text: /*
   *  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 <stddef.h>
  
  #include "rtc_base/checks.h"
  #include "rtc_base/location.h"
  #include "rtc_base/thread.h"
  #include "rtc_base/timeutils.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
  