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.cc b/audio/audio_receive_stream.cc
index 74f1773..3c53818 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -68,22 +68,16 @@
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log)
- : config_(config), audio_state_(audio_state) {
- RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
- RTC_DCHECK_NE(config_.voe_channel_id, -1);
+ : audio_state_(audio_state) {
+ RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.ToString();
+ RTC_DCHECK_NE(config.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
RTC_DCHECK(packet_router);
module_process_thread_checker_.DetachFromThread();
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
- channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
- channel_proxy_->SetRtcEventLog(event_log);
- channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
- // TODO(solenberg): Config NACK history window (which is a packet count),
- // using the actual packet size for the configured codec.
- channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
- config_.rtp.nack.rtp_history_ms / 20);
+ channel_proxy_ = voe_impl->GetChannelProxy(config.voe_channel_id);
// TODO(ossu): This is where we'd like to set the decoder factory to
// use. However, since it needs to be included when constructing Channel, we
@@ -94,25 +88,18 @@
RTC_CHECK_EQ(config.decoder_factory,
channel_proxy_->GetAudioDecoderFactory());
+ channel_proxy_->SetRtcEventLog(event_log);
channel_proxy_->RegisterTransport(config.rtcp_send_transport);
- channel_proxy_->SetReceiveCodecs(config.decoder_map);
- for (const auto& extension : config.rtp.extensions) {
- if (extension.uri == RtpExtension::kAudioLevelUri) {
- channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
- } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
- channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
- } else {
- RTC_NOTREACHED() << "Unsupported RTP extension.";
- }
- }
// Configure bandwidth estimation.
channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
// Register with transport.
rtp_stream_receiver_ =
- receiver_controller->CreateReceiver(config_.rtp.remote_ssrc,
+ receiver_controller->CreateReceiver(config.rtp.remote_ssrc,
channel_proxy_.get());
+
+ ConfigureStream(this, config, true);
}
AudioReceiveStream::~AudioReceiveStream() {
@@ -125,6 +112,13 @@
channel_proxy_->SetRtcEventLog(nullptr);
}
+void AudioReceiveStream::Reconfigure(
+ const webrtc::AudioReceiveStream::Config& config) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_LOG(LS_INFO) << "AudioReceiveStream::Reconfigure: " << config_.ToString();
+ ConfigureStream(this, config, false);
+}
+
void AudioReceiveStream::Start() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (playing_) {
@@ -329,5 +323,67 @@
RTC_DCHECK(audio_state);
return audio_state;
}
+
+AudioReceiveStream::ExtensionIds AudioReceiveStream::FindExtensionIds(
+ const std::vector<RtpExtension>& extensions) {
+ ExtensionIds ids;
+ for (const auto& extension : extensions) {
+ if (extension.uri == RtpExtension::kAudioLevelUri) {
+ ids.audio_level = extension.id;
+ } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
+ ids.transport_sequence_number = extension.id;
+ }
+ }
+ return ids;
+}
+
+void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
+ const Config& new_config,
+ bool first_time) {
+ RTC_DCHECK(stream);
+ const auto& channel_proxy = stream->channel_proxy_;
+ const auto& old_config = stream->config_;
+
+ // Configuration parameters which cannot be changed.
+ RTC_DCHECK(first_time ||
+ old_config.voe_channel_id == new_config.voe_channel_id);
+ RTC_DCHECK(first_time ||
+ old_config.rtp.remote_ssrc == new_config.rtp.remote_ssrc);
+ RTC_DCHECK(first_time ||
+ old_config.rtcp_send_transport == new_config.rtcp_send_transport);
+ // Decoder factory cannot be changed because it is configured at
+ // voe::Channel construction time.
+ RTC_DCHECK(first_time ||
+ old_config.decoder_factory == new_config.decoder_factory);
+
+ if (first_time || old_config.rtp.local_ssrc != new_config.rtp.local_ssrc) {
+ channel_proxy->SetLocalSSRC(new_config.rtp.local_ssrc);
+ }
+ // TODO(solenberg): Config NACK history window (which is a packet count),
+ // using the actual packet size for the configured codec.
+ if (first_time || old_config.rtp.nack.rtp_history_ms !=
+ new_config.rtp.nack.rtp_history_ms) {
+ channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
+ new_config.rtp.nack.rtp_history_ms / 20);
+ }
+ if (first_time || old_config.decoder_map != new_config.decoder_map) {
+ channel_proxy->SetReceiveCodecs(new_config.decoder_map);
+ }
+
+ // RTP Header Extensions.
+ const ExtensionIds old_ids = FindExtensionIds(old_config.rtp.extensions);
+ const ExtensionIds new_ids = FindExtensionIds(new_config.rtp.extensions);
+ if (first_time || new_ids.audio_level != old_ids.audio_level) {
+ channel_proxy->SetReceiveAudioLevelIndicationStatus(
+ new_ids.audio_level != 0, new_ids.audio_level);
+ }
+ if (first_time || new_ids.transport_sequence_number !=
+ old_ids.transport_sequence_number) {
+ channel_proxy->EnableReceiveTransportSequenceNumber(
+ new_ids.transport_sequence_number);
+ }
+
+ stream->config_ = new_config;
+}
} // namespace internal
} // namespace webrtc