Shigemasa Watanabe bcb19c00ba Allow sending to separate payload types for each simulcast index. ...

This change is for mixed-codec simulcast.

By obtaining the payload type via RtpConfig::GetStreamConfig(),
the correct payload type can be retrieved regardless of whether
RtpConfig::stream_configs is initialized or not.

Bug: webrtc:362277533
Change-Id: I6b2a1ae66356b20a832565ce6729c3ce9e73a161
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364760
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43197}

2024-10-08 18:45:27 +00:00

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adaptation

Migrate absl::optional to std::optional

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test

Migrate absl::optional to std::optional

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audio_receive_stream.cc

Rename AudioReceiveStream to AudioReceiveStreamInterface

2022-05-23 08:44:26 +00:00

audio_receive_stream.h

rename timestamps to show epoch

2024-10-08 16:27:58 +00:00

audio_send_stream.cc

Apply include-cleaner to call/

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audio_send_stream.h

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audio_sender.h

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audio_state.cc

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audio_state.h

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bitrate_allocator_unittest.cc

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bitrate_allocator.cc

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bitrate_allocator.h

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bitrate_estimator_tests.cc

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BUILD.gn

Prune packets from pacer when sending deactivated via bitrate allocation

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call_config.cc

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call_config.h

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call_perf_tests.cc

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call_unittest.cc

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call.cc

Use PayloadTypePicker in WebRtcVoiceEngine

2024-09-09 18:44:21 +00:00

call.h

Move the payload type picker to call/

2024-09-03 12:36:50 +00:00

create_call.cc

Migrate absl::optional to std::optional

2024-09-02 12:16:47 +00:00

DEPS

Move the payload type picker to call/

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fake_network_pipe_unittest.cc

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fake_network_pipe.cc

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fake_network_pipe.h

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flexfec_receive_stream_impl.cc

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flexfec_receive_stream_impl.h

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flexfec_receive_stream_unittest.cc

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flexfec_receive_stream.cc

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flexfec_receive_stream.h

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OWNERS

Make perkj owner of call/

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packet_receiver.h

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payload_type_picker_unittest.cc

Move the payload type picker to call/

2024-09-03 12:36:50 +00:00

payload_type_picker.cc

Use PayloadTypePicker in WebRtcVoiceEngine

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payload_type_picker.h

Move the payload type picker to call/

2024-09-03 12:36:50 +00:00

payload_type.h

Use PayloadTypePicker in WebRtcVoiceEngine

2024-09-09 18:44:21 +00:00

rampup_tests.cc

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rampup_tests.h

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receive_stream.h

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receive_time_calculator_unittest.cc

Migrate absl::optional to std::optional

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receive_time_calculator.cc

Apply include-cleaner to call/

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receive_time_calculator.h

WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf

2022-03-29 10:14:00 +00:00

rtp_bitrate_configurator_unittest.cc

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rtp_bitrate_configurator.cc

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rtp_bitrate_configurator.h

Migrate absl::optional to std::optional

2024-09-02 12:16:47 +00:00

rtp_config.cc

Allow sending to separate payload types for each simulcast index.

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rtp_config.h

Allow sending to separate payload types for each simulcast index.

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rtp_demuxer_unittest.cc

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rtp_demuxer.cc

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rtp_demuxer.h

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rtp_packet_sink_interface.h

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rtp_payload_params_unittest.cc

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rtp_payload_params.cc

Add FrameInstrumentationData to RTPVideoHeader and CodecSpecificInfo

2024-09-04 07:21:02 +00:00

rtp_payload_params.h

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rtp_stream_receiver_controller_interface.h

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rtp_stream_receiver_controller.cc

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rtp_stream_receiver_controller.h

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rtp_transport_config.h

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rtp_transport_controller_send_factory_interface.h

Use Environment in RtpTransportyControllerSend

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rtp_transport_controller_send_factory.h

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rtp_transport_controller_send_interface.h

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rtp_transport_controller_send.cc

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rtp_transport_controller_send.h

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rtp_video_sender_interface.h

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rtp_video_sender_unittest.cc

Allow sending to separate payload types for each simulcast index.

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rtp_video_sender.cc

Allow sending to separate payload types for each simulcast index.

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rtp_video_sender.h

Prune packets from pacer when sending deactivated via bitrate allocation

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rtx_receive_stream_unittest.cc

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rtx_receive_stream.cc

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rtx_receive_stream.h

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simulated_network.h

Move call/simulated_network to test/network

2024-04-29 09:55:06 +00:00

simulated_packet_receiver.h

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syncable.cc

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syncable.h

Migrate absl::optional to std::optional

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version.cc

Update WebRTC code version (2024-10-08T04:06:02).

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version.h

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video_receive_stream.cc

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video_receive_stream.h

rename timestamps to show epoch

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video_send_stream.cc

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video_send_stream.h

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