mirror of
https://github.com/mollyim/webrtc.git
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2e00abc98e
Reason for reland: Downstream project fixed. Original change's description: > [cleanup] Remove useless includes. > > Manual cleanup guided by include-what-you-use diagnostic. > > Bug: webrtc:8311 > Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68 > Reviewed-on: https://webrtc-review.googlesource.com/c/103320 > Commit-Queue: Yves Gerey <yvesg@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25013} Bug: webrtc:8311 Change-Id: Id6ec4aeb798886a90ace640a190eaf16497ba31b Reviewed-on: https://webrtc-review.googlesource.com/c/104120 Commit-Queue: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25034}
67 lines
2.2 KiB
C++
67 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 "rtc_base/logging.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/timeutils.h" // for TimeMillis
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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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: 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_.CalledOnValidThread());
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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_.CalledOnValidThread());
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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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