/* * Copyright (c) 2014 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/codecs/red/audio_encoder_copy_red.h" #include #include #include #include "rtc_base/byte_order.h" #include "rtc_base/checks.h" namespace webrtc { // RED packets must be less than 1024 bytes to fit the 10 bit block length. static constexpr const int kRedMaxPacketSize = 1 << 10; // The typical MTU is 1200 bytes. static constexpr const size_t kAudioMaxRtpPacketLen = 1200; AudioEncoderCopyRed::Config::Config() = default; AudioEncoderCopyRed::Config::Config(Config&&) = default; AudioEncoderCopyRed::Config::~Config() = default; AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config) : speech_encoder_(std::move(config.speech_encoder)), max_packet_length_(kAudioMaxRtpPacketLen), red_payload_type_(config.payload_type) { RTC_CHECK(speech_encoder_) << "Speech encoder not provided."; } AudioEncoderCopyRed::~AudioEncoderCopyRed() = default; int AudioEncoderCopyRed::SampleRateHz() const { return speech_encoder_->SampleRateHz(); } size_t AudioEncoderCopyRed::NumChannels() const { return speech_encoder_->NumChannels(); } int AudioEncoderCopyRed::RtpTimestampRateHz() const { return speech_encoder_->RtpTimestampRateHz(); } size_t AudioEncoderCopyRed::Num10MsFramesInNextPacket() const { return speech_encoder_->Num10MsFramesInNextPacket(); } size_t AudioEncoderCopyRed::Max10MsFramesInAPacket() const { return speech_encoder_->Max10MsFramesInAPacket(); } int AudioEncoderCopyRed::GetTargetBitrate() const { return speech_encoder_->GetTargetBitrate(); } size_t AudioEncoderCopyRed::CalculateHeaderLength(size_t encoded_bytes) const { size_t header_size = 1; size_t bytes_available = max_packet_length_ - encoded_bytes; if (secondary_info_.encoded_bytes > 0 && secondary_info_.encoded_bytes < bytes_available) { header_size += 4; bytes_available -= secondary_info_.encoded_bytes; } if (tertiary_info_.encoded_bytes > 0 && tertiary_info_.encoded_bytes < bytes_available) { header_size += 4; } return header_size > 1 ? header_size : 0; } AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl( uint32_t rtp_timestamp, rtc::ArrayView audio, rtc::Buffer* encoded) { rtc::Buffer primary_encoded; EncodedInfo info = speech_encoder_->Encode(rtp_timestamp, audio, &primary_encoded); RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders."; RTC_DCHECK_EQ(primary_encoded.size(), info.encoded_bytes); if (info.encoded_bytes == 0 || info.encoded_bytes > kRedMaxPacketSize) { return info; } RTC_DCHECK_GT(max_packet_length_, info.encoded_bytes); // Allocate room for RFC 2198 header if there is redundant data. // Otherwise this will send the primary payload type without // wrapping in RED. const size_t header_length_bytes = CalculateHeaderLength(info.encoded_bytes); encoded->SetSize(header_length_bytes); size_t header_offset = 0; size_t bytes_available = max_packet_length_ - info.encoded_bytes; if (tertiary_info_.encoded_bytes > 0 && tertiary_info_.encoded_bytes + secondary_info_.encoded_bytes < bytes_available) { encoded->AppendData(tertiary_encoded_); const uint32_t timestamp_delta = info.encoded_timestamp - tertiary_info_.encoded_timestamp; encoded->data()[header_offset] = tertiary_info_.payload_type | 0x80; rtc::SetBE16(static_cast(encoded->data()) + header_offset + 1, (timestamp_delta << 2) | (tertiary_info_.encoded_bytes >> 8)); encoded->data()[header_offset + 3] = tertiary_info_.encoded_bytes & 0xff; header_offset += 4; bytes_available -= tertiary_info_.encoded_bytes; } if (secondary_info_.encoded_bytes > 0 && secondary_info_.encoded_bytes < bytes_available) { encoded->AppendData(secondary_encoded_); const uint32_t timestamp_delta = info.encoded_timestamp - secondary_info_.encoded_timestamp; encoded->data()[header_offset] = secondary_info_.payload_type | 0x80; rtc::SetBE16(static_cast(encoded->data()) + header_offset + 1, (timestamp_delta << 2) | (secondary_info_.encoded_bytes >> 8)); encoded->data()[header_offset + 3] = secondary_info_.encoded_bytes & 0xff; header_offset += 4; bytes_available -= secondary_info_.encoded_bytes; } encoded->AppendData(primary_encoded); if (header_length_bytes > 0) { RTC_DCHECK_EQ(header_offset, header_length_bytes - 1); encoded->data()[header_offset] = info.payload_type; } // |info| will be implicitly cast to an EncodedInfoLeaf struct, effectively // discarding the (empty) vector of redundant information. This is // intentional. info.redundant.push_back(info); RTC_DCHECK_EQ(info.redundant.size(), 1); RTC_DCHECK_EQ(info.speech, info.redundant[0].speech); if (secondary_info_.encoded_bytes > 0) { info.redundant.push_back(secondary_info_); RTC_DCHECK_EQ(info.redundant.size(), 2); } if (tertiary_info_.encoded_bytes > 0) { info.redundant.push_back(tertiary_info_); RTC_DCHECK_EQ(info.redundant.size(), 2 + (secondary_info_.encoded_bytes > 0 ? 1 : 0)); } // Save secondary to tertiary. tertiary_encoded_.SetData(secondary_encoded_); tertiary_info_ = secondary_info_; // Save primary to secondary. secondary_encoded_.SetData(primary_encoded); secondary_info_ = info; // Update main EncodedInfo. if (header_length_bytes > 0) { info.payload_type = red_payload_type_; } info.encoded_bytes = encoded->size(); return info; } void AudioEncoderCopyRed::Reset() { speech_encoder_->Reset(); secondary_encoded_.Clear(); secondary_info_.encoded_bytes = 0; } bool AudioEncoderCopyRed::SetFec(bool enable) { return speech_encoder_->SetFec(enable); } bool AudioEncoderCopyRed::SetDtx(bool enable) { return speech_encoder_->SetDtx(enable); } bool AudioEncoderCopyRed::GetDtx() const { return speech_encoder_->GetDtx(); } bool AudioEncoderCopyRed::SetApplication(Application application) { return speech_encoder_->SetApplication(application); } void AudioEncoderCopyRed::SetMaxPlaybackRate(int frequency_hz) { speech_encoder_->SetMaxPlaybackRate(frequency_hz); } bool AudioEncoderCopyRed::EnableAudioNetworkAdaptor( const std::string& config_string, RtcEventLog* event_log) { return speech_encoder_->EnableAudioNetworkAdaptor(config_string, event_log); } void AudioEncoderCopyRed::DisableAudioNetworkAdaptor() { speech_encoder_->DisableAudioNetworkAdaptor(); } void AudioEncoderCopyRed::OnReceivedUplinkPacketLossFraction( float uplink_packet_loss_fraction) { speech_encoder_->OnReceivedUplinkPacketLossFraction( uplink_packet_loss_fraction); } void AudioEncoderCopyRed::OnReceivedUplinkBandwidth( int target_audio_bitrate_bps, absl::optional bwe_period_ms) { speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps, bwe_period_ms); } void AudioEncoderCopyRed::OnReceivedUplinkAllocation( BitrateAllocationUpdate update) { speech_encoder_->OnReceivedUplinkAllocation(update); } absl::optional> AudioEncoderCopyRed::GetFrameLengthRange() const { return speech_encoder_->GetFrameLengthRange(); } void AudioEncoderCopyRed::OnReceivedRtt(int rtt_ms) { speech_encoder_->OnReceivedRtt(rtt_ms); } void AudioEncoderCopyRed::OnReceivedOverhead(size_t overhead_bytes_per_packet) { max_packet_length_ = kAudioMaxRtpPacketLen - overhead_bytes_per_packet; return speech_encoder_->OnReceivedOverhead(overhead_bytes_per_packet); } void AudioEncoderCopyRed::SetReceiverFrameLengthRange(int min_frame_length_ms, int max_frame_length_ms) { return speech_encoder_->SetReceiverFrameLengthRange(min_frame_length_ms, max_frame_length_ms); } ANAStats AudioEncoderCopyRed::GetANAStats() const { return speech_encoder_->GetANAStats(); } rtc::ArrayView> AudioEncoderCopyRed::ReclaimContainedEncoders() { return rtc::ArrayView>(&speech_encoder_, 1); } } // namespace webrtc