Remove default receive channel from WVoE; baby step 1.

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 05b98ec154402c2de2d94c8716def60bb0692e8d..8607a7324347b19f78a455c0cf4376b8a759e8c8 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -148,6 +148,18 @@ const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
 const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
 #endif
 
+bool ValidateStreamParams(const StreamParams& sp) {

[pthatcher1, 2015/10/06 18:19:52]

To go along with VerifyUniquePayloadTypes, you might call this VerifyOneSsrc.

[the sun, 2015/10/07 10:50:25]

That may not be the only thing it checks, going forward. I'm using the same name
as we do in WebRtcVideoChannel2.

+  if (sp.ssrcs.empty()) {
+    LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
+    return false;
+  }
+  if (sp.ssrcs.size() > 1) {
+    LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: " << sp.ToString();
+    return false;
+  }
+  return true;
+}
+
 // Dumps an AudioCodec in RFC 2327-ish format.
 std::string ToString(const AudioCodec& codec) {
   std::stringstream ss;
@@ -221,6 +233,19 @@ bool FindCodec(const std::vector<AudioCodec>& codecs,
   return false;
 }
 
+bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) {
+  if (codecs.empty()) {
+    return true;
+  }
+  std::vector<int> payload_types;
+  for (const AudioCodec& codec : codecs) {
+    payload_types.push_back(codec.id);
+  }
+  std::sort(payload_types.begin(), payload_types.end());
+  auto it = std::unique(payload_types.begin(), payload_types.end());
+  return it == payload_types.end();
+}
+
 bool IsNackEnabled(const AudioCodec& codec) {
   return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
                                               kParamValueEmpty));
@@ -1461,14 +1486,39 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
     }
   }
 
+  if (!SetRecvOptions(voe_channel(), options)) {
+    return false;
+  }
+  for (const auto& ch : receive_channels_) {
+    if (!SetRecvOptions(ch.second->channel(), options)) {
+      return false;
+    }
+  }
+  if (dscp_option_changed) {
+    rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
+    if (options_.dscp.GetWithDefaultIfUnset(false))
+      dscp = kAudioDscpValue;
+    if (MediaChannel::SetDscp(dscp) != 0) {
+      LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
+    }
+  }
+  RecreateAudioReceiveStreams();
+  LOG(LS_INFO) << "Set voice channel options.  Current options: "
+               << options_.ToString();
+  return true;
+}
+
+bool WebRtcVoiceMediaChannel::SetRecvOptions(int channel_id,
+                                             const AudioOptions& options) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
   // Receiver-side auto gain control happens per channel, so set it here from
-  // options. Note that, like conference mode, setting it on the engine won't
-  // have the desired effect, since voice channels don't inherit options from
-  // the media engine when those options are applied per-channel.
+  // options. Note that voice channels don't inherit options from the media
+  // engine when those options are applied per-channel.
   bool rx_auto_gain_control;
   if (options.rx_auto_gain_control.Get(&rx_auto_gain_control)) {
     if (engine()->voe()->processing()->SetRxAgcStatus(
-            voe_channel(), rx_auto_gain_control,
+            channel_id, rx_auto_gain_control,
             webrtc::kAgcFixedDigital) == -1) {
       LOG_RTCERR1(SetRxAgcStatus, rx_auto_gain_control);
       return false;
@@ -1487,9 +1537,9 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
         !options.rx_agc_digital_compression_gain.IsSet() ||
         !options.rx_agc_limiter.IsSet()) {
       if (engine()->voe()->processing()->GetRxAgcConfig(
-              voe_channel(), config) != 0) {
+              channel_id, config) != 0) {
         LOG(LS_ERROR) << "Failed to get default rx agc configuration for "
-                      << "channel " << voe_channel() << ". Since not all rx "
+                      << "channel " << channel_id << ". Since not all rx "
                       << "agc options are specified, unable to safely set rx "
                       << "agc options.";
         return false;
@@ -1504,33 +1554,25 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
     config.limiterEnable = options.rx_agc_limiter.GetWithDefaultIfUnset(
         config.limiterEnable);
     if (engine()->voe()->processing()->SetRxAgcConfig(
-            voe_channel(), config) == -1) {
-      LOG_RTCERR4(SetRxAgcConfig, voe_channel(), config.targetLeveldBOv,
+            channel_id, config) == -1) {
+      LOG_RTCERR4(SetRxAgcConfig, channel_id, config.targetLeveldBOv,
                   config.digitalCompressionGaindB, config.limiterEnable);
       return false;
     }
   }
-  if (dscp_option_changed) {
-    rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
-    if (options_.dscp.GetWithDefaultIfUnset(false))
-      dscp = kAudioDscpValue;
-    if (MediaChannel::SetDscp(dscp) != 0) {
-      LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
-    }
-  }
-
-  RecreateAudioReceiveStreams();
-
-  LOG(LS_INFO) << "Set voice channel options.  Current options: "
-               << options_.ToString();
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::SetRecvCodecs(
     const std::vector<AudioCodec>& codecs) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   // Set the payload types to be used for incoming media.
-  LOG(LS_INFO) << "Setting receive voice codecs:";
+  LOG(LS_INFO) << "Setting receive voice codecs.";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
+  if (!VerifyUniquePayloadTypes(codecs)) {
+    LOG(LS_ERROR) << "Codec payload types overlap.";
+    return false;
+  }

[pthatcher1, 2015/10/06 18:19:52]

Can we add a unit test for this?

[the sun, 2015/10/07 10:50:25]

There is one already:
WebRtcVoiceEngineTestFake.SetRecvCodecsDuplicatePayloadType

 
   std::vector<AudioCodec> new_codecs;
   // Find all new codecs. We allow adding new codecs but don't allow changing
@@ -2228,17 +2270,18 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
 
-  rtc::CritScope lock(&receive_channels_cs_);
-
-  if (!VERIFY(sp.ssrcs.size() == 1))
+  if (!ValidateStreamParams(sp)) {
     return false;
-  uint32 ssrc = sp.first_ssrc();
+  }
 
+  uint32 ssrc = sp.first_ssrc();
   if (ssrc == 0) {
-    LOG(LS_WARNING) << "AddRecvStream with 0 ssrc is not supported.";
+    LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
     return false;
   }
 
+  rtc::CritScope lock(&receive_channels_cs_);
+
   if (receive_channels_.find(ssrc) != receive_channels_.end()) {
     LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
     return false;
@@ -2294,6 +2337,10 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
     return false;
   }
 
+  if (!SetRecvOptions(channel, options_)) {
+    return false;
+  }

[pthatcher1, 2015/10/06 18:19:52]

Can we add a unit test to cover that we are setting the options for all receive
streams and not just the default one?

+
   // Use the same SSRC as our default channel (so the RTCP reports are correct).
   unsigned int send_ssrc = 0;
   webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp();
@@ -2663,16 +2710,19 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
     return;
   }
 
-  // If it is a sender report, find the channel that is listening.
+  // If it is a sender report, find the receive channel that is listening.
   bool has_sent_to_default_channel = false;
   if (type == kRtcpTypeSR) {
-    int which_channel =
-        GetReceiveChannelId(ParseSsrc(packet->data(), packet->size(), true));
-    if (which_channel != -1) {
+    uint32 ssrc = 0;
+    if (!GetRtcpSsrc(packet->data(), packet->size(), &ssrc)) {
+      return;
+    }
+    int recv_channel_id = GetReceiveChannelId(ssrc);
+    if (recv_channel_id != -1) {
       engine()->voe()->network()->ReceivedRTCPPacket(
-          which_channel, packet->data(), packet->size());
+          recv_channel_id, packet->data(), packet->size());
 
-      if (IsDefaultChannel(which_channel))
+      if (IsDefaultChannel(recv_channel_id))
         has_sent_to_default_channel = true;
     }
   }