/* * Copyright (c) 2017 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 "audio/null_audio_poller.h" #include "rtc_base/logging.h" #include "rtc_base/thread.h" namespace webrtc { namespace internal { namespace { constexpr int64_t kPollDelayMs = 10; // WebRTC uses 10ms by default constexpr size_t kNumChannels = 1; constexpr uint32_t kSamplesPerSecond = 48000; // 48kHz constexpr size_t kNumSamples = kSamplesPerSecond / 100; // 10ms of samples } // namespace NullAudioPoller::NullAudioPoller(AudioTransport* audio_transport) : audio_transport_(audio_transport), reschedule_at_(rtc::TimeMillis() + kPollDelayMs) { RTC_DCHECK(audio_transport); OnMessage(nullptr); // Start the poll loop. } NullAudioPoller::~NullAudioPoller() { RTC_DCHECK(thread_checker_.CalledOnValidThread()); rtc::Thread::Current()->Clear(this); } void NullAudioPoller::OnMessage(rtc::Message* msg) { RTC_DCHECK(thread_checker_.CalledOnValidThread()); // Buffer to hold the audio samples. int16_t buffer[kNumSamples * kNumChannels]; // Output variables from |NeedMorePlayData|. size_t n_samples; int64_t elapsed_time_ms; int64_t ntp_time_ms; audio_transport_->NeedMorePlayData(kNumSamples, sizeof(int16_t), kNumChannels, kSamplesPerSecond, buffer, n_samples, &elapsed_time_ms, &ntp_time_ms); // Reschedule the next poll iteration. If, for some reason, the given // reschedule time has already passed, reschedule as soon as possible. int64_t now = rtc::TimeMillis(); if (reschedule_at_ < now) { reschedule_at_ = now; } rtc::Thread::Current()->PostAt(RTC_FROM_HERE, reschedule_at_, this, 0); // Loop after next will be kPollDelayMs later. reschedule_at_ += kPollDelayMs; } } // namespace internal } // namespace webrtc