137e9fc160
Thus removing RTPHeader in favor RtpPacketReceived in this api Bug: webrtc:42225366 Change-Id: I90ef081b2bdace17311af32eae050975b1bd033c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404180 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45331}
104 lines
3.2 KiB
C++
104 lines
3.2 KiB
C++
/*
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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 "modules/audio_coding/neteq/tools/encode_neteq_input.h"
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#include <cstddef>
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#include <cstdint>
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#include <memory>
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#include <optional>
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#include <utility>
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/units/timestamp.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/safe_conversions.h"
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namespace webrtc {
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namespace test {
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EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator,
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std::unique_ptr<AudioEncoder> encoder,
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int64_t input_duration_ms)
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: generator_(std::move(generator)),
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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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EncodeNetEqInput::~EncodeNetEqInput() = default;
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std::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
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RTC_DCHECK(packet_data_);
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return packet_data_->arrival_time().ms();
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}
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std::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
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return next_output_event_ms_;
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}
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std::unique_ptr<RtpPacketReceived> 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<RtpPacketReceived> 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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bool EncodeNetEqInput::ended() const {
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return next_output_event_ms_ > input_duration_ms_;
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}
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const RtpPacketReceived* EncodeNetEqInput::NextPacket() const {
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RTC_DCHECK(packet_data_);
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return packet_data_.get();
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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_ = std::make_unique<RtpPacketReceived>();
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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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Buffer payload;
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while (payload.empty() && !info.send_even_if_empty) {
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const size_t num_samples = CheckedDivExact(
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static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
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info = encoder_->Encode(rtp_timestamp_, generator_->Generate(num_samples),
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&payload);
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rtp_timestamp_ +=
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dchecked_cast<uint32_t>(num_samples * encoder_->RtpTimestampRateHz() /
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encoder_->SampleRateHz());
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++num_blocks;
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}
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packet_data_->SetPayload(payload);
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packet_data_->SetTimestamp(info.encoded_timestamp);
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packet_data_->SetPayloadType(info.payload_type);
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packet_data_->SetSequenceNumber(sequence_number_++);
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packet_data_->set_arrival_time(Timestamp::Millis(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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