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Extends the RED implementation to support a distance of two, i.e. two packets redundancy. BUG=webrtc:11640 Change-Id: I5113a97a4e3d45d836d7952a0c19c5381069c158 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178565 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31625}
203 lines
6.4 KiB
C++
203 lines
6.4 KiB
C++
/*
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* Copyright (c) 2014 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/codecs/red/audio_encoder_copy_red.h"
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#include <string.h>
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#include <utility>
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#include <vector>
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#include "rtc_base/byte_order.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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static const int kRedMaxPacketSize = 1 << 10;
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AudioEncoderCopyRed::Config::Config() = default;
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AudioEncoderCopyRed::Config::Config(Config&&) = default;
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AudioEncoderCopyRed::Config::~Config() = default;
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AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config)
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: speech_encoder_(std::move(config.speech_encoder)),
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red_payload_type_(config.payload_type) {
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RTC_CHECK(speech_encoder_) << "Speech encoder not provided.";
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}
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AudioEncoderCopyRed::~AudioEncoderCopyRed() = default;
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int AudioEncoderCopyRed::SampleRateHz() const {
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return speech_encoder_->SampleRateHz();
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}
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size_t AudioEncoderCopyRed::NumChannels() const {
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return speech_encoder_->NumChannels();
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}
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int AudioEncoderCopyRed::RtpTimestampRateHz() const {
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return speech_encoder_->RtpTimestampRateHz();
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}
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size_t AudioEncoderCopyRed::Num10MsFramesInNextPacket() const {
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return speech_encoder_->Num10MsFramesInNextPacket();
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}
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size_t AudioEncoderCopyRed::Max10MsFramesInAPacket() const {
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return speech_encoder_->Max10MsFramesInAPacket();
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}
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int AudioEncoderCopyRed::GetTargetBitrate() const {
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return speech_encoder_->GetTargetBitrate();
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}
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size_t AudioEncoderCopyRed::CalculateHeaderLength() const {
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size_t header_size = 1;
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if (secondary_info_.encoded_bytes > 0) {
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header_size += 4;
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}
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if (tertiary_info_.encoded_bytes > 0) {
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header_size += 4;
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}
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return header_size > 1 ? header_size : 0;
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}
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AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
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uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) {
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rtc::Buffer primary_encoded;
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EncodedInfo info =
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speech_encoder_->Encode(rtp_timestamp, audio, &primary_encoded);
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RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
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RTC_DCHECK_EQ(primary_encoded.size(), info.encoded_bytes);
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if (info.encoded_bytes == 0) {
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return info;
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}
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// Allocate room for RFC 2198 header if there is redundant data.
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// Otherwise this will send the primary payload type without
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// wrapping in RED.
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const size_t header_length_bytes = CalculateHeaderLength();
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encoded->SetSize(header_length_bytes);
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size_t header_offset = 0;
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if (tertiary_info_.encoded_bytes > 0 &&
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tertiary_info_.encoded_bytes < kRedMaxPacketSize) {
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encoded->AppendData(tertiary_encoded_);
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const uint32_t timestamp_delta =
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info.encoded_timestamp - tertiary_info_.encoded_timestamp;
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encoded->data()[header_offset] = tertiary_info_.payload_type | 0x80;
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rtc::SetBE16(static_cast<uint8_t*>(encoded->data()) + header_offset + 1,
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(timestamp_delta << 2) | (tertiary_info_.encoded_bytes >> 8));
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encoded->data()[header_offset + 3] = tertiary_info_.encoded_bytes & 0xff;
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header_offset += 4;
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}
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if (secondary_info_.encoded_bytes > 0 &&
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secondary_info_.encoded_bytes < kRedMaxPacketSize) {
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encoded->AppendData(secondary_encoded_);
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const uint32_t timestamp_delta =
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info.encoded_timestamp - secondary_info_.encoded_timestamp;
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encoded->data()[header_offset] = secondary_info_.payload_type | 0x80;
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rtc::SetBE16(static_cast<uint8_t*>(encoded->data()) + header_offset + 1,
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(timestamp_delta << 2) | (secondary_info_.encoded_bytes >> 8));
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encoded->data()[header_offset + 3] = secondary_info_.encoded_bytes & 0xff;
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header_offset += 4;
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}
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encoded->AppendData(primary_encoded);
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if (header_length_bytes > 0) {
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RTC_DCHECK_EQ(header_offset, header_length_bytes - 1);
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encoded->data()[header_offset] = info.payload_type;
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}
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// |info| will be implicitly cast to an EncodedInfoLeaf struct, effectively
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// discarding the (empty) vector of redundant information. This is
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// intentional.
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info.redundant.push_back(info);
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RTC_DCHECK_EQ(info.redundant.size(), 1);
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RTC_DCHECK_EQ(info.speech, info.redundant[0].speech);
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if (secondary_info_.encoded_bytes > 0) {
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info.redundant.push_back(secondary_info_);
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RTC_DCHECK_EQ(info.redundant.size(), 2);
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}
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if (tertiary_info_.encoded_bytes > 0) {
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info.redundant.push_back(tertiary_info_);
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RTC_DCHECK_EQ(info.redundant.size(),
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2 + (secondary_info_.encoded_bytes > 0 ? 1 : 0));
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}
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// Save secondary to tertiary.
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tertiary_encoded_.SetData(secondary_encoded_);
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tertiary_info_ = secondary_info_;
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// Save primary to secondary.
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secondary_encoded_.SetData(primary_encoded);
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secondary_info_ = info;
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// Update main EncodedInfo.
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if (header_length_bytes > 0) {
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info.payload_type = red_payload_type_;
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}
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info.encoded_bytes = encoded->size();
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return info;
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}
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void AudioEncoderCopyRed::Reset() {
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speech_encoder_->Reset();
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secondary_encoded_.Clear();
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secondary_info_.encoded_bytes = 0;
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}
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bool AudioEncoderCopyRed::SetFec(bool enable) {
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return speech_encoder_->SetFec(enable);
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}
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bool AudioEncoderCopyRed::SetDtx(bool enable) {
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return speech_encoder_->SetDtx(enable);
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}
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bool AudioEncoderCopyRed::SetApplication(Application application) {
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return speech_encoder_->SetApplication(application);
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}
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void AudioEncoderCopyRed::SetMaxPlaybackRate(int frequency_hz) {
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speech_encoder_->SetMaxPlaybackRate(frequency_hz);
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}
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rtc::ArrayView<std::unique_ptr<AudioEncoder>>
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AudioEncoderCopyRed::ReclaimContainedEncoders() {
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return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
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}
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void AudioEncoderCopyRed::OnReceivedUplinkPacketLossFraction(
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float uplink_packet_loss_fraction) {
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speech_encoder_->OnReceivedUplinkPacketLossFraction(
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uplink_packet_loss_fraction);
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}
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void AudioEncoderCopyRed::OnReceivedUplinkBandwidth(
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int target_audio_bitrate_bps,
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absl::optional<int64_t> bwe_period_ms) {
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speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
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bwe_period_ms);
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}
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absl::optional<std::pair<TimeDelta, TimeDelta>>
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AudioEncoderCopyRed::GetFrameLengthRange() const {
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return speech_encoder_->GetFrameLengthRange();
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}
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} // namespace webrtc
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