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Inlines NullAudioPoller functionality into AudioState class.

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As part of this, we also use TaskQueue and RepeatedTask rather
than rtc::Thread + rtc::MessageHandler. With the ultimate goal of
deprecating rtc::Thread.

Bug: webrtc:9883
Change-Id: I2fb851ac31ee2431435d51de78ff446572512201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167528
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30430}
This commit is contained in:
Sebastian Jansson
2020-01-30 16:30:18 +01:00
committed by Commit Bot
parent 8e998f17e0
commit 0e96535be9
5 changed files with 28 additions and 118 deletions
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3
audio/BUILD.gn
View File

@ -29,8 +29,6 @@ rtc_library("audio") {
"channel_send.cc",
"channel_send.h",
"conversion.h",
"null_audio_poller.cc",
"null_audio_poller.h",
"remix_resample.cc",
"remix_resample.h",
]
@ -82,6 +80,7 @@ rtc_library("audio") {
"../rtc_base:rtc_task_queue",
"../rtc_base:safe_minmax",
"../rtc_base/experiments:field_trial_parser",
"../rtc_base/task_utils:repeating_task",
"../system_wrappers",
"../system_wrappers:field_trial",
"../system_wrappers:metrics",

28
audio/audio_state.cc
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@ -38,6 +38,7 @@ AudioState::~AudioState() {
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(receiving_streams_.empty());
RTC_DCHECK(sending_streams_.empty());
null_audio_poller_.Stop();
}
AudioProcessing* AudioState::audio_processing() {
@ -176,10 +177,31 @@ void AudioState::UpdateNullAudioPollerState() {
// Run NullAudioPoller when there are receiving streams and playout is
// disabled.
if (!receiving_streams_.empty() && !playout_enabled_) {
if (!null_audio_poller_)
null_audio_poller_ = std::make_unique<NullAudioPoller>(&audio_transport_);
if (!null_audio_poller_.Running()) {
// TODO(srte): Replace current thread with an explicit task queue
// instance.
null_audio_poller_ =
RepeatingTaskHandle::Start(rtc::Thread::Current(), [this] {
// WebRTC uses 10ms audio windows by default
constexpr TimeDelta kPollInterval = TimeDelta::ms(10);
constexpr Frequency kSampleRate = Frequency::kHz(48);
constexpr size_t kSamplesPerPoll =
static_cast<size_t>(kSampleRate * kPollInterval);
constexpr size_t kNumChannels = 1;
int16_t audio_sample_buffer[kSamplesPerPoll * kNumChannels];
// Output variables from |NeedMorePlayData|.
size_t n_samples;
int64_t elapsed_time_ms;
int64_t ntp_time_ms;
audio_transport_.NeedMorePlayData(kSamplesPerPoll, sizeof(int16_t),
kNumChannels, kSampleRate.hertz(),
audio_sample_buffer, n_samples,
&elapsed_time_ms, &ntp_time_ms);
return kPollInterval;
});
}
} else {
null_audio_poller_.reset();
null_audio_poller_.Stop();
}
}
} // namespace internal

4
audio/audio_state.h
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@ -16,11 +16,11 @@
#include <unordered_set>
#include "audio/audio_transport_impl.h"
#include "audio/null_audio_poller.h"
#include "call/audio_state.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
@ -75,7 +75,7 @@ class AudioState : public webrtc::AudioState {
// Null audio poller is used to continue polling the audio streams if audio
// playout is disabled so that audio processing still happens and the audio
// stats are still updated.
std::unique_ptr<NullAudioPoller> null_audio_poller_;
RepeatingTaskHandle null_audio_poller_;
std::unordered_set<webrtc::AudioReceiveStream*> receiving_streams_;
struct StreamProperties {

71
audio/null_audio_poller.cc
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@ -1,71 +0,0 @@
/*
* 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

40
audio/null_audio_poller.h
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@ -1,40 +0,0 @@
/*
* 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.
*/
#ifndef AUDIO_NULL_AUDIO_POLLER_H_
#define AUDIO_NULL_AUDIO_POLLER_H_
#include <stdint.h>
#include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/message_handler.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
namespace internal {
class NullAudioPoller final : public rtc::MessageHandler {
public:
explicit NullAudioPoller(AudioTransport* audio_transport);
~NullAudioPoller() override;
protected:
void OnMessage(rtc::Message* msg) override;
private:
rtc::ThreadChecker thread_checker_;
AudioTransport* const audio_transport_;
int64_t reschedule_at_;
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_NULL_AUDIO_POLLER_H_