Remove default receive channel from WVoE; baby step 1.

Rx AGC config bits copied from https://codereview.webrtc.org/1315903004.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1388723002

Cr-Commit-Position: refs/heads/master@{#10233}

talk/media/base/mediachannel.h

@@ -160,7 +160,6 @@ struct AudioOptions {
   void SetAll(const AudioOptions& change) {
     echo_cancellation.SetFrom(change.echo_cancellation);
     auto_gain_control.SetFrom(change.auto_gain_control);
-    rx_auto_gain_control.SetFrom(change.rx_auto_gain_control);
     noise_suppression.SetFrom(change.noise_suppression);
     highpass_filter.SetFrom(change.highpass_filter);
     stereo_swapping.SetFrom(change.stereo_swapping);
@@ -181,10 +180,6 @@ struct AudioOptions {
     tx_agc_digital_compression_gain.SetFrom(
         change.tx_agc_digital_compression_gain);
     tx_agc_limiter.SetFrom(change.tx_agc_limiter);
-    rx_agc_target_dbov.SetFrom(change.rx_agc_target_dbov);
-    rx_agc_digital_compression_gain.SetFrom(
-        change.rx_agc_digital_compression_gain);
-    rx_agc_limiter.SetFrom(change.rx_agc_limiter);
     recording_sample_rate.SetFrom(change.recording_sample_rate);
     playout_sample_rate.SetFrom(change.playout_sample_rate);
     dscp.SetFrom(change.dscp);
@@ -194,7 +189,6 @@ struct AudioOptions {
   bool operator==(const AudioOptions& o) const {
     return echo_cancellation == o.echo_cancellation &&
         auto_gain_control == o.auto_gain_control &&
-        rx_auto_gain_control == o.rx_auto_gain_control &&
         noise_suppression == o.noise_suppression &&
         highpass_filter == o.highpass_filter &&
         stereo_swapping == o.stereo_swapping &&
@@ -213,9 +207,6 @@ struct AudioOptions {
         tx_agc_target_dbov == o.tx_agc_target_dbov &&
         tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
         tx_agc_limiter == o.tx_agc_limiter &&
-        rx_agc_target_dbov == o.rx_agc_target_dbov &&
-        rx_agc_digital_compression_gain == o.rx_agc_digital_compression_gain &&
-        rx_agc_limiter == o.rx_agc_limiter &&
         recording_sample_rate == o.recording_sample_rate &&
         playout_sample_rate == o.playout_sample_rate &&
         dscp == o.dscp &&
@@ -227,7 +218,6 @@ struct AudioOptions {
     ost << "AudioOptions {";
     ost << ToStringIfSet("aec", echo_cancellation);
     ost << ToStringIfSet("agc", auto_gain_control);
-    ost << ToStringIfSet("rx_agc", rx_auto_gain_control);
     ost << ToStringIfSet("ns", noise_suppression);
     ost << ToStringIfSet("hf", highpass_filter);
     ost << ToStringIfSet("swap", stereo_swapping);
@@ -248,10 +238,6 @@ struct AudioOptions {
     ost << ToStringIfSet("tx_agc_digital_compression_gain",
         tx_agc_digital_compression_gain);
     ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
-    ost << ToStringIfSet("rx_agc_target_dbov", rx_agc_target_dbov);
-    ost << ToStringIfSet("rx_agc_digital_compression_gain",
-        rx_agc_digital_compression_gain);
-    ost << ToStringIfSet("rx_agc_limiter", rx_agc_limiter);
     ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
     ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
     ost << ToStringIfSet("dscp", dscp);
@@ -265,8 +251,6 @@ struct AudioOptions {
   Settable<bool> echo_cancellation;
   // Audio processing to adjust the sensitivity of the local mic dynamically.
   Settable<bool> auto_gain_control;
-  // Audio processing to apply gain to the remote audio.
-  Settable<bool> rx_auto_gain_control;
   // Audio processing to filter out background noise.
   Settable<bool> noise_suppression;
   // Audio processing to remove background noise of lower frequencies.
@@ -291,9 +275,6 @@ struct AudioOptions {
   Settable<uint16_t> tx_agc_target_dbov;
   Settable<uint16_t> tx_agc_digital_compression_gain;
   Settable<bool> tx_agc_limiter;
-  Settable<uint16_t> rx_agc_target_dbov;
-  Settable<uint16_t> rx_agc_digital_compression_gain;
-  Settable<bool> rx_agc_limiter;
   Settable<uint32_t> recording_sample_rate;
   Settable<uint32_t> playout_sample_rate;
   // Set DSCP value for packet sent from audio channel.

talk/media/webrtc/fakewebrtcvoiceengine.h

@@ -208,8 +208,6 @@ class FakeWebRtcVoiceEngine
           opus_dtx(false),
           red(false),
           nack(false),
-          rx_agc_enabled(false),
-          rx_agc_mode(webrtc::kAgcDefault),
           cn8_type(13),
           cn16_type(105),
           dtmf_type(106),
@@ -224,7 +222,6 @@ class FakeWebRtcVoiceEngine
           neteq_capacity(-1),
           neteq_fast_accelerate(false) {
       memset(&send_codec, 0, sizeof(send_codec));
-      memset(&rx_agc_config, 0, sizeof(rx_agc_config));
     }
     bool external_transport;
     bool send;
@@ -238,9 +235,6 @@ class FakeWebRtcVoiceEngine
     bool opus_dtx;
     bool red;
     bool nack;
-    bool rx_agc_enabled;
-    webrtc::AgcModes rx_agc_mode;
-    webrtc::AgcConfig rx_agc_config;
     int cn8_type;
     int cn16_type;
     int dtmf_type;
@@ -1001,27 +995,12 @@ class FakeWebRtcVoiceEngine
   WEBRTC_STUB(SetRxNsStatus, (int channel, bool enable, webrtc::NsModes mode));
   WEBRTC_STUB(GetRxNsStatus, (int channel, bool& enabled,
                               webrtc::NsModes& mode));
-  WEBRTC_FUNC(SetRxAgcStatus, (int channel, bool enable,
+  WEBRTC_STUB(SetRxAgcStatus, (int channel, bool enable,
-                               webrtc::AgcModes mode)) {
+                               webrtc::AgcModes mode));
-    channels_[channel]->rx_agc_enabled = enable;
+  WEBRTC_STUB(GetRxAgcStatus, (int channel, bool& enabled,
-    channels_[channel]->rx_agc_mode = mode;
+                               webrtc::AgcModes& mode));
-    return 0;
+  WEBRTC_STUB(SetRxAgcConfig, (int channel, webrtc::AgcConfig config));
-  }
+  WEBRTC_STUB(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config));
-  WEBRTC_FUNC(GetRxAgcStatus, (int channel, bool& enabled,
-                               webrtc::AgcModes& mode)) {
-    enabled = channels_[channel]->rx_agc_enabled;
-    mode = channels_[channel]->rx_agc_mode;
-    return 0;
-  }
-
-  WEBRTC_FUNC(SetRxAgcConfig, (int channel, webrtc::AgcConfig config)) {
-    channels_[channel]->rx_agc_config = config;
-    return 0;
-  }
-  WEBRTC_FUNC(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config)) {
-    config = channels_[channel]->rx_agc_config;
-    return 0;
-  }
 
   WEBRTC_STUB(RegisterRxVadObserver, (int, webrtc::VoERxVadCallback&));
   WEBRTC_STUB(DeRegisterRxVadObserver, (int channel));

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) {
+  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));
@@ -1445,9 +1470,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
   // Check if DSCP value is changed from previous.
   bool dscp_option_changed = (options_.dscp != options.dscp);
 
-  // TODO(xians): Add support to set different options for different send
-  // streams after we support multiple APMs.
-
   // We retain all of the existing options, and apply the given ones
   // on top.  This means there is no way to "clear" options such that
   // they go back to the engine default.
@@ -1461,55 +1483,6 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
     }
   }
 
-  // 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.
-  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,
-            webrtc::kAgcFixedDigital) == -1) {
-      LOG_RTCERR1(SetRxAgcStatus, rx_auto_gain_control);
-      return false;
-    } else {
-      LOG(LS_VERBOSE) << "Rx auto gain set to " << rx_auto_gain_control
-                      << " with mode " << webrtc::kAgcFixedDigital;
-    }
-  }
-  if (options.rx_agc_target_dbov.IsSet() ||
-      options.rx_agc_digital_compression_gain.IsSet() ||
-      options.rx_agc_limiter.IsSet()) {
-    webrtc::AgcConfig config;
-    // If only some of the options are being overridden, get the current
-    // settings for the channel and bail if they aren't available.
-    if (!options.rx_agc_target_dbov.IsSet() ||
-        !options.rx_agc_digital_compression_gain.IsSet() ||
-        !options.rx_agc_limiter.IsSet()) {
-      if (engine()->voe()->processing()->GetRxAgcConfig(
-              voe_channel(), config) != 0) {
-        LOG(LS_ERROR) << "Failed to get default rx agc configuration for "
-                      << "channel " << voe_channel() << ". Since not all rx "
-                      << "agc options are specified, unable to safely set rx "
-                      << "agc options.";
-        return false;
-      }
-    }
-    config.targetLeveldBOv =
-        options.rx_agc_target_dbov.GetWithDefaultIfUnset(
-            config.targetLeveldBOv);
-    config.digitalCompressionGaindB =
-        options.rx_agc_digital_compression_gain.GetWithDefaultIfUnset(
-            config.digitalCompressionGaindB);
-    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,
-                  config.digitalCompressionGaindB, config.limiterEnable);
-      return false;
-    }
-  }
   if (dscp_option_changed) {
     rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
     if (options_.dscp.GetWithDefaultIfUnset(false))
@@ -1518,9 +1491,7 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
       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;
@@ -1528,9 +1499,14 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
 
 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;
+  }
 
   std::vector<AudioCodec> new_codecs;
   // Find all new codecs. We allow adding new codecs but don't allow changing
@@ -2229,17 +2205,18 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
 
-  rtc::CritScope lock(&receive_channels_cs_);
+  if (!ValidateStreamParams(sp)) {
-
-  if (!VERIFY(sp.ssrcs.size() == 1))
-    return false;
-  uint32_t ssrc = sp.first_ssrc();
-
-  if (ssrc == 0) {
-    LOG(LS_WARNING) << "AddRecvStream with 0 ssrc is not supported.";
     return false;
   }
 
+  uint32_t ssrc = sp.first_ssrc();
+  if (ssrc == 0) {
+    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;
@@ -2667,16 +2644,19 @@ 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 =
+    uint32_t ssrc = 0;
-        GetReceiveChannelId(ParseSsrc(packet->data(), packet->size(), true));
+    if (!GetRtcpSsrc(packet->data(), packet->size(), &ssrc)) {
-    if (which_channel != -1) {
+      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;
     }
   }

talk/media/webrtc/webrtcvoiceengine_unittest.cc

@@ -2219,23 +2219,6 @@ TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
   EXPECT_EQ(13, agc_config.targetLeveldBOv);
 }
 
-TEST_F(WebRtcVoiceEngineTestFake, RxAgcConfigViaOptions) {
-  EXPECT_TRUE(SetupEngine());
-  int channel_num = voe_.GetLastChannel();
-  send_parameters_.options.rx_agc_target_dbov.Set(6);
-  send_parameters_.options.rx_agc_digital_compression_gain.Set(0);
-  send_parameters_.options.rx_agc_limiter.Set(true);
-  send_parameters_.options.rx_auto_gain_control.Set(true);
-  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-
-  webrtc::AgcConfig agc_config;
-  EXPECT_EQ(0, engine_.voe()->processing()->GetRxAgcConfig(
-      channel_num, agc_config));
-  EXPECT_EQ(6, agc_config.targetLeveldBOv);
-  EXPECT_EQ(0, agc_config.digitalCompressionGaindB);
-  EXPECT_TRUE(agc_config.limiterEnable);
-}
-
 TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) {
   EXPECT_TRUE(SetupEngine());
   cricket::AudioOptions options;