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This reverts commitfab3460a82
. Reason for revert: fix downstream instead Original change's description: > Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" > > This reverts commit9973933d2e
. > > Reason for revert: breaking downstream projects and not reviewed by direct owners > > Original change's description: > > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > This reverts commit24192c267a
. > > > > Reason for revert: Analyzed the performance regression in more detail. > > > > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit1796a820f6
has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac. > > > > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall. > > > > Original change's description: > > > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > > > This reverts commit3e8ef940fe
. > > > > > > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260. > > > > > > Original change's description: > > > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. > > > > > > > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time. > > > > > > > > Bug: webrtc:10668 > > > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890 > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Reviewed-by: Minyue Li <minyue@webrtc.org> > > > > Commit-Queue: Chen Xing <chxg@google.com> > > > > Cr-Commit-Position: refs/heads/master@{#28434} > > > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > > > Bug: webrtc:10668, chromium:982260 > > > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339 > > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28561} > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:10668, chromium:982260 > > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Chen Xing <chxg@google.com> > > Cr-Commit-Position: refs/heads/master@{#28664} > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10668, chromium:982260 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28671} TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10668, chromium:982260 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28672}
127 lines
4.3 KiB
C++
127 lines
4.3 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_TEST_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
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#include <fstream>
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#include <map>
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#include <memory>
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#include <string>
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#include <utility>
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#include "absl/types/optional.h"
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#include "api/audio_codecs/audio_decoder_factory.h"
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#include "api/test/neteq_simulator.h"
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#include "modules/audio_coding/neteq/include/neteq.h"
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "modules/audio_coding/neteq/tools/neteq_input.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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namespace test {
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class NetEqTestErrorCallback {
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public:
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virtual ~NetEqTestErrorCallback() = default;
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virtual void OnInsertPacketError(const NetEqInput::PacketData& packet) {}
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virtual void OnGetAudioError() {}
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};
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class DefaultNetEqTestErrorCallback : public NetEqTestErrorCallback {
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void OnInsertPacketError(const NetEqInput::PacketData& packet) override;
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void OnGetAudioError() override;
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};
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class NetEqPostInsertPacket {
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public:
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virtual ~NetEqPostInsertPacket() = default;
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virtual void AfterInsertPacket(const NetEqInput::PacketData& packet,
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NetEq* neteq) = 0;
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};
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class NetEqGetAudioCallback {
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public:
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virtual ~NetEqGetAudioCallback() = default;
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virtual void BeforeGetAudio(NetEq* neteq) = 0;
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virtual void AfterGetAudio(int64_t time_now_ms,
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const AudioFrame& audio_frame,
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bool muted,
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NetEq* neteq) = 0;
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};
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class NetEqSimulationEndedCallback {
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public:
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virtual ~NetEqSimulationEndedCallback() = default;
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virtual void SimulationEnded(int64_t simulation_time_ms) = 0;
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};
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// Class that provides an input--output test for NetEq. The input (both packets
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// and output events) is provided by a NetEqInput object, while the output is
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// directed to an AudioSink object.
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class NetEqTest : public NetEqSimulator {
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public:
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using DecoderMap = std::map<int, SdpAudioFormat>;
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struct Callbacks {
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NetEqTestErrorCallback* error_callback = nullptr;
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NetEqPostInsertPacket* post_insert_packet = nullptr;
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NetEqGetAudioCallback* get_audio_callback = nullptr;
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NetEqSimulationEndedCallback* simulation_ended_callback = nullptr;
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};
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// Sets up the test with given configuration, codec mappings, input, ouput,
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// and callback objects for error reporting.
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NetEqTest(const NetEq::Config& config,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
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const DecoderMap& codecs,
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std::unique_ptr<std::ofstream> text_log,
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std::unique_ptr<NetEqInput> input,
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std::unique_ptr<AudioSink> output,
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Callbacks callbacks);
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~NetEqTest() override;
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// Runs the test. Returns the duration of the produced audio in ms.
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int64_t Run();
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// Runs the simulation until we hit the next GetAudio event. If the simulation
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// is finished, is_simulation_finished will be set to true in the returned
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// SimulationStepResult.
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SimulationStepResult RunToNextGetAudio() override;
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void SetNextAction(Action next_operation) override;
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NetEqState GetNetEqState() override;
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// Returns the statistics from NetEq.
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NetEqNetworkStatistics SimulationStats();
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NetEqLifetimeStatistics LifetimeStats() const;
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static DecoderMap StandardDecoderMap();
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private:
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void RegisterDecoders(const DecoderMap& codecs);
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SimulatedClock clock_;
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absl::optional<Action> next_action_;
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absl::optional<int> last_packet_time_ms_;
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std::unique_ptr<NetEq> neteq_;
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std::unique_ptr<NetEqInput> input_;
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std::unique_ptr<AudioSink> output_;
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Callbacks callbacks_;
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int sample_rate_hz_;
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NetEqState current_state_;
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NetEqOperationsAndState prev_ops_state_;
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NetEqLifetimeStatistics prev_lifetime_stats_;
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absl::optional<uint32_t> last_packet_timestamp_;
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std::unique_ptr<std::ofstream> text_log_;
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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_TEST_H_
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