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webrtc_m130/modules/rtp_rtcp/source
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Tommi 08be9baaa3 Don't recreate the audio receive stream when updating the local_ssrc. 
Bug: webrtc:11993
Change-Id: Ic5d8a8a8b7c12fb1d906e0b3cbdf657fd9e8eafc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222042
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34299}
2021-06-16 10:03:31 +00:00
..
deprecated
Reland "Correctly handle retransmissions/padding in early loss detection."
2021-06-16 08:14:27 +00:00
rtcp_packet
Clean up previously deleted RTCP VOIP metrics block.
2021-02-04 18:34:28 +00:00
absolute_capture_time_interpolator_unittest.cc
Make local to capturer clock offset a separate entry in PacketInfo.
2021-05-20 13:42:57 +00:00
absolute_capture_time_interpolator.cc
Make local to capturer clock offset a separate entry in PacketInfo.
2021-05-20 13:42:57 +00:00
absolute_capture_time_interpolator.h
Make local to capturer clock offset a separate entry in PacketInfo.
2021-05-20 13:42:57 +00:00
absolute_capture_time_receiver.cc
Make local to capturer clock offset a separate entry in PacketInfo.
2021-05-20 13:42:57 +00:00
absolute_capture_time_receiver.h
Make local to capturer clock offset a separate entry in PacketInfo.
2021-05-20 13:42:57 +00:00
absolute_capture_time_sender_unittest.cc
Add helper classes to send and receive abs-capture-time extensions.
2019-08-22 09:09:17 +00:00
absolute_capture_time_sender.cc
Migrate modules/rtp_rtcp to webrtc::Mutex.
2020-07-07 12:13:47 +00:00
absolute_capture_time_sender.h
Migrate modules/rtp_rtcp to webrtc::Mutex.
2020-07-07 12:13:47 +00:00
active_decode_targets_helper_unittest.cc
In DependencyDescriptor rtp header extension drop partial chain support
2020-07-21 14:01:27 +00:00
active_decode_targets_helper.cc
In DependencyDescriptor rtp header extension drop partial chain support
2020-07-21 14:01:27 +00:00
active_decode_targets_helper.h
Propagate active decode targets bitmask into DependencyDescriptor
2020-06-29 12:54:43 +00:00
byte_io_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
byte_io.h
Format almost everything.
2019-07-08 13:45:15 +00:00
capture_clock_offset_updater_unittest.cc
Make local to capturer clock offset a separate entry in PacketInfo.
2021-05-20 13:42:57 +00:00
capture_clock_offset_updater.cc
Make local to capturer clock offset a separate entry in PacketInfo.
2021-05-20 13:42:57 +00:00
capture_clock_offset_updater.h
Make local to capturer clock offset a separate entry in PacketInfo.
2021-05-20 13:42:57 +00:00
create_video_rtp_depacketizer.cc
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
create_video_rtp_depacketizer.h
Add implementations of the VideoRtpDepacketizer interface
2019-12-06 15:20:29 +00:00
dtmf_queue.cc
Migrate modules/rtp_rtcp to webrtc::Mutex.
2020-07-07 12:13:47 +00:00
dtmf_queue.h
Migrate modules/rtp_rtcp to webrtc::Mutex.
2020-07-07 12:13:47 +00:00
fec_private_tables_bursty_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
fec_private_tables_bursty.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fec_private_tables_bursty.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
fec_private_tables_random.cc
…
fec_private_tables_random.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
fec_test_helper.cc
Updates ulpfec reader to accept padding on media packets.
2021-03-16 19:47:09 +00:00
fec_test_helper.h
Fix: don't use recovered packets in UlpFEC recovery
2020-03-11 12:49:11 +00:00
flexfec_header_reader_writer_unittest.cc
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +00:00
flexfec_header_reader_writer.cc
Increase FEC receiver's protected packet queue size.
2021-05-12 06:26:38 +00:00
flexfec_header_reader_writer.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
flexfec_receiver_unittest.cc
Increase FEC receiver's protected packet queue size.
2021-05-12 06:26:38 +00:00
flexfec_receiver.cc
Add Slice method to CopyOnWriteBuffer and use it in FEC code.
2019-09-26 09:48:07 +00:00
flexfec_sender_unittest.cc
Refactors FEC in preparation for deferred packet generation.
2020-05-05 13:59:14 +00:00
flexfec_sender.cc
Migrate modules/rtp_rtcp to webrtc::Mutex.
2020-07-07 12:13:47 +00:00
forward_error_correction_internal.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
forward_error_correction_internal.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
forward_error_correction.cc
Purge old FEC packets from receiver's queue before media sequence numbers wrap around
2021-04-21 07:33:14 +00:00
forward_error_correction.h
Purge old FEC packets from receiver's queue before media sequence numbers wrap around
2021-04-21 07:33:14 +00:00
nack_rtx_unittest.cc
Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 17:47:32 +00:00
packet_loss_stats_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
packet_loss_stats.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
packet_loss_stats.h
Format almost everything.
2019-07-08 13:45:15 +00:00
packet_sequencer.cc
Extract sequencing from RtpSender
2021-02-22 14:00:06 +00:00
packet_sequencer.h
Extract sequencing from RtpSender
2021-02-22 14:00:06 +00:00
receive_statistics_impl.cc
Cleanup ReceiveStatistics collecting ReportBlock
2021-05-17 14:15:45 +00:00
receive_statistics_impl.h
Cleanup ReceiveStatistics collecting ReportBlock
2021-05-17 14:15:45 +00:00
receive_statistics_unittest.cc
Add a mutex free implementation of webrtc::ReceiveStatistics
2021-03-10 14:16:38 +00:00
remote_ntp_time_estimator_unittest.cc
Stop trying to compensate for the offset between the different NTP clocks.
2021-04-08 14:48:20 +00:00
remote_ntp_time_estimator.cc
Stop trying to compensate for the offset between the different NTP clocks.
2021-04-08 14:48:20 +00:00
rtcp_nack_stats_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_nack_stats.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
rtcp_nack_stats.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcp_packet_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_packet.cc
Enable padding bit in TransportFeedback packets
2019-02-12 11:55:34 +00:00
rtcp_packet.h
Replace virtual RtcpPacket::SetSenderSsrc with base member
2019-10-10 09:14:11 +00:00
rtcp_receiver_unittest.cc
Use webrtc::Clock to query for the NTP time and to convert timestamps
2021-05-21 14:33:00 +00:00
rtcp_receiver.cc
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtcp_receiver.h
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtcp_sender_unittest.cc
Use webrtc::Clock to query for the NTP time and to convert timestamps
2021-05-21 14:33:00 +00:00
rtcp_sender.cc
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtcp_sender.h
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtcp_transceiver_config.cc
Migrate RtcpTransceiver to use webrtc::TaskQueueBase instead of rtc::TaskQueue
2020-03-06 11:26:51 +00:00
rtcp_transceiver_config.h
Add a clock to be used for getting the NTP time in RtcpTransceiverConfig.
2021-04-08 12:43:27 +00:00
rtcp_transceiver_impl_unittest.cc
Reland "Avoid sending empty receiver reports with RtcpTransceiver"
2021-06-15 12:57:56 +00:00
rtcp_transceiver_impl.cc
Reland "Avoid sending empty receiver reports with RtcpTransceiver"
2021-06-15 12:57:56 +00:00
rtcp_transceiver_impl.h
Delete the old flavor of RtcpTransceiverImpl::ReceivePacket
2021-05-21 19:06:22 +00:00
rtcp_transceiver_unittest.cc
Use the injected clock in rtcp_transciever.
2021-05-21 17:34:33 +00:00
rtcp_transceiver.cc
Use the injected clock in rtcp_transciever.
2021-05-21 17:34:33 +00:00
rtcp_transceiver.h
Use the injected clock in rtcp_transciever.
2021-05-21 17:34:33 +00:00
rtp_dependency_descriptor_extension_unittest.cc
Ignore inactive chains when writing DependencyDescriptor rtp header extension.
2020-06-25 14:59:38 +00:00
rtp_dependency_descriptor_extension.cc
Ignore inactive chains when writing DependencyDescriptor rtp header extension.
2020-06-25 14:59:38 +00:00
rtp_dependency_descriptor_extension.h
Ignore inactive chains when writing DependencyDescriptor rtp header extension.
2020-06-25 14:59:38 +00:00
rtp_dependency_descriptor_reader.cc
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
rtp_dependency_descriptor_reader.h
Publish DependencyDescriptor structures in the api
2020-01-20 15:05:48 +00:00
rtp_dependency_descriptor_writer.cc
In DependencyDescriptor rtp header extension drop partial chain support
2020-07-21 14:01:27 +00:00
rtp_dependency_descriptor_writer.h
Ignore inactive chains when writing DependencyDescriptor rtp header extension.
2020-06-25 14:59:38 +00:00
rtp_descriptor_authentication.cc
Do not propagate generic descriptor on receiving frame
2020-02-17 14:52:03 +00:00
rtp_descriptor_authentication.h
Do not propagate generic descriptor on receiving frame
2020-02-17 14:52:03 +00:00
rtp_fec_unittest.cc
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +00:00
rtp_format_h264_unittest.cc
Ignore fragmentation header when packetizing H264
2020-07-16 16:12:33 +00:00
rtp_format_h264.cc
h264: s/StrapA/STAP-A
2020-12-03 13:09:04 +00:00
rtp_format_h264.h
Ignore fragmentation header when packetizing H264
2020-07-16 16:12:33 +00:00
rtp_format_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_format_video_generic_unittest.cc
Remove RTPFragmentationHeader creation and propagation through webrtc
2020-08-17 16:37:33 +00:00
rtp_format_video_generic.cc
Pass picture_id from generic packetizer through codec-specific field
2020-01-28 19:26:28 +00:00
rtp_format_video_generic.h
Delete RtpGenericDepacketizer as no longer used
2020-01-13 13:45:37 +00:00
rtp_format_vp8_test_helper.cc
Cleanup RtpPacketizerVP8 tests
2018-08-31 08:13:45 +00:00
rtp_format_vp8_test_helper.h
Do not propage RTPFragmentationHeader into rtp_rtcp
2020-07-21 14:37:08 +00:00
rtp_format_vp8_unittest.cc
Delete RtpDepacketizerVp8 in favor of VideoRtpDepacketizerVp8
2020-01-09 12:10:19 +00:00
rtp_format_vp8.cc
Delete RtpDepacketizerVp8 in favor of VideoRtpDepacketizerVp8
2020-01-09 12:10:19 +00:00
rtp_format_vp8.h
Delete RtpDepacketizerVp8 in favor of VideoRtpDepacketizerVp8
2020-01-09 12:10:19 +00:00
rtp_format_vp9_unittest.cc
Reland "[VP9] Shift spatial layers on RTP level to always start from 0."
2020-02-04 10:06:44 +00:00
rtp_format_vp9.cc
Reland "[VP9] Shift spatial layers on RTP level to always start from 0."
2020-02-04 10:06:44 +00:00
rtp_format_vp9.h
Delete RtpDepacketizerVp9 in favor of VideoRtpDepacketizerVp9
2020-01-09 13:07:44 +00:00
rtp_format.cc
In Av1 packetizer set marker bit with respect of end_of_picture flag
2020-11-16 11:08:48 +00:00
rtp_format.h
Do not propage RTPFragmentationHeader into rtp_rtcp
2020-07-21 14:37:08 +00:00
rtp_generic_frame_descriptor_extension_unittest.cc
Delete generic frame descriptor v1 trait and enum value
2020-04-16 17:29:18 +00:00
rtp_generic_frame_descriptor_extension.cc
Delete generic frame descriptor v1 trait and enum value
2020-04-16 17:29:18 +00:00
rtp_generic_frame_descriptor_extension.h
Delete generic frame descriptor v1 trait and enum value
2020-04-16 17:29:18 +00:00
rtp_generic_frame_descriptor.cc
Do not propagate generic descriptor on receiving frame
2020-02-17 14:52:03 +00:00
rtp_generic_frame_descriptor.h
Delete generic frame descriptor v1 trait and enum value
2020-04-16 17:29:18 +00:00
rtp_header_extension_map_unittest.cc
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file
2018-10-03 17:25:31 +00:00
rtp_header_extension_map.cc
Implement the mixer-to-client per CSRC audio level extension (RFC 6465).
2021-05-24 14:11:28 +00:00
rtp_header_extension_size_unittest.cc
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file
2018-10-03 17:25:31 +00:00
rtp_header_extension_size.cc
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
rtp_header_extension_size.h
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file
2018-10-03 17:25:31 +00:00
rtp_header_extensions.cc
Implement the mixer-to-client per CSRC audio level extension (RFC 6465).
2021-05-24 14:11:28 +00:00
rtp_header_extensions.h
Implement the mixer-to-client per CSRC audio level extension (RFC 6465).
2021-05-24 14:11:28 +00:00
rtp_packet_history_unittest.cc
Add ability to disable padding prioritization.
2020-03-30 09:01:51 +00:00
rtp_packet_history.cc
Remove tautological 'unsigned expr < 0' comparisons
2021-04-12 11:40:14 +00:00
rtp_packet_history.h
Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
2020-09-29 10:19:20 +00:00
rtp_packet_received.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtp_packet_received.h
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtp_packet_to_send.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
rtp_packet_to_send.h
Refactor and improve RtpSender packet history test.
2021-05-25 09:53:27 +00:00
rtp_packet_unittest.cc
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
rtp_packet.cc
Add rtp_config() accessor to ReceiveStream.
2021-06-14 17:57:57 +00:00
rtp_packet.h
Add rtp_config() accessor to ReceiveStream.
2021-06-14 17:57:57 +00:00
rtp_packetizer_av1_unittest.cc
In Av1 packetizer set marker bit with respect of end_of_picture flag
2020-11-16 11:08:48 +00:00
rtp_packetizer_av1.cc
In Av1 packetizer set marker bit with respect of end_of_picture flag
2020-11-16 11:08:48 +00:00
rtp_packetizer_av1.h
In Av1 packetizer set marker bit with respect of end_of_picture flag
2020-11-16 11:08:48 +00:00
rtp_rtcp_config.h
Delete unused constants from rtp_rtcp_config.h
2018-08-09 08:38:51 +00:00
rtp_rtcp_impl2_unittest.cc
Add an UlpFec test to RtpRtcp unit tests.
2021-05-19 15:26:24 +00:00
rtp_rtcp_impl2.cc
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtp_rtcp_impl2.h
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtp_rtcp_impl_unittest.cc
Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket
2021-06-03 12:29:09 +00:00
rtp_rtcp_impl.cc
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtp_rtcp_impl.h
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtp_rtcp_interface.h
Don't recreate the audio receive stream when updating the local_ssrc.
2021-06-16 10:03:31 +00:00
rtp_sender_audio_unittest.cc
Use the clock to convert absolute capture timestamps to NTP times.
2021-05-21 12:41:50 +00:00
rtp_sender_audio.cc
Use the clock to convert absolute capture timestamps to NTP times.
2021-05-21 12:41:50 +00:00
rtp_sender_audio.h
Use the clock to convert absolute capture timestamps to NTP times.
2021-05-21 12:41:50 +00:00
rtp_sender_egress_unittest.cc
Reland "Correctly handle retransmissions/padding in early loss detection."
2021-06-16 08:14:27 +00:00
rtp_sender_egress.cc
Reland "Correctly handle retransmissions/padding in early loss detection."
2021-06-16 08:14:27 +00:00
rtp_sender_egress.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
rtp_sender_unittest.cc
Clean up RtpSenderTest and remove RtpSenderEgress dependencies.
2021-05-26 15:25:58 +00:00
rtp_sender_video_frame_transformer_delegate.cc
[InsertableStreams] Ensure encoder queue is valid in OnTransformedFrame
2021-04-28 16:46:22 +00:00
rtp_sender_video_frame_transformer_delegate.h
Add RtpVideoSender::SendVideoLayersAllocation
2020-10-14 08:10:03 +00:00
rtp_sender_video_unittest.cc
Use the clock to convert absolute capture timestamps to NTP times.
2021-05-21 12:41:50 +00:00
rtp_sender_video.cc
Use the clock to convert absolute capture timestamps to NTP times.
2021-05-21 12:41:50 +00:00
rtp_sender_video.h
Use the clock to convert absolute capture timestamps to NTP times.
2021-05-21 12:41:50 +00:00
rtp_sender.cc
Implement the mixer-to-client per CSRC audio level extension (RFC 6465).
2021-05-24 14:11:28 +00:00
rtp_sender.h
Extract sequencing from RtpSender
2021-02-22 14:00:06 +00:00
rtp_sequence_number_map_unittest.cc
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
2019-09-24 08:56:24 +00:00
rtp_sequence_number_map.cc
Add RtpSequenceNumberMap::InsertFrame()
2019-04-08 12:39:08 +00:00
rtp_sequence_number_map.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_utility_unittest.cc
Add writing and parsing of the abs-capture-time RTP header extension.
2019-07-03 14:07:36 +00:00
rtp_utility.cc
Implement the mixer-to-client per CSRC audio level extension (RFC 6465).
2021-05-24 14:11:28 +00:00
rtp_utility.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_video_header.cc
Add default values for PlayoutDelay in RTPVideoHeader.
2018-11-16 12:10:23 +00:00
rtp_video_header.h
Send and Receive VideoFrameTrackingid RTP header extension.
2021-03-25 21:57:29 +00:00
rtp_video_layers_allocation_extension_unittest.cc
Fix RtpVideoLayersAllocationExtension::Write of invalid allocation
2021-03-24 13:53:13 +00:00
rtp_video_layers_allocation_extension.cc
Remove tautological 'unsigned expr < 0' comparisons
2021-04-12 11:40:14 +00:00
rtp_video_layers_allocation_extension.h
Add experimental extension RtpVideoLayersAllocation
2020-10-05 13:38:13 +00:00
source_tracker_unittest.cc
Include all RTP packet infos from the mix list when updating the audio frame for mixing.
2021-05-18 11:05:37 +00:00
source_tracker.cc
Migrate modules/rtp_rtcp to webrtc::Mutex.
2020-07-07 12:13:47 +00:00
source_tracker.h
Migrate modules/rtp_rtcp to webrtc::Mutex.
2020-07-07 12:13:47 +00:00
time_util_unittest.cc
Delete NtpOffsetMs and TimeMicrosToNtp methods.
2021-05-21 19:32:42 +00:00
time_util.cc
Delete NtpOffsetMs and TimeMicrosToNtp methods.
2021-05-21 19:32:42 +00:00
time_util.h
Delete NtpOffsetMs and TimeMicrosToNtp methods.
2021-05-21 19:32:42 +00:00
tmmbr_help.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
tmmbr_help.h
Format almost everything.
2019-07-08 13:45:15 +00:00
ulpfec_generator_unittest.cc
Update how FEC handles protection parameters for key vs delta frames.
2020-12-07 13:36:03 +00:00
ulpfec_generator.cc
Update how FEC handles protection parameters for key vs delta frames.
2020-12-07 13:36:03 +00:00
ulpfec_generator.h
Update how FEC handles protection parameters for key vs delta frames.
2020-12-07 13:36:03 +00:00
ulpfec_header_reader_writer_unittest.cc
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +00:00
ulpfec_header_reader_writer.cc
Increase FEC receiver's protected packet queue size.
2021-05-12 06:26:38 +00:00
ulpfec_header_reader_writer.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
ulpfec_receiver_impl.cc
Updates ulpfec reader to accept padding on media packets.
2021-03-16 19:47:09 +00:00
ulpfec_receiver_impl.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
ulpfec_receiver_unittest.cc
Increase FEC receiver's protected packet queue size.
2021-05-12 06:26:38 +00:00
video_fec_generator.h
Refactors FEC in preparation for deferred packet generation.
2020-05-05 13:59:14 +00:00
video_rtp_depacketizer_av1_unittest.cc
Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface.
2020-03-24 15:55:00 +00:00
video_rtp_depacketizer_av1.cc
Change Av1 depacketizer to implement VideoRtpDepacketizer interface
2020-01-15 10:16:03 +00:00
video_rtp_depacketizer_av1.h
Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface.
2020-03-24 15:55:00 +00:00
video_rtp_depacketizer_generic_unittest.cc
Pass picture_id from generic packetizer through codec-specific field
2020-01-28 19:26:28 +00:00
video_rtp_depacketizer_generic.cc
Pass picture_id from generic packetizer through codec-specific field
2020-01-28 19:26:28 +00:00
video_rtp_depacketizer_generic.h
Add VideoRtpDepacketizerGeneric
2020-01-07 09:27:34 +00:00
video_rtp_depacketizer_h264_unittest.cc
Remove RTPFragmentationHeader creation and propagation through webrtc
2020-08-17 16:37:33 +00:00
video_rtp_depacketizer_h264.cc
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +00:00
video_rtp_depacketizer_h264.h
Change H264 depacketizer to implement VideoRtpDepacketizer interface
2020-01-15 12:26:55 +00:00
video_rtp_depacketizer_raw_unittest.cc
Add implementations of the VideoRtpDepacketizer interface
2019-12-06 15:20:29 +00:00
video_rtp_depacketizer_raw.cc
Add implementations of the VideoRtpDepacketizer interface
2019-12-06 15:20:29 +00:00
video_rtp_depacketizer_raw.h
Add implementations of the VideoRtpDepacketizer interface
2019-12-06 15:20:29 +00:00
video_rtp_depacketizer_vp8_unittest.cc
Move vp8 rtp depacketization to VideoRtpDepacketizerVp8
2019-12-13 15:10:46 +00:00
video_rtp_depacketizer_vp8.cc
Move vp8 rtp depacketization to VideoRtpDepacketizerVp8
2019-12-13 15:10:46 +00:00
video_rtp_depacketizer_vp8.h
Move vp8 rtp depacketization to VideoRtpDepacketizerVp8
2019-12-13 15:10:46 +00:00
video_rtp_depacketizer_vp9_unittest.cc
Move vp9 rtp depacketization to VideoRtpDepacketizerVp9
2019-12-16 17:11:13 +00:00
video_rtp_depacketizer_vp9.cc
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
video_rtp_depacketizer_vp9.h
Move vp9 rtp depacketization to VideoRtpDepacketizerVp9
2019-12-16 17:11:13 +00:00
video_rtp_depacketizer.cc
Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface.
2020-03-24 15:55:00 +00:00
video_rtp_depacketizer.h
Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface.
2020-03-24 15:55:00 +00:00