Only three items in the (rather large) header were actually used after InsertPacket: payloadType, timestamp and sequenceNumber. They are now put directly into Packet. This saves 129 bytes per Packet that no longer need to be allocated and deallocated. This also works towards decoupling NetEq from RTP. As part of that, I've moved the NACK code earlier in InsertPacketInternal, together with other things that directly reference the RTPHeader. BUG=webrtc:6549 Review-Url: https://codereview.webrtc.org/2411183003 Cr-Commit-Position: refs/heads/master@{#14658}
316 lines
9.6 KiB
C++
316 lines
9.6 KiB
C++
/*
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* Copyright (c) 2012 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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// This is the implementation of the PacketBuffer class. It is mostly based on
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// an STL list. The list is kept sorted at all times so that the next packet to
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// decode is at the beginning of the list.
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#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
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#include <algorithm> // find_if()
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#include "webrtc/base/logging.h"
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#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
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#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
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#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
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namespace webrtc {
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namespace {
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// Predicate used when inserting packets in the buffer list.
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// Operator() returns true when |packet| goes before |new_packet|.
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class NewTimestampIsLarger {
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public:
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explicit NewTimestampIsLarger(const Packet* new_packet)
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: new_packet_(new_packet) {
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}
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bool operator()(Packet* packet) {
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return (*new_packet_ >= *packet);
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}
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private:
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const Packet* new_packet_;
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};
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// Returns true if both payload types are known to the decoder database, and
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// have the same sample rate.
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bool EqualSampleRates(uint8_t pt1,
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uint8_t pt2,
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const DecoderDatabase& decoder_database) {
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auto di1 = decoder_database.GetDecoderInfo(pt1);
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auto di2 = decoder_database.GetDecoderInfo(pt2);
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return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
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}
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} // namespace
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PacketBuffer::PacketBuffer(size_t max_number_of_packets,
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const TickTimer* tick_timer)
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: max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {}
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// Destructor. All packets in the buffer will be destroyed.
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PacketBuffer::~PacketBuffer() {
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Flush();
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}
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// Flush the buffer. All packets in the buffer will be destroyed.
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void PacketBuffer::Flush() {
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DeleteAllPackets(&buffer_);
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}
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bool PacketBuffer::Empty() const {
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return buffer_.empty();
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}
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int PacketBuffer::InsertPacket(Packet* packet) {
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if (!packet || packet->empty()) {
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if (packet) {
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delete packet;
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}
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LOG(LS_WARNING) << "InsertPacket invalid packet";
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return kInvalidPacket;
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}
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RTC_DCHECK_GE(packet->priority.codec_level, 0);
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RTC_DCHECK_GE(packet->priority.red_level, 0);
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int return_val = kOK;
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packet->waiting_time = tick_timer_->GetNewStopwatch();
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if (buffer_.size() >= max_number_of_packets_) {
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// Buffer is full. Flush it.
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Flush();
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LOG(LS_WARNING) << "Packet buffer flushed";
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return_val = kFlushed;
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}
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// Get an iterator pointing to the place in the buffer where the new packet
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// should be inserted. The list is searched from the back, since the most
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// likely case is that the new packet should be near the end of the list.
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PacketList::reverse_iterator rit = std::find_if(
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buffer_.rbegin(), buffer_.rend(),
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NewTimestampIsLarger(packet));
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// The new packet is to be inserted to the right of |rit|. If it has the same
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// timestamp as |rit|, which has a higher priority, do not insert the new
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// packet to list.
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if (rit != buffer_.rend() && packet->timestamp == (*rit)->timestamp) {
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delete packet;
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return return_val;
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}
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// The new packet is to be inserted to the left of |it|. If it has the same
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// timestamp as |it|, which has a lower priority, replace |it| with the new
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// packet.
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PacketList::iterator it = rit.base();
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if (it != buffer_.end() && packet->timestamp == (*it)->timestamp) {
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delete *it;
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it = buffer_.erase(it);
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}
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buffer_.insert(it, packet); // Insert the packet at that position.
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return return_val;
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}
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int PacketBuffer::InsertPacketList(
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PacketList* packet_list,
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const DecoderDatabase& decoder_database,
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rtc::Optional<uint8_t>* current_rtp_payload_type,
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rtc::Optional<uint8_t>* current_cng_rtp_payload_type) {
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bool flushed = false;
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while (!packet_list->empty()) {
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Packet* packet = packet_list->front();
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if (decoder_database.IsComfortNoise(packet->payload_type)) {
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if (*current_cng_rtp_payload_type &&
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**current_cng_rtp_payload_type != packet->payload_type) {
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// New CNG payload type implies new codec type.
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*current_rtp_payload_type = rtc::Optional<uint8_t>();
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Flush();
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flushed = true;
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}
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*current_cng_rtp_payload_type =
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rtc::Optional<uint8_t>(packet->payload_type);
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} else if (!decoder_database.IsDtmf(packet->payload_type)) {
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// This must be speech.
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if ((*current_rtp_payload_type &&
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**current_rtp_payload_type != packet->payload_type) ||
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(*current_cng_rtp_payload_type &&
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!EqualSampleRates(packet->payload_type,
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**current_cng_rtp_payload_type,
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decoder_database))) {
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*current_cng_rtp_payload_type = rtc::Optional<uint8_t>();
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Flush();
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flushed = true;
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}
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*current_rtp_payload_type = rtc::Optional<uint8_t>(packet->payload_type);
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}
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int return_val = InsertPacket(packet);
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packet_list->pop_front();
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if (return_val == kFlushed) {
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// The buffer flushed, but this is not an error. We can still continue.
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flushed = true;
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} else if (return_val != kOK) {
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// An error occurred. Delete remaining packets in list and return.
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DeleteAllPackets(packet_list);
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return return_val;
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}
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}
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return flushed ? kFlushed : kOK;
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}
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int PacketBuffer::NextTimestamp(uint32_t* next_timestamp) const {
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if (Empty()) {
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return kBufferEmpty;
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}
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if (!next_timestamp) {
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return kInvalidPointer;
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}
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*next_timestamp = buffer_.front()->timestamp;
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return kOK;
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}
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int PacketBuffer::NextHigherTimestamp(uint32_t timestamp,
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uint32_t* next_timestamp) const {
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if (Empty()) {
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return kBufferEmpty;
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}
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if (!next_timestamp) {
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return kInvalidPointer;
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}
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PacketList::const_iterator it;
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for (it = buffer_.begin(); it != buffer_.end(); ++it) {
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if ((*it)->timestamp >= timestamp) {
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// Found a packet matching the search.
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*next_timestamp = (*it)->timestamp;
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return kOK;
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}
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}
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return kNotFound;
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}
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const Packet* PacketBuffer::PeekNextPacket() const {
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return buffer_.empty() ? nullptr : buffer_.front();
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}
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Packet* PacketBuffer::GetNextPacket(size_t* discard_count) {
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if (Empty()) {
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// Buffer is empty.
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return NULL;
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}
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Packet* packet = buffer_.front();
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// Assert that the packet sanity checks in InsertPacket method works.
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RTC_DCHECK(packet && !packet->empty());
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buffer_.pop_front();
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// Discard other packets with the same timestamp. These are duplicates or
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// redundant payloads that should not be used.
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size_t discards = 0;
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while (!Empty() && buffer_.front()->timestamp == packet->timestamp) {
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if (DiscardNextPacket() != kOK) {
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assert(false); // Must be ok by design.
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}
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++discards;
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}
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// The way of inserting packet should not cause any packet discarding here.
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// TODO(minyue): remove |discard_count|.
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assert(discards == 0);
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if (discard_count)
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*discard_count = discards;
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return packet;
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}
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int PacketBuffer::DiscardNextPacket() {
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if (Empty()) {
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return kBufferEmpty;
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}
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// Assert that the packet sanity checks in InsertPacket method works.
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RTC_DCHECK(buffer_.front());
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RTC_DCHECK(!buffer_.front()->empty());
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DeleteFirstPacket(&buffer_);
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return kOK;
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}
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int PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
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uint32_t horizon_samples) {
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while (!Empty() && timestamp_limit != buffer_.front()->timestamp &&
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IsObsoleteTimestamp(buffer_.front()->timestamp, timestamp_limit,
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horizon_samples)) {
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if (DiscardNextPacket() != kOK) {
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assert(false); // Must be ok by design.
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}
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}
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return 0;
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}
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int PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit) {
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return DiscardOldPackets(timestamp_limit, 0);
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}
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void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type) {
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for (auto it = buffer_.begin(); it != buffer_.end(); /* */) {
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Packet* packet = *it;
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if (packet->payload_type == payload_type) {
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delete packet;
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it = buffer_.erase(it);
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} else {
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++it;
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}
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}
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}
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size_t PacketBuffer::NumPacketsInBuffer() const {
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return buffer_.size();
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}
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size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
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size_t num_samples = 0;
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size_t last_duration = last_decoded_length;
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for (Packet* packet : buffer_) {
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if (packet->frame) {
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// TODO(hlundin): Verify that it's fine to count all packets and remove
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// this check.
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if (packet->priority != Packet::Priority(0, 0)) {
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continue;
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}
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size_t duration = packet->frame->Duration();
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if (duration > 0) {
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last_duration = duration; // Save the most up-to-date (valid) duration.
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}
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}
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num_samples += last_duration;
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}
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return num_samples;
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}
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bool PacketBuffer::DeleteFirstPacket(PacketList* packet_list) {
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if (packet_list->empty()) {
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return false;
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}
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Packet* first_packet = packet_list->front();
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delete first_packet;
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packet_list->pop_front();
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return true;
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}
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void PacketBuffer::DeleteAllPackets(PacketList* packet_list) {
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while (DeleteFirstPacket(packet_list)) {
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// Continue while the list is not empty.
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}
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}
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void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
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*num_packets = static_cast<int>(buffer_.size());
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*max_num_packets = static_cast<int>(max_number_of_packets_);
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}
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} // namespace webrtc
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