Replace rtc::Optional with absl::optional in api
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}diff --git a/api/rtpparameters.h b/api/rtpparameters.h
index e2405d3..84da811 100644
--- a/api/rtpparameters.h
+++ b/api/rtpparameters.h
@@ -15,8 +15,8 @@
#include <unordered_map>
#include <vector>
+#include "absl/types/optional.h"
#include "api/mediatypes.h"
-#include "api/optional.h"
namespace webrtc {
@@ -94,7 +94,7 @@
// 1. It's an enum instead of a string.
// 2. Generic NACK feedback is represented by a GENERIC_NACK message type,
// rather than an unset "parameter" value.
- rtc::Optional<RtcpFeedbackMessageType> message_type;
+ absl::optional<RtcpFeedbackMessageType> message_type;
// Constructors for convenience.
RtcpFeedback();
@@ -125,23 +125,23 @@
cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
// Clock rate in Hertz. If unset, the codec is applicable to any clock rate.
- rtc::Optional<int> clock_rate;
+ absl::optional<int> clock_rate;
// Default payload type for this codec. Mainly needed for codecs that use
// that have statically assigned payload types.
- rtc::Optional<int> preferred_payload_type;
+ absl::optional<int> preferred_payload_type;
// Maximum packetization time supported by an RtpReceiver for this codec.
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> max_ptime;
+ absl::optional<int> max_ptime;
// Preferred packetization time for an RtpReceiver or RtpSender of this
// codec.
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> ptime;
+ absl::optional<int> ptime;
// The number of audio channels supported. Unused for video codecs.
- rtc::Optional<int> num_channels;
+ absl::optional<int> num_channels;
// Feedback mechanisms supported for this codec.
std::vector<RtcpFeedback> rtcp_feedback;
@@ -204,7 +204,7 @@
std::string uri;
// Preferred value of ID that goes in the packet.
- rtc::Optional<int> preferred_id;
+ absl::optional<int> preferred_id;
// If true, it's preferred that the value in the header is encrypted.
// TODO(deadbeef): Not implemented.
@@ -313,7 +313,7 @@
struct RtpFecParameters {
// If unset, a value is chosen by the implementation.
// Works just like RtpEncodingParameters::ssrc.
- rtc::Optional<uint32_t> ssrc;
+ absl::optional<uint32_t> ssrc;
FecMechanism mechanism = FecMechanism::RED;
@@ -332,7 +332,7 @@
struct RtpRtxParameters {
// If unset, a value is chosen by the implementation.
// Works just like RtpEncodingParameters::ssrc.
- rtc::Optional<uint32_t> ssrc;
+ absl::optional<uint32_t> ssrc;
// Constructors for convenience.
RtpRtxParameters();
@@ -353,7 +353,7 @@
// may change due to an SSRC conflict, in which case the conflict is handled
// internally without any event. Another way of looking at this is that an
// unset SSRC acts as a "wildcard" SSRC.
- rtc::Optional<uint32_t> 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
@@ -361,23 +361,23 @@
// prepare to receive any codec (for a receiver).
// TODO(deadbeef): Not implemented. Implementation of RtpSender will always
// choose the first codec from the list.
- rtc::Optional<int> codec_payload_type;
+ 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.
- rtc::Optional<RtpFecParameters> fec;
+ absl::optional<RtpFecParameters> fec;
// Specifies the RTX parameters, if set.
// TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
- rtc::Optional<RtpRtxParameters> rtx;
+ 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.
- rtc::Optional<DtxStatus> dtx;
+ 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
@@ -394,7 +394,7 @@
// creates a ptime for a specific codec, which is later changed in the
// RtpEncodingParameters by the application.
// TODO(bugs.webrtc.org/8819): Not implemented.
- rtc::Optional<int> ptime;
+ absl::optional<int> ptime;
// If set, this represents the Transport Independent Application Specific
// maximum bandwidth defined in RFC3890. If unset, there is no maximum
@@ -407,23 +407,23 @@
// bandwidth for the entire bandwidth estimator (audio and video). This is
// just always how "b=AS" was handled, but it's not correct and should be
// fixed.
- rtc::Optional<int> max_bitrate_bps;
+ absl::optional<int> max_bitrate_bps;
// Specifies the minimum bitrate in bps for video.
// TODO(asapersson): Not implemented for ORTC API.
// TODO(asapersson): Not implemented for single layer.
- rtc::Optional<int> min_bitrate_bps;
+ absl::optional<int> min_bitrate_bps;
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> max_framerate;
+ absl::optional<int> max_framerate;
// For video, scale the resolution down by this factor.
// TODO(deadbeef): Not implemented.
- rtc::Optional<double> scale_resolution_down_by;
+ absl::optional<double> scale_resolution_down_by;
// Scale the framerate down by this factor.
// TODO(deadbeef): Not implemented.
- rtc::Optional<double> scale_framerate_down_by;
+ 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
@@ -478,24 +478,24 @@
int payload_type = 0;
// If unset, the implementation default is used.
- rtc::Optional<int> clock_rate;
+ absl::optional<int> clock_rate;
// The number of audio channels used. Unset for video codecs. If unset for
// audio, the implementation default is used.
// TODO(deadbeef): The "implementation default" part isn't fully implemented.
// Only defaults to 1, even though some codecs (such as opus) should really
// default to 2.
- rtc::Optional<int> num_channels;
+ absl::optional<int> num_channels;
// The maximum packetization time to be used by an RtpSender.
// If |ptime| is also set, this will be ignored.
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> max_ptime;
+ absl::optional<int> max_ptime;
// The packetization time to be used by an RtpSender.
// If unset, will use any time up to max_ptime.
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> ptime;
+ absl::optional<int> ptime;
// Feedback mechanisms to be used for this codec.
// TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
@@ -551,7 +551,7 @@
// The SSRC to be used in the "SSRC of packet sender" field. If not set, one
// will be chosen by the implementation.
// TODO(deadbeef): Not implemented.
- rtc::Optional<uint32_t> ssrc;
+ absl::optional<uint32_t> ssrc;
// The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
//