Index: webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..6903e1e5639d86d9a7eb0a93d78b553f965cdcd8 |
--- /dev/null |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc |
@@ -0,0 +1,118 @@ |
+/* |
+ * Copyright (c) 2014 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 <string> |
+ |
+#include "webrtc/base/logging.h" |
+#include "webrtc/modules/include/module_common_types.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" |
+ |
+namespace webrtc { |
+ |
+static const size_t kStereoHeaderLength = 1; |
+ |
+RtpPacketizerStereo::RtpPacketizerStereo( |
+ size_t max_payload_len, |
+ size_t last_packet_reduction_len, |
+ const RTPVideoTypeHeader* rtp_type_header, |
+ const RTPVideoStereoInfo* stereoInfo) |
+ : max_payload_len_(max_payload_len - kStereoHeaderLength), |
+ last_packet_reduction_len_(last_packet_reduction_len), |
+ frame_index_(0), |
+ stereoInfo_(stereoInfo) { |
+ packetizers_.emplace_back(RtpPacketizer::Create( |
+ stereoInfo->stereoCodecType, max_payload_len_, last_packet_reduction_len_, |
+ rtp_type_header, stereoInfo, kVideoFrameDelta)); |
+ for (int i = 0; i < stereoInfo->num_frames; ++i) { |
+ packetizers_.emplace_back( |
+ RtpPacketizer::Create(stereoInfo->stereoCodecType, max_payload_len_, |
+ last_packet_reduction_len_, |
+ &(stereoInfo->rtp_video_headers[i]->codecHeader), |
+ stereoInfo, kVideoFrameDelta)); |
+ } |
+} |
+ |
+RtpPacketizerStereo::~RtpPacketizerStereo() {} |
+ |
+size_t RtpPacketizerStereo::SetPayloadData( |
+ const uint8_t* payload_data, |
+ size_t payload_size, |
+ const RTPFragmentationHeader* fragmentation) { |
+ size_t num_packets = packetizers_[0]->SetPayloadData( |
+ payload_data, payload_size, fragmentation); |
+ for (size_t i = 0; i < packetizers_.size() - 1; ++i) { |
+ num_packets += packetizers_[i + 1]->SetPayloadData( |
+ stereoInfo_->encoded_images[i]->_buffer, |
+ stereoInfo_->encoded_images[i]->_length, |
+ stereoInfo_->fragmentations[i]); |
+ } |
+ return num_packets; |
+} |
+ |
+bool RtpPacketizerStereo::NextPacket(RtpPacketToSend* packet) { |
+ RTC_DCHECK(packet); |
+ RTC_DCHECK_LE(frame_index_, static_cast<uint8_t>(packetizers_.size())); |
+ const bool rv = packetizers_[frame_index_]->NextPacket(packet); |
+ RTC_CHECK(rv); |
+ |
+ const bool last_packet = packet->Marker(); |
+ if (last_packet && |
+ frame_index_ < static_cast<uint8_t>(packetizers_.size()) - 1) { |
+ packet->SetMarker(false); |
+ } |
+ |
+ std::unique_ptr<RtpPacketToSend> packet_copy(new RtpPacketToSend(*packet)); |
+ uint8_t* wrapped_payload = |
+ packet->AllocatePayload(kStereoHeaderLength + packet->payload_size()); |
+ RTC_DCHECK(wrapped_payload); |
+ wrapped_payload[0] = frame_index_; |
+ auto payload = packet_copy->payload(); |
+ memcpy(&wrapped_payload[kStereoHeaderLength], payload.data(), payload.size()); |
+ |
+ frame_index_ = last_packet ? frame_index_ + 1 : frame_index_; |
+ return rv; |
+} |
+ |
+ProtectionType RtpPacketizerStereo::GetProtectionType() { |
+ return kProtectedPacket; |
+} |
+ |
+StorageType RtpPacketizerStereo::GetStorageType( |
+ uint32_t retransmission_settings) { |
+ return kDontRetransmit; |
+} |
+ |
+std::string RtpPacketizerStereo::ToString() { |
+ return "RtpPacketizerStereo"; |
+} |
+ |
+bool RtpDepacketizerStereo::Parse(ParsedPayload* parsed_payload, |
+ const uint8_t* payload_data, |
+ size_t payload_data_length) { |
+ assert(parsed_payload != NULL); |
+ if (payload_data_length == 0) { |
+ LOG(LS_ERROR) << "Empty payload."; |
+ return false; |
+ } |
+ |
+ uint8_t frame_index = *payload_data++; |
+ --payload_data_length; |
+ |
+ const bool rv = |
+ depacketizer_.Parse(parsed_payload, payload_data, payload_data_length); |
+ parsed_payload->type.Video.stereoInfo.frame_index = frame_index; |
+ RTC_DCHECK(rv); |
+ if (frame_index > 0) |
+ parsed_payload->type.Video.is_first_packet_in_frame = false; |
+ parsed_payload->type.Video.codec = kRtpVideoStereo; |
+ return rv; |
+} |
+} // namespace webrtc |