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Philipp Hancke 622ca1a011 stats: fix remote-outbound-rtp id for video 
which was using the audio type to generate the id. Safe change
since the id is supposed to be random.

BUG=webrtc:12529

Change-Id: I9909c6d320f6f9239f0466599eba1f0eacf00adf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347683
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42142}
2024-04-22 18:22:02 +00:00
..
g3doc
Document how bitrate probing works from a RTP perspective
2023-06-09 14:14:12 +00:00
scenario_tests
…
test
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
audio_rtp_receiver_unittest.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
audio_rtp_receiver.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
audio_rtp_receiver.h
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
audio_track.cc
…
audio_track.h
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
BUILD.gn
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
channel_interface.h
Remove all split channels related code
2023-06-29 09:32:04 +00:00
channel_unittest.cc
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback
2024-02-27 15:55:02 +00:00
channel.cc
Revert "Make Codec::Matches also consider packetization"
2023-11-30 14:06:01 +00:00
channel.h
Use webrtc::TaskQueueBase type instead of rtc::Thread
2023-10-30 14:43:46 +00:00
connection_context.cc
Cleanup ConnectionContext dependency on field trials
2023-12-06 18:01:06 +00:00
connection_context.h
Cleanup ConnectionContext dependency on field trials
2023-12-06 18:01:06 +00:00
data_channel_controller_unittest.cc
Expose bufferedAmountLowThreshold
2024-03-21 19:59:39 +00:00
data_channel_controller.cc
Expose bufferedAmountLowThreshold
2024-03-21 19:59:39 +00:00
data_channel_controller.h
Expose bufferedAmountLowThreshold
2024-03-21 19:59:39 +00:00
data_channel_integrationtest.cc
pc: Increase timeout for EndToEndCallWithSctpDataChannelFullBuffer
2024-03-27 11:09:05 +00:00
data_channel_unittest.cc
pc: Remove additional buffering in SctpDataChannel
2024-03-22 09:25:11 +00:00
data_channel_utils.cc
…
data_channel_utils.h
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
DEPS
Remove the SctpDataChannel::config_ member variable.
2023-03-12 17:28:14 +00:00
dtls_srtp_transport_unittest.cc
Convert signals in rtp_transport_internal.h to CallbackList
2023-09-05 11:37:32 +00:00
dtls_srtp_transport.cc
Reduce logging verbosity of DTLS-SRTP RTCP transport
2023-08-22 13:48:09 +00:00
dtls_srtp_transport.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
dtls_transport_unittest.cc
…
dtls_transport.cc
Reduce locking in DtlsTransport
2023-12-14 21:01:04 +00:00
dtls_transport.h
Reduce locking in DtlsTransport
2023-12-14 21:01:04 +00:00
dtmf_sender_unittest.cc
Remove sigslot usage from DtmfProviderInterface
2022-08-03 14:16:35 +00:00
dtmf_sender.cc
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
dtmf_sender.h
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
external_hmac.cc
…
external_hmac.h
…
ice_server_parsing_unittest.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
ice_server_parsing.cc
Reject stun urls with query parameters
2023-04-21 11:19:37 +00:00
ice_server_parsing.h
Reland "ice server parsing: return RTCError with more details"
2022-10-12 11:23:22 +00:00
ice_transport_unittest.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
ice_transport.cc
…
ice_transport.h
…
jitter_buffer_delay_unittest.cc
…
jitter_buffer_delay.cc
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
jitter_buffer_delay.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
jsep_ice_candidate.cc
…
jsep_session_description_unittest.cc
Reland "Deprecate old constructors and set_type() in Candidate and Port"
2024-03-08 20:39:59 +00:00
jsep_session_description.cc
Move candidate type preference defaults to the Candidate class
2024-01-22 18:27:38 +00:00
jsep_transport_collection.cc
…
jsep_transport_collection.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
jsep_transport_controller_unittest.cc
sctp: Pass webrtc::Environment to DcSctpTransport
2024-03-08 09:45:12 +00:00
jsep_transport_controller.cc
Convert P2PtransportChannel.GatheringState to CallbackList
2024-04-22 12:29:56 +00:00
jsep_transport_controller.h
sctp: Pass webrtc::Environment to DcSctpTransport
2024-03-08 09:45:12 +00:00
jsep_transport_unittest.cc
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback
2024-02-27 15:55:02 +00:00
jsep_transport.cc
pc: Provide DtlsTransport to SctpTransport constr
2024-04-03 10:13:33 +00:00
jsep_transport.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
legacy_stats_collector_interface.h
Delete api/stats_types.h in favor of api/legacy_stats_types.h
2022-11-14 12:10:06 +00:00
legacy_stats_collector_unittest.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
legacy_stats_collector.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
legacy_stats_collector.h
Use is_* getters when checking the Candidate type
2024-01-19 15:37:32 +00:00
local_audio_source_unittest.cc
…
local_audio_source.cc
…
local_audio_source.h
…
media_factory.h
Delete deprecated call_factory and media_engine
2023-12-04 14:42:44 +00:00
media_protocol_names.cc
…
media_protocol_names.h
…
media_session_unittest.cc
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
media_session.cc
sdp: set content to rejected if the list of common codecs is empty
2024-02-29 08:02:26 +00:00
media_session.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
media_stream_observer.cc
…
media_stream_observer.h
…
media_stream_proxy.h
…
media_stream_track_proxy.h
…
media_stream_unittest.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
media_stream.cc
…
media_stream.h
…
OWNERS
Enforce policy that SDP munging requires special approval
2023-01-30 20:09:07 +00:00
peer_connection_adaptation_integrationtest.cc
New PeerConnectionFactory::CreateVideoTrack with refcounted source
2023-03-22 09:10:27 +00:00
peer_connection_bundle_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_crypto_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_data_channel_unittest.cc
Migrate PeerConnection tests to EnableMedia api
2023-11-03 12:32:14 +00:00
peer_connection_encodings_integrationtest.cc
Report proper VP9 scalability mode with layer activation.
2024-04-05 14:02:59 +00:00
peer_connection_end_to_end_unittest.cc
Reland "Unify access to SDP codec parameters"
2024-01-03 12:03:11 +00:00
peer_connection_factory_proxy.h
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
peer_connection_factory_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_factory.cc
Remove unused WebRTC-Bwe-InjectedCongestionController
2024-04-19 08:05:25 +00:00
peer_connection_factory.h
Reland "FrameCadenceAdapter: align video encoding to metronome"
2024-01-08 13:54:56 +00:00
peer_connection_field_trial_tests.cc
Disable InjectDependencyDescriptor on Windows.
2024-03-19 10:25:18 +00:00
peer_connection_header_extension_unittest.cc
Deprecate RtcEventLogFactory constructor taking unused parameter
2023-12-07 21:46:56 +00:00
peer_connection_histogram_unittest.cc
Delete CallFactoryInterface as no longer needed
2023-12-05 15:44:43 +00:00
peer_connection_ice_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_integrationtest.cc
Reland "Take out Fuchsia-only SDES-enabling parameters"
2024-02-02 17:02:29 +00:00
peer_connection_interface_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_internal.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_jsep_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_media_unittest.cc
Add killswitch for receive-only setCodecPreferences change
2024-02-29 12:43:05 +00:00
peer_connection_message_handler.cc
Rename api/stats_types.h to api/legacy_stats_types.h.
2022-11-11 10:29:25 +00:00
peer_connection_message_handler.h
Rename api/stats_types.h to api/legacy_stats_types.h.
2022-11-11 10:29:25 +00:00
peer_connection_proxy.h
Add PeerConnectionInterface::ReconfigureBandwidthEstimation
2024-02-07 14:10:02 +00:00
peer_connection_rampup_tests.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_rtp_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_signaling_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_simulcast_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_svc_integrationtest.cc
Reland "Make setCodecPreferences only look at receive codecs"
2024-02-26 10:52:23 +00:00
peer_connection_wrapper.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
peer_connection_wrapper.h
Enable the use of CreateDataChannel with a DataChannelInit config.
2023-02-10 12:24:47 +00:00
peer_connection.cc
sctp: Pass webrtc::Environment to DcSctpTransport
2024-03-08 09:45:12 +00:00
peer_connection.h
Add PeerConnectionInterface::ReconfigureBandwidthEstimation
2024-02-07 14:10:02 +00:00
proxy_unittest.cc
…
proxy.h
Replace proxy ScopedEvent with TRACE_EVENT
2024-03-18 09:57:36 +00:00
remote_audio_source.cc
Split audio and video channels into Send and Receive APIs.
2022-12-14 11:00:17 +00:00
remote_audio_source.h
Split audio and video channels into Send and Receive APIs.
2022-12-14 11:00:17 +00:00
rtc_stats_collector_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
rtc_stats_collector.cc
stats: fix remote-outbound-rtp id for video
2024-04-22 18:22:02 +00:00
rtc_stats_collector.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
rtc_stats_integrationtest.cc
stats: implement remote-outbound-rtp for video
2024-04-15 15:10:54 +00:00
rtc_stats_traversal_unittest.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
rtc_stats_traversal.cc
[Stats] Migrate from the RTCStatsMember type alias to absl::optional.
2024-01-25 21:56:08 +00:00
rtc_stats_traversal.h
…
rtcp_mux_filter_unittest.cc
…
rtcp_mux_filter.cc
…
rtcp_mux_filter.h
…
rtp_media_utils_unittest.cc
…
rtp_media_utils.cc
…
rtp_media_utils.h
…
rtp_parameters_conversion_unittest.cc
Reland "Merge the codec types"
2023-06-15 15:53:29 +00:00
rtp_parameters_conversion.cc
Reland "Merge the codec types"
2023-06-15 15:53:29 +00:00
rtp_parameters_conversion.h
Reland "Merge the codec types"
2023-06-15 15:53:29 +00:00
rtp_receiver_proxy.h
…
rtp_receiver.cc
Guard GenerateUniqueId aginst concurrent access.
2024-01-25 12:29:42 +00:00
rtp_receiver.h
Reland "Remove 'trackId' dependency in stats selector algorithm."
2023-01-05 09:04:12 +00:00
rtp_sender_proxy.h
Remove public GenerateKeyFrame(list-of-rids) API from RtpSender
2023-08-30 14:54:17 +00:00
rtp_sender_receiver_unittest.cc
Keep Environment instead of test field trials in FakeCall test object
2024-04-09 11:53:18 +00:00
rtp_sender.cc
Reland "Make setCodecPreferences only look at receive codecs"
2024-02-26 10:52:23 +00:00
rtp_sender.h
Reland "Make setCodecPreferences only look at receive codecs"
2024-02-26 10:52:23 +00:00
rtp_transceiver_unittest.cc
Reland "Enable DD and VLA header extensions by default for Simulcast/SVC"
2024-01-26 06:08:28 +00:00
rtp_transceiver.cc
Add killswitch for receive-only setCodecPreferences change
2024-02-29 12:43:05 +00:00
rtp_transceiver.h
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
rtp_transmission_manager.cc
Reland "Ensure RTCRtpSenders are always created with one encoding"
2023-01-19 15:49:04 +00:00
rtp_transmission_manager.h
Delete some unused sigslot includes
2023-08-28 12:36:39 +00:00
rtp_transport_internal.h
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
rtp_transport_unittest.cc
Propagate ECN to RtpPacketReceived
2024-03-15 08:58:28 +00:00
rtp_transport.cc
Propagate ECN to RtpPacketReceived
2024-03-15 08:58:28 +00:00
rtp_transport.h
Propagate ECN to RtpPacketReceived
2024-03-15 08:58:28 +00:00
sctp_data_channel.cc
pc: Close the data channel association after sending messages in closing state
2024-03-25 18:36:09 +00:00
sctp_data_channel.h
pc: Remove additional buffering in SctpDataChannel
2024-03-22 09:25:11 +00:00
sctp_transport_unittest.cc
pc: Provide DtlsTransport to SctpTransport constr
2024-04-03 10:13:33 +00:00
sctp_transport.cc
pc: Provide DtlsTransport to SctpTransport constr
2024-04-03 10:13:33 +00:00
sctp_transport.h
pc: Provide DtlsTransport to SctpTransport constr
2024-04-03 10:13:33 +00:00
sctp_utils_unittest.cc
pc: Simplify StreamId class
2024-03-12 10:57:56 +00:00
sctp_utils.cc
Deprecate char* functions on ByteBufferReader
2023-11-22 11:46:25 +00:00
sctp_utils.h
pc: Simplify StreamId class
2024-03-12 10:57:56 +00:00
sdp_offer_answer_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
sdp_offer_answer.cc
Demote RTC_CHECK for sctp_mid() to RTC_LOG(LS_ERROR) if unavailable
2024-02-23 14:20:23 +00:00
sdp_offer_answer.h
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
sdp_state_provider.h
…
sdp_utils.cc
…
sdp_utils.h
…
session_description_unittest.cc
…
session_description.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
session_description.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
simulcast_description.cc
…
simulcast_description.h
…
simulcast_sdp_serializer_unittest.cc
Rename simulcast SDP serializer
2023-09-18 10:09:02 +00:00
simulcast_sdp_serializer.cc
Rename simulcast SDP serializer
2023-09-18 10:09:02 +00:00
simulcast_sdp_serializer.h
Rename simulcast SDP serializer
2023-09-18 10:09:02 +00:00
slow_peer_connection_integration_test.cc
Use rtc::ReceivedPacket in Stun and TurnServer
2023-11-23 10:40:56 +00:00
srtp_session_unittest.cc
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
srtp_session.cc
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
srtp_session.h
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
srtp_transport_unittest.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
srtp_transport.cc
Propagate ECN to RtpPacketReceived
2024-03-15 08:58:28 +00:00
srtp_transport.h
Propagate rtc::ReceivedPacket further in RtpTransport
2024-02-27 17:46:18 +00:00
stream_collection.h
…
track_media_info_map_unittest.cc
Delete deprecated 'track' and 'stream' metrics from WebRTC.
2023-05-24 12:26:56 +00:00
track_media_info_map.cc
Delete deprecated 'track' and 'stream' metrics from WebRTC.
2023-05-24 12:26:56 +00:00
track_media_info_map.h
Delete deprecated 'track' and 'stream' metrics from WebRTC.
2023-05-24 12:26:56 +00:00
transceiver_list.cc
…
transceiver_list.h
…
transport_stats.cc
…
transport_stats.h
Add histogram for DTLS peer signature algorithm
2023-10-06 12:25:37 +00:00
usage_pattern.cc
…
usage_pattern.h
…
used_ids_unittest.cc
Dynamically assign ids to header extensions not enabled by default
2024-02-08 12:52:58 +00:00
used_ids.h
Dynamically assign ids to header extensions not enabled by default
2024-02-08 12:52:58 +00:00
video_rtp_receiver_unittest.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
video_rtp_receiver.cc
Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters
2023-09-01 08:12:55 +00:00
video_rtp_receiver.h
Implement GetParameters/GetSources support for unsignaled SSRCs.
2023-01-10 06:44:27 +00:00
video_rtp_track_source_unittest.cc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
video_rtp_track_source.cc
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
video_rtp_track_source.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
video_track_source_proxy.cc
…
video_track_source_proxy.h
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
video_track_source.cc
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
video_track_source.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
video_track_unittest.cc
…
video_track.cc
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
video_track.h
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
webrtc_sdp_unittest.cc
Reland "Deprecate old constructors and set_type() in Candidate and Port"
2024-03-08 20:39:59 +00:00
webrtc_sdp.cc
Reland "Deprecate old constructors and set_type() in Candidate and Port"
2024-03-08 20:39:59 +00:00
webrtc_sdp.h
Reland "Unify access to SDP codec parameters"
2024-01-03 12:03:11 +00:00
webrtc_session_description_factory.cc
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
webrtc_session_description_factory.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00