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}
diff --git a/pc/session_description.h b/pc/session_description.h
index 91b2a8b..3405acc 100644
--- a/pc/session_description.h
+++ b/pc/session_description.h
@@ -126,10 +126,6 @@
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 @@
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 ad005ff..af58479 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -225,8 +225,7 @@
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 @@
// 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 @@
// 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 67bcb2e..7b83c86 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -2189,18 +2189,16 @@
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 @@
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 @@
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;
