RTP demo with RawAudio

Hello friends, I'm trying to get microphone input (via

Membrane.PortAudio.Source

Membrane.PortAudio.Source

Membrane.PortAudio.Source

Membrane.PortAudio.Source) packaged into an RTP stream and sent to a server and can't quite seem to get it right.

Excerpt below based on the demo in

membrane-demo/rtp

membrane-demo/rtp

membrane-demo/rtp

membrane-demo/rtp but with microphone input substituted and newer syntax.

alias Membrane.{RTP, UDP, PortAudio, RawAudio}
...
links = [
      child(:mic_input, %PortAudio.Source{
        channels: 2,
        sample_rate: 24_000
      })
      |> child(:encoder, %Membrane.Opus.Encoder{
        application: :audio,
        input_stream_format: %RawAudio{
          channels: 2,
          sample_format: :s16le,
          sample_rate: 24_000
        }
      })
      |> via_in(Pad.ref(:input, audio_ssrc), options: [payloader: RTP.Opus.Payloader])
      |> child(:rtp, %RTP.SessionBin{
        secure?: secure?,
        srtp_policies: [
          %ExLibSRTP.Policy{
            ssrc: :any_inbound,
            key: srtp_key
          }
        ]
      })
      |> via_out(Pad.ref(:rtp_output, audio_ssrc), options: [encoding: :OPUS])
      |> child(:audio_realtimer, Membrane.Realtimer)
      |> child(:audio_sink, %UDP.Sink{
        destination_port_no: destination_port,
        destination_address: destination_address
      })
    ]

alias Membrane.{RTP, UDP, PortAudio, RawAudio}
...
links = [
      child(:mic_input, %PortAudio.Source{
        channels: 2,
        sample_rate: 24_000
      })
      |> child(:encoder, %Membrane.Opus.Encoder{
        application: :audio,
        input_stream_format: %RawAudio{
          channels: 2,
          sample_format: :s16le,
          sample_rate: 24_000
        }
      })
      |> via_in(Pad.ref(:input, audio_ssrc), options: [payloader: RTP.Opus.Payloader])
      |> child(:rtp, %RTP.SessionBin{
        secure?: secure?,
        srtp_policies: [
          %ExLibSRTP.Policy{
            ssrc: :any_inbound,
            key: srtp_key
          }
        ]
      })
      |> via_out(Pad.ref(:rtp_output, audio_ssrc), options: [encoding: :OPUS])
      |> child(:audio_realtimer, Membrane.Realtimer)
      |> child(:audio_sink, %UDP.Sink{
        destination_port_no: destination_port,
        destination_address: destination_address
      })
    ]

alias Membrane.{RTP, UDP, PortAudio, RawAudio}
...
links = [
      child(:mic_input, %PortAudio.Source{
        channels: 2,
        sample_rate: 24_000
      })
      |> child(:encoder, %Membrane.Opus.Encoder{
        application: :audio,
        input_stream_format: %RawAudio{
          channels: 2,
          sample_format: :s16le,
          sample_rate: 24_000
        }
      })
      |> via_in(Pad.ref(:input, audio_ssrc), options: [payloader: RTP.Opus.Payloader])
      |> child(:rtp, %RTP.SessionBin{
        secure?: secure?,
        srtp_policies: [
          %ExLibSRTP.Policy{
            ssrc: :any_inbound,
            key: srtp_key
          }
        ]
      })
      |> via_out(Pad.ref(:rtp_output, audio_ssrc), options: [encoding: :OPUS])
      |> child(:audio_realtimer, Membrane.Realtimer)
      |> child(:audio_sink, %UDP.Sink{
        destination_port_no: destination_port,
        destination_address: destination_address
      })
    ]

alias Membrane.{RTP, UDP, PortAudio, RawAudio}
...
links = [
      child(:mic_input, %PortAudio.Source{
        channels: 2,
        sample_rate: 24_000
      })
      |> child(:encoder, %Membrane.Opus.Encoder{
        application: :audio,
        input_stream_format: %RawAudio{
          channels: 2,
          sample_format: :s16le,
          sample_rate: 24_000
        }
      })
      |> via_in(Pad.ref(:input, audio_ssrc), options: [payloader: RTP.Opus.Payloader])
      |> child(:rtp, %RTP.SessionBin{
        secure?: secure?,
        srtp_policies: [
          %ExLibSRTP.Policy{
            ssrc: :any_inbound,
            key: srtp_key
          }
        ]
      })
      |> via_out(Pad.ref(:rtp_output, audio_ssrc), options: [encoding: :OPUS])
      |> child(:audio_realtimer, Membrane.Realtimer)
      |> child(:audio_sink, %UDP.Sink{
        destination_port_no: destination_port,
        destination_address: destination_address
      })
    ]

This seems to throw an error when generating the headers because buffer.pts is

nil

nil

nil

nil:

[error] <0.815.0>/:rtp/{:stream_send_bin, 1236}/:payloader/:header_generator Error handling action {:split, {:handle_buffer, [[:input, %Membrane.Buffer{payload: <<220, 255, 254>>, pts: nil, dts: nil, metadata: %{}}]]}} returned by callback Membrane.RTP.HeaderGenerator.handle_buffers_batch

[error] GenServer #PID<0.849.0> terminating
** (FunctionClauseError) no function clause matching in Ratio.mult/2
    (ratio 2.4.2) lib/ratio.ex:418: Ratio.mult(nil, 48000)
    (membrane_rtp_plugin 0.23.0) lib/membrane/rtp/header_generator.ex:71: Membrane.RTP.HeaderGenerator.handle_process/4

[error] <0.815.0>/:rtp/{:stream_send_bin, 1236}/:payloader/:header_generator Error handling action {:split, {:handle_buffer, [[:input, %Membrane.Buffer{payload: <<220, 255, 254>>, pts: nil, dts: nil, metadata: %{}}]]}} returned by callback Membrane.RTP.HeaderGenerator.handle_buffers_batch

[error] GenServer #PID<0.849.0> terminating
** (FunctionClauseError) no function clause matching in Ratio.mult/2
    (ratio 2.4.2) lib/ratio.ex:418: Ratio.mult(nil, 48000)
    (membrane_rtp_plugin 0.23.0) lib/membrane/rtp/header_generator.ex:71: Membrane.RTP.HeaderGenerator.handle_process/4

[error] <0.815.0>/:rtp/{:stream_send_bin, 1236}/:payloader/:header_generator Error handling action {:split, {:handle_buffer, [[:input, %Membrane.Buffer{payload: <<220, 255, 254>>, pts: nil, dts: nil, metadata: %{}}]]}} returned by callback Membrane.RTP.HeaderGenerator.handle_buffers_batch

[error] GenServer #PID<0.849.0> terminating
** (FunctionClauseError) no function clause matching in Ratio.mult/2
    (ratio 2.4.2) lib/ratio.ex:418: Ratio.mult(nil, 48000)
    (membrane_rtp_plugin 0.23.0) lib/membrane/rtp/header_generator.ex:71: Membrane.RTP.HeaderGenerator.handle_process/4

[error] <0.815.0>/:rtp/{:stream_send_bin, 1236}/:payloader/:header_generator Error handling action {:split, {:handle_buffer, [[:input, %Membrane.Buffer{payload: <<220, 255, 254>>, pts: nil, dts: nil, metadata: %{}}]]}} returned by callback Membrane.RTP.HeaderGenerator.handle_buffers_batch

[error] GenServer #PID<0.849.0> terminating
** (FunctionClauseError) no function clause matching in Ratio.mult/2
    (ratio 2.4.2) lib/ratio.ex:418: Ratio.mult(nil, 48000)
    (membrane_rtp_plugin 0.23.0) lib/membrane/rtp/header_generator.ex:71: Membrane.RTP.HeaderGenerator.handle_process/4

Does anyone have any tips on what I might be doing wrong?

CalmsOP•7/21/23, 5:58 AM

Ah, just stumbled across this issue:
https://github.com/membraneframework/membrane_opus_plugin/issues/22
Looks like I wasn't going mad after all - opus plugin doesn't support it yet.

CalmsOP•7/21/23, 6:21 AM

Although I don't quite understand why I'm hitting this issue when membrane-demo/rtp works fine - perhaps a difference with the underlying RawAudio stream instead of a file?

varsill•7/21/23, 9:35 AM

Hello @calms! I believe that adding child(:parser, Membrane.Opus.Parser)child(:parser, Membrane.Opus.Parser) right after the encoder should solve your problem, since the Opus.ParserOpus.Parser adds some metadata and PTS to the buffers flowing in the stream.

CalmsOP•7/21/23, 9:50 AM

Ah, that does indeed solve the problem with buffer.ptsbuffer.pts!

Many thanks!

CalmsOP•7/21/23, 9:50 AM

With that in place, this now works (although it's super quiet...) - happy to mark this as solved.

If you had any tips on volume, I'd be all ears though!

RTP demo with RawAudio

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