Delete ssrc book-keeping in NetEq
The ssrc for a given NetEq instance shouldn't change.
Bug: webrtc:7135
Change-Id: Iee0d4cd8bd5d917e819fa2ecf45a40e203c6d9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/111661
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25825}diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 2a025f3..436e60a 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -102,7 +102,6 @@
new_codec_(false),
timestamp_(0),
reset_decoder_(false),
- ssrc_(0),
first_packet_(true),
enable_fast_accelerate_(config.enable_fast_accelerate),
nack_enabled_(false),
@@ -533,8 +532,7 @@
return packet;
}());
- bool update_sample_rate_and_channels =
- first_packet_ || (rtp_header.ssrc != ssrc_);
+ bool update_sample_rate_and_channels = first_packet_;
if (update_sample_rate_and_channels) {
// Reset timestamp scaling.
@@ -561,9 +559,6 @@
packet_buffer_->Flush();
dtmf_buffer_->Flush();
- // Store new SSRC.
- ssrc_ = rtp_header.ssrc;
-
// Update audio buffer timestamp.
sync_buffer_->IncreaseEndTimestamp(main_timestamp - timestamp_);