diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc | 90 |
1 files changed, 90 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc new file mode 100644 index 0000000000..55a5653238 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc @@ -0,0 +1,90 @@ +/* + * Copyright (c) 2016 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/audio_coding/neteq/tools/neteq_packet_source_input.h" + +#include <algorithm> +#include <limits> + +#include "absl/strings/string_view.h" +#include "modules/audio_coding/neteq/tools/rtp_file_source.h" +#include "rtc_base/checks.h" + +namespace webrtc { +namespace test { + +NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {} + +absl::optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const { + return packet_ + ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms())) + : absl::nullopt; +} + +absl::optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const { + return packet_ ? absl::optional<RTPHeader>(packet_->header()) : absl::nullopt; +} + +void NetEqPacketSourceInput::LoadNextPacket() { + packet_ = source()->NextPacket(); +} + +std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() { + if (!packet_) { + return std::unique_ptr<PacketData>(); + } + std::unique_ptr<PacketData> packet_data(new PacketData); + packet_data->header = packet_->header(); + if (packet_->payload_length_bytes() == 0 && + packet_->virtual_payload_length_bytes() > 0) { + // This is a header-only "dummy" packet. Set the payload to all zeros, with + // length according to the virtual length. + packet_data->payload.SetSize(packet_->virtual_payload_length_bytes()); + std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0); + } else { + packet_data->payload.SetData(packet_->payload(), + packet_->payload_length_bytes()); + } + packet_data->time_ms = packet_->time_ms(); + + LoadNextPacket(); + + return packet_data; +} + +NetEqRtpDumpInput::NetEqRtpDumpInput(absl::string_view file_name, + const RtpHeaderExtensionMap& hdr_ext_map, + absl::optional<uint32_t> ssrc_filter) + : source_(RtpFileSource::Create(file_name, ssrc_filter)) { + for (const auto& ext_pair : hdr_ext_map) { + source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first); + } + LoadNextPacket(); +} + +absl::optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const { + return next_output_event_ms_; +} + +void NetEqRtpDumpInput::AdvanceOutputEvent() { + if (next_output_event_ms_) { + *next_output_event_ms_ += kOutputPeriodMs; + } + if (!NextPacketTime()) { + next_output_event_ms_ = absl::nullopt; + } +} + +PacketSource* NetEqRtpDumpInput::source() { + return source_.get(); +} + +} // namespace test +} // namespace webrtc |