Support N unsignaled audio streams.
BUG=webrtc:7175, webrtc:5208
Review-Url: https://codereview.webrtc.org/2685893002
Cr-Commit-Position: refs/heads/master@{#16952}
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index d774f78..996bc5a 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -40,12 +40,15 @@
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/include/field_trial.h"
+#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/transmit_mixer.h"
namespace cricket {
namespace {
+constexpr size_t kMaxUnsignaledRecvStreams = 50;
+
const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo |
webrtc::kTraceWarning | webrtc::kTraceError |
webrtc::kTraceCritical;
@@ -2235,12 +2238,10 @@
return false;
}
- // If the default receive stream was created with this ssrc, we unmark it as
- // being the default stream, and possibly recreate the AudioReceiveStream, if
- // sync_label has changed.
- if (IsDefaultRecvStream(ssrc)) {
+ // If this stream was previously received unsignaled, we promote it, possibly
+ // recreating the AudioReceiveStream, if sync_label has changed.
+ if (MaybeDeregisterUnsignaledRecvStream(ssrc)) {
recv_streams_[ssrc]->MaybeRecreateAudioReceiveStream(sp.sync_label);
- default_recv_ssrc_ = -1;
return true;
}
@@ -2304,10 +2305,7 @@
return false;
}
- // Deregister default channel, if that's the one being destroyed.
- if (IsDefaultRecvStream(ssrc)) {
- default_recv_ssrc_ = -1;
- }
+ MaybeDeregisterUnsignaledRecvStream(ssrc);
const int channel = it->second->channel();
@@ -2367,21 +2365,21 @@
bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ std::vector<uint32_t> ssrcs(1, ssrc);
if (ssrc == 0) {
default_recv_volume_ = volume;
- if (default_recv_ssrc_ == -1) {
- return true;
+ ssrcs = unsignaled_recv_ssrcs_;
+ }
+ for (uint32_t ssrc : ssrcs) {
+ const auto it = recv_streams_.find(ssrc);
+ if (it == recv_streams_.end()) {
+ LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc;
+ return false;
}
- ssrc = static_cast<uint32_t>(default_recv_ssrc_);
+ it->second->SetOutputVolume(volume);
+ LOG(LS_INFO) << "SetOutputVolume() to " << volume
+ << " for recv stream with ssrc " << ssrc;
}
- const auto it = recv_streams_.find(ssrc);
- if (it == recv_streams_.end()) {
- LOG(LS_WARNING) << "SetOutputVolume: no recv stream" << ssrc;
- return false;
- }
- it->second->SetOutputVolume(volume);
- LOG(LS_INFO) << "SetOutputVolume() to " << volume
- << " for recv stream with ssrc " << ssrc;
return true;
}
@@ -2432,35 +2430,53 @@
return;
}
- // Create a default receive stream for this unsignalled and previously not
- // received ssrc. If there already is a default receive stream, delete it.
+ // Create an unsignaled receive stream for this previously not received ssrc.
+ // If there already is N unsignaled receive streams, delete the oldest.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
uint32_t ssrc = 0;
if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
return;
}
+ RTC_DCHECK(std::find(unsignaled_recv_ssrcs_.begin(),
+ unsignaled_recv_ssrcs_.end(), ssrc) == unsignaled_recv_ssrcs_.end());
+ // Add new stream.
StreamParams sp;
sp.ssrcs.push_back(ssrc);
- LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
+ LOG(LS_INFO) << "Creating unsignaled receive stream for SSRC=" << ssrc;
if (!AddRecvStream(sp)) {
- LOG(LS_WARNING) << "Could not create default receive stream.";
+ LOG(LS_WARNING) << "Could not create unsignaled receive stream.";
return;
}
- if (default_recv_ssrc_ != -1) {
- LOG(LS_INFO) << "Removing default receive stream with ssrc "
- << default_recv_ssrc_;
- RTC_DCHECK_NE(ssrc, default_recv_ssrc_);
- RemoveRecvStream(default_recv_ssrc_);
- }
- default_recv_ssrc_ = ssrc;
+ unsignaled_recv_ssrcs_.push_back(ssrc);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.NumOfUnsignaledStreams", unsignaled_recv_ssrcs_.size(), 1,
+ 100, 101);
- SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
+ // Remove oldest unsignaled stream, if we have too many.
+ if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
+ uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front();
+ LOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
+ << remove_ssrc;
+ RemoveRecvStream(remove_ssrc);
+ }
+ RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size());
+
+ SetOutputVolume(ssrc, default_recv_volume_);
+
+ // The default sink can only be attached to one stream at a time, so we hook
+ // it up to the *latest* unsignaled stream we've seen, in order to support the
+ // case where the SSRC of one unsignaled stream changes.
if (default_sink_) {
+ for (uint32_t drop_ssrc : unsignaled_recv_ssrcs_) {
+ auto it = recv_streams_.find(drop_ssrc);
+ it->second->SetRawAudioSink(nullptr);
+ }
std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
new ProxySink(default_sink_.get()));
- SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
+ SetRawAudioSink(ssrc, std::move(proxy_sink));
}
+
delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
packet->cdata(),
packet->size(),
@@ -2625,17 +2641,17 @@
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
<< " " << (sink ? "(ptr)" : "NULL");
if (ssrc == 0) {
- if (default_recv_ssrc_ != -1) {
+ if (!unsignaled_recv_ssrcs_.empty()) {
std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
sink ? new ProxySink(sink.get()) : nullptr);
- SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
+ SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink));
}
default_sink_ = std::move(sink);
return;
}
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
- LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;
+ LOG(LS_WARNING) << "SetRawAudioSink: no recv stream " << ssrc;
return;
}
it->second->SetRawAudioSink(std::move(sink));
@@ -2664,6 +2680,19 @@
}
return -1;
}
+
+bool WebRtcVoiceMediaChannel::
+ MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ auto it = std::find(unsignaled_recv_ssrcs_.begin(),
+ unsignaled_recv_ssrcs_.end(),
+ ssrc);
+ if (it != unsignaled_recv_ssrcs_.end()) {
+ unsignaled_recv_ssrcs_.erase(it);
+ return true;
+ }
+ return false;
+}
} // namespace cricket
#endif // HAVE_WEBRTC_VOICE
