Delete webrtc::PacketTime and backwards compatibility.
This is a followup to https://webrtc-review.googlesource.com/c/src/+/91840, which needed transitional methods while updating downstream code. This cl completes the deletion, and can be landed after downstream code is updated. Bug: webtrc:9584 Change-Id: I4d3654748973a4757a8d79bb93f524c630a0eca3 Reviewed-on: https://webrtc-review.googlesource.com/93285 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24329}
This commit is contained in:
@ -20,7 +20,6 @@ rtc_source_set("call_interfaces") {
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"call_config.h",
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"flexfec_receive_stream.cc",
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"flexfec_receive_stream.h",
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"packet_receiver.cc",
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"packet_receiver.h",
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"syncable.cc",
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"syncable.h",
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@ -44,23 +44,6 @@ NetworkPacket::NetworkPacket(rtc::CopyOnWriteBuffer packet,
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media_type_(media_type),
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packet_time_us_(packet_time_us) {}
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NetworkPacket::NetworkPacket(rtc::CopyOnWriteBuffer packet,
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int64_t send_time,
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int64_t arrival_time,
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absl::optional<PacketOptions> packet_options,
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bool is_rtcp,
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MediaType media_type,
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absl::optional<PacketTime> packet_time)
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: NetworkPacket(packet,
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send_time,
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arrival_time,
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packet_options,
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is_rtcp,
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media_type,
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packet_time
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? absl::optional<int64_t>(packet_time->timestamp)
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: absl::nullopt) {}
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NetworkPacket::NetworkPacket(NetworkPacket&& o)
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: packet_(std::move(o.packet_)),
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send_time_(o.send_time_),
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@ -215,25 +198,11 @@ bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
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bool is_rtcp,
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MediaType media_type,
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absl::optional<int64_t> packet_time_us) {
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absl::optional<PacketTime> packet_time;
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if (packet_time_us) {
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packet_time = PacketTime(*packet_time_us, -1);
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}
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return EnqueuePacket(packet, options, is_rtcp, media_type, packet_time);
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}
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bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
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absl::optional<PacketOptions> options,
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bool is_rtcp,
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MediaType media_type,
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absl::optional<PacketTime> packet_time) {
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int64_t time_now_us = clock_->TimeInMicroseconds();
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rtc::CritScope crit(&process_lock_);
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size_t packet_size = packet.size();
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NetworkPacket net_packet(
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std::move(packet), time_now_us, time_now_us, options, is_rtcp, media_type,
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packet_time ? absl::optional<int64_t>(packet_time->timestamp)
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: absl::nullopt);
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NetworkPacket net_packet(std::move(packet), time_now_us, time_now_us, options,
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is_rtcp, media_type, packet_time_us);
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packets_in_flight_.emplace_back(StoredPacket(std::move(net_packet)));
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int64_t packet_id = reinterpret_cast<uint64_t>(&packets_in_flight_.back());
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@ -43,14 +43,6 @@ class NetworkPacket {
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bool is_rtcp,
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MediaType media_type,
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absl::optional<int64_t> packet_time_us);
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// TODO(nisse): Deprecated.
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NetworkPacket(rtc::CopyOnWriteBuffer packet,
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int64_t send_time,
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int64_t arrival_time,
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absl::optional<PacketOptions> packet_options,
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bool is_rtcp,
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MediaType media_type,
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absl::optional<PacketTime> packet_time);
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// Disallow copy constructor and copy assignment (no deep copies of |data_|).
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NetworkPacket(const NetworkPacket&) = delete;
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@ -74,10 +66,6 @@ class NetworkPacket {
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bool is_rtcp() const { return is_rtcp_; }
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MediaType media_type() const { return media_type_; }
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absl::optional<int64_t> packet_time_us() const { return packet_time_us_; }
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// TODO(nisse): Deprecated.
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PacketTime packet_time() const {
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return PacketTime(packet_time_us_.value_or(-1), -1);
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}
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private:
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rtc::CopyOnWriteBuffer packet_;
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@ -218,19 +206,11 @@ class FakeNetworkPipe : public Transport, public PacketReceiver, public Module {
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MediaType media_type,
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absl::optional<int64_t> packet_time_us);
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// TODO(nisse): Deprecated. Delete as soon as overrides in downstream code are
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// updated.
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virtual bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
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absl::optional<PacketOptions> options,
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bool is_rtcp,
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MediaType media_type,
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absl::optional<PacketTime> packet_time);
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bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
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absl::optional<PacketOptions> options,
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bool is_rtcp,
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MediaType media_type) {
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return EnqueuePacket(packet, options, is_rtcp, media_type,
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absl::optional<PacketTime>());
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return EnqueuePacket(packet, options, is_rtcp, media_type, absl::nullopt);
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}
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void DeliverNetworkPacket(NetworkPacket* packet)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(config_lock_);
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@ -1,31 +0,0 @@
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/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "call/packet_receiver.h"
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namespace webrtc {
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PacketReceiver::DeliveryStatus PacketReceiver::DeliverPacket(
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MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) {
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return DeliverPacket(media_type, packet, PacketTime(packet_time_us, -1));
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}
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// TODO(bugs.webrtc.org/9584): Deprecated. Over the transition, default
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// implementations are used, and subclasses must override one or the other.
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PacketReceiver::DeliveryStatus PacketReceiver::DeliverPacket(
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MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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const PacketTime& packet_time) {
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return DeliverPacket(media_type, packet, packet_time.timestamp);
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}
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} // namespace webrtc
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@ -31,13 +31,7 @@ class PacketReceiver {
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virtual DeliveryStatus DeliverPacket(MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us);
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// TODO(bugs.webrtc.org/9584): Deprecated. Over the transition, default
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// implementations are used, and subclasses must override one or the other.
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virtual DeliveryStatus DeliverPacket(MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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const PacketTime& packet_time);
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int64_t packet_time_us) = 0;
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protected:
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virtual ~PacketReceiver() {}
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@ -404,24 +404,6 @@ struct OverUseDetectorOptions {
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double initial_var_noise;
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};
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// TODO(nisse): This struct is phased out, delete as soon as down stream code is
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// updated.
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// This structure will have the information about when packet is actually
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// received by socket.
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struct PacketTime {
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PacketTime() : timestamp(-1), not_before(-1) {}
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PacketTime(int64_t timestamp, int64_t not_before)
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: timestamp(timestamp), not_before(not_before) {}
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int64_t timestamp; // Receive time after socket delivers the data.
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int64_t not_before; // Earliest possible time the data could have arrived,
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// indicating the potential error in the |timestamp|
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// value,in case the system is busy.
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// For example, the time of the last select() call.
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// If unknown, this value will be set to zero.
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};
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// Minimum and maximum playout delay values from capture to render.
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// These are best effort values.
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//
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