mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 05:40:42 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions

m123 merge fixes

Browse source
This commit is contained in:
Jim Gustafson 2024-06-13 10:17:08 -07:00
parent c43adafcd5
commit 4cf759baf3
7 changed files with 72 additions and 54 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

5
api/neteq/neteq.cc
View file

@ -26,7 +26,10 @@ std::string NetEq::Config::ToString() const {
rtc::SimpleStringBuilder ss(buf);
ss << "sample_rate_hz=" << sample_rate_hz
<< ", max_packets_in_buffer=" << max_packets_in_buffer
<< ", min_delay_ms=" << min_delay_ms << ", enable_fast_accelerate="
<< ", min_delay_ms=" << min_delay_ms
// RingRTC change to configure the jitter buffer's max target delay.
<< ", max_delay_ms=" << max_delay_ms
<< ", enable_fast_accelerate="
<< (enable_fast_accelerate ? "true" : "false")
<< ", enable_muted_state=" << (enable_muted_state ? "true" : "false")
<< ", enable_rtx_handling=" << (enable_rtx_handling ? "true" : "false");

13
audio/channel_receive.cc
View file

@ -72,7 +72,9 @@ acm2::AcmReceiver::Config AcmConfig(
absl::optional<AudioCodecPairId> codec_pair_id,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms) {
int jitter_buffer_min_delay_ms,
// RingRTC change to configure the jitter buffer's max target delay.
int jitter_buffer_max_target_delay_ms) {
acm2::AcmReceiver::Config acm_config;
acm_config.neteq_factory = neteq_factory;
acm_config.decoder_factory = decoder_factory;
@ -81,6 +83,8 @@ acm2::AcmReceiver::Config AcmConfig(
acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout;
acm_config.neteq_config.enable_muted_state = true;
acm_config.neteq_config.min_delay_ms = jitter_buffer_min_delay_ms;
// RingRTC change to configure the jitter buffer's max target delay.
acm_config.neteq_config.max_delay_ms = jitter_buffer_max_target_delay_ms;
return acm_config;
}
@ -561,7 +565,9 @@ ChannelReceive::ChannelReceive(
codec_pair_id,
jitter_buffer_max_packets,
jitter_buffer_fast_playout,
jitter_buffer_min_delay_ms)),
jitter_buffer_min_delay_ms,
// RingRTC change to configure the jitter buffer's max target delay.
jitter_buffer_max_target_delay_ms)),
_outputAudioLevel(),
clock_(clock),
ntp_estimator_(clock),
@ -579,9 +585,6 @@ ChannelReceive::ChannelReceive(
network_thread_checker_.Detach();
// RingRTC change to configure the jitter buffer's max target delay.
acm_receiver_.SetMaximumDelay(jitter_buffer_max_target_delay_ms);
rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true);
RtpRtcpInterface::Configuration configuration;
configuration.clock = clock;

10
p2p/BUILD.gn
View file

@ -67,6 +67,8 @@ rtc_library("rtc_p2p") {
":ice_controller_factory_interface",
":ice_controller_interface",
":ice_credentials_iterator",
# RingRTC change to add ICE forking
":ice_gatherer",
":ice_switch_reason",
":ice_transport_internal",
":p2p_constants",
@ -505,6 +507,14 @@ rtc_library("ice_credentials_iterator") {
]
}
# RingRTC change to add ICE forking
rtc_library("ice_gatherer") {
sources = [
"base/ice_gatherer.cc",
"base/ice_gatherer.h",
]
}
rtc_library("ice_switch_reason") {
sources = [
"base/ice_switch_reason.cc",

1
p2p/base/p2p_transport_channel.cc
View file

@ -1277,6 +1277,7 @@ void P2PTransportChannel::OnRoleConflict(PortInterface* port) {
// from Transport.
}
// RingRTC change to add ICE forking
void P2PTransportChannel::OnRoleConflictIgnored(PortInterface* port) {
RTC_LOG(LS_ERROR) << "Ignored conflict. This is bad.";
}

1
p2p/base/p2p_transport_channel.h
View file

@ -359,6 +359,7 @@ class RTC_EXPORT P2PTransportChannel : public IceTransportInternal,
// Returns true if the port is found and removed from `ports_`.
bool PrunePort(PortInterface* port);
void OnRoleConflict(PortInterface* port);
// RingRTC change to add ICE forking
void OnRoleConflictIgnored(PortInterface* port);
void OnConnectionStateChange(Connection* connection);

6
pc/media_session.cc
View file

@ -2180,12 +2180,12 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForAnswer(
}
// If this section is part of a bundle, bundle_transport is non-null.
// Then require_transport_attributes is false - we can handle sections
// without the DTLS parameters. Otherwise, transport attributes MUST
// be present.
// without the DTLS parameters. For rejected m-lines it does not matter.
// Otherwise, transport attributes MUST be present.
std::unique_ptr<TransportDescription> transport = CreateTransportAnswer(
media_description_options.mid, offer_description,
media_description_options.transport_options, current_description,
bundle_transport == nullptr, ice_credentials);
!offer_content->rejected && bundle_transport == nullptr, ice_credentials);
if (!transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,

90
ringrtc/rffi/src/stats_observer.cc
View file

@ -40,23 +40,23 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
if (*stat->kind == "audio" && (*stat->mid == "audio" || absl::StartsWith(*stat->mid, "local-audio"))) {
AudioSenderStatistics audio_sender = {0};
audio_sender.ssrc = stat->ssrc.ValueOrDefault(0);
audio_sender.packets_sent = stat->packets_sent.ValueOrDefault(0);
audio_sender.bytes_sent = stat->bytes_sent.ValueOrDefault(0);
audio_sender.ssrc = stat->ssrc.value_or(0);
audio_sender.packets_sent = stat->packets_sent.value_or(0);
audio_sender.bytes_sent = stat->bytes_sent.value_or(0);
if (stat->remote_id.has_value()) {
auto remote_stat = report->GetAs<RTCRemoteInboundRtpStreamStats>(*stat->remote_id);
if (remote_stat) {
audio_sender.remote_packets_lost = remote_stat->packets_lost.ValueOrDefault(0);
audio_sender.remote_jitter = remote_stat->jitter.ValueOrDefault(0.0);
audio_sender.remote_round_trip_time = remote_stat->round_trip_time.ValueOrDefault(0.0);
audio_sender.remote_packets_lost = remote_stat->packets_lost.value_or(0);
audio_sender.remote_jitter = remote_stat->jitter.value_or(0.0);
audio_sender.remote_round_trip_time = remote_stat->round_trip_time.value_or(0.0);
}
}
if (stat->media_source_id.has_value()) {
auto audio_source_stat = report->GetAs<RTCAudioSourceStats>(*stat->media_source_id);
if (audio_source_stat) {
audio_sender.total_audio_energy = audio_source_stat->total_audio_energy.ValueOrDefault(0.0);
audio_sender.total_audio_energy = audio_source_stat->total_audio_energy.value_or(0.0);
}
}
@ -64,19 +64,19 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
} else if (*stat->kind == "video" && (*stat->mid == "video" || absl::StartsWith(*stat->mid, "local-video"))) {
VideoSenderStatistics video_sender = {0};
video_sender.ssrc = stat->ssrc.ValueOrDefault(0);
video_sender.packets_sent = stat->packets_sent.ValueOrDefault(0);
video_sender.bytes_sent = stat->bytes_sent.ValueOrDefault(0);
video_sender.frames_encoded = stat->frames_encoded.ValueOrDefault(0);
video_sender.key_frames_encoded = stat->key_frames_encoded.ValueOrDefault(0);
video_sender.total_encode_time = stat->total_encode_time.ValueOrDefault(0.0);
video_sender.frame_width = stat->frame_width.ValueOrDefault(0);
video_sender.frame_height = stat->frame_height.ValueOrDefault(0);
video_sender.retransmitted_packets_sent = stat->retransmitted_packets_sent.ValueOrDefault(0);
video_sender.retransmitted_bytes_sent = stat->retransmitted_bytes_sent.ValueOrDefault(0);
video_sender.total_packet_send_delay = stat->total_packet_send_delay.ValueOrDefault(0.0);
video_sender.nack_count = stat->nack_count.ValueOrDefault(0);
video_sender.pli_count = stat->pli_count.ValueOrDefault(0);
video_sender.ssrc = stat->ssrc.value_or(0);
video_sender.packets_sent = stat->packets_sent.value_or(0);
video_sender.bytes_sent = stat->bytes_sent.value_or(0);
video_sender.frames_encoded = stat->frames_encoded.value_or(0);
video_sender.key_frames_encoded = stat->key_frames_encoded.value_or(0);
video_sender.total_encode_time = stat->total_encode_time.value_or(0.0);
video_sender.frame_width = stat->frame_width.value_or(0);
video_sender.frame_height = stat->frame_height.value_or(0);
video_sender.retransmitted_packets_sent = stat->retransmitted_packets_sent.value_or(0);
video_sender.retransmitted_bytes_sent = stat->retransmitted_bytes_sent.value_or(0);
video_sender.total_packet_send_delay = stat->total_packet_send_delay.value_or(0.0);
video_sender.nack_count = stat->nack_count.value_or(0);
video_sender.pli_count = stat->pli_count.value_or(0);
if (stat->quality_limitation_reason.has_value()) {
// "none" = 0 (the default)
if (*stat->quality_limitation_reason == "cpu") {
@ -87,14 +87,14 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
video_sender.quality_limitation_reason = 3;
}
}
video_sender.quality_limitation_resolution_changes = stat->quality_limitation_resolution_changes.ValueOrDefault(0);
video_sender.quality_limitation_resolution_changes = stat->quality_limitation_resolution_changes.value_or(0);
if (stat->remote_id.has_value()) {
auto remote_stat = report->GetAs<RTCRemoteInboundRtpStreamStats>(*stat->remote_id);
if (remote_stat) {
video_sender.remote_packets_lost = remote_stat->packets_lost.ValueOrDefault(0);
video_sender.remote_jitter = remote_stat->jitter.ValueOrDefault(0.0);
video_sender.remote_round_trip_time = remote_stat->round_trip_time.ValueOrDefault(0.0);
video_sender.remote_packets_lost = remote_stat->packets_lost.value_or(0);
video_sender.remote_jitter = remote_stat->jitter.value_or(0.0);
video_sender.remote_round_trip_time = remote_stat->round_trip_time.value_or(0.0);
}
}
@ -106,28 +106,28 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
if (*stat->kind == "audio" && (*stat->mid == "audio" || absl::StartsWith(*stat->mid, "remote-audio"))) {
AudioReceiverStatistics audio_receiver = {0};
audio_receiver.ssrc = stat->ssrc.ValueOrDefault(0);
audio_receiver.packets_received = stat->packets_received.ValueOrDefault(0);
audio_receiver.packets_lost = stat->packets_lost.ValueOrDefault(0);
audio_receiver.bytes_received = stat->bytes_received.ValueOrDefault(0);
audio_receiver.jitter = stat->jitter.ValueOrDefault(0.0);
audio_receiver.total_audio_energy = stat->total_audio_energy.ValueOrDefault(0.0);
audio_receiver.jitter_buffer_delay = stat->jitter_buffer_delay.ValueOrDefault(0.0);
audio_receiver.jitter_buffer_emitted_count = stat->jitter_buffer_emitted_count.ValueOrDefault(0);
audio_receiver.ssrc = stat->ssrc.value_or(0);
audio_receiver.packets_received = stat->packets_received.value_or(0);
audio_receiver.packets_lost = stat->packets_lost.value_or(0);
audio_receiver.bytes_received = stat->bytes_received.value_or(0);
audio_receiver.jitter = stat->jitter.value_or(0.0);
audio_receiver.total_audio_energy = stat->total_audio_energy.value_or(0.0);
audio_receiver.jitter_buffer_delay = stat->jitter_buffer_delay.value_or(0.0);
audio_receiver.jitter_buffer_emitted_count = stat->jitter_buffer_emitted_count.value_or(0);
this->audio_receiver_statistics_.push_back(audio_receiver);
} else if (*stat->kind == "video" && (*stat->mid == "video" || absl::StartsWith(*stat->mid, "remote-video"))) {
VideoReceiverStatistics video_receiver = {0};
video_receiver.ssrc = stat->ssrc.ValueOrDefault(0);
video_receiver.packets_received = stat->packets_received.ValueOrDefault(0);
video_receiver.packets_lost = stat->packets_lost.ValueOrDefault(0);
video_receiver.bytes_received = stat->bytes_received.ValueOrDefault(0);
video_receiver.frames_decoded = stat->frames_decoded.ValueOrDefault(0);
video_receiver.key_frames_decoded = stat->key_frames_decoded.ValueOrDefault(0);
video_receiver.total_decode_time = stat->total_decode_time.ValueOrDefault(0.0);
video_receiver.frame_width = stat->frame_width.ValueOrDefault(0);
video_receiver.frame_height = stat->frame_height.ValueOrDefault(0);
video_receiver.ssrc = stat->ssrc.value_or(0);
video_receiver.packets_received = stat->packets_received.value_or(0);
video_receiver.packets_lost = stat->packets_lost.value_or(0);
video_receiver.bytes_received = stat->bytes_received.value_or(0);
video_receiver.frames_decoded = stat->frames_decoded.value_or(0);
video_receiver.key_frames_decoded = stat->key_frames_decoded.value_or(0);
video_receiver.total_decode_time = stat->total_decode_time.value_or(0.0);
video_receiver.frame_width = stat->frame_width.value_or(0);
video_receiver.frame_height = stat->frame_height.value_or(0);
this->video_receiver_statistics_.push_back(video_receiver);
}
@ -139,11 +139,11 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
for (const auto& stat : candidate_pair_stats) {
// We'll only look at the pair that is nominated with the highest priority, usually
// that has useful values (there does not seem to be a 'in_use' type of flag).
uint64_t current_priority = stat->priority.ValueOrDefault(0);
if (*stat->nominated && stat->priority.ValueOrDefault(0) > highest_priority) {
uint64_t current_priority = stat->priority.value_or(0);
if (*stat->nominated && stat->priority.value_or(0) > highest_priority) {
highest_priority = current_priority;
connection_statistics.current_round_trip_time = stat->current_round_trip_time.ValueOrDefault(0.0);
connection_statistics.available_outgoing_bitrate = stat->available_outgoing_bitrate.ValueOrDefault(0.0);
connection_statistics.current_round_trip_time = stat->current_round_trip_time.value_or(0.0);
connection_statistics.available_outgoing_bitrate = stat->available_outgoing_bitrate.value_or(0.0);
}
}