Commit 93fe814f authored by Michael Froman's avatar Michael Froman
Browse files

Bug 1828517 - Vendor libwebrtc from b3046c25aa 

Upstream commit: https://webrtc.googlesource.com/src/+/b3046c25aad89c20e435bc54a085ace622930665
    Use PacketReceiver::DeliverRtpPaket in scenario tests

    Change-Id: I36db102d023e4b716ce33a0afcff38b79b59b622

    Bug: webrtc:7135
    Change-Id: I36db102d023e4b716ce33a0afcff38b79b59b622
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290883


    Reviewed-by: default avatarErik Språng <sprang@webrtc.org>
    Commit-Queue: Per Kjellander <perkj@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#39090}
parent a7837ece

third_party/libwebrtc/README.moz-ff-commit

+3 −0
Original line number Diff line number Diff line
@@ -20886,3 +20886,6 @@ b613d62285 
# MOZ_LIBWEBRTC_SRC=/home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh

# base of lastest vendoring

5b7896be29

# MOZ_LIBWEBRTC_SRC=/home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh

# base of lastest vendoring

b3046c25aa

third_party/libwebrtc/README.mozilla

+2 −0
Original line number Diff line number Diff line
@@ -13946,3 +13946,5 @@ libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-l 
libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-21T03:54:38.724155.

# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc

libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-21T03:55:47.728276.

# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc

libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-21T03:56:44.643140.

third_party/libwebrtc/test/scenario/BUILD.gn

+1 −0
Original line number Diff line number Diff line
@@ -74,6 +74,7 @@ if (rtc_include_tests && !build_with_chromium) { 
      "../:test_common",

      "../:test_support",

      "../:video_test_common",

      "../../api:array_view",

      "../../api:create_frame_generator",

      "../../api:fec_controller_api",

      "../../api:frame_generator_api",

third_party/libwebrtc/test/scenario/audio_stream.cc

+16 −10
Original line number Diff line number Diff line
@@ -67,6 +67,20 @@ absl::optional<std::string> CreateAdaptationString( 
}

}  // namespace



std::vector<RtpExtension> GetAudioRtpExtensions(

    const AudioStreamConfig& config) {

  std::vector<RtpExtension> extensions;

  if (config.stream.in_bandwidth_estimation) {

    extensions.push_back({RtpExtension::kTransportSequenceNumberUri,

                          kTransportSequenceNumberExtensionId});

  }

  if (config.stream.abs_send_time) {

    extensions.push_back(

        {RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId});

  }

  return extensions;

}



SendAudioStream::SendAudioStream(

    CallClient* sender,

    AudioStreamConfig config,
@@ -120,13 +134,8 @@ SendAudioStream::SendAudioStream( 


  if (config.stream.in_bandwidth_estimation) {

    send_config.send_codec_spec->transport_cc_enabled = true;

    send_config.rtp.extensions = {{RtpExtension::kTransportSequenceNumberUri,

                                   kTransportSequenceNumberExtensionId}};

  }

  if (config.stream.abs_send_time) {

    send_config.rtp.extensions.push_back(

        {RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId});

  }

  send_config.rtp.extensions = GetAudioRtpExtensions(config);



  sender_->SendTask([&] {

    send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
@@ -179,10 +188,7 @@ ReceiveAudioStream::ReceiveAudioStream( 
  recv_config.rtcp_send_transport = feedback_transport;

  recv_config.rtp.remote_ssrc = send_stream->ssrc_;

  receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::AUDIO;

  if (config.stream.in_bandwidth_estimation) {

    recv_config.rtp.extensions = {{RtpExtension::kTransportSequenceNumberUri,

                                   kTransportSequenceNumberExtensionId}};

  }

  recv_config.rtp.extensions = GetAudioRtpExtensions(config);

  recv_config.decoder_factory = decoder_factory;

  recv_config.decoder_map = {

      {CallTest::kAudioSendPayloadType, {"opus", 48000, 2}}};

third_party/libwebrtc/test/scenario/audio_stream.h

+4 −0
Original line number Diff line number Diff line
@@ -100,6 +100,10 @@ class AudioStreamPair { 
  SendAudioStream send_stream_;

  ReceiveAudioStream receive_stream_;

};



std::vector<RtpExtension> GetAudioRtpExtensions(

    const AudioStreamConfig& config);



}  // namespace test

}  // namespace webrtc