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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
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@ -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);