blob: 190449b3ddfe6bdde9561cc16a0ce6d1387ed767 [file] [log] [blame]
/*
* 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.
*/
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include "webrtc/base/logging.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
namespace webrtc {
using RtpUtility::Payload;
RtpReceiver* RtpReceiver::CreateVideoReceiver(
Clock* clock,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry) {
if (!incoming_payload_callback)
incoming_payload_callback = NullObjectRtpData();
if (!incoming_messages_callback)
incoming_messages_callback = NullObjectRtpFeedback();
return new RtpReceiverImpl(
clock, incoming_messages_callback, rtp_payload_registry,
RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
}
RtpReceiver* RtpReceiver::CreateAudioReceiver(
Clock* clock,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry) {
if (!incoming_payload_callback)
incoming_payload_callback = NullObjectRtpData();
if (!incoming_messages_callback)
incoming_messages_callback = NullObjectRtpFeedback();
return new RtpReceiverImpl(
clock, incoming_messages_callback, rtp_payload_registry,
RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
}
RtpReceiverImpl::RtpReceiverImpl(
Clock* clock,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry,
RTPReceiverStrategy* rtp_media_receiver)
: clock_(clock),
rtp_payload_registry_(rtp_payload_registry),
rtp_media_receiver_(rtp_media_receiver),
cb_rtp_feedback_(incoming_messages_callback),
last_receive_time_(0),
last_received_payload_length_(0),
ssrc_(0),
num_csrcs_(0),
current_remote_csrc_(),
last_received_timestamp_(0),
last_received_frame_time_ms_(-1),
last_received_sequence_number_(0) {
assert(incoming_messages_callback);
memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
}
RtpReceiverImpl::~RtpReceiverImpl() {
for (int i = 0; i < num_csrcs_; ++i) {
cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
}
}
int32_t RtpReceiverImpl::RegisterReceivePayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
const uint32_t frequency,
const size_t channels,
const uint32_t rate) {
rtc::CritScope lock(&critical_section_rtp_receiver_);
// TODO(phoglund): Try to streamline handling of the RED codec and some other
// cases which makes it necessary to keep track of whether we created a
// payload or not.
bool created_new_payload = false;
int32_t result = rtp_payload_registry_->RegisterReceivePayload(
payload_name, payload_type, frequency, channels, rate,
&created_new_payload);
if (created_new_payload) {
if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type,
frequency) != 0) {
LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/"
<< static_cast<int>(payload_type);
return -1;
}
}
return result;
}
int32_t RtpReceiverImpl::DeRegisterReceivePayload(
const int8_t payload_type) {
rtc::CritScope lock(&critical_section_rtp_receiver_);
return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
}
uint32_t RtpReceiverImpl::SSRC() const {
rtc::CritScope lock(&critical_section_rtp_receiver_);
return ssrc_;
}
// Get remote CSRC.
int32_t RtpReceiverImpl::CSRCs(uint32_t array_of_csrcs[kRtpCsrcSize]) const {
rtc::CritScope lock(&critical_section_rtp_receiver_);
assert(num_csrcs_ <= kRtpCsrcSize);
if (num_csrcs_ > 0) {
memcpy(array_of_csrcs, current_remote_csrc_, sizeof(uint32_t)*num_csrcs_);
}
return num_csrcs_;
}
int32_t RtpReceiverImpl::Energy(
uint8_t array_of_energy[kRtpCsrcSize]) const {
return rtp_media_receiver_->Energy(array_of_energy);
}
bool RtpReceiverImpl::IncomingRtpPacket(
const RTPHeader& rtp_header,
const uint8_t* payload,
size_t payload_length,
PayloadUnion payload_specific,
bool in_order) {
// Trigger our callbacks.
CheckSSRCChanged(rtp_header);
int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
bool is_red = false;
if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red,
&payload_specific) == -1) {
if (payload_length == 0) {
// OK, keep-alive packet.
return true;
}
LOG(LS_WARNING) << "Receiving invalid payload type.";
return false;
}
WebRtcRTPHeader webrtc_rtp_header;
memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
webrtc_rtp_header.header = rtp_header;
CheckCSRC(webrtc_rtp_header);
size_t payload_data_length = payload_length - rtp_header.paddingLength;
bool is_first_packet_in_frame = false;
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
if (HaveReceivedFrame()) {
is_first_packet_in_frame =
last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
last_received_timestamp_ != rtp_header.timestamp;
} else {
is_first_packet_in_frame = true;
}
}
int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
&webrtc_rtp_header, payload_specific, is_red, payload, payload_length,
clock_->TimeInMilliseconds(), is_first_packet_in_frame);
if (ret_val < 0) {
return false;
}
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
last_receive_time_ = clock_->TimeInMilliseconds();
last_received_payload_length_ = payload_data_length;
if (in_order) {
if (last_received_timestamp_ != rtp_header.timestamp) {
last_received_timestamp_ = rtp_header.timestamp;
last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
}
last_received_sequence_number_ = rtp_header.sequenceNumber;
}
}
return true;
}
TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
return rtp_media_receiver_->GetTelephoneEventHandler();
}
bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
rtc::CritScope lock(&critical_section_rtp_receiver_);
if (!HaveReceivedFrame())
return false;
*timestamp = last_received_timestamp_;
return true;
}
bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
rtc::CritScope lock(&critical_section_rtp_receiver_);
if (!HaveReceivedFrame())
return false;
*receive_time_ms = last_received_frame_time_ms_;
return true;
}
bool RtpReceiverImpl::HaveReceivedFrame() const {
return last_received_frame_time_ms_ >= 0;
}
// Implementation note: must not hold critsect when called.
void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
bool new_ssrc = false;
bool re_initialize_decoder = false;
char payload_name[RTP_PAYLOAD_NAME_SIZE];
size_t channels = 1;
uint32_t rate = 0;
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
int8_t last_received_payload_type =
rtp_payload_registry_->last_received_payload_type();
if (ssrc_ != rtp_header.ssrc ||
(last_received_payload_type == -1 && ssrc_ == 0)) {
// We need the payload_type_ to make the call if the remote SSRC is 0.
new_ssrc = true;
last_received_timestamp_ = 0;
last_received_sequence_number_ = 0;
last_received_frame_time_ms_ = -1;
// Do we have a SSRC? Then the stream is restarted.
if (ssrc_ != 0) {
// Do we have the same codec? Then re-initialize coder.
if (rtp_header.payloadType == last_received_payload_type) {
re_initialize_decoder = true;
const Payload* payload = rtp_payload_registry_->PayloadTypeToPayload(
rtp_header.payloadType);
if (!payload) {
return;
}
payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
if (payload->audio) {
channels = payload->typeSpecific.Audio.channels;
rate = payload->typeSpecific.Audio.rate;
}
}
}
ssrc_ = rtp_header.ssrc;
}
}
if (new_ssrc) {
// We need to get this to our RTCP sender and receiver.
// We need to do this outside critical section.
cb_rtp_feedback_->OnIncomingSSRCChanged(rtp_header.ssrc);
}
if (re_initialize_decoder) {
if (-1 ==
cb_rtp_feedback_->OnInitializeDecoder(
rtp_header.payloadType, payload_name,
rtp_header.payload_type_frequency, channels, rate)) {
// New stream, same codec.
LOG(LS_ERROR) << "Failed to create decoder for payload type: "
<< static_cast<int>(rtp_header.payloadType);
}
}
}
// Implementation note: must not hold critsect when called.
// TODO(phoglund): Move as much as possible of this code path into the media
// specific receivers. Basically this method goes through a lot of trouble to
// compute something which is only used by the media specific parts later. If
// this code path moves we can get rid of some of the rtp_receiver ->
// media_specific interface (such as CheckPayloadChange, possibly get/set
// last known payload).
int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
const int8_t first_payload_byte,
bool* is_red,
PayloadUnion* specific_payload) {
bool re_initialize_decoder = false;
char payload_name[RTP_PAYLOAD_NAME_SIZE];
int8_t payload_type = rtp_header.payloadType;
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
int8_t last_received_payload_type =
rtp_payload_registry_->last_received_payload_type();
// TODO(holmer): Remove this code when RED parsing has been broken out from
// RtpReceiverAudio.
if (payload_type != last_received_payload_type) {
if (rtp_payload_registry_->red_payload_type() == payload_type) {
// Get the real codec payload type.
payload_type = first_payload_byte & 0x7f;
*is_red = true;
if (rtp_payload_registry_->red_payload_type() == payload_type) {
// Invalid payload type, traced by caller. If we proceeded here,
// this would be set as |_last_received_payload_type|, and we would no
// longer catch corrupt packets at this level.
return -1;
}
// When we receive RED we need to check the real payload type.
if (payload_type == last_received_payload_type) {
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
return 0;
}
}
bool should_discard_changes = false;
rtp_media_receiver_->CheckPayloadChanged(
payload_type, specific_payload,
&should_discard_changes);
if (should_discard_changes) {
*is_red = false;
return 0;
}
const Payload* payload =
rtp_payload_registry_->PayloadTypeToPayload(payload_type);
if (!payload) {
// Not a registered payload type.
return -1;
}
payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
rtp_payload_registry_->set_last_received_payload_type(payload_type);
re_initialize_decoder = true;
rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific);
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
if (!payload->audio) {
bool media_type_unchanged =
rtp_payload_registry_->ReportMediaPayloadType(payload_type);
if (media_type_unchanged) {
// Only reset the decoder if the media codec type has changed.
re_initialize_decoder = false;
}
}
} else {
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
*is_red = false;
}
} // End critsect.
if (re_initialize_decoder) {
if (-1 ==
rtp_media_receiver_->InvokeOnInitializeDecoder(
cb_rtp_feedback_, payload_type, payload_name, *specific_payload)) {
return -1; // Wrong payload type.
}
}
return 0;
}
// Implementation note: must not hold critsect when called.
void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
int32_t num_csrcs_diff = 0;
uint32_t old_remote_csrc[kRtpCsrcSize];
uint8_t old_num_csrcs = 0;
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
if (!rtp_media_receiver_->ShouldReportCsrcChanges(
rtp_header.header.payloadType)) {
return;
}
old_num_csrcs = num_csrcs_;
if (old_num_csrcs > 0) {
// Make a copy of old.
memcpy(old_remote_csrc, current_remote_csrc_,
num_csrcs_ * sizeof(uint32_t));
}
const uint8_t num_csrcs = rtp_header.header.numCSRCs;
if ((num_csrcs > 0) && (num_csrcs <= kRtpCsrcSize)) {
// Copy new.
memcpy(current_remote_csrc_,
rtp_header.header.arrOfCSRCs,
num_csrcs * sizeof(uint32_t));
}
if (num_csrcs > 0 || old_num_csrcs > 0) {
num_csrcs_diff = num_csrcs - old_num_csrcs;
num_csrcs_ = num_csrcs; // Update stored CSRCs.
} else {
// No change.
return;
}
} // End critsect.
bool have_called_callback = false;
// Search for new CSRC in old array.
for (uint8_t i = 0; i < rtp_header.header.numCSRCs; ++i) {
const uint32_t csrc = rtp_header.header.arrOfCSRCs[i];
bool found_match = false;
for (uint8_t j = 0; j < old_num_csrcs; ++j) {
if (csrc == old_remote_csrc[j]) { // old list
found_match = true;
break;
}
}
if (!found_match && csrc) {
// Didn't find it, report it as new.
have_called_callback = true;
cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, true);
}
}
// Search for old CSRC in new array.
for (uint8_t i = 0; i < old_num_csrcs; ++i) {
const uint32_t csrc = old_remote_csrc[i];
bool found_match = false;
for (uint8_t j = 0; j < rtp_header.header.numCSRCs; ++j) {
if (csrc == rtp_header.header.arrOfCSRCs[j]) {
found_match = true;
break;
}
}
if (!found_match && csrc) {
// Did not find it, report as removed.
have_called_callback = true;
cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, false);
}
}
if (!have_called_callback) {
// If the CSRC list contain non-unique entries we will end up here.
// Using CSRC 0 to signal this event, not interop safe, other
// implementations might have CSRC 0 as a valid value.
if (num_csrcs_diff > 0) {
cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
} else if (num_csrcs_diff < 0) {
cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
}
}
}
} // namespace webrtc