Cleanup of unused RTP structs and packetizer for stereo codec
This CL is a followup to https://webrtc-review.googlesource.com/c/src/+/38481.
With the new approach we can just use the generic RTP packetizer to pass frames
over the wire as the specific info is contained within the bitstream. This makes
the new codec more modular and reduces its footprint.
Bug: webrtc:7671
Change-Id: Ib07f72a9d338e3cbfdbf39cba9891a959b5f7552
Reviewed-on: https://webrtc-review.googlesource.com/43220
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21753}diff --git a/modules/include/module_common_types.h b/modules/include/module_common_types.h
index 94ccbdc..5860b59 100644
--- a/modules/include/module_common_types.h
+++ b/modules/include/module_common_types.h
@@ -23,7 +23,6 @@
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types_public.h"
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
-#include "modules/video_coding/codecs/stereo/include/stereo_globals.h"
#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
#include "rtc_base/constructormagic.h"
@@ -46,20 +45,13 @@
kRtpVideoGeneric = 1,
kRtpVideoVp8 = 2,
kRtpVideoVp9 = 3,
- kRtpVideoH264 = 4,
- kRtpVideoStereo = 5
-};
-
-struct RTPVideoHeaderStereo {
- RtpVideoCodecTypes associated_codec_type;
- StereoIndices indices;
+ kRtpVideoH264 = 4
};
union RTPVideoTypeHeader {
RTPVideoHeaderVP8 VP8;
RTPVideoHeaderVP9 VP9;
RTPVideoHeaderH264 H264;
- RTPVideoHeaderStereo stereo;
};
// Since RTPVideoHeader is used as a member of a union, it can't have a
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index c2dafe1..d755148 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -139,8 +139,6 @@
"source/rtp_format_h264.h",
"source/rtp_format_video_generic.cc",
"source/rtp_format_video_generic.h",
- "source/rtp_format_video_stereo.cc",
- "source/rtp_format_video_stereo.h",
"source/rtp_format_vp8.cc",
"source/rtp_format_vp8.h",
"source/rtp_format_vp9.cc",
@@ -368,7 +366,6 @@
"source/rtp_fec_unittest.cc",
"source/rtp_format_h264_unittest.cc",
"source/rtp_format_video_generic_unittest.cc",
- "source/rtp_format_video_stereo_unittest.cc",
"source/rtp_format_vp8_test_helper.cc",
"source/rtp_format_vp8_test_helper.h",
"source/rtp_format_vp8_unittest.cc",
diff --git a/modules/rtp_rtcp/source/rtp_format.cc b/modules/rtp_rtcp/source/rtp_format.cc
index c5f5750..05dc900 100644
--- a/modules/rtp_rtcp/source/rtp_format.cc
+++ b/modules/rtp_rtcp/source/rtp_format.cc
@@ -14,7 +14,6 @@
#include "modules/rtp_rtcp/source/rtp_format_h264.h"
#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
-#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
@@ -37,10 +36,6 @@
RTC_CHECK(rtp_type_header);
return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len,
last_packet_reduction_len);
- case kRtpVideoStereo:
- return new RtpPacketizerStereo(rtp_type_header->stereo, frame_type,
- max_payload_len,
- last_packet_reduction_len);
case kRtpVideoGeneric:
return new RtpPacketizerGeneric(frame_type, max_payload_len,
last_packet_reduction_len);
@@ -58,8 +53,6 @@
return new RtpDepacketizerVp8();
case kRtpVideoVp9:
return new RtpDepacketizerVp9();
- case kRtpVideoStereo:
- return new RtpDepacketizerStereo();
case kRtpVideoGeneric:
return new RtpDepacketizerGeneric();
case kRtpVideoNone:
diff --git a/modules/rtp_rtcp/source/rtp_format_video_stereo.cc b/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
deleted file mode 100644
index 7c3a933..0000000
--- a/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
+++ /dev/null
@@ -1,164 +0,0 @@
-/*
- * Copyright (c) 2017 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 <memory>
-#include <string>
-
-#include "modules/include/module_common_types.h"
-#include "modules/rtp_rtcp/source/byte_io.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 {
-
-namespace {
-// Write the Stereo header descriptor.
-// 0 1 2 3 4 5 6 7 8
-// +-+-+-+-+-+-+-+-+-+
-// | VideoCodecType | (optional)
-// +-+-+-+-+-+-+-+-+-+
-// | frame_index | (optional)
-// +-+-+-+-+-+-+-+-+-+
-// | frame_count | (optional)
-// +-+-+-+-+-+-+-+-+-+
-// | picture_index | (optional)
-// | (16 bits) |
-// +-+-+-+-+-+-+-+-+-+
-// | HeaderMarker | (mandatory)
-// +-+-+-+-+-+-+-+-+-+
-constexpr size_t kStereoHeaderLength =
- sizeof(uint8_t) + sizeof(uint8_t) + sizeof(uint8_t) + sizeof(uint16_t);
-
-constexpr size_t kHeaderMarkerLength = 1;
-constexpr uint8_t kHeaderMarkerBit = 0x02;
-} // namespace
-
-RtpPacketizerStereo::RtpPacketizerStereo(const RTPVideoHeaderStereo& header,
- FrameType frame_type,
- size_t max_payload_len,
- size_t last_packet_reduction_len)
- : header_(header),
- max_payload_len_(max_payload_len - kHeaderMarkerLength),
- last_packet_reduction_len_(last_packet_reduction_len +
- kStereoHeaderLength),
- packetizer_(frame_type, max_payload_len_, last_packet_reduction_len_) {}
-
-RtpPacketizerStereo::~RtpPacketizerStereo() {}
-
-size_t RtpPacketizerStereo::SetPayloadData(
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation) {
- num_packets_remaining_ =
- packetizer_.SetPayloadData(payload_data, payload_size, fragmentation);
- return num_packets_remaining_;
-}
-
-bool RtpPacketizerStereo::NextPacket(RtpPacketToSend* packet) {
- if (max_payload_len_ <= last_packet_reduction_len_) {
- RTC_LOG(LS_ERROR) << "Payload length not large enough.";
- return false;
- }
-
- RTC_DCHECK(packet);
- if (!packetizer_.NextPacket(packet))
- return false;
-
- RTC_DCHECK_GT(num_packets_remaining_, 0);
- const bool last_packet = --num_packets_remaining_ == 0;
- const size_t header_length = last_packet
- ? kHeaderMarkerLength + kStereoHeaderLength
- : kHeaderMarkerLength;
-
- const uint8_t* payload_ptr = packet->payload().data();
- const size_t payload_size = packet->payload_size();
- uint8_t* padded_payload_ptr =
- packet->SetPayloadSize(header_length + packet->payload_size());
- RTC_DCHECK(padded_payload_ptr);
-
- padded_payload_ptr += payload_size;
- if (last_packet) {
- ByteWriter<uint8_t>::WriteBigEndian(
- padded_payload_ptr,
- static_cast<uint8_t>(header_.associated_codec_type));
- padded_payload_ptr += sizeof(uint8_t);
- ByteWriter<uint8_t>::WriteBigEndian(padded_payload_ptr,
- header_.indices.frame_index);
- padded_payload_ptr += sizeof(uint8_t);
- ByteWriter<uint8_t>::WriteBigEndian(padded_payload_ptr,
- header_.indices.frame_count);
- padded_payload_ptr += sizeof(uint8_t);
- ByteWriter<uint16_t>::WriteBigEndian(padded_payload_ptr,
- header_.indices.picture_index);
- padded_payload_ptr += sizeof(uint16_t);
- RTC_DCHECK_EQ(payload_size + kStereoHeaderLength,
- padded_payload_ptr - payload_ptr);
- }
- padded_payload_ptr[0] = last_packet ? kHeaderMarkerBit : 0;
- return true;
-}
-
-std::string RtpPacketizerStereo::ToString() {
- return "RtpPacketizerStereo";
-}
-
-RtpDepacketizerStereo::~RtpDepacketizerStereo() {}
-
-bool RtpDepacketizerStereo::Parse(ParsedPayload* parsed_payload,
- const uint8_t* payload_data,
- size_t payload_data_length) {
- RTC_DCHECK(parsed_payload);
- if (payload_data_length == 0) {
- RTC_LOG(LS_ERROR) << "Empty payload.";
- return false;
- }
-
- uint8_t header_marker = payload_data[payload_data_length - 1];
- --payload_data_length;
- const bool last_packet = (header_marker & kHeaderMarkerBit) != 0;
-
- if (last_packet) {
- if (payload_data_length <= kStereoHeaderLength) {
- RTC_LOG(LS_WARNING) << "Payload not large enough.";
- return false;
- }
- size_t offset = payload_data_length - kStereoHeaderLength;
- uint8_t associated_codec_type =
- ByteReader<uint8_t>::ReadBigEndian(&payload_data[offset]);
- switch (associated_codec_type) {
- case kRtpVideoVp8:
- case kRtpVideoVp9:
- case kRtpVideoH264:
- break;
- default:
- RTC_LOG(LS_WARNING) << "Unexpected codec type.";
- return false;
- }
- parsed_payload->type.Video.codecHeader.stereo.associated_codec_type =
- static_cast<RtpVideoCodecTypes>(associated_codec_type);
- offset += sizeof(uint8_t);
- parsed_payload->type.Video.codecHeader.stereo.indices.frame_index =
- ByteReader<uint8_t>::ReadBigEndian(&payload_data[offset]);
- offset += sizeof(uint8_t);
- parsed_payload->type.Video.codecHeader.stereo.indices.frame_count =
- ByteReader<uint8_t>::ReadBigEndian(&payload_data[offset]);
- offset += sizeof(uint8_t);
- parsed_payload->type.Video.codecHeader.stereo.indices.picture_index =
- ByteReader<uint16_t>::ReadBigEndian(&payload_data[offset]);
- RTC_DCHECK_EQ(payload_data_length, offset + sizeof(uint16_t));
- payload_data_length -= kStereoHeaderLength;
- }
- if (!depacketizer_.Parse(parsed_payload, payload_data, payload_data_length))
- return false;
- parsed_payload->type.Video.codec = kRtpVideoStereo;
- return true;
-}
-} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_format_video_stereo.h b/modules/rtp_rtcp/source/rtp_format_video_stereo.h
deleted file mode 100644
index f054498..0000000
--- a/modules/rtp_rtcp/source/rtp_format_video_stereo.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2017 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.
- */
-#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_
-#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_
-
-#include <string>
-
-#include "modules/rtp_rtcp/source/rtp_format.h"
-#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class RtpPacketizerStereo : public RtpPacketizer {
- public:
- RtpPacketizerStereo(const RTPVideoHeaderStereo& header,
- FrameType frame_type,
- size_t max_payload_len,
- size_t last_packet_reduction_len);
-
- ~RtpPacketizerStereo() override;
-
- size_t SetPayloadData(const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation) override;
-
- // Get the next payload with generic payload header.
- // Write payload and set marker bit of the |packet|.
- // Returns true on success, false otherwise.
- bool NextPacket(RtpPacketToSend* packet) override;
-
- std::string ToString() override;
-
- private:
- const RTPVideoHeaderStereo header_;
- const size_t max_payload_len_;
- const size_t last_packet_reduction_len_;
- size_t num_packets_remaining_ = 0;
- // TODO(emircan): Use codec specific packetizers. If not possible, refactor
- // this class to have similar logic to generic packetizer.
- RtpPacketizerGeneric packetizer_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerStereo);
-};
-
-class RtpDepacketizerStereo : public RtpDepacketizer {
- public:
- ~RtpDepacketizerStereo() override;
-
- bool Parse(ParsedPayload* parsed_payload,
- const uint8_t* payload_data,
- size_t payload_data_length) override;
-
- private:
- RtpDepacketizerGeneric depacketizer_;
-};
-} // namespace webrtc
-#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_
diff --git a/modules/rtp_rtcp/source/rtp_format_video_stereo_unittest.cc b/modules/rtp_rtcp/source/rtp_format_video_stereo_unittest.cc
deleted file mode 100644
index f75dbce..0000000
--- a/modules/rtp_rtcp/source/rtp_format_video_stereo_unittest.cc
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * Copyright (c) 2017 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 <vector>
-
-#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "test/gmock.h"
-#include "test/gtest.h"
-
-namespace webrtc {
-
-namespace {
-
-using ::testing::Each;
-using ::testing::ElementsAreArray;
-using ::testing::Le;
-using ::testing::SizeIs;
-
-constexpr RtpVideoCodecTypes kTestAssociatedCodecType = kRtpVideoVp9;
-constexpr uint8_t kTestFrameIndex = 23;
-constexpr uint8_t kTestFrameCount = 34;
-constexpr uint16_t kTestPictureIndex = 123;
-
-RTPVideoHeaderStereo GenerateTestStereoHeader() {
- RTPVideoHeaderStereo header;
- header.associated_codec_type = kTestAssociatedCodecType;
- header.indices.frame_index = kTestFrameIndex;
- header.indices.frame_count = kTestFrameCount;
- header.indices.picture_index = kTestPictureIndex;
- return header;
-}
-
-std::vector<uint8_t> GenerateTestPayload() {
- const size_t kPayloadSize = 68;
- return std::vector<uint8_t>(kPayloadSize, 0);
-}
-
-std::vector<size_t> NextPacketFillPayloadSizes(
- RtpPacketizerStereo* packetizer) {
- RtpPacketToSend packet(nullptr);
- std::vector<size_t> result;
- while (packetizer->NextPacket(&packet))
- result.push_back(packet.payload_size());
- return result;
-}
-
-} // namespace
-
-TEST(RtpPacketizerVideoStereo, SmallMaxPayloadSizeThrowsErrors) {
- const size_t kMaxPayloadLen = 7;
- const size_t kLastPacketReductionLen = 2;
- RtpPacketizerStereo packetizer(GenerateTestStereoHeader(), kVideoFrameKey,
- kMaxPayloadLen, kLastPacketReductionLen);
- const std::vector<uint8_t>& test_payload = GenerateTestPayload();
- packetizer.SetPayloadData(test_payload.data(), test_payload.size(), nullptr);
- RtpPacketToSend packet(nullptr);
- EXPECT_FALSE(packetizer.NextPacket(&packet));
-}
-
-TEST(RtpPacketizerVideoStereo, AllPacketsRespectMaxPayloadSize) {
- const size_t kMaxPayloadLen = 34;
- const size_t kLastPacketReductionLen = 2;
- RtpPacketizerStereo packetizer(GenerateTestStereoHeader(), kVideoFrameKey,
- kMaxPayloadLen, kLastPacketReductionLen);
- const std::vector<uint8_t>& test_payload = GenerateTestPayload();
- size_t num_packets = packetizer.SetPayloadData(test_payload.data(),
- test_payload.size(), nullptr);
- std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
- EXPECT_THAT(payload_sizes, SizeIs(num_packets));
- EXPECT_THAT(payload_sizes, Each(Le(kMaxPayloadLen)));
-}
-
-TEST(RtpPacketizerVideoStereo, PreservesTypeAndHeader) {
- const size_t kMaxPayloadLen = 34;
- const size_t kLastPacketReductionLen = 2;
- const auto kFrameType = kVideoFrameKey;
- RtpPacketizerStereo packetizer(GenerateTestStereoHeader(), kFrameType,
- kMaxPayloadLen, kLastPacketReductionLen);
- const std::vector<uint8_t>& test_payload = GenerateTestPayload();
- packetizer.SetPayloadData(test_payload.data(), test_payload.size(), nullptr);
- RtpPacketToSend packet(nullptr);
- std::vector<RtpPacketToSend> result;
- while (packetizer.NextPacket(&packet)) {
- result.push_back(packet);
- packet = RtpPacketToSend(nullptr);
- }
-
- RtpDepacketizerStereo depacketizer;
- const auto& first_payload = result.front().payload();
- RtpDepacketizer::ParsedPayload parsed_payload;
- ASSERT_TRUE(depacketizer.Parse(&parsed_payload, first_payload.data(),
- first_payload.size()));
- EXPECT_TRUE(parsed_payload.type.Video.is_first_packet_in_frame);
-
- const auto& last_payload = result.back().payload();
- ASSERT_TRUE(depacketizer.Parse(&parsed_payload, last_payload.data(),
- last_payload.size()));
- EXPECT_EQ(kFrameType, parsed_payload.frame_type);
- EXPECT_EQ(kRtpVideoStereo, parsed_payload.type.Video.codec);
- EXPECT_EQ(kTestAssociatedCodecType,
- parsed_payload.type.Video.codecHeader.stereo.associated_codec_type);
- EXPECT_EQ(kTestFrameIndex,
- parsed_payload.type.Video.codecHeader.stereo.indices.frame_index);
- EXPECT_EQ(kTestFrameCount,
- parsed_payload.type.Video.codecHeader.stereo.indices.frame_count);
- EXPECT_EQ(kTestPictureIndex,
- parsed_payload.type.Video.codecHeader.stereo.indices.picture_index);
-}
-
-} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_payload_registry.cc b/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 60c8c75..631ffae 100644
--- a/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -58,11 +58,14 @@
case kVideoCodecRED:
case kVideoCodecULPFEC:
return kRtpVideoNone;
+ case kVideoCodecI420:
+ case kVideoCodecFlexfec:
+ case kVideoCodecGeneric:
case kVideoCodecStereo:
- return kRtpVideoStereo;
- default:
+ case kVideoCodecUnknown:
return kRtpVideoGeneric;
}
+ return kRtpVideoGeneric;
}
RtpUtility::Payload CreatePayloadType(const VideoCodec& video_codec) {
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index da75688..bb8c795 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -90,7 +90,7 @@
} else if (RtpUtility::StringCompare(payload_name, "I420", 4)) {
video_type = kRtpVideoGeneric;
} else if (RtpUtility::StringCompare(payload_name, "stereo", 6)) {
- video_type = kRtpVideoStereo;
+ video_type = kRtpVideoGeneric;
} else {
video_type = kRtpVideoGeneric;
}
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index d493e4e..05aa634 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -160,7 +160,6 @@
sources = [
"codecs/h264/include/h264_globals.h",
"codecs/interface/common_constants.h",
- "codecs/stereo/include/stereo_globals.h",
"codecs/vp8/include/vp8_globals.h",
"codecs/vp9/include/vp9_globals.h",
]
diff --git a/modules/video_coding/codecs/stereo/include/stereo_globals.h b/modules/video_coding/codecs/stereo/include/stereo_globals.h
deleted file mode 100644
index 9f9ad0e..0000000
--- a/modules/video_coding/codecs/stereo/include/stereo_globals.h
+++ /dev/null
@@ -1,24 +0,0 @@
-/*
- * Copyright (c) 2017 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.
- */
-
-#ifndef MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_GLOBALS_H_
-#define MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_GLOBALS_H_
-
-namespace webrtc {
-
-struct StereoIndices {
- uint8_t frame_index;
- uint8_t frame_count;
- uint16_t picture_index;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_GLOBALS_H_
diff --git a/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc b/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc
index b44165e..3c1cc53 100644
--- a/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc
+++ b/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc
@@ -212,8 +212,6 @@
const EncodedImage& encodedImage,
const CodecSpecificInfo* codecSpecificInfo,
const RTPFragmentationHeader* fragmentation) {
- CodecSpecificInfo codec_info = *codecSpecificInfo;
- codec_info.codecType = kVideoCodecStereo;
const auto& stashed_image_itr = stashed_images_.find(encodedImage._timeStamp);
const auto& stashed_image_next_itr = std::next(stashed_image_itr, 1);
RTC_DCHECK(stashed_image_itr != stashed_images_.end());
@@ -246,6 +244,10 @@
delete[] combined_image_._buffer;
combined_image_ =
MultiplexEncodedImagePacker::PackAndRelease(iter->second);
+
+ CodecSpecificInfo codec_info = *codecSpecificInfo;
+ codec_info.codecType = kVideoCodecStereo;
+ codec_info.codecSpecific.generic.simulcast_idx = 0;
encoded_complete_callback_->OnEncodedImage(combined_image_, &codec_info,
fragmentation);
iter++;
diff --git a/modules/video_coding/encoded_frame.cc b/modules/video_coding/encoded_frame.cc
index 2d8716c..9950012 100644
--- a/modules/video_coding/encoded_frame.cc
+++ b/modules/video_coding/encoded_frame.cc
@@ -193,29 +193,8 @@
_codecSpecificInfo.codecType = kVideoCodecH264;
break;
}
- case kRtpVideoStereo: {
- _codecSpecificInfo.codecType = kVideoCodecStereo;
- VideoCodecType associated_codec_type = kVideoCodecUnknown;
- switch (header->codecHeader.stereo.associated_codec_type) {
- case kRtpVideoVp8:
- associated_codec_type = kVideoCodecVP8;
- break;
- case kRtpVideoVp9:
- associated_codec_type = kVideoCodecVP9;
- break;
- case kRtpVideoH264:
- associated_codec_type = kVideoCodecH264;
- break;
- default:
- RTC_NOTREACHED();
- }
- _codecSpecificInfo.codecSpecific.stereo.associated_codec_type =
- associated_codec_type;
- _codecSpecificInfo.codecSpecific.stereo.indices =
- header->codecHeader.stereo.indices;
- break;
- }
- default: {
+ case kRtpVideoNone:
+ case kRtpVideoGeneric: {
_codecSpecificInfo.codecType = kVideoCodecUnknown;
break;
}
diff --git a/modules/video_coding/frame_object.cc b/modules/video_coding/frame_object.cc
index 6eb28de..6a31cfd 100644
--- a/modules/video_coding/frame_object.cc
+++ b/modules/video_coding/frame_object.cc
@@ -43,14 +43,9 @@
frame_type_ = first_packet->frameType;
codec_type_ = first_packet->codec;
- // Stereo codec appends CopyCodecSpecific to last packet to avoid copy.
- VCMPacket* packet_with_codec_specific =
- codec_type_ == kVideoCodecStereo ? packet_buffer_->GetPacket(last_seq_num)
- : first_packet;
-
// TODO(philipel): Remove when encoded image is replaced by FrameObject.
// VCMEncodedFrame members
- CopyCodecSpecific(&packet_with_codec_specific->video_header);
+ CopyCodecSpecific(&first_packet->video_header);
_completeFrame = true;
_payloadType = first_packet->payloadType;
_timeStamp = first_packet->timestamp;
diff --git a/modules/video_coding/include/video_codec_interface.h b/modules/video_coding/include/video_codec_interface.h
index ef52d8bf..6616053 100644
--- a/modules/video_coding/include/video_codec_interface.h
+++ b/modules/video_coding/include/video_codec_interface.h
@@ -73,17 +73,11 @@
H264PacketizationMode packetization_mode;
};
-struct CodecSpecificInfoStereo {
- VideoCodecType associated_codec_type;
- StereoIndices indices;
-};
-
union CodecSpecificInfoUnion {
CodecSpecificInfoGeneric generic;
CodecSpecificInfoVP8 VP8;
CodecSpecificInfoVP9 VP9;
CodecSpecificInfoH264 H264;
- CodecSpecificInfoStereo stereo;
};
// Note: if any pointers are added to this struct or its sub-structs, it
diff --git a/modules/video_coding/packet.cc b/modules/video_coding/packet.cc
index 9ae5ba0..f176194 100644
--- a/modules/video_coding/packet.cc
+++ b/modules/video_coding/packet.cc
@@ -133,9 +133,6 @@
}
codec = kVideoCodecH264;
return;
- case kRtpVideoStereo:
- codec = kVideoCodecStereo;
- return;
case kRtpVideoGeneric:
codec = kVideoCodecGeneric;
return;
diff --git a/video/payload_router.cc b/video/payload_router.cc
index 68d4ee8..743abc5 100644
--- a/video/payload_router.cc
+++ b/video/payload_router.cc
@@ -81,26 +81,7 @@
rtp->codecHeader.H264.packetization_mode =
info->codecSpecific.H264.packetization_mode;
return;
- case kVideoCodecStereo: {
- rtp->codec = kRtpVideoStereo;
- RtpVideoCodecTypes associated_codec_type = kRtpVideoNone;
- switch (info->codecSpecific.stereo.associated_codec_type) {
- case kVideoCodecVP8:
- associated_codec_type = kRtpVideoVp8;
- break;
- case kVideoCodecVP9:
- associated_codec_type = kRtpVideoVp9;
- break;
- case kVideoCodecH264:
- associated_codec_type = kRtpVideoH264;
- break;
- default:
- RTC_NOTREACHED();
- }
- rtp->codecHeader.stereo.associated_codec_type = associated_codec_type;
- rtp->codecHeader.stereo.indices = info->codecSpecific.stereo.indices;
- return;
- }
+ case kVideoCodecStereo:
case kVideoCodecGeneric:
rtp->codec = kRtpVideoGeneric;
rtp->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
