mirror of
https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
265 lines
7.8 KiB
C++
265 lines
7.8 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/modules/audio_coding/neteq/decoder_database.h"
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#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
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namespace webrtc {
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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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PacketBuffer::PacketBuffer(size_t max_number_of_packets)
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: max_number_of_packets_(max_number_of_packets) {}
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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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int PacketBuffer::InsertPacket(Packet* packet) {
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if (!packet || !packet->payload) {
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if (packet) {
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delete packet;
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}
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return kInvalidPacket;
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}
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int return_val = kOK;
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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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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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buffer_.insert(rit.base(), 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(PacketList* packet_list,
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const DecoderDatabase& decoder_database,
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uint8_t* current_rtp_payload_type,
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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->header.payloadType)) {
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if (*current_cng_rtp_payload_type != 0xFF &&
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*current_cng_rtp_payload_type != packet->header.payloadType) {
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// New CNG payload type implies new codec type.
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*current_rtp_payload_type = 0xFF;
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Flush();
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flushed = true;
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}
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*current_cng_rtp_payload_type = packet->header.payloadType;
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} else if (!decoder_database.IsDtmf(packet->header.payloadType)) {
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// This must be speech.
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if (*current_rtp_payload_type != 0xFF &&
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*current_rtp_payload_type != packet->header.payloadType) {
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*current_cng_rtp_payload_type = 0xFF;
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Flush();
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flushed = true;
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}
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*current_rtp_payload_type = packet->header.payloadType;
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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()->header.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)->header.timestamp >= timestamp) {
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// Found a packet matching the search.
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*next_timestamp = (*it)->header.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 RTPHeader* PacketBuffer::NextRtpHeader() const {
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if (Empty()) {
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return NULL;
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}
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return const_cast<const RTPHeader*>(&(buffer_.front()->header));
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}
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Packet* PacketBuffer::GetNextPacket(int* 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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assert(packet && packet->payload);
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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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if (discard_count) {
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*discard_count = 0;
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}
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while (!Empty() &&
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buffer_.front()->header.timestamp == packet->header.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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if (discard_count) {
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++(*discard_count);
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}
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}
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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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assert(buffer_.front());
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assert(buffer_.front()->payload);
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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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while (!Empty() &&
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timestamp_limit != buffer_.front()->header.timestamp &&
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static_cast<uint32_t>(timestamp_limit
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- buffer_.front()->header.timestamp) <
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0xFFFFFFFF / 2) {
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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::NumSamplesInBuffer(DecoderDatabase* decoder_database,
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int last_decoded_length) const {
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PacketList::const_iterator it;
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int num_samples = 0;
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int last_duration = last_decoded_length;
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for (it = buffer_.begin(); it != buffer_.end(); ++it) {
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Packet* packet = (*it);
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AudioDecoder* decoder =
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decoder_database->GetDecoder(packet->header.payloadType);
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if (decoder) {
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int duration;
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if (packet->sync_packet) {
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duration = last_duration;
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} else if (packet->primary) {
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duration =
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decoder->PacketDuration(packet->payload, packet->payload_length);
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} else {
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continue;
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}
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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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void PacketBuffer::IncrementWaitingTimes(int inc) {
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PacketList::iterator it;
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for (it = buffer_.begin(); it != buffer_.end(); ++it) {
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(*it)->waiting_time += inc;
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}
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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->payload;
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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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