Fix some typos. Don't set sender_ssrc in record_packet. Refactor test.

elixir-webrtc · Jan 31, 2025 · b36e75d · b36e75d
1 parent f5a0198
commit b36e75d
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Showing 5 changed files with 10 additions and 7 deletions.
diff --git a/lib/ex_webrtc/peer_connection.ex b/lib/ex_webrtc/peer_connection.ex
@@ -165,7 +165,7 @@ defmodule ExWebRTC.PeerConnection do
   should be used for sending.
 
   Once the first RTP packet is sent (via `send_rtp/4`), `set_sender_codec/3`
-  can only be called with a codec with the clock rate.
+  can only be called with a codec with the same clock rate.
 
   Although very unlikely, keep in mind that after renegotiation,
   the selected codec may no longer be supported by the remote side and you might
@@ -177,6 +177,10 @@ defmodule ExWebRTC.PeerConnection do
   {:ok, pc} = PeerConnection.start_link()
   {:ok, rtp_sender} = PeerConnection.add_track(MediaStreamTrack.new(:video))
 
+  # exchange SDP with the remote side
+  # {:ok, offer} = PeerConnection.create_offer(pc)
+  # ...
+
   tr =
     pc
     |> PeerConnection.get_transceivers()

diff --git a/lib/ex_webrtc/rtp_sender.ex b/lib/ex_webrtc/rtp_sender.ex
@@ -263,8 +263,8 @@ defmodule ExWebRTC.RTPSender do
   defp rtx?(codec), do: String.ends_with?(codec.mime_type, "rtx")
   defp supported?(sender, codec), do: codec in sender.codecs
 
-  # As long as report recorder is not initialized i.e. we have not send any RTP packet
-  # allow for codec changes. Once we start sending RTP packet, require the same clock rate.
+  # As long as report recorder is not initialized i.e. we have not sent any RTP packet,
+  # allow for codec changes. Once we start sending RTP packets, require the same clock rate.
   defp same_clock_rate?(%{report_recorder: %{clock_rate: nil}}, _codec), do: true
   defp same_clock_rate?(sender, codec), do: sender.report_recorder.clock_rate == codec.clock_rate
 

diff --git a/lib/ex_webrtc/rtp_sender/report_recorder.ex b/lib/ex_webrtc/rtp_sender/report_recorder.ex
@@ -50,8 +50,7 @@ defmodule ExWebRTC.RTPSender.ReportRecorder do
   def record_packet(%{last_seq_no: nil} = recorder, packet, time) do
     %__MODULE__{
       recorder
-      | sender_ssrc: packet.ssrc,
-        last_rtp_timestamp: packet.timestamp,
+      | last_rtp_timestamp: packet.timestamp,
         last_seq_no: packet.sequence_number,
         last_timestamp: time,
         packet_count: 1,

diff --git a/test/ex_webrtc/rtp_sender/report_recorder_test.exs b/test/ex_webrtc/rtp_sender/report_recorder_test.exs
@@ -10,7 +10,7 @@ defmodule ExWebRTC.RTPSender.ReportRecorderTest do
   @rtp_ts 234_444
   @clock_rate 90_000
   @packet ExRTP.Packet.new(<<>>, sequence_number: @seq_no, timestamp: @rtp_ts)
-  @recorder %ReportRecorder{sender_ssrc: 123_467, clock_rate: @clock_rate}
+  @recorder ReportRecorder.init(%ReportRecorder{}, 123_467, @clock_rate)
 
   @ntp_offset 2_208_988_800
   @max_u32 0xFFFFFFFF

diff --git a/test/ex_webrtc/rtp_sender_test.exs b/test/ex_webrtc/rtp_sender_test.exs
@@ -49,7 +49,7 @@ defmodule ExWebRTC.RTPSenderTest do
     %{sender: sender}
   end
 
-  test "new/6" do
+  test "new/7" do
     track = MediaStreamTrack.new(:video)
 
     codecs = [@vp8, @av1, @av1_45khz, @rtx]