Remove unused non-standard RtpEncodingParameters members
Bug: webrtc:7580
Change-Id: Ic1a6e52f25eb35c797e669bffe8040ec84fec386
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160415
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29983}
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index 77db960..124abc9 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -380,30 +380,6 @@
// unset SSRC acts as a "wildcard" SSRC.
absl::optional<uint32_t> ssrc;
- // Can be used to reference a codec in the |codecs| member of the
- // RtpParameters that contains this RtpEncodingParameters. If unset, the
- // implementation will choose the first possible codec (if a sender), or
- // prepare to receive any codec (for a receiver).
- // TODO(deadbeef): Not implemented. Implementation of RtpSender will always
- // choose the first codec from the list.
- absl::optional<int> codec_payload_type;
-
- // Specifies the FEC mechanism, if set.
- // TODO(deadbeef): Not implemented. Current implementation will use whatever
- // FEC codecs are available, including red+ulpfec.
- absl::optional<RtpFecParameters> fec;
-
- // Specifies the RTX parameters, if set.
- // TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
- absl::optional<RtpRtxParameters> rtx;
-
- // Only used for audio. If set, determines whether or not discontinuous
- // transmission will be used, if an available codec supports it. If not
- // set, the implementation default setting will be used.
- // TODO(deadbeef): Not implemented. Current implementation will use a CN
- // codec as long as it's present.
- absl::optional<DtxStatus> dtx;
-
// The relative bitrate priority of this encoding. Currently this is
// implemented for the entire rtp sender by using the value of the first
// encoding parameter.
@@ -421,14 +397,6 @@
// TODO(http://crbug.com/webrtc/8630): Implement this per encoding parameter.
double network_priority = kDefaultBitratePriority;
- // Indicates the preferred duration of media represented by a packet in
- // milliseconds for this encoding. If set, this will take precedence over the
- // ptime set in the RtpCodecParameters. This could happen if SDP negotiation
- // creates a ptime for a specific codec, which is later changed in the
- // RtpEncodingParameters by the application.
- // TODO(bugs.webrtc.org/8819): Not implemented.
- absl::optional<int> ptime;
-
// If set, this represents the Transport Independent Application Specific
// maximum bandwidth defined in RFC3890. If unset, there is no maximum
// bitrate. Currently this is implemented for the entire rtp sender by using
@@ -443,7 +411,6 @@
absl::optional<int> max_bitrate_bps;
// Specifies the minimum bitrate in bps for video.
- // TODO(asapersson): Not implemented for ORTC API.
absl::optional<int> min_bitrate_bps;
// Specifies the maximum framerate in fps for video.
@@ -462,10 +429,6 @@
// For video, scale the resolution down by this factor.
absl::optional<double> scale_resolution_down_by;
- // Scale the framerate down by this factor.
- // TODO(deadbeef): Not implemented.
- absl::optional<double> scale_framerate_down_by;
-
// For an RtpSender, set to true to cause this encoding to be encoded and
// sent, and false for it not to be encoded and sent. This allows control
// across multiple encodings of a sender for turning simulcast layers on and
@@ -478,24 +441,15 @@
// Called "encodingId" in ORTC.
std::string rid;
- // RIDs of encodings on which this layer depends.
- // Called "dependencyEncodingIds" in ORTC spec.
- // TODO(deadbeef): Not implemented.
- std::vector<std::string> dependency_rids;
-
bool operator==(const RtpEncodingParameters& o) const {
- return ssrc == o.ssrc && codec_payload_type == o.codec_payload_type &&
- fec == o.fec && rtx == o.rtx && dtx == o.dtx &&
- bitrate_priority == o.bitrate_priority &&
- network_priority == o.network_priority && ptime == o.ptime &&
+ return ssrc == o.ssrc && bitrate_priority == o.bitrate_priority &&
+ network_priority == o.network_priority &&
max_bitrate_bps == o.max_bitrate_bps &&
min_bitrate_bps == o.min_bitrate_bps &&
max_framerate == o.max_framerate &&
num_temporal_layers == o.num_temporal_layers &&
scale_resolution_down_by == o.scale_resolution_down_by &&
- scale_framerate_down_by == o.scale_framerate_down_by &&
- active == o.active && rid == o.rid &&
- dependency_rids == o.dependency_rids;
+ active == o.active && rid == o.rid;
}
bool operator!=(const RtpEncodingParameters& o) const {
return !(*this == o);
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index b709992..a1f50c5 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -1460,53 +1460,6 @@
.error()
.type());
init.send_encodings = default_send_encodings;
-
- init.send_encodings[0].codec_payload_type = 1;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init)
- .error()
- .type());
- init.send_encodings = default_send_encodings;
-
- init.send_encodings[0].fec = RtpFecParameters();
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init)
- .error()
- .type());
- init.send_encodings = default_send_encodings;
-
- init.send_encodings[0].rtx = RtpRtxParameters();
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init)
- .error()
- .type());
- init.send_encodings = default_send_encodings;
-
- init.send_encodings[0].dtx = DtxStatus::ENABLED;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init)
- .error()
- .type());
- init.send_encodings = default_send_encodings;
-
- init.send_encodings[0].ptime = 1;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init)
- .error()
- .type());
- init.send_encodings = default_send_encodings;
-
- init.send_encodings[0].dependency_rids.push_back("dummy_rid");
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init)
- .error()
- .type());
}
// Test that AddTransceiver fails if trying to use invalid RTP encoding
diff --git a/pc/rtp_parameters_conversion.cc b/pc/rtp_parameters_conversion.cc
index b7fb691..363fa06 100644
--- a/pc/rtp_parameters_conversion.cc
+++ b/pc/rtp_parameters_conversion.cc
@@ -234,17 +234,9 @@
}
cricket::StreamParamsVec cricket_streams;
const RtpEncodingParameters& encoding = encodings[0];
- if (encoding.rtx && encoding.rtx->ssrc && !encoding.ssrc) {
- LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
- "Setting an RTX SSRC explicitly while leaving the "
- "primary SSRC unset is not currently supported.");
- }
if (encoding.ssrc) {
cricket::StreamParams stream_params;
stream_params.add_ssrc(*encoding.ssrc);
- if (encoding.rtx && encoding.rtx->ssrc) {
- stream_params.AddFidSsrc(*encoding.ssrc, *encoding.rtx->ssrc);
- }
cricket_streams.push_back(std::move(stream_params));
}
return std::move(cricket_streams);
@@ -308,11 +300,6 @@
for (const cricket::StreamParams& stream_param : stream_params) {
RtpEncodingParameters rtp_encoding;
rtp_encoding.ssrc.emplace(stream_param.first_ssrc());
- uint32_t rtx_ssrc = 0;
- if (stream_param.GetFidSsrc(stream_param.first_ssrc(), &rtx_ssrc)) {
- RtpRtxParameters rtx_param(rtx_ssrc);
- rtp_encoding.rtx.emplace(rtx_param);
- }
rtp_encodings.push_back(std::move(rtp_encoding));
}
return rtp_encodings;
diff --git a/pc/rtp_parameters_conversion_unittest.cc b/pc/rtp_parameters_conversion_unittest.cc
index 3d64d62..44dc0df 100644
--- a/pc/rtp_parameters_conversion_unittest.cc
+++ b/pc/rtp_parameters_conversion_unittest.cc
@@ -346,23 +346,6 @@
EXPECT_EQ(0xbaadf00d, result.value()[0].first_ssrc());
}
-TEST(RtpParametersConversionTest, ToCricketStreamParamsVecWithRtx) {
- std::vector<RtpEncodingParameters> encodings;
- RtpEncodingParameters encoding;
- // Test a corner case SSRC of 0.
- encoding.ssrc.emplace(0u);
- encoding.rtx.emplace(0xdeadbeef);
- encodings.push_back(encoding);
- auto result = ToCricketStreamParamsVec(encodings);
- ASSERT_TRUE(result.ok());
- ASSERT_EQ(1u, result.value().size());
- EXPECT_EQ(2u, result.value()[0].ssrcs.size());
- EXPECT_EQ(0u, result.value()[0].first_ssrc());
- uint32_t rtx_ssrc = 0;
- EXPECT_TRUE(result.value()[0].GetFidSsrc(0u, &rtx_ssrc));
- EXPECT_EQ(0xdeadbeef, rtx_ssrc);
-}
-
// No encodings should be accepted; an endpoint may want to prepare a
// decoder/encoder without having something to receive/send yet.
TEST(RtpParametersConversionTest, ToCricketStreamParamsVecNoEncodings) {
@@ -377,21 +360,11 @@
TEST(RtpParametersConversionTest, ToCricketStreamParamsVecMissingSsrcs) {
std::vector<RtpEncodingParameters> encodings = {{}};
// Creates RtxParameters with empty SSRC.
- encodings[0].rtx.emplace();
auto result = ToCricketStreamParamsVec(encodings);
ASSERT_TRUE(result.ok());
EXPECT_EQ(0u, result.value().size());
}
-// The media engine doesn't have a way of receiving an RTX SSRC that's known
-// with a primary SSRC that's unknown, so this should produce an error.
-TEST(RtpParametersConversionTest, ToStreamParamsWithPrimarySsrcSetAndRtxUnset) {
- std::vector<RtpEncodingParameters> encodings = {{}};
- encodings[0].rtx.emplace(0xdeadbeef);
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- ToCricketStreamParamsVec(encodings).error().type());
-}
-
// TODO(deadbeef): Update this test when we support multiple encodings.
TEST(RtpParametersConversionTest, ToCricketStreamParamsVecMultipleEncodings) {
std::vector<RtpEncodingParameters> encodings = {{}, {}};
@@ -511,11 +484,9 @@
cricket::StreamParamsVec streams;
cricket::StreamParams stream1;
stream1.ssrcs.push_back(1111u);
- stream1.AddFidSsrc(1111u, 0xaaaaaaaa);
cricket::StreamParams stream2;
stream2.ssrcs.push_back(2222u);
- stream2.AddFidSsrc(2222u, 0xaaaaaaab);
streams.push_back(stream1);
streams.push_back(stream2);
@@ -523,9 +494,7 @@
auto rtp_encodings = ToRtpEncodings(streams);
ASSERT_EQ(2u, rtp_encodings.size());
EXPECT_EQ(1111u, rtp_encodings[0].ssrc);
- EXPECT_EQ(0xaaaaaaaa, rtp_encodings[0].rtx->ssrc);
EXPECT_EQ(2222u, rtp_encodings[1].ssrc);
- EXPECT_EQ(0xaaaaaaab, rtp_encodings[1].rtx->ssrc);
}
TEST(RtpParametersConversionTest, ToAudioRtpCodecParameters) {
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 9eaed31..402ad97 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -38,20 +38,6 @@
return ++g_unique_id;
}
-// Returns an true if any RtpEncodingParameters member that isn't implemented
-// contains a value.
-bool UnimplementedRtpEncodingParameterHasValue(
- const RtpEncodingParameters& encoding_params) {
- if (encoding_params.codec_payload_type.has_value() ||
- encoding_params.fec.has_value() || encoding_params.rtx.has_value() ||
- encoding_params.dtx.has_value() || encoding_params.ptime.has_value() ||
- encoding_params.scale_framerate_down_by.has_value() ||
- !encoding_params.dependency_rids.empty()) {
- return true;
- }
- return false;
-}
-
// Returns true if a "per-sender" encoding parameter contains a value that isn't
// its default. Currently max_bitrate_bps and bitrate_priority both are
// implemented "per-sender," meaning that these encoding parameters
@@ -109,9 +95,6 @@
return true;
}
for (size_t i = 0; i < parameters.encodings.size(); ++i) {
- if (UnimplementedRtpEncodingParameterHasValue(parameters.encodings[i])) {
- return true;
- }
// Encoding parameters that are per-sender should only contain value at
// index 0.
if (i != 0 &&
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index 9026cfc..b9c07ef 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -968,46 +968,6 @@
DestroyAudioRtpSender();
}
-TEST_F(RtpSenderReceiverTest,
- AudioSenderCantSetUnimplementedRtpEncodingParameters) {
- CreateAudioRtpSender();
- RtpParameters params = audio_rtp_sender_->GetParameters();
- EXPECT_EQ(1u, params.encodings.size());
-
- // Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
- // scale_framerate_down_by, dependency_rids.
- params.encodings[0].codec_payload_type = 1;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- audio_rtp_sender_->SetParameters(params).type());
- params = audio_rtp_sender_->GetParameters();
-
- params.encodings[0].fec = RtpFecParameters();
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- audio_rtp_sender_->SetParameters(params).type());
- params = audio_rtp_sender_->GetParameters();
-
- params.encodings[0].rtx = RtpRtxParameters();
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- audio_rtp_sender_->SetParameters(params).type());
- params = audio_rtp_sender_->GetParameters();
-
- params.encodings[0].dtx = DtxStatus::ENABLED;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- audio_rtp_sender_->SetParameters(params).type());
- params = audio_rtp_sender_->GetParameters();
-
- params.encodings[0].ptime = 1;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- audio_rtp_sender_->SetParameters(params).type());
- params = audio_rtp_sender_->GetParameters();
-
- params.encodings[0].dependency_rids.push_back("dummy_rid");
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- audio_rtp_sender_->SetParameters(params).type());
-
- DestroyAudioRtpSender();
-}
-
TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
CreateAudioRtpSender();
@@ -1245,46 +1205,6 @@
DestroyVideoRtpSender();
}
-TEST_F(RtpSenderReceiverTest,
- VideoSenderCantSetUnimplementedEncodingParameters) {
- CreateVideoRtpSender();
- RtpParameters params = video_rtp_sender_->GetParameters();
- EXPECT_EQ(1u, params.encodings.size());
-
- // Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
- // scale_framerate_down_by, dependency_rids.
- params.encodings[0].codec_payload_type = 1;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[0].fec = RtpFecParameters();
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[0].rtx = RtpRtxParameters();
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[0].dtx = DtxStatus::ENABLED;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[0].ptime = 1;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[0].dependency_rids.push_back("dummy_rid");
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
-
- DestroyVideoRtpSender();
-}
-
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetScaleResolutionDownBy) {
CreateVideoRtpSender();
@@ -1309,49 +1229,6 @@
DestroyVideoRtpSender();
}
-TEST_F(RtpSenderReceiverTest,
- VideoSenderCantSetUnimplementedEncodingParametersWithSimulcast) {
- CreateVideoRtpSenderWithSimulcast();
- RtpParameters params = video_rtp_sender_->GetParameters();
- EXPECT_EQ(kVideoSimulcastLayerCount, params.encodings.size());
-
- // Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
- // scale_framerate_down_by, dependency_rids.
- for (size_t i = 0; i < params.encodings.size(); i++) {
- params.encodings[i].codec_payload_type = 1;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[i].fec = RtpFecParameters();
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[i].rtx = RtpRtxParameters();
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[i].dtx = DtxStatus::ENABLED;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[i].ptime = 1;
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
-
- params.encodings[i].dependency_rids.push_back("dummy_rid");
- EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
- video_rtp_sender_->SetParameters(params).type());
- params = video_rtp_sender_->GetParameters();
- }
-
- DestroyVideoRtpSender();
-}
-
// A video sender can have multiple simulcast layers, in which case it will
// contain multiple RtpEncodingParameters. This tests that if this is the case
// (simulcast), then we can't set the bitrate_priority, or max_bitrate_bps
