abdb470d00
As documented in webrtc:11908 this cleanup is fairly invasive and when a part of a frequently executed code path, can be quite costly in terms of performance overhead. This is currently the case with synchronous calls between threads (Thread) as well with our proxy api classes. With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup or calling MessageHandler(false) explicitly. Next steps will be to update external code to either depend on the AutoCleanup variant, or call MessageHandler(false). Changing the proxy classes to use TaskQueue set of concepts instead of MessageHandler. This avoids the perf overhead related to the cleanup above as well as incompatibility with the thread policy checks in Thread that some current external users of the proxies would otherwise run into (if we were to use Thread::Send() for synchronous call). Following this we'll move the cleanup step into the AutoCleanup class and an RTC_DCHECK that all calls to the MessageHandler are setting the flag to false, before eventually removing the flag and make MessageHandler pure virtual. Bug: webrtc:11908 Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32049}
73 lines
2.2 KiB
C++
73 lines
2.2 KiB
C++
/*
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* Copyright (c) 2017 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 "audio/null_audio_poller.h"
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#include <stddef.h>
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#include "rtc_base/checks.h"
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#include "rtc_base/location.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/time_utils.h"
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namespace webrtc {
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namespace internal {
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namespace {
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constexpr int64_t kPollDelayMs = 10; // WebRTC uses 10ms by default
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constexpr size_t kNumChannels = 1;
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constexpr uint32_t kSamplesPerSecond = 48000; // 48kHz
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constexpr size_t kNumSamples = kSamplesPerSecond / 100; // 10ms of samples
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} // namespace
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NullAudioPoller::NullAudioPoller(AudioTransport* audio_transport)
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: MessageHandler(false),
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audio_transport_(audio_transport),
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reschedule_at_(rtc::TimeMillis() + kPollDelayMs) {
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RTC_DCHECK(audio_transport);
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OnMessage(nullptr); // Start the poll loop.
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}
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NullAudioPoller::~NullAudioPoller() {
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RTC_DCHECK(thread_checker_.IsCurrent());
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rtc::Thread::Current()->Clear(this);
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}
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void NullAudioPoller::OnMessage(rtc::Message* msg) {
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RTC_DCHECK(thread_checker_.IsCurrent());
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// Buffer to hold the audio samples.
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int16_t buffer[kNumSamples * kNumChannels];
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// Output variables from |NeedMorePlayData|.
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size_t n_samples;
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int64_t elapsed_time_ms;
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int64_t ntp_time_ms;
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audio_transport_->NeedMorePlayData(kNumSamples, sizeof(int16_t), kNumChannels,
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kSamplesPerSecond, buffer, n_samples,
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&elapsed_time_ms, &ntp_time_ms);
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// Reschedule the next poll iteration. If, for some reason, the given
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// reschedule time has already passed, reschedule as soon as possible.
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int64_t now = rtc::TimeMillis();
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if (reschedule_at_ < now) {
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reschedule_at_ = now;
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}
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rtc::Thread::Current()->PostAt(RTC_FROM_HERE, reschedule_at_, this, 0);
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// Loop after next will be kPollDelayMs later.
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reschedule_at_ += kPollDelayMs;
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}
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} // namespace internal
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} // namespace webrtc
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