Set local ssrc at construction (audio)
Changing the ssrc for a module is intended to be removed, and will in
the future require creating a new instance.
Bug: webrtc:10774
Change-Id: Ie96daa4a8cf00223ea040509037582f6b1c8eb19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145205
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28571}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index e7cee58..447dabe 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -97,7 +97,8 @@
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
- int rtcp_report_interval_ms);
+ int rtcp_report_interval_ms,
+ uint32_t ssrc);
~ChannelSend() override;
@@ -640,7 +641,8 @@
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
- int rtcp_report_interval_ms)
+ int rtcp_report_interval_ms,
+ uint32_t ssrc)
: event_log_(rtc_event_log),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
@@ -695,6 +697,8 @@
configuration.extmap_allow_mixed = extmap_allow_mixed;
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
+ configuration.media_send_ssrc = ssrc;
+
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
@@ -1256,12 +1260,13 @@
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
- int rtcp_report_interval_ms) {
+ int rtcp_report_interval_ms,
+ uint32_t ssrc) {
return absl::make_unique<ChannelSend>(
clock, task_queue_factory, module_process_thread, media_transport_config,
overhead_observer, rtp_transport, rtcp_rtt_stats, rtc_event_log,
frame_encryptor, crypto_options, extmap_allow_mixed,
- rtcp_report_interval_ms);
+ rtcp_report_interval_ms, ssrc);
}
} // namespace voe