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This reverts commit d93b7b91e0
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Reason for revert: Breaks downstream tests
Original change's description:
> Unifying the handling of the events in NetEqInput.
>
> Bug: webrtc:14763
> Change-Id: I9615a9ce41c9b577c4ebd4cdcc9885bfbc5dac48
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293040
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39706}
Bug: webrtc:14763
Change-Id: If076c8fc59a38f011dfa20829f2dd91dd2f914b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299420
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39711}
135 lines
4.9 KiB
C++
135 lines
4.9 KiB
C++
/*
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* Copyright (c) 2016 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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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
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#include <algorithm>
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#include <memory>
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#include <string>
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#include "absl/types/optional.h"
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "rtc_base/buffer.h"
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namespace webrtc {
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namespace test {
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// Interface class for input to the NetEqTest class.
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class NetEqInput {
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public:
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struct PacketData {
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PacketData();
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~PacketData();
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std::string ToString() const;
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RTPHeader header;
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rtc::Buffer payload;
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int64_t time_ms;
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};
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struct SetMinimumDelayInfo {
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SetMinimumDelayInfo(int64_t timestamp_ms_in, int delay_ms_in)
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: timestamp_ms(timestamp_ms_in), delay_ms(delay_ms_in) {}
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int64_t timestamp_ms;
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int delay_ms;
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};
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virtual ~NetEqInput() = default;
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// Returns at what time (in ms) NetEq::InsertPacket should be called next, or
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// empty if the source is out of packets.
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virtual absl::optional<int64_t> NextPacketTime() const = 0;
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// Returns at what time (in ms) NetEq::GetAudio should be called next, or
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// empty if no more output events are available.
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virtual absl::optional<int64_t> NextOutputEventTime() const = 0;
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// Returns the information related to the next NetEq set minimum delay event
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// if available.
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virtual absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo()
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const = 0;
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// Returns the time (in ms) for the next event (packet, output or set minimum
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// delay event) or empty if there are no more events.
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absl::optional<int64_t> NextEventTime() const {
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absl::optional<int64_t> next_event_time = NextPacketTime();
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const auto next_output_time = NextOutputEventTime();
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// Return the minimum of non-empty `a` and `b`, or empty if both are empty.
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if (next_output_time) {
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next_event_time = next_event_time ? std::min(next_event_time.value(),
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next_output_time.value())
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: next_output_time;
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}
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const auto next_neteq_minimum_delay = NextSetMinimumDelayInfo();
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if (next_neteq_minimum_delay) {
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next_event_time =
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next_event_time
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? std::min(next_event_time.value(),
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next_neteq_minimum_delay.value().timestamp_ms)
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: next_neteq_minimum_delay.value().timestamp_ms;
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}
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return next_event_time;
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}
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// Returns the next packet to be inserted into NetEq. The packet following the
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// returned one is pre-fetched in the NetEqInput object, such that future
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// calls to NextPacketTime() or NextHeader() will return information from that
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// packet.
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virtual std::unique_ptr<PacketData> PopPacket() = 0;
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// Move to the next output event. This will make NextOutputEventTime() return
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// a new value (potentially the same if several output events share the same
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// time).
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virtual void AdvanceOutputEvent() = 0;
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// Move to the next NetEq set minimum delay. This will make
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// `NextSetMinimumDelayInfo` return a new value.
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virtual void AdvanceSetMinimumDelay() = 0;
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// Returns true if the source has come to an end. An implementation must
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// eventually return true from this method, or the test will end up in an
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// infinite loop.
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virtual bool ended() const = 0;
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// Returns the RTP header for the next packet, i.e., the packet that will be
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// delivered next by PopPacket().
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virtual absl::optional<RTPHeader> NextHeader() const = 0;
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};
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// Wrapper class to impose a time limit on a NetEqInput object, typically
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// another time limit than what the object itself provides. For example, an
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// input taken from a file can be cut shorter by wrapping it in this class.
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class TimeLimitedNetEqInput : public NetEqInput {
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public:
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TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
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~TimeLimitedNetEqInput() override;
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absl::optional<int64_t> NextPacketTime() const override;
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absl::optional<int64_t> NextOutputEventTime() const override;
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absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override;
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std::unique_ptr<PacketData> PopPacket() override;
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void AdvanceOutputEvent() override;
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void AdvanceSetMinimumDelay() override;
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bool ended() const override;
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absl::optional<RTPHeader> NextHeader() const override;
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private:
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void MaybeSetEnded();
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std::unique_ptr<NetEqInput> input_;
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const absl::optional<int64_t> start_time_ms_;
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const int64_t duration_ms_;
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bool ended_ = false;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
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