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Remove unused non-standard RtpEncodingParameters members

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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}
This commit is contained in:
Florent Castelli 2019-11-28 15:48:24 +01:00 committed by Commit Bot
parent 6c0e94650e
commit a8c2f5180f
6 changed files with 3 additions and 280 deletions
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52
api/rtp_parameters.h
View file

@ -380,30 +380,6 @@ struct RTC_EXPORT RtpEncodingParameters {
// 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 @@ struct RTC_EXPORT RtpEncodingParameters {
// 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 @@ struct RTC_EXPORT RtpEncodingParameters {
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 @@ struct RTC_EXPORT RtpEncodingParameters {
// 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 @@ struct RTC_EXPORT RtpEncodingParameters {
// 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);

47
pc/peer_connection_rtp_unittest.cc
View file

@ -1460,53 +1460,6 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan,
.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

13
pc/rtp_parameters_conversion.cc
View file

@ -234,17 +234,9 @@ RTCErrorOr<cricket::StreamParamsVec> ToCricketStreamParamsVec(
}
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 @@ std::vector<RtpEncodingParameters> ToRtpEncodings(
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;

31
pc/rtp_parameters_conversion_unittest.cc
View file

@ -346,23 +346,6 @@ TEST(RtpParametersConversionTest, ToCricketStreamParamsVecSimple) {
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, ToCricketStreamParamsVecNoEncodings) {
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 @@ TEST(RtpParametersConversionTest, ToRtpEncodingsWithMultipleStreamParams) {
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 @@ TEST(RtpParametersConversionTest, ToRtpEncodingsWithMultipleStreamParams) {
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) {

17
pc/rtp_sender.cc
View file

@ -38,20 +38,6 @@ int GenerateUniqueId() {
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 @@ bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters) {
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 &&

123
pc/rtp_sender_receiver_unittest.cc
View file

@ -968,46 +968,6 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCantSetUnimplementedRtpParameters) {
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 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCantSetUnimplementedRtpParameters) {
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 @@ TEST_F(RtpSenderReceiverTest, VideoSenderDetectInvalidScaleResolutionDownBy) {
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