| /* |
| * Copyright (c) 2012 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. |
| */ |
| |
| // This is the implementation of the PacketBuffer class. It is mostly based on |
| // an STL list. The list is kept sorted at all times so that the next packet to |
| // decode is at the beginning of the list. |
| |
| #include "modules/audio_coding/neteq/packet_buffer.h" |
| |
| #include <algorithm> |
| #include <list> |
| #include <memory> |
| #include <type_traits> |
| #include <utility> |
| |
| #include "api/audio_codecs/audio_decoder.h" |
| #include "modules/audio_coding/neteq/decoder_database.h" |
| #include "modules/audio_coding/neteq/statistics_calculator.h" |
| #include "modules/audio_coding/neteq/tick_timer.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| |
| namespace webrtc { |
| namespace { |
| // Predicate used when inserting packets in the buffer list. |
| // Operator() returns true when |packet| goes before |new_packet|. |
| class NewTimestampIsLarger { |
| public: |
| explicit NewTimestampIsLarger(const Packet& new_packet) |
| : new_packet_(new_packet) {} |
| bool operator()(const Packet& packet) { return (new_packet_ >= packet); } |
| |
| private: |
| const Packet& new_packet_; |
| }; |
| |
| // Returns true if both payload types are known to the decoder database, and |
| // have the same sample rate. |
| bool EqualSampleRates(uint8_t pt1, |
| uint8_t pt2, |
| const DecoderDatabase& decoder_database) { |
| auto* di1 = decoder_database.GetDecoderInfo(pt1); |
| auto* di2 = decoder_database.GetDecoderInfo(pt2); |
| return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz(); |
| } |
| |
| void LogPacketDiscarded(int codec_level, StatisticsCalculator* stats) { |
| RTC_CHECK(stats); |
| if (codec_level > 0) { |
| stats->SecondaryPacketsDiscarded(1); |
| } else { |
| stats->PacketsDiscarded(1); |
| } |
| } |
| |
| } // namespace |
| |
| PacketBuffer::PacketBuffer(size_t max_number_of_packets, |
| const TickTimer* tick_timer) |
| : max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {} |
| |
| // Destructor. All packets in the buffer will be destroyed. |
| PacketBuffer::~PacketBuffer() { |
| Flush(); |
| } |
| |
| // Flush the buffer. All packets in the buffer will be destroyed. |
| void PacketBuffer::Flush() { |
| buffer_.clear(); |
| } |
| |
| bool PacketBuffer::Empty() const { |
| return buffer_.empty(); |
| } |
| |
| int PacketBuffer::InsertPacket(Packet&& packet, StatisticsCalculator* stats) { |
| if (packet.empty()) { |
| RTC_LOG(LS_WARNING) << "InsertPacket invalid packet"; |
| return kInvalidPacket; |
| } |
| |
| RTC_DCHECK_GE(packet.priority.codec_level, 0); |
| RTC_DCHECK_GE(packet.priority.red_level, 0); |
| |
| int return_val = kOK; |
| |
| packet.waiting_time = tick_timer_->GetNewStopwatch(); |
| |
| if (buffer_.size() >= max_number_of_packets_) { |
| // Buffer is full. Flush it. |
| Flush(); |
| stats->FlushedPacketBuffer(); |
| RTC_LOG(LS_WARNING) << "Packet buffer flushed"; |
| return_val = kFlushed; |
| } |
| |
| // Get an iterator pointing to the place in the buffer where the new packet |
| // should be inserted. The list is searched from the back, since the most |
| // likely case is that the new packet should be near the end of the list. |
| PacketList::reverse_iterator rit = std::find_if( |
| buffer_.rbegin(), buffer_.rend(), NewTimestampIsLarger(packet)); |
| |
| // The new packet is to be inserted to the right of |rit|. If it has the same |
| // timestamp as |rit|, which has a higher priority, do not insert the new |
| // packet to list. |
| if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) { |
| LogPacketDiscarded(packet.priority.codec_level, stats); |
| return return_val; |
| } |
| |
| // The new packet is to be inserted to the left of |it|. If it has the same |
| // timestamp as |it|, which has a lower priority, replace |it| with the new |
| // packet. |
| PacketList::iterator it = rit.base(); |
| if (it != buffer_.end() && packet.timestamp == it->timestamp) { |
| LogPacketDiscarded(it->priority.codec_level, stats); |
| it = buffer_.erase(it); |
| } |
| buffer_.insert(it, std::move(packet)); // Insert the packet at that position. |
| |
| return return_val; |
| } |
| |
| int PacketBuffer::InsertPacketList( |
| PacketList* packet_list, |
| const DecoderDatabase& decoder_database, |
| absl::optional<uint8_t>* current_rtp_payload_type, |
| absl::optional<uint8_t>* current_cng_rtp_payload_type, |
| StatisticsCalculator* stats) { |
| RTC_DCHECK(stats); |
| bool flushed = false; |
| for (auto& packet : *packet_list) { |
| if (decoder_database.IsComfortNoise(packet.payload_type)) { |
| if (*current_cng_rtp_payload_type && |
| **current_cng_rtp_payload_type != packet.payload_type) { |
| // New CNG payload type implies new codec type. |
| *current_rtp_payload_type = absl::nullopt; |
| Flush(); |
| flushed = true; |
| } |
| *current_cng_rtp_payload_type = packet.payload_type; |
| } else if (!decoder_database.IsDtmf(packet.payload_type)) { |
| // This must be speech. |
| if ((*current_rtp_payload_type && |
| **current_rtp_payload_type != packet.payload_type) || |
| (*current_cng_rtp_payload_type && |
| !EqualSampleRates(packet.payload_type, |
| **current_cng_rtp_payload_type, |
| decoder_database))) { |
| *current_cng_rtp_payload_type = absl::nullopt; |
| Flush(); |
| flushed = true; |
| } |
| *current_rtp_payload_type = packet.payload_type; |
| } |
| int return_val = InsertPacket(std::move(packet), stats); |
| if (return_val == kFlushed) { |
| // The buffer flushed, but this is not an error. We can still continue. |
| flushed = true; |
| } else if (return_val != kOK) { |
| // An error occurred. Delete remaining packets in list and return. |
| packet_list->clear(); |
| return return_val; |
| } |
| } |
| packet_list->clear(); |
| return flushed ? kFlushed : kOK; |
| } |
| |
| int PacketBuffer::NextTimestamp(uint32_t* next_timestamp) const { |
| if (Empty()) { |
| return kBufferEmpty; |
| } |
| if (!next_timestamp) { |
| return kInvalidPointer; |
| } |
| *next_timestamp = buffer_.front().timestamp; |
| return kOK; |
| } |
| |
| int PacketBuffer::NextHigherTimestamp(uint32_t timestamp, |
| uint32_t* next_timestamp) const { |
| if (Empty()) { |
| return kBufferEmpty; |
| } |
| if (!next_timestamp) { |
| return kInvalidPointer; |
| } |
| PacketList::const_iterator it; |
| for (it = buffer_.begin(); it != buffer_.end(); ++it) { |
| if (it->timestamp >= timestamp) { |
| // Found a packet matching the search. |
| *next_timestamp = it->timestamp; |
| return kOK; |
| } |
| } |
| return kNotFound; |
| } |
| |
| const Packet* PacketBuffer::PeekNextPacket() const { |
| return buffer_.empty() ? nullptr : &buffer_.front(); |
| } |
| |
| absl::optional<Packet> PacketBuffer::GetNextPacket() { |
| if (Empty()) { |
| // Buffer is empty. |
| return absl::nullopt; |
| } |
| |
| absl::optional<Packet> packet(std::move(buffer_.front())); |
| // Assert that the packet sanity checks in InsertPacket method works. |
| RTC_DCHECK(!packet->empty()); |
| buffer_.pop_front(); |
| |
| return packet; |
| } |
| |
| int PacketBuffer::DiscardNextPacket(StatisticsCalculator* stats) { |
| if (Empty()) { |
| return kBufferEmpty; |
| } |
| // Assert that the packet sanity checks in InsertPacket method works. |
| const Packet& packet = buffer_.front(); |
| RTC_DCHECK(!packet.empty()); |
| LogPacketDiscarded(packet.priority.codec_level, stats); |
| buffer_.pop_front(); |
| return kOK; |
| } |
| |
| void PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit, |
| uint32_t horizon_samples, |
| StatisticsCalculator* stats) { |
| buffer_.remove_if([timestamp_limit, horizon_samples, stats](const Packet& p) { |
| if (timestamp_limit == p.timestamp || |
| !IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples)) { |
| return false; |
| } |
| LogPacketDiscarded(p.priority.codec_level, stats); |
| return true; |
| }); |
| } |
| |
| void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit, |
| StatisticsCalculator* stats) { |
| DiscardOldPackets(timestamp_limit, 0, stats); |
| } |
| |
| void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type, |
| StatisticsCalculator* stats) { |
| buffer_.remove_if([payload_type, stats](const Packet& p) { |
| if (p.payload_type != payload_type) { |
| return false; |
| } |
| LogPacketDiscarded(p.priority.codec_level, stats); |
| return true; |
| }); |
| } |
| |
| size_t PacketBuffer::NumPacketsInBuffer() const { |
| return buffer_.size(); |
| } |
| |
| size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const { |
| size_t num_samples = 0; |
| size_t last_duration = last_decoded_length; |
| for (const Packet& packet : buffer_) { |
| if (packet.frame) { |
| // TODO(hlundin): Verify that it's fine to count all packets and remove |
| // this check. |
| if (packet.priority != Packet::Priority(0, 0)) { |
| continue; |
| } |
| size_t duration = packet.frame->Duration(); |
| if (duration > 0) { |
| last_duration = duration; // Save the most up-to-date (valid) duration. |
| } |
| } |
| num_samples += last_duration; |
| } |
| return num_samples; |
| } |
| |
| size_t PacketBuffer::GetSpanSamples(size_t last_decoded_length) const { |
| if (buffer_.size() == 0) { |
| return 0; |
| } |
| |
| size_t span = buffer_.back().timestamp - buffer_.front().timestamp; |
| if (buffer_.back().frame && buffer_.back().frame->Duration() > 0) { |
| span += buffer_.back().frame->Duration(); |
| } else { |
| span += last_decoded_length; |
| } |
| return span; |
| } |
| |
| bool PacketBuffer::ContainsDtxOrCngPacket( |
| const DecoderDatabase* decoder_database) const { |
| RTC_DCHECK(decoder_database); |
| for (const Packet& packet : buffer_) { |
| if ((packet.frame && packet.frame->IsDtxPacket()) || |
| decoder_database->IsComfortNoise(packet.payload_type)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| } // namespace webrtc |