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Unified Diff: talk/app/webrtc/jsepsessiondescription.cc

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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Index: talk/app/webrtc/jsepsessiondescription.cc
diff --git a/talk/app/webrtc/jsepsessiondescription.cc b/talk/app/webrtc/jsepsessiondescription.cc
deleted file mode 100644
index 226432db6956d63852d536fcb40fc68c778a8ec4..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/jsepsessiondescription.cc
+++ /dev/null
@@ -1,203 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/app/webrtc/jsepsessiondescription.h"
-
-#include "talk/app/webrtc/webrtcsdp.h"
-#include "talk/session/media/mediasession.h"
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/stringencode.h"
-
-using rtc::scoped_ptr;
-using cricket::SessionDescription;
-
-namespace webrtc {
-
-static const char* kSupportedTypes[] = {
- JsepSessionDescription::kOffer,
- JsepSessionDescription::kPrAnswer,
- JsepSessionDescription::kAnswer
-};
-
-static bool IsTypeSupported(const std::string& type) {
- bool type_supported = false;
- for (size_t i = 0; i < arraysize(kSupportedTypes); ++i) {
- if (kSupportedTypes[i] == type) {
- type_supported = true;
- break;
- }
- }
- return type_supported;
-}
-
-const char SessionDescriptionInterface::kOffer[] = "offer";
-const char SessionDescriptionInterface::kPrAnswer[] = "pranswer";
-const char SessionDescriptionInterface::kAnswer[] = "answer";
-
-const int JsepSessionDescription::kDefaultVideoCodecId = 100;
-// This is effectively a max value of the frame rate. 30 is default from camera.
-const int JsepSessionDescription::kDefaultVideoCodecFramerate = 60;
-const char JsepSessionDescription::kDefaultVideoCodecName[] = "VP8";
-// Used as default max video codec size before we have it in signaling.
-#if defined(ANDROID) || defined(WEBRTC_IOS)
-// Limit default max video codec size for Android to avoid
-// HW VP8 codec initialization failure for resolutions higher
-// than 1280x720 or 720x1280.
-// Same patch for iOS to support 720P in portrait mode.
-const int JsepSessionDescription::kMaxVideoCodecWidth = 1280;
-const int JsepSessionDescription::kMaxVideoCodecHeight = 1280;
-#else
-const int JsepSessionDescription::kMaxVideoCodecWidth = 1920;
-const int JsepSessionDescription::kMaxVideoCodecHeight = 1080;
-#endif
-const int JsepSessionDescription::kDefaultVideoCodecPreference = 1;
-
-SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
- const std::string& sdp,
- SdpParseError* error) {
- if (!IsTypeSupported(type)) {
- return NULL;
- }
-
- JsepSessionDescription* jsep_desc = new JsepSessionDescription(type);
- if (!jsep_desc->Initialize(sdp, error)) {
- delete jsep_desc;
- return NULL;
- }
- return jsep_desc;
-}
-
-JsepSessionDescription::JsepSessionDescription(const std::string& type)
- : type_(type) {
-}
-
-JsepSessionDescription::~JsepSessionDescription() {}
-
-bool JsepSessionDescription::Initialize(
- cricket::SessionDescription* description,
- const std::string& session_id,
- const std::string& session_version) {
- if (!description)
- return false;
-
- session_id_ = session_id;
- session_version_ = session_version;
- description_.reset(description);
- candidate_collection_.resize(number_of_mediasections());
- return true;
-}
-
-bool JsepSessionDescription::Initialize(const std::string& sdp,
- SdpParseError* error) {
- return SdpDeserialize(sdp, this, error);
-}
-
-bool JsepSessionDescription::AddCandidate(
- const IceCandidateInterface* candidate) {
- if (!candidate || candidate->sdp_mline_index() < 0)
- return false;
- size_t mediasection_index = 0;
- if (!GetMediasectionIndex(candidate, &mediasection_index)) {
- return false;
- }
- if (mediasection_index >= number_of_mediasections())
- return false;
- const std::string& content_name =
- description_->contents()[mediasection_index].name;
- const cricket::TransportInfo* transport_info =
- description_->GetTransportInfoByName(content_name);
- if (!transport_info) {
- return false;
- }
-
- cricket::Candidate updated_candidate = candidate->candidate();
- if (updated_candidate.username().empty()) {
- updated_candidate.set_username(transport_info->description.ice_ufrag);
- }
- if (updated_candidate.password().empty()) {
- updated_candidate.set_password(transport_info->description.ice_pwd);
- }
-
- scoped_ptr<JsepIceCandidate> updated_candidate_wrapper(
- new JsepIceCandidate(candidate->sdp_mid(),
- static_cast<int>(mediasection_index),
- updated_candidate));
- if (!candidate_collection_[mediasection_index].HasCandidate(
- updated_candidate_wrapper.get()))
- candidate_collection_[mediasection_index].add(
- updated_candidate_wrapper.release());
-
- return true;
-}
-
-size_t JsepSessionDescription::number_of_mediasections() const {
- if (!description_)
- return 0;
- return description_->contents().size();
-}
-
-const IceCandidateCollection* JsepSessionDescription::candidates(
- size_t mediasection_index) const {
- if (mediasection_index >= candidate_collection_.size())
- return NULL;
- return &candidate_collection_[mediasection_index];
-}
-
-bool JsepSessionDescription::ToString(std::string* out) const {
- if (!description_ || !out)
- return false;
- *out = SdpSerialize(*this);
- return !out->empty();
-}
-
-bool JsepSessionDescription::GetMediasectionIndex(
- const IceCandidateInterface* candidate,
- size_t* index) {
- if (!candidate || !index) {
- return false;
- }
- *index = static_cast<size_t>(candidate->sdp_mline_index());
- if (description_ && !candidate->sdp_mid().empty()) {
- bool found = false;
- // Try to match the sdp_mid with content name.
- for (size_t i = 0; i < description_->contents().size(); ++i) {
- if (candidate->sdp_mid() == description_->contents().at(i).name) {
- *index = i;
- found = true;
- break;
- }
- }
- if (!found) {
- // If the sdp_mid is presented but we can't find a match, we consider
- // this as an error.
- return false;
- }
- }
- return true;
-}
-
-} // namespace webrtc
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