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/*
* Copyright (c) 2015 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 "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/test/layer_filtering_transport.h"
namespace webrtc {
namespace test {
LayerFilteringTransport::LayerFilteringTransport(
const FakeNetworkPipe::Config& config,
Call* send_call,
uint8_t vp8_video_payload_type,
uint8_t vp9_video_payload_type,
int selected_tl,
int selected_sl)
: test::DirectTransport(config, send_call),
vp8_video_payload_type_(vp8_video_payload_type),
vp9_video_payload_type_(vp9_video_payload_type),
selected_tl_(selected_tl),
selected_sl_(selected_sl),
discarded_last_packet_(false) {}
bool LayerFilteringTransport::DiscardedLastPacket() const {
return discarded_last_packet_;
}
bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) {
if (selected_tl_ == -1 && selected_sl_ == -1) {
// Nothing to change, forward the packet immediately.
return test::DirectTransport::SendRtp(packet, length, options);
}
bool set_marker_bit = false;
RtpUtility::RtpHeaderParser parser(packet, length);
RTPHeader header;
parser.Parse(&header);
RTC_DCHECK_LE(length, static_cast<size_t>(IP_PACKET_SIZE));
uint8_t temp_buffer[IP_PACKET_SIZE];
memcpy(temp_buffer, packet, length);
if (header.payloadType == vp8_video_payload_type_ ||
header.payloadType == vp9_video_payload_type_) {
const uint8_t* payload = packet + header.headerLength;
RTC_DCHECK_GT(length, header.headerLength);
const size_t payload_length = length - header.headerLength;
RTC_DCHECK_GT(payload_length, header.paddingLength);
const size_t payload_data_length = payload_length - header.paddingLength;
const bool is_vp8 = header.payloadType == vp8_video_payload_type_;
rtc::scoped_ptr<RtpDepacketizer> depacketizer(
RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
RtpDepacketizer::ParsedPayload parsed_payload;
if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) {
const int temporal_idx = static_cast<int>(
is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
: parsed_payload.type.Video.codecHeader.VP9.temporal_idx);
const int spatial_idx = static_cast<int>(
is_vp8 ? kNoSpatialIdx
: parsed_payload.type.Video.codecHeader.VP9.spatial_idx);
if (selected_sl_ >= 0 && spatial_idx == selected_sl_ &&
parsed_payload.type.Video.codecHeader.VP9.end_of_frame) {
// This layer is now the last in the superframe.
set_marker_bit = true;
} else if ((selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
temporal_idx > selected_tl_) ||
(selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
spatial_idx > selected_sl_)) {
// Truncate packet to a padding packet.
length = header.headerLength + 1;
temp_buffer[0] |= (1 << 5); // P = 1.
temp_buffer[1] &= 0x7F; // M = 0.
discarded_last_packet_ = true;
temp_buffer[header.headerLength] = 1; // One byte of padding.
}
} else {
RTC_NOTREACHED() << "Parse error";
}
}
// We are discarding some of the packets (specifically, whole layers), so
// make sure the marker bit is set properly, and that sequence numbers are
// continuous.
if (set_marker_bit)
temp_buffer[1] |= kRtpMarkerBitMask;
return test::DirectTransport::SendRtp(temp_buffer, length, options);
}
} // namespace test
} // namespace webrtc
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