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webrtc/modules/rtp_rtcp/source
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Taylor Brandstetter 1204448a68 Revert "Reland "Rename stereo video codec to multiplex"" 
This reverts commit 4954a77cf8.

Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(

Original change's description:
> Reland "Rename stereo video codec to multiplex"
> 
> This is a reland of bbdabe50db.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
> 
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > This CL only does the rename from"stereo" to multiplex". With this we have a
> > better name that doesn't clash with audio's usage of stereo.
> >
> > Bug: webrtc:7671
> > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> > Reviewed-on: https://webrtc-review.googlesource.com/43242
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21769}
> 
> TBR=niklas.enbom@webrtc.org
> 
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}

TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org

Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
2018-01-27 00:45:20 +00:00
..
rtcp_packet Reland "Make RTCP cumulative_lost be a signed value" 2017-12-08 08:47:09 +00:00
byte_io.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
byte_io_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
dtmf_queue.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
dtmf_queue.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fec_private_tables_bursty.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
fec_private_tables_random.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
fec_test_helper.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fec_test_helper.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
flexfec_header_reader_writer.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
flexfec_header_reader_writer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
flexfec_header_reader_writer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
flexfec_receiver.cc Avoid lifetime issues with FlexfecReceiver packet buffer. 2017-12-12 10:12:47 +00:00
flexfec_receiver_unittest.cc Avoid lifetime issues with FlexfecReceiver packet buffer. 2017-12-12 10:12:47 +00:00
flexfec_sender.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
flexfec_sender_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
forward_error_correction.cc Do not send 48 empty FEC packets when there is a large media packet seq. num. gap. 2017-12-07 11:22:30 +00:00
forward_error_correction.h Change ForwardErrorCorrection class to accept one received packet at a time. 2017-09-18 14:58:59 +00:00
forward_error_correction_internal.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
forward_error_correction_internal.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
nack_rtx_unittest.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
packet_loss_stats.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
packet_loss_stats.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
packet_loss_stats_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
playout_delay_oracle.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
playout_delay_oracle.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
playout_delay_oracle_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
receive_statistics_impl.cc Support more ssrcs in ReceiveStatistics than retrieved per RtcpReportBlocks call 2018-01-16 12:02:24 +00:00
receive_statistics_impl.h Support more ssrcs in ReceiveStatistics than retrieved per RtcpReportBlocks call 2018-01-16 12:02:24 +00:00
receive_statistics_unittest.cc Support more ssrcs in ReceiveStatistics than retrieved per RtcpReportBlocks call 2018-01-16 12:02:24 +00:00
remote_ntp_time_estimator.cc Remove WebRTC-ClockEstimation experiment and make new clock estimation always enabled 2017-12-05 09:49:32 +00:00
remote_ntp_time_estimator_unittest.cc Remove WebRTC-ClockEstimation experiment and make new clock estimation always enabled 2017-12-05 09:49:32 +00:00
rtcp_nack_stats.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcp_nack_stats.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcp_nack_stats_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcp_packet.cc Change RtcpPacket::PacketReadyCallback to rtc::FunctionView 2017-12-07 11:20:08 +00:00
rtcp_packet.h Change RtcpPacket::PacketReadyCallback to rtc::FunctionView 2017-12-07 11:20:08 +00:00
rtcp_packet_unittest.cc Change RtcpPacket::PacketReadyCallback to rtc::FunctionView 2017-12-07 11:20:08 +00:00
rtcp_receiver.cc Reland "Make RTCP cumulative_lost be a signed value" 2017-12-08 08:47:09 +00:00
rtcp_receiver.h Trigger rtt and stats update on report block rather than receiver report. 2017-09-28 10:29:59 +00:00
rtcp_receiver_unittest.cc Trigger rtt and stats update on report block rather than receiver report. 2017-09-28 10:29:59 +00:00
rtcp_sender.cc Change RtpRtcp::SetRemb signature to match RtcpTransceiver::SetRemb 2017-12-13 14:40:01 +00:00
rtcp_sender.h Change RtpRtcp::SetRemb signature to match RtcpTransceiver::SetRemb 2017-12-13 14:40:01 +00:00
rtcp_sender_unittest.cc Reland "Make RTCP cumulative_lost be a signed value" 2017-12-08 08:47:09 +00:00
rtcp_transceiver.cc Propagate media receiver rtcp observers to RtcpTransceiver 2017-12-14 17:39:13 +00:00
rtcp_transceiver.h Narrow interface PacketRouter use to send Remb and TransportFeedback 2017-12-15 15:58:17 +00:00
rtcp_transceiver_config.cc Calculate RTT using ExtendedReports in RtcpTransceiver 2017-11-30 14:34:40 +00:00
rtcp_transceiver_config.h Add rtcp observers for media receiver to RtcpTransceiverImpl 2017-12-13 12:22:41 +00:00
rtcp_transceiver_impl.cc Propagate media receiver rtcp observers to RtcpTransceiver 2017-12-14 17:39:13 +00:00
rtcp_transceiver_impl.h Propagate media receiver rtcp observers to RtcpTransceiver 2017-12-14 17:39:13 +00:00
rtcp_transceiver_impl_unittest.cc Propagate media receiver rtcp observers to RtcpTransceiver 2017-12-14 17:39:13 +00:00
rtcp_transceiver_unittest.cc Propagate media receiver rtcp observers to RtcpTransceiver 2017-12-14 17:39:13 +00:00
rtp_fec_unittest.cc Do not send 48 empty FEC packets when there is a large media packet seq. num. gap. 2017-12-07 11:22:30 +00:00
rtp_format.cc Cleanup of unused RTP structs and packetizer for stereo codec 2018-01-25 01:25:56 +00:00
rtp_format.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_format_h264.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_format_h264.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_format_h264_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_format_video_generic.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_format_video_generic.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_format_video_generic_unittest.cc Reland "Add stereo codec header and pass it through RTP" 2017-11-30 01:44:19 +00:00
rtp_format_vp8.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_format_vp8.h Delete unused VP8 packetization modes. 2017-10-27 09:18:17 +00:00
rtp_format_vp8_test_helper.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_format_vp8_test_helper.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_format_vp8_unittest.cc Delete unused VP8 packetization modes. 2017-10-27 09:18:17 +00:00
rtp_format_vp9.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_format_vp9.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_format_vp9_unittest.cc Reland "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices." 2017-11-07 16:34:20 +00:00
rtp_header_extension_map.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_header_extension_map_unittest.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_header_extensions.cc Support writing network timestamp delta fields into VideoTimingExtension 2017-11-01 10:15:56 +00:00
rtp_header_extensions.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_header_parser.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_packet.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
rtp_packet.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_packet_history.cc Don't overwrite RTP packets in history within one second or 3x RTT. 2018-01-23 17:08:28 +00:00
rtp_packet_history.h Don't overwrite packets in rtp packet history too early 2018-01-18 22:41:18 +00:00
rtp_packet_history_unittest.cc Don't overwrite RTP packets in history within one second or 3x RTT. 2018-01-23 17:08:28 +00:00
rtp_packet_received.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
rtp_packet_received.h Add application extension field to RtpPacketReceived. 2017-10-24 14:22:18 +00:00
rtp_packet_to_send.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_packet_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_payload_registry.cc Revert "Reland "Rename stereo video codec to multiplex"" 2018-01-27 00:45:20 +00:00
rtp_payload_registry_unittest.cc RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs 2017-10-04 11:30:14 +00:00
rtp_receiver_audio.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_receiver_audio.h RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs 2017-10-04 11:30:14 +00:00
rtp_receiver_impl.cc Optional: Use nullopt and implicit construction in /modules/rtp_rtcp 2017-11-23 11:36:08 +00:00
rtp_receiver_impl.h Delete in_order argument to RtpReceiver::IncomingRtpPacket 2017-10-05 07:19:20 +00:00
rtp_receiver_strategy.cc Convert PayloadUnion from a union to a class, step 1 2017-09-28 18:32:37 +00:00
rtp_receiver_strategy.h RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs 2017-10-04 11:30:14 +00:00
rtp_receiver_unittest.cc Delete in_order argument to RtpReceiver::IncomingRtpPacket 2017-10-05 07:19:20 +00:00
rtp_receiver_video.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_receiver_video.h RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs 2017-10-04 11:30:14 +00:00
rtp_rtcp_config.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_rtcp_impl.cc Don't overwrite packets in rtp packet history too early 2018-01-18 22:41:18 +00:00
rtp_rtcp_impl.h Change RtpRtcp::SetRemb signature to match RtcpTransceiver::SetRemb 2017-12-13 14:40:01 +00:00
rtp_rtcp_impl_unittest.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_sender.cc Don't overwrite packets in rtp packet history too early 2018-01-18 22:41:18 +00:00
rtp_sender.h Don't overwrite packets in rtp packet history too early 2018-01-18 22:41:18 +00:00
rtp_sender_audio.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_sender_audio.h Remove use of RTPFragmentationHeader from RTPSenderAudio 2017-10-30 09:56:19 +00:00
rtp_sender_unittest.cc Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead 2017-10-03 15:26:56 +00:00
rtp_sender_video.cc Cleanup of unused RTP structs and packetizer for stereo codec 2018-01-25 01:25:56 +00:00
rtp_sender_video.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_utility.cc Fixes to build WebRTC for Fuchsia 2017-12-12 23:37:28 +00:00
rtp_utility.h Convert PayloadUnion from a union to a class, step 3 2017-10-02 08:53:30 +00:00
rtp_utility_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_util.cc Add TimeMicrosToNtp to calculate current NtpTime without Clock 2017-11-28 10:11:58 +00:00
time_util.h Add TimeMicrosToNtp to calculate current NtpTime without Clock 2017-11-28 10:11:58 +00:00
time_util_unittest.cc Disable TimeUtilTest.TimeMicrosToNtpMatchRealTimeClockInitially on ios 2017-12-04 18:09:20 +00:00
tmmbr_help.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
tmmbr_help.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
ulpfec_generator.cc Use correct RTP header length in RED generation for ULPFEC packets. 2018-01-22 15:12:08 +00:00
ulpfec_generator.h Use correct RTP header length in RED generation for ULPFEC packets. 2018-01-22 15:12:08 +00:00
ulpfec_generator_unittest.cc Use correct RTP header length in RED generation for ULPFEC packets. 2018-01-22 15:12:08 +00:00
ulpfec_header_reader_writer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ulpfec_header_reader_writer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ulpfec_header_reader_writer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ulpfec_receiver_impl.cc Avoid infinite recursion if a RED packet encapsulate a RED packet. 2018-01-22 14:51:34 +00:00
ulpfec_receiver_impl.h Change ForwardErrorCorrection class to accept one received packet at a time. 2017-09-18 14:58:59 +00:00
ulpfec_receiver_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
video_codec_information.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00