generic-library/webrtc
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Rename RTP-extension constants.

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BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-11-20 11:48:56 +00:00
parent 53c8573525
commit ce90eff345
6 changed files with 27 additions and 5 deletions
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17
webrtc/config.cc Normal file
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@ -0,0 +1,17 @@
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/config.h"
namespace webrtc {
const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
const char* RtpExtension::kAbsSendTime =
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
} // namespace webrtc

4
webrtc/config.h
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@ -16,6 +16,8 @@
#include <string>
#include <vector>
#include "webrtc/typedefs.h"
namespace webrtc {
struct RtpStatistics {
@ -67,6 +69,8 @@ struct RtxConfig {
// RTP header extension to use for the video stream, see RFC 5285.
struct RtpExtension {
static const char* kTOffset;
static const char* kAbsSendTime;
RtpExtension(const char* name, int id) : name(name), id(id) {}
// TODO(mflodman) Add API to query supported extensions.
std::string name;

2
webrtc/video/rampup_tests.cc
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@ -155,7 +155,7 @@ TEST_P(RampUpTest, RampUpWithPadding) {
send_config.codec.plType = 125;
send_config.pacing = GetParam();
send_config.rtp.extensions.push_back(
RtpExtension("toffset", kTOffsetExtensionId));
RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);

4
webrtc/video/video_send_stream.cc
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@ -123,10 +123,10 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
const std::string& extension = config_.rtp.extensions[i].name;
int id = config_.rtp.extensions[i].id;
if (extension == "toffset") {
if (extension == RtpExtension::kTOffset) {
if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
abort();
} else if (extension == "abs-send-time") {
} else if (extension == RtpExtension::kAbsSendTime) {
if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0)
abort();
} else {

4
webrtc/video/video_send_stream_tests.cc
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@ -170,7 +170,7 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.extensions.push_back(
RtpExtension("abs-send-time", kAbsSendTimeExtensionId));
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
RunSendTest(call.get(), send_config, &observer);
}
@ -216,7 +216,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.encoder = &encoder;
send_config.rtp.extensions.push_back(
RtpExtension("toffset", kTOffsetExtensionId));
RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
RunSendTest(call.get(), send_config, &observer);
}

1
webrtc/webrtc.gyp
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@ -61,6 +61,7 @@
'sources': [
'call.cc',
'call.h',
'config.cc',
'config.h',
'frame_callback.h',
'transport.h',