Reconfigure, not reconstruct, AudioReceiveStreams.
In preparation of moving ownership of voe::Channel to the audio stream
classes, semantics for changing configuration properties on the receive
streams need to change, otherwise RTP, audio decoding and NetEq state
will be discarded when streams are recreated. The same pattern as for
AudioSendStream is applied, and the reconfigurable information is kept
to a minimum.
AudioReceiveStream:s may still be recreated when an unsignaled stream
is 'promoted' to signaled state, and the sync label changes at the
same time.
Bug: webrtc:4690
Change-Id: Ibad282965310c3c8174a91e05a659fa3e1827607
Reviewed-on: https://webrtc-review.googlesource.com/38300
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21560}diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 4d4af2e..e03fbd5 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -394,5 +394,46 @@
// Stop stream before it is being destructed.
recv_stream2.Stop();
}
+
+TEST(AudioReceiveStreamTest, ReconfigureWithSameConfig) {
+ ConfigHelper helper;
+ internal::AudioReceiveStream recv_stream(
+ helper.rtp_stream_receiver_controller(),
+ helper.packet_router(),
+ helper.config(), helper.audio_state(), helper.event_log());
+ recv_stream.Reconfigure(helper.config());
+}
+
+TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
+ ConfigHelper helper;
+ internal::AudioReceiveStream recv_stream(
+ helper.rtp_stream_receiver_controller(),
+ helper.packet_router(),
+ helper.config(), helper.audio_state(), helper.event_log());
+
+ auto new_config = helper.config();
+ new_config.rtp.local_ssrc = kLocalSsrc + 1;
+ new_config.rtp.nack.rtp_history_ms = 300 + 20;
+ new_config.rtp.extensions.clear();
+ new_config.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
+ new_config.rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberId + 1));
+ new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));
+
+ MockVoEChannelProxy& channel_proxy = *helper.channel_proxy();
+ EXPECT_CALL(channel_proxy, SetLocalSSRC(kLocalSsrc + 1)).Times(1);
+ EXPECT_CALL(channel_proxy, SetNACKStatus(true, 15 + 1)).Times(1);
+ EXPECT_CALL(channel_proxy, SetReceiveCodecs(new_config.decoder_map));
+ EXPECT_CALL(channel_proxy,
+ SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId + 1))
+ .Times(1);
+ EXPECT_CALL(channel_proxy,
+ EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId + 1))
+ .Times(1);
+
+ recv_stream.Reconfigure(new_config);
+}
} // namespace test
} // namespace webrtc