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Unified Diff: modules/rtp_rtcp/source/rtp_format_video_stereo.cc

Issue 2951033003: [EXPERIMENTAL] Generic stereo codec with index header sending single frames
Patch Set: Rebase and add external codec support. Created 3 years, 3 months ago
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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
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