_t>(num_samples * encoder_->RtpTimestampRateHz() /
encoder_->SampleRateHz());
++num_blocks;
}
packet_data_->SetPayload(payload);
packet_data_->SetTimestamp(info.encoded_timestamp);
packet_data_->SetPayloadType(info.payload_type);
packet_data_->SetSequenceNumber(sequence_number_++);
packet_data_->set_arrival_time(Timestamp::Millis(next_packet_time_ms_));
next_packet_time_ms_ += num_blocks * kOutputPeriodMs;
}
} // namespace test
} // namespace webrtc
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