Remove RtcpPacket dependency on rtcp_utility
and thus IP_PACKET_SIZE constant: Build() use BlockLength() instead of constant IP_PACKET_SIZE for packet capacity, adding extra checks about packet generation in tests. Build(callback) removed as unused. definitions reordered to follow style. BUG=webrtc:5260 Review-Url: https://codereview.webrtc.org/2270753002 Cr-Commit-Position: refs/heads/master@{#14647}
This commit is contained in:
@ -11,50 +11,23 @@
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
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namespace webrtc {
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namespace webrtc {
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namespace rtcp {
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namespace rtcp {
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namespace {
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constexpr size_t RtcpPacket::kHeaderLength;
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void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) {
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buffer[(*offset)++] = value;
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}
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void AssignUWord16(uint8_t* buffer, size_t* offset, uint16_t value) {
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ByteWriter<uint16_t>::WriteBigEndian(buffer + *offset, value);
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*offset += 2;
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}
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} // namespace
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rtc::Buffer RtcpPacket::Build() const {
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rtc::Buffer RtcpPacket::Build() const {
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rtc::Buffer packet(BlockLength());
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size_t length = 0;
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size_t length = 0;
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rtc::Buffer packet(IP_PACKET_SIZE);
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bool created = Create(packet.data(), &length, packet.capacity(), nullptr);
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RTC_DCHECK(created) << "Invalid packet is not supported.";
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RTC_DCHECK_EQ(length, packet.size())
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<< "BlockLength mispredicted size used by Create";
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class PacketVerifier : public PacketReadyCallback {
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public:
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explicit PacketVerifier(rtc::Buffer* packet)
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: called_(false), packet_(packet) {}
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virtual ~PacketVerifier() {}
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void OnPacketReady(uint8_t* data, size_t length) override {
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RTC_CHECK(!called_) << "Fragmentation not supported.";
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called_ = true;
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packet_->SetSize(length);
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}
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private:
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bool called_;
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rtc::Buffer* const packet_;
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} verifier(&packet);
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Create(packet.data(), &length, packet.capacity(), &verifier);
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OnBufferFull(packet.data(), &length, &verifier);
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return packet;
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return packet;
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}
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}
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bool RtcpPacket::Build(PacketReadyCallback* callback) const {
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uint8_t buffer[IP_PACKET_SIZE];
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return BuildExternalBuffer(buffer, IP_PACKET_SIZE, callback);
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}
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bool RtcpPacket::BuildExternalBuffer(uint8_t* buffer,
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bool RtcpPacket::BuildExternalBuffer(uint8_t* buffer,
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size_t max_length,
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size_t max_length,
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PacketReadyCallback* callback) const {
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PacketReadyCallback* callback) const {
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@ -66,9 +39,10 @@ bool RtcpPacket::BuildExternalBuffer(uint8_t* buffer,
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bool RtcpPacket::OnBufferFull(uint8_t* packet,
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bool RtcpPacket::OnBufferFull(uint8_t* packet,
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size_t* index,
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size_t* index,
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RtcpPacket::PacketReadyCallback* callback) const {
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PacketReadyCallback* callback) const {
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if (*index == 0)
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if (*index == 0)
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return false;
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return false;
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RTC_DCHECK(callback) << "Fragmentation not supported.";
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callback->OnPacketReady(packet, *index);
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callback->OnPacketReady(packet, *index);
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*index = 0;
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*index = 0;
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return true;
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return true;
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@ -76,9 +50,10 @@ bool RtcpPacket::OnBufferFull(uint8_t* packet,
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size_t RtcpPacket::HeaderLength() const {
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size_t RtcpPacket::HeaderLength() const {
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size_t length_in_bytes = BlockLength();
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size_t length_in_bytes = BlockLength();
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// Length in 32-bit words minus 1.
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RTC_DCHECK_GT(length_in_bytes, 0u);
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assert(length_in_bytes > 0);
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RTC_DCHECK_EQ(length_in_bytes % 4, 0u) << "Padding not supported";
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return ((length_in_bytes + 3) / 4) - 1;
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// Length in 32-bit words without common header.
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return (length_in_bytes - kHeaderLength) / 4;
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}
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}
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// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
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// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
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@ -89,18 +64,21 @@ size_t RtcpPacket::HeaderLength() const {
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// |V=2|P| RC/FMT | PT | length |
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// |V=2|P| RC/FMT | PT | length |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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void RtcpPacket::CreateHeader(
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void RtcpPacket::CreateHeader(
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uint8_t count_or_format, // Depends on packet type.
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uint8_t count_or_format, // Depends on packet type.
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uint8_t packet_type,
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uint8_t packet_type,
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size_t length,
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size_t length,
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uint8_t* buffer,
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uint8_t* buffer,
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size_t* pos) {
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size_t* pos) {
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assert(length <= 0xffff);
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RTC_DCHECK_LE(length, 0xffffU);
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const uint8_t kVersion = 2;
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RTC_DCHECK_LE(count_or_format, 0x1f);
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AssignUWord8(buffer, pos, (kVersion << 6) + count_or_format);
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constexpr uint8_t kVersionBits = 2 << 6;
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AssignUWord8(buffer, pos, packet_type);
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constexpr uint8_t kNoPaddingBit = 0 << 5;
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AssignUWord16(buffer, pos, length);
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buffer[*pos + 0] = kVersionBits | kNoPaddingBit | count_or_format;
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buffer[*pos + 1] = packet_type;
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buffer[*pos + 2] = (length >> 8) & 0xff;
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buffer[*pos + 3] = length & 0xff;
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*pos += kHeaderLength;
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}
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}
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} // namespace rtcp
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} // namespace rtcp
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@ -8,12 +8,11 @@
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* be found in the AUTHORS file in the root of the source tree.
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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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*
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*/
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/buffer.h"
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#include "webrtc/base/buffer.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
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namespace webrtc {
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namespace webrtc {
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namespace rtcp {
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namespace rtcp {
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@ -46,27 +45,24 @@ namespace rtcp {
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class RtcpPacket {
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class RtcpPacket {
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public:
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public:
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virtual ~RtcpPacket() {}
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// Callback used to signal that an RTCP packet is ready. Note that this may
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// Callback used to signal that an RTCP packet is ready. Note that this may
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// not contain all data in this RtcpPacket; if a packet cannot fit in
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// not contain all data in this RtcpPacket; if a packet cannot fit in
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// max_length bytes, it will be fragmented and multiple calls to this
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// max_length bytes, it will be fragmented and multiple calls to this
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// callback will be made.
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// callback will be made.
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class PacketReadyCallback {
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class PacketReadyCallback {
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public:
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public:
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virtual void OnPacketReady(uint8_t* data, size_t length) = 0;
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protected:
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PacketReadyCallback() {}
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PacketReadyCallback() {}
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virtual ~PacketReadyCallback() {}
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virtual ~PacketReadyCallback() {}
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virtual void OnPacketReady(uint8_t* data, size_t length) = 0;
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};
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};
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// Convenience method mostly used for test. Max length of IP_PACKET_SIZE is
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virtual ~RtcpPacket() {}
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// used, will cause assertion error if fragmentation occurs.
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rtc::Buffer Build() const;
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// Returns true if call to Create succeeded. A buffer of size
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// Convenience method mostly used for test. Creates packet without
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// IP_PACKET_SIZE will be allocated and reused between calls to callback.
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// fragmentation using BlockLength() to allocate big enough buffer.
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bool Build(PacketReadyCallback* callback) const;
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rtc::Buffer Build() const;
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// Returns true if call to Create succeeded. Provided buffer reference
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// Returns true if call to Create succeeded. Provided buffer reference
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// will be used for all calls to callback.
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// will be used for all calls to callback.
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@ -86,21 +82,21 @@ class RtcpPacket {
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PacketReadyCallback* callback) const = 0;
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PacketReadyCallback* callback) const = 0;
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protected:
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protected:
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// Size of the rtcp common header.
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static constexpr size_t kHeaderLength = 4;
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RtcpPacket() {}
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RtcpPacket() {}
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static void CreateHeader(uint8_t count_or_format,
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static void CreateHeader(uint8_t count_or_format,
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uint8_t packet_type,
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uint8_t packet_type,
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size_t block_length, // Size in 32bit words - 1.
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size_t block_length, // Payload size in 32bit words.
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uint8_t* buffer,
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uint8_t* buffer,
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size_t* pos);
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size_t* pos);
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bool OnBufferFull(uint8_t* packet,
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bool OnBufferFull(uint8_t* packet,
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size_t* index,
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size_t* index,
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RtcpPacket::PacketReadyCallback* callback) const;
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PacketReadyCallback* callback) const;
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// Size of the rtcp packet as written in header.
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size_t HeaderLength() const;
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size_t HeaderLength() const;
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static const size_t kHeaderLength = 4;
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};
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};
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} // namespace rtcp
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} // namespace rtcp
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} // namespace webrtc
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} // namespace webrtc
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