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/*
* 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/time_utils.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_.IsCurrent());
rtc::Thread::Current()->Clear(this);
}
void NullAudioPoller::OnMessage(rtc::Message* msg) {
RTC_DCHECK(thread_checker_.IsCurrent());
// 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