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00eb12a20c
This change enables NetEq to use the packet concealment audio (aka PLC) produced by a decoder. The change also includes a new API to the AudioDecoder interface, which lets the decoder implementation generate and deliver concealment audio. Bug: webrtc:9180 Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4 Reviewed-on: https://webrtc-review.googlesource.com/96340 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24738}
94 lines
3 KiB
C++
94 lines
3 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 <utility>
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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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absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
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RTC_DCHECK(packet_data_);
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return static_cast<int64_t>(packet_data_->time_ms);
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}
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absl::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<NetEqInput::PacketData> 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<PacketData> 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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absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
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RTC_DCHECK(packet_data_);
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return packet_data_->header;
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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_.reset(new NetEqInput::PacketData);
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RTC_DCHECK_EQ(packet_data_->payload.size(), 0);
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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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while (packet_data_->payload.size() == 0 && !info.send_even_if_empty) {
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const size_t num_samples = rtc::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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&packet_data_->payload);
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rtp_timestamp_ += rtc::dchecked_cast<uint32_t>(
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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_->header.timestamp = info.encoded_timestamp;
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packet_data_->header.payloadType = info.payload_type;
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packet_data_->header.sequenceNumber = sequence_number_++;
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packet_data_->time_ms = 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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