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Delete RtpGenericDepacketizer as no longer used

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Bug: webrtc:11152
Change-Id: I275765e1aa013d8188d43e2911e8ab022563d1d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165394
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30234}
This commit is contained in:
Danil Chapovalov 2020-01-09 14:50:45 +01:00 committed by Commit Bot
parent 48148dc840
commit 7d43801a07
5 changed files with 0 additions and 130 deletions
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46
modules/rtp_rtcp/source/rtp_format_video_generic.cc
View File

@ -97,50 +97,4 @@ void RtpPacketizerGeneric::BuildHeader(const RTPVideoHeader& rtp_video_header) {
header_size_ += kExtendedHeaderLength;
}
}
RtpDepacketizerGeneric::RtpDepacketizerGeneric(bool generic_header_enabled)
: generic_header_enabled_(generic_header_enabled) {}
RtpDepacketizerGeneric::~RtpDepacketizerGeneric() = default;
bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) {
assert(parsed_payload != NULL);
if (payload_data_length == 0) {
RTC_LOG(LS_WARNING) << "Empty payload.";
return false;
}
if (generic_header_enabled_) {
uint8_t generic_header = *payload_data++;
--payload_data_length;
parsed_payload->video_header().frame_type =
((generic_header & RtpFormatVideoGeneric::kKeyFrameBit) != 0)
? VideoFrameType::kVideoFrameKey
: VideoFrameType::kVideoFrameDelta;
parsed_payload->video_header().is_first_packet_in_frame =
(generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0;
parsed_payload->video_header().codec = kVideoCodecGeneric;
parsed_payload->video_header().width = 0;
parsed_payload->video_header().height = 0;
if (generic_header & RtpFormatVideoGeneric::kExtendedHeaderBit) {
if (payload_data_length < kExtendedHeaderLength) {
RTC_LOG(LS_WARNING) << "Too short payload for generic header.";
return false;
}
parsed_payload->video_header().generic.emplace();
parsed_payload->video_header().generic->frame_id =
((payload_data[0] & 0x7F) << 8) | payload_data[1];
payload_data += kExtendedHeaderLength;
payload_data_length -= kExtendedHeaderLength;
}
}
parsed_payload->payload = payload_data;
parsed_payload->payload_length = payload_data_length;
return true;
}
} // namespace webrtc

16
modules/rtp_rtcp/source/rtp_format_video_generic.h
View File

@ -67,21 +67,5 @@ class RtpPacketizerGeneric : public RtpPacketizer {
RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
};
// Depacketizer for generic codec.
class RtpDepacketizerGeneric : public RtpDepacketizer {
public:
// Parses the generic payload header if |generic_header_enabled| is true,
// returns raw payload otherwise.
explicit RtpDepacketizerGeneric(bool generic_header_enabled);
~RtpDepacketizerGeneric() override;
bool Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) override;
private:
bool generic_header_enabled_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_

37
modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
View File

@ -165,42 +165,5 @@ TEST(RtpPacketizerVideoGeneric, DoesNotWriteHeaderForRawPayload) {
EXPECT_THAT(payload, ElementsAreArray(kPayload));
}
TEST(RtpDepacketizerVideoGeneric, NonExtendedHeaderNoFrameId) {
const size_t kPayloadLen = 1;
uint8_t payload[kPayloadLen] = {0x01};
RtpDepacketizerGeneric depacketizer(/*generic_header_enabled=*/true);
RtpDepacketizer::ParsedPayload parsed_payload;
depacketizer.Parse(&parsed_payload, payload, kPayloadLen);
EXPECT_FALSE(parsed_payload.video_header().generic);
}
TEST(RtpDepacketizerVideoGeneric, ExtendedHeaderParsesFrameId) {
const size_t kPayloadLen = 3;
uint8_t payload[kPayloadLen] = {0x05, 0x13, 0x37};
RtpDepacketizerGeneric depacketizer(/*generic_header_enabled=*/true);
RtpDepacketizer::ParsedPayload parsed_payload;
depacketizer.Parse(&parsed_payload, payload, kPayloadLen);
ASSERT_TRUE(parsed_payload.video_header().generic);
EXPECT_EQ(0x1337, parsed_payload.video_header().generic->frame_id);
}
TEST(RtpDepacketizerVideoGeneric, DoesNotParseHeaderForRawPayload) {
const uint8_t kPayload[] = {0x05, 0x25, 0x52};
const size_t kPayloadLen = sizeof(kPayload);
RtpDepacketizerGeneric depacketizer(/*generic_header_enabled=*/false);
RtpDepacketizer::ParsedPayload parsed_payload;
depacketizer.Parse(&parsed_payload, kPayload, kPayloadLen);
EXPECT_FALSE(parsed_payload.video_header().generic);
EXPECT_THAT(rtc::MakeArrayView<const uint8_t>(parsed_payload.payload,
parsed_payload.payload_length),
ElementsAreArray(kPayload));
}
} // namespace
} // namespace webrtc

9
test/fuzzers/BUILD.gn
View File

@ -63,15 +63,6 @@ webrtc_fuzzer_test("h264_depacketizer_fuzzer") {
]
}
webrtc_fuzzer_test("generic_depacketizer_fuzzer") {
sources = [
"generic_depacketizer_fuzzer.cc",
]
deps = [
"../../modules/rtp_rtcp",
]
}
webrtc_fuzzer_test("vp8_depacketizer_fuzzer") {
sources = [
"vp8_depacketizer_fuzzer.cc",

22
test/fuzzers/generic_depacketizer_fuzzer.cc
View File

@ -1,22 +0,0 @@
/*
* Copyright (c) 2019 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 "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
namespace webrtc {
void FuzzOneInput(const uint8_t* data, size_t size) {
RtpDepacketizerGeneric depacketizer(/*generic_header_enabled=*/true);
RtpDepacketizer::ParsedPayload parsed_payload;
depacketizer.Parse(&parsed_payload, data, size);
}
} // namespace webrtc