Revert "sdp: parse and serialize b=TIAS"

This reverts commit c6801d4522ab94f965e258e68259fde312023654.

Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.

One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.

Original change's description:
> sdp: parse and serialize b=TIAS
> 
> BUG=webrtc:5788
> 
> Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31729}

TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:5788
Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31762}

3 files changed, 43 insertions, 88 deletions

diff --git a/pc/session_description.h b/pc/session_description.h
index 91b2a8bbe4..3405accbf3 100644
--- a/pc/session_description.h
+++ b/pc/session_description.h
@@ -126,10 +126,6 @@ class MediaContentDescription {
 
   virtual int bandwidth() const { return bandwidth_; }
   virtual void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
-  virtual std::string bandwidth_type() const { return bandwidth_type_; }
-  virtual void set_bandwidth_type(std::string bandwidth_type) {
-    bandwidth_type_ = bandwidth_type;
-  }
 
   virtual const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
   virtual void AddCrypto(const CryptoParams& params) {
@@ -255,7 +251,6 @@ class MediaContentDescription {
   bool rtcp_reduced_size_ = false;
   bool remote_estimate_ = false;
   int bandwidth_ = kAutoBandwidth;
-  std::string bandwidth_type_;
   std::string protocol_;
   std::vector<CryptoParams> cryptos_;
   std::vector<webrtc::RtpExtension> rtp_header_extensions_;
diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc
index ad005ff9b6..af584791be 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -225,8 +225,7 @@ static const char kMediaPortRejected[] = "0";
 static const char kDummyAddress[] = "0.0.0.0";
 static const char kDummyPort[] = "9";
 // RFC 3556
-static const char kApplicationSpecificBandwidth[] = "AS";
-static const char kTransportSpecificBandwidth[] = "TIAS";
+static const char kApplicationSpecificMaximum[] = "AS";
 
 static const char kDefaultSctpmapProtocol[] = "webrtc-datachannel";
 
@@ -1438,14 +1437,9 @@ void BuildMediaDescription(const ContentInfo* content_info,
 
   // RFC 4566
   // b=AS:<bandwidth>
-  int bandwidth = media_desc->bandwidth();
-  if (bandwidth >= 1000) {
-    std::string bandwidth_type = media_desc->bandwidth_type();
-    InitLine(kLineTypeSessionBandwidth, bandwidth_type, &os);
-    if (bandwidth_type == kApplicationSpecificBandwidth) {
-      bandwidth /= 1000;
-    }
-    os << kSdpDelimiterColon << bandwidth;
+  if (media_desc->bandwidth() >= 1000) {
+    InitLine(kLineTypeSessionBandwidth, kApplicationSpecificMaximum, &os);
+    os << kSdpDelimiterColon << (media_desc->bandwidth() / 1000);
     AddLine(os.str(), message);
   }
 
@@ -2989,60 +2983,46 @@ bool ParseContent(const std::string& message,
     // b=* (zero or more bandwidth information lines)
     if (IsLineType(line, kLineTypeSessionBandwidth)) {
       std::string bandwidth;
-      std::string bandwidth_type;
-      if (HasAttribute(line, kApplicationSpecificBandwidth)) {
-        if (!GetValue(line, kApplicationSpecificBandwidth, &bandwidth, error)) {
+      if (HasAttribute(line, kApplicationSpecificMaximum)) {
+        if (!GetValue(line, kApplicationSpecificMaximum, &bandwidth, error)) {
           return false;
+        } else {
+          int b = 0;
+          if (!GetValueFromString(line, bandwidth, &b, error)) {
+            return false;
+          }
+          // TODO(deadbeef): Historically, applications may be setting a value
+          // of -1 to mean "unset any previously set bandwidth limit", even
+          // though ommitting the "b=AS" entirely will do just that. Once we've
+          // transitioned applications to doing the right thing, it would be
+          // better to treat this as a hard error instead of just ignoring it.
+          if (b == -1) {
+            RTC_LOG(LS_WARNING)
+                << "Ignoring \"b=AS:-1\"; will be treated as \"no "
+                   "bandwidth limit\".";
+            continue;
+          }
+          if (b < 0) {
+            return ParseFailed(line, "b=AS value can't be negative.", error);
+          }
+          // We should never use more than the default bandwidth for RTP-based
+          // data channels. Don't allow SDP to set the bandwidth, because
+          // that would give JS the opportunity to "break the Internet".
+          // See: https://code.google.com/p/chromium/issues/detail?id=280726
+          if (media_type == cricket::MEDIA_TYPE_DATA &&
+              cricket::IsRtpProtocol(protocol) &&
+              b > cricket::kDataMaxBandwidth / 1000) {
+            rtc::StringBuilder description;
+            description << "RTP-based data channels may not send more than "
+                        << cricket::kDataMaxBandwidth / 1000 << "kbps.";
+            return ParseFailed(line, description.str(), error);
+          }
+          // Prevent integer overflow.
+          b = std::min(b, INT_MAX / 1000);
+          media_desc->set_bandwidth(b * 1000);
         }
-        bandwidth_type = kApplicationSpecificBandwidth;
-      } else if (HasAttribute(line, kTransportSpecificBandwidth)) {
-        if (!GetValue(line, kTransportSpecificBandwidth, &bandwidth, error)) {
-          return false;
-        }
-        bandwidth_type = kTransportSpecificBandwidth;
-      } else {
-        continue;
-      }
-      int b = 0;
-      if (!GetValueFromString(line, bandwidth, &b, error)) {
-        return false;
-      }
-      // TODO(deadbeef): Historically, applications may be setting a value
-      // of -1 to mean "unset any previously set bandwidth limit", even
-      // though ommitting the "b=AS" entirely will do just that. Once we've
-      // transitioned applications to doing the right thing, it would be
-      // better to treat this as a hard error instead of just ignoring it.
-      if (bandwidth_type == kApplicationSpecificBandwidth && b == -1) {
-        RTC_LOG(LS_WARNING) << "Ignoring \"b=AS:-1\"; will be treated as \"no "
-                               "bandwidth limit\".";
-        continue;
-      }
-      if (b < 0) {
-        return ParseFailed(
-            line, "b=" + bandwidth_type + " value can't be negative.", error);
-      }
-      // We should never use more than the default bandwidth for RTP-based
-      // data channels. Don't allow SDP to set the bandwidth, because
-      // that would give JS the opportunity to "break the Internet".
-      // See: https://code.google.com/p/chromium/issues/detail?id=280726
-      // Disallow TIAS since that is not generated for this.
-      if (media_type == cricket::MEDIA_TYPE_DATA &&
-          cricket::IsRtpProtocol(protocol) &&
-          (b > cricket::kDataMaxBandwidth / 1000 ||
-           bandwidth_type == kTransportSpecificBandwidth)) {
-        rtc::StringBuilder description;
-        description << "RTP-based data channels may not send more than "
-                    << cricket::kDataMaxBandwidth / 1000 << "kbps.";
-        return ParseFailed(line, description.str(), error);
       }
-      // Convert values. Prevent integer overflow.
-      if (bandwidth_type == kApplicationSpecificBandwidth) {
-        b = std::min(b, INT_MAX / 1000) * 1000;
-      } else {
-        b = std::min(b, INT_MAX);
-      }
-      media_desc->set_bandwidth(b);
-      media_desc->set_bandwidth_type(bandwidth_type);
+      continue;
     }
 
     // Parse the media level connection data.
diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index 67bcb2ec26..7b83c86ab1 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -2189,18 +2189,16 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBundle) {
 
 TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBandwidth) {
   VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
-  vcd->set_bandwidth(100 * 1000 + 755);  // Integer division will drop the 755.
-  vcd->set_bandwidth_type("AS");
+  vcd->set_bandwidth(100 * 1000);
   AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
   acd->set_bandwidth(50 * 1000);
-  acd->set_bandwidth_type("TIAS");
   ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
                                 jdesc_.session_version()));
   std::string message = webrtc::SdpSerialize(jdesc_);
   std::string sdp_with_bandwidth = kSdpFullString;
   InjectAfter("c=IN IP4 74.125.224.39\r\n", "b=AS:100\r\n",
               &sdp_with_bandwidth);
-  InjectAfter("c=IN IP4 74.125.127.126\r\n", "b=TIAS:50000\r\n",
+  InjectAfter("c=IN IP4 74.125.127.126\r\n", "b=AS:50\r\n",
               &sdp_with_bandwidth);
   EXPECT_EQ(sdp_with_bandwidth, message);
 }
@@ -2311,7 +2309,6 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
   JsepSessionDescription jsep_desc(kDummyType);
   AddRtpDataChannel();
   data_desc_->set_bandwidth(100 * 1000);
-  data_desc_->set_bandwidth_type("AS");
   MakeDescriptionWithoutCandidates(&jsep_desc);
   std::string message = webrtc::SdpSerialize(jsep_desc);
 
@@ -2615,23 +2612,6 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBandwidth) {
   EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bandwidth));
 }
 
-TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithTiasBandwidth) {
-  JsepSessionDescription jdesc_with_bandwidth(kDummyType);
-  std::string sdp_with_bandwidth = kSdpFullString;
-  InjectAfter("a=mid:video_content_name\r\na=sendrecv\r\n", "b=TIAS:100000\r\n",
-              &sdp_with_bandwidth);
-  InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n", "b=TIAS:50000\r\n",
-              &sdp_with_bandwidth);
-  EXPECT_TRUE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
-  VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
-  vcd->set_bandwidth(100 * 1000);
-  AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
-  acd->set_bandwidth(50 * 1000);
-  ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
-                                jdesc_.session_version()));
-  EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bandwidth));
-}
-
 TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithIceOptions) {
   JsepSessionDescription jdesc_with_ice_options(kDummyType);
   std::string sdp_with_ice_options = kSdpFullString;