Index: modules/rtp_rtcp/source/rtp_format_video_stereo.cc |
diff --git a/modules/rtp_rtcp/source/rtp_format_video_stereo.cc b/modules/rtp_rtcp/source/rtp_format_video_stereo.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..451a92929dc62814911fd04732fba8949117c992 |
--- /dev/null |
+++ b/modules/rtp_rtcp/source/rtp_format_video_stereo.cc |
@@ -0,0 +1,116 @@ |
+/* |
+ * 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 "modules/include/module_common_types.h" |
+#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h" |
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" |
+#include "rtc_base/logging.h" |
+ |
+namespace webrtc { |
+ |
+static const size_t kStereoHeaderMarkerLength = 1; |
+static const size_t kStereoHeaderLength = sizeof(RTPVideoStereoInfo); |
+ |
+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 - kStereoHeaderMarkerLength - |
+ kStereoHeaderLength), |
+ last_packet_reduction_len_(last_packet_reduction_len), |
+ packetizer_(RtpPacketizer::Create(stereoInfo->stereoCodecType, |
+ max_payload_len_, |
+ last_packet_reduction_len_, |
+ rtp_type_header, |
+ stereoInfo, |
+ kVideoFrameDelta)), |
+ stereoInfo_(stereoInfo) {} |
+ |
+RtpPacketizerStereo::~RtpPacketizerStereo() {} |
+ |
+size_t RtpPacketizerStereo::SetPayloadData( |
+ const uint8_t* payload_data, |
+ size_t payload_size, |
+ const RTPFragmentationHeader* fragmentation) { |
+ header_marker_ = RtpFormatVideoStereo::kFirstPacketBit; |
+ return packetizer_->SetPayloadData(payload_data, payload_size, fragmentation); |
+} |
+ |
+bool RtpPacketizerStereo::NextPacket(RtpPacketToSend* packet) { |
+ RTC_DCHECK(packet); |
+ const bool rv = packetizer_->NextPacket(packet); |
+ RTC_CHECK(rv); |
+ |
+ const bool first_packet = |
+ header_marker_ == RtpFormatVideoStereo::kFirstPacketBit; |
+ size_t header_length = first_packet |
+ ? kStereoHeaderMarkerLength + kStereoHeaderLength |
+ : kStereoHeaderMarkerLength; |
+ |
+ std::unique_ptr<RtpPacketToSend> copied_packet(new RtpPacketToSend(*packet)); |
+ uint8_t* wrapped_payload = |
+ packet->AllocatePayload(header_length + packet->payload_size()); |
+ RTC_DCHECK(wrapped_payload); |
+ wrapped_payload[0] = header_marker_; |
+ header_marker_ &= ~RtpFormatVideoStereo::kFirstPacketBit; |
+ if (first_packet) { |
+ memcpy(&wrapped_payload[kStereoHeaderMarkerLength], stereoInfo_, |
+ kStereoHeaderLength); |
+ } |
+ auto payload = copied_packet->payload(); |
+ memcpy(&wrapped_payload[header_length], payload.data(), payload.size()); |
+ 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 marker_header = *payload_data++; |
+ --payload_data_length; |
+ const bool first_packet = |
+ (marker_header & RtpFormatVideoStereo::kFirstPacketBit) != 0; |
+ |
+ if (first_packet) { |
+ memcpy(&parsed_payload->type.Video.stereoInfo, payload_data, |
+ kStereoHeaderLength); |
+ payload_data += kStereoHeaderLength; |
+ payload_data_length -= kStereoHeaderLength; |
+ } |
+ const bool rv = |
+ depacketizer_.Parse(parsed_payload, payload_data, payload_data_length); |
+ RTC_DCHECK(rv); |
+ RTC_DCHECK_EQ(parsed_payload->type.Video.is_first_packet_in_frame, |
+ first_packet); |
+ parsed_payload->type.Video.codec = kRtpVideoStereo; |
+ return rv; |
+} |
+} // namespace webrtc |