update the rtp_to_hls example

membraneframework · Apr 22, 2024 · a395527 · a395527
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diff --git a/rtp_to_hls/README.md b/rtp_to_hls/README.md
@@ -12,13 +12,10 @@ To run the demo, you'll need to have [Elixir installed](https://elixir-lang.org/
 elixir rtp_to_hls.exs
 ```
 
-After a while, the server will start listening for UDP connections on port 5000.
-
-After that, you can start sending any H264 video and AAC audio stream via RTP. Below you can see an example of how to generate sample streams with [GStreamer](https://gstreamer.freedesktop.org/).
+After a while, the server will start listening for UDP connections on port 5000. Then, start the RTP stream with
 
 ```shell
-gst-launch-1.0 -v audiotestsrc ! audio/x-raw,rate=44100 ! faac ! rtpmp4gpay  pt=127 ! udpsink host=127.0.0.1 port=5000 \
-    videotestsrc ! video/x-raw,format=I420 ! x264enc key-int-max=10 tune=zerolatency ! rtph264pay pt=96 ! udpsink host=127.0.0.1 port=5000
+elixir send.exs
 ```
 
 When the server prints that playback is available, visit `http://localhost:8000/stream.html` and you should see the stream there. The stream can be also played with players other than the browser, like `vlc` or `ffplay`, for example
@@ -27,6 +24,14 @@ When the server prints that playback is available, visit `http://localhost:8000/
 ffplay http://localhost:8000/output/index.m3u8
 ```
 
+The RTP stream can be sent with other tools as well, for example with [GStreamer](https://gstreamer.freedesktop.org/):
+
+```shell
+gst-launch-1.0 -v audiotestsrc ! audio/x-raw,rate=44100 ! faac ! rtpmp4gpay  pt=127 ! udpsink host=127.0.0.1 port=5000 \
+    videotestsrc ! video/x-raw,format=I420 ! x264enc key-int-max=10 tune=zerolatency ! rtph264pay pt=96 ! udpsink host=127.0.0.1 port=5000
+```
+
+
 ## Copyright and License
 
 Copyright 2020, [Software Mansion](https://swmansion.com/?utm_source=git&utm_medium=readme&utm_campaign=membrane)

diff --git a/rtp_to_hls/rtp_to_hls.exs b/rtp_to_hls/rtp_to_hls.exs
@@ -3,12 +3,13 @@ Logger.configure(level: :info)
 
 Mix.install([
   {:membrane_core, "~> 1.0"},
-  {:membrane_udp_plugin, "~> 0.12.0"},
-  {:membrane_rtp_plugin, "~> 0.24.0"},
+  {:membrane_udp_plugin, "~> 0.13.0"},
+  {:membrane_rtp_plugin, "~> 0.27.1"},
   {:membrane_rtp_aac_plugin, "~> 0.8.0"},
   {:membrane_rtp_h264_plugin, "~> 0.19.0"},
-  {:membrane_http_adaptive_stream_plugin, "~> 0.18.0"},
-  {:membrane_fake_plugin, "~> 0.11.0"}
+  {:membrane_http_adaptive_stream_plugin, "~> 0.18.4"},
+  {:membrane_fake_plugin, "~> 0.11.0"},
+  {:membrane_h26x_plugin, "~> 0.10.0"}
 ])
 
 defmodule RtpToHls do
@@ -24,7 +25,7 @@ defmodule RtpToHls do
       |> child(:rtp, Membrane.RTP.SessionBin),
       child(:hls, %Membrane.HTTPAdaptiveStream.SinkBin{
         manifest_module: Membrane.HTTPAdaptiveStream.HLS,
-        target_window_duration: Membrane.Time.seconds(10),
+        target_window_duration: Membrane.Time.seconds(15),
         storage: %Membrane.HTTPAdaptiveStream.Storages.FileStorage{directory: "output"}
       })
     ]
@@ -36,7 +37,9 @@ defmodule RtpToHls do
   def handle_child_notification({:new_rtp_stream, ssrc, 96, _ext}, :rtp, _ctx, state) do
     spec =
       get_child(:rtp)
-      |> via_out(Pad.ref(:output, ssrc), options: [depayloader: Membrane.RTP.H264.Depayloader])
+      |> via_out(Pad.ref(:output, ssrc),
+        options: [depayloader: Membrane.RTP.H264.Depayloader]
+      )
       |> via_in(Pad.ref(:input, :video),
         options: [encoding: :H264, segment_duration: Membrane.Time.seconds(4)]
       )
@@ -49,7 +52,9 @@ defmodule RtpToHls do
   def handle_child_notification({:new_rtp_stream, ssrc, 127, _ext}, :rtp, _ctx, state) do
     spec =
       get_child(:rtp)
-      |> via_out(Pad.ref(:output, ssrc), options: [depayloader: Membrane.RTP.AAC.Depayloader])
+      |> via_out(Pad.ref(:output, ssrc),
+        options: [depayloader: Membrane.RTP.AAC.Depayloader]
+      )
       |> via_in(Pad.ref(:input, :audio),
         options: [encoding: :AAC, segment_duration: Membrane.Time.seconds(4)]
       )

diff --git a/rtp_to_hls/send.exs b/rtp_to_hls/send.exs
@@ -0,0 +1,100 @@
+require Logger
+Logger.configure(level: :info)
+
+Mix.install([
+  {:membrane_core, "~> 1.0"},
+  {:membrane_hackney_plugin, "~> 0.11.0"},
+  {:membrane_realtimer_plugin, "~> 0.9.0"},
+  {:membrane_h26x_plugin, "~> 0.10.0"},
+  {:membrane_aac_plugin, "~> 0.18.1"},
+  {:membrane_funnel_plugin, "~> 0.9.0"},
+  {:membrane_mp4_plugin, "~> 0.34.1"},
+  {:membrane_udp_plugin, "~> 0.13.0"},
+  {:membrane_rtp_plugin, "~> 0.27.1"},
+  {:membrane_rtp_aac_plugin, "~> 0.9.0"},
+  {:membrane_rtp_h264_plugin, "~> 0.19.1"}
+])
+
+defmodule SendRTP do
+  use Membrane.Pipeline
+
+  @samples_url "https://raw.githubusercontent.com/membraneframework/static/gh-pages/samples/big-buck-bunny/"
+
+  @mp4_url @samples_url <> "bun33s.mp4"
+
+  @impl true
+  def handle_init(_ctx, _opts) do
+    spec =
+      child(:video_src, %Membrane.Hackney.Source{
+        location: @mp4_url,
+        hackney_opts: [follow_redirect: true]
+      })
+      |> child(:mp4, Membrane.MP4.Demuxer.ISOM)
+
+    {[spec: spec], %{}}
+  end
+
+  @impl true
+  def handle_child_notification({:new_tracks, tracks}, :mp4, _ctx, state) do
+    audio_ssrc = 1234
+    video_ssrc = 1235
+    {audio_id, _format} = Enum.find(tracks, fn {_id, %format{}} -> format == Membrane.AAC end)
+    {video_id, _format} = Enum.find(tracks, fn {_id, %format{}} -> format == Membrane.H264 end)
+
+    spec = [
+      get_child(:mp4)
+      |> via_out(Pad.ref(:output, video_id))
+      |> child(:video_parser, %Membrane.H264.Parser{
+        output_stream_structure: :annexb,
+        output_alignment: :nalu
+      })
+      |> child(:video_realtimer, Membrane.Realtimer)
+      |> via_in(Pad.ref(:input, video_ssrc),
+        options: [payloader: Membrane.RTP.H264.Payloader]
+      )
+      |> child(:rtp, Membrane.RTP.SessionBin)
+      |> via_out(Pad.ref(:rtp_output, video_ssrc), options: [encoding: :H264])
+      |> get_child(:funnel),
+      get_child(:mp4)
+      |> via_out(Pad.ref(:output, audio_id))
+      |> child(:audio_realtimer, Membrane.Realtimer)
+      |> child(:audio_parser, %Membrane.AAC.Parser{out_encapsulation: :none})
+      |> via_in(Pad.ref(:input, audio_ssrc),
+        options: [payloader: %Membrane.RTP.AAC.Payloader{frames_per_packet: 1, mode: :hbr}]
+      )
+      |> get_child(:rtp)
+      |> via_out(Pad.ref(:rtp_output, audio_ssrc), options: [encoding: :AAC])
+      |> get_child(:funnel),
+      child(:funnel, Membrane.Funnel)
+      |> child(:udp, %Membrane.UDP.Sink{
+        destination_port_no: 5000,
+        destination_address: {127, 0, 0, 1}
+      })
+    ]
+
+    {[spec: spec], state}
+  end
+
+  @impl true
+  def handle_child_notification(_notification, _child, _ctx, state) do
+    {[], state}
+  end
+
+  @impl true
+  def handle_element_end_of_stream(:udp, :input, _ctx, state) do
+    {[terminate: :normal], state}
+  end
+
+  @impl true
+  def handle_element_end_of_stream(_element, _pad, _ctx, state) do
+    {[], state}
+  end
+end
+
+{:ok, supervisor, _pipeline} = Membrane.Pipeline.start_link(SendRTP)
+
+monitor = Process.monitor(supervisor)
+
+receive do
+  {:DOWN, ^monitor, _kind, _pid, :normal} -> :ok
+end