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Set local ssrc at construction (audio)

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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}
This commit is contained in:
Erik Språng
2019-07-11 15:20:15 +02:00
committed by Commit Bot
parent f89110d679
commit 4c2c412d7e
3 changed files with 18 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

12
audio/audio_send_stream.cc
View File

@ -113,7 +113,8 @@ AudioSendStream::AudioSendStream(
config.frame_encryptor,
config.crypto_options,
config.rtp.extmap_allow_mixed,
config.rtcp_report_interval_ms)) {}
config.rtcp_report_interval_ms,
config.rtp.ssrc)) {}
AudioSendStream::AudioSendStream(
Clock* clock,
@ -239,11 +240,12 @@ void AudioSendStream::ConfigureStream(
RTC_DCHECK(first_time ||
old_config.send_transport == new_config.send_transport);
if (first_time || old_config.rtp.ssrc != new_config.rtp.ssrc) {
if (old_config.rtp.ssrc != new_config.rtp.ssrc) {
channel_send->SetLocalSSRC(new_config.rtp.ssrc);
if (stream->suspended_rtp_state_) {
stream->rtp_rtcp_module_->SetRtpState(*stream->suspended_rtp_state_);
}
}
if (stream->suspended_rtp_state_ &&
(first_time || old_config.rtp.ssrc != new_config.rtp.ssrc)) {
stream->rtp_rtcp_module_->SetRtpState(*stream->suspended_rtp_state_);
}
if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
channel_send->SetRTCP_CNAME(new_config.rtp.c_name);

13
audio/channel_send.cc
View File

@ -97,7 +97,8 @@ class ChannelSend : public ChannelSendInterface,
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 @@ ChannelSend::ChannelSend(Clock* clock,
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 @@ ChannelSend::ChannelSend(Clock* clock,
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 @@ std::unique_ptr<ChannelSendInterface> CreateChannelSend(
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

3
audio/channel_send.h
View File

@ -140,7 +140,8 @@ std::unique_ptr<ChannelSendInterface> CreateChannelSend(
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);
} // namespace voe
} // namespace webrtc