codecs: Change interface, use SDP information for dynamic codecs

RTP static registered payload types fully describe the codec,
including clock rate and channels. For the payload formats that
use dynamic payload type number assignment, while some formats
have a fixed (or normal) clock rate and number of audio channels,
there are some codecs that can accept several possible values.

Change the codec plugin interface to accept these parameters,
and move the codec-specific state to a member of a new struct.

As an example, use this to implement the L16 media type for other
clock rates and number of audio channels, both the standard PT=10
stereo type as well as other clock rates negotiated via a dynamic
type. (See sip-rtp-l16.pcap on the SampleCaptures wiki page for
an example.)

Note that RTP Player doesn't support codecs returning output with
more than one channel currently, so downmix to mono.

The next step is adding the format parameters from fmtp to this.
(See #17608)

1 files changed, 2 insertions, 0 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index a63815b9ae..5ceb809fc7 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -303,6 +303,8 @@ void RtpAudioStream::decodeAudio(QAudioDeviceInfo out_device)
         }
 
         size_t decoded_bytes = decode_rtp_packet(rtp_packet, &decode_buff, decoders_hash_, &channels, &sample_rate);
+        // XXX: We don't actually *do* anything with channels, and just treat
+        // everything as if it were mono
 
         unsigned rtp_clock_rate = sample_rate;
         if (rtp_packet->info->info_payload_type == PT_G722) {