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Changes name of RtpTransceiverInit's stream_labels to stream_ids.

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The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.

Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
This commit is contained in:
Seth Hampson 2018-03-06 09:35:56 -08:00 committed by Commit Bot
parent 8e20f17d1d
commit 513449eab9
12 changed files with 26 additions and 29 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
api/rtptransceiverinterface.h
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@ -43,7 +43,7 @@ struct RtpTransceiverInit final {
// TODO(shampson): Change name to stream_id & update native wrapper's naming
// as well.
// TODO(bugs.webrtc.org/7600): Not implemented.
std::vector<std::string> stream_labels;
std::vector<std::string> stream_ids;
// TODO(bugs.webrtc.org/7600): Not implemented.
std::vector<RtpEncodingParameters> send_encodings;

4
examples/objc/AppRTCMobile/ARDAppClient.m
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@ -692,10 +692,10 @@ static int const kKbpsMultiplier = 1000;
RTCAudioSource *source = [_factory audioSourceWithConstraints:constraints];
RTCAudioTrack *track = [_factory audioTrackWithSource:source
trackId:kARDAudioTrackId];
[_peerConnection addTrack:track streamLabels:@[ kARDMediaStreamId ]];
[_peerConnection addTrack:track streamIds:@[ kARDMediaStreamId ]];
_localVideoTrack = [self createLocalVideoTrack];
if (_localVideoTrack) {
[_peerConnection addTrack:_localVideoTrack streamLabels:@[ kARDMediaStreamId ]];
[_peerConnection addTrack:_localVideoTrack streamIds:@[ kARDMediaStreamId ]];
// We can set up rendering for the remote track right away since the transceiver already has an
// RTCRtpReceiver with a track. The track will automatically get unmuted and produce frames
// once RTP is received.

4
examples/peerconnection/client/conductor.cc
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@ -408,7 +408,7 @@ std::unique_ptr<cricket::VideoCapturer> Conductor::OpenVideoCaptureDevice() {
}
void Conductor::AddStreams() {
if (active_streams_.find(kStreamLabel) != active_streams_.end())
if (active_streams_.find(kStreamId) != active_streams_.end())
return; // Already added.
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
@ -429,7 +429,7 @@ void Conductor::AddStreams() {
}
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
peer_connection_factory_->CreateLocalMediaStream(kStreamLabel);
peer_connection_factory_->CreateLocalMediaStream(kStreamId);
stream->AddTrack(audio_track);
if (video_track)

2
examples/peerconnection/client/defaults.cc
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@ -23,7 +23,7 @@
const char kAudioLabel[] = "audio_label";
const char kVideoLabel[] = "video_label";
const char kStreamLabel[] = "stream_label";
const char kStreamId[] = "stream_id";
const uint16_t kDefaultServerPort = 8888;
std::string GetEnvVarOrDefault(const char* env_var_name,

2
examples/peerconnection/client/defaults.h
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@ -17,7 +17,7 @@
extern const char kAudioLabel[];
extern const char kVideoLabel[];
extern const char kStreamLabel[];
extern const char kStreamId[];
extern const uint16_t kDefaultServerPort;
std::string GetEnvVarOrDefault(const char* env_var_name,

8
examples/unityplugin/simple_peer_connection.cc
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@ -33,10 +33,10 @@
#include "sdk/android/src/jni/jni_helpers.h"
#endif
// Names used for media stream labels.
// Names used for media stream ids.
const char kAudioLabel[] = "audio_label";
const char kVideoLabel[] = "video_label";
const char kStreamLabel[] = "stream_label";
const char kStreamId[] = "stream_id";
namespace {
static int g_peer_count = 0;
@ -418,11 +418,11 @@ SimplePeerConnection::OpenVideoCaptureDevice() {
}
void SimplePeerConnection::AddStreams(bool audio_only) {
if (active_streams_.find(kStreamLabel) != active_streams_.end())
if (active_streams_.find(kStreamId) != active_streams_.end())
return; // Already added.
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
g_peer_connection_factory->CreateLocalMediaStream(kStreamLabel);
g_peer_connection_factory->CreateLocalMediaStream(kStreamId);
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
g_peer_connection_factory->CreateAudioTrack(

7
pc/peerconnection.cc
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@ -1351,14 +1351,13 @@ PeerConnection::AddTransceiver(
// TODO(bugs.webrtc.org/7600): Verify init.
if (init.stream_labels.size() > 1u) {
if (init.stream_ids.size() > 1u) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"More than one stream is not yet supported.");
}
std::vector<std::string> stream_ids = {!init.stream_labels.empty()
? init.stream_labels[0]
: rtc::CreateRandomUuid()};
std::vector<std::string> stream_ids = {
!init.stream_ids.empty() ? init.stream_ids[0] : rtc::CreateRandomUuid()};
auto sender = CreateSender(media_type, track, stream_ids);
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());

2
sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream.mm
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@ -43,7 +43,7 @@
}
- (NSString *)streamId {
return [NSString stringForStdString:_nativeMediaStream->label()];
return [NSString stringForStdString:_nativeMediaStream->id()];
}
- (void)addAudioTrack:(RTCAudioTrack *)audioTrack {

11
sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
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@ -367,14 +367,13 @@ void PeerConnectionDelegateAdapter::OnAddTrack(
[_localStreams removeObject:stream];
}
- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track
streamLabels:(NSArray<NSString *> *)streamLabels {
std::vector<std::string> nativeStreamLabels;
for (NSString *label in streamLabels) {
nativeStreamLabels.push_back([label UTF8String]);
- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track streamIds:(NSArray<NSString *> *)streamIds {
std::vector<std::string> nativeStreamIds;
for (NSString *streamId in streamIds) {
nativeStreamIds.push_back([streamId UTF8String]);
}
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenderOrError =
_peerConnection->AddTrack(track.nativeTrack, nativeStreamLabels);
_peerConnection->AddTrack(track.nativeTrack, nativeStreamIds);
if (!nativeSenderOrError.ok()) {
RTCLogError(@"Failed to add track %@: %s", track, nativeSenderOrError.error().message());
return nil;

6
sdk/objc/Framework/Classes/PeerConnection/RTCRtpTransceiver.mm
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@ -20,7 +20,7 @@
@implementation RTCRtpTransceiverInit
@synthesize direction = _direction;
@synthesize streamLabels = _streamLabels;
@synthesize streamIds = _streamIds;
@synthesize sendEncodings = _sendEncodings;
- (instancetype)init {
@ -33,8 +33,8 @@
- (webrtc::RtpTransceiverInit)nativeInit {
webrtc::RtpTransceiverInit init;
init.direction = [RTCRtpTransceiver nativeRtpTransceiverDirectionFromDirection:_direction];
for (NSString *streamLabel in _streamLabels) {
init.stream_labels.push_back([streamLabel UTF8String]);
for (NSString *streamId in _streamIds) {
init.stream_ids.push_back([streamId UTF8String]);
}
for (RTCRtpEncodingParameters *sendEncoding in _sendEncodings) {
init.send_encodings.push_back(sendEncoding.nativeParameters);

5
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
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@ -201,14 +201,13 @@ RTC_EXPORT
/** Add a new media stream track to be sent on this peer connection, and return
* the newly created RTCRtpSender. The RTCRtpSender will be associated with
* the streams specified in the |streamLabels| list.
* the streams specified in the |streamIds| list.
*
* Errors: If an error occurs, returns nil. An error can occur if:
* - A sender already exists for the track.
* - The peer connection is closed.
*/
- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track
streamLabels:(NSArray<NSString *> *)streamLabels;
- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track streamIds:(NSArray<NSString *> *)streamIds;
/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
*

2
sdk/objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h
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@ -34,7 +34,7 @@ typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
@property(nonatomic) RTCRtpTransceiverDirection direction;
/** The added RTCRtpTransceiver will be added to these streams. */
@property(nonatomic) NSArray<NSString *> *streamLabels;
@property(nonatomic) NSArray<NSString *> *streamIds;
/** TODO(bugs.webrtc.org/7600): Not implemented. */
@property(nonatomic) NSArray<RTCRtpEncodingParameters *> *sendEncodings;