diff options
author | Sebastian Jansson <srte@webrtc.org> | 2020-01-30 18:13:54 +0000 |
---|---|---|
committer | Commit Bot <commit-bot@chromium.org> | 2020-01-30 18:14:11 +0000 |
commit | fdbbada4d15740bfc400c0c1b49314f4590d330d (patch) | |
tree | 68024dbae27b53fa2115a8777921dbc4e47d21a5 | |
parent | 0e96535be97916d8fcaa9873ffab3c636539f9d8 (diff) | |
download | webrtc-fdbbada4d15740bfc400c0c1b49314f4590d330d.tar.gz |
Revert "Inlines NullAudioPoller functionality into AudioState class."
This reverts commit 0e96535be97916d8fcaa9873ffab3c636539f9d8.
Reason for revert: Downstream test failure
Original change's description:
> Inlines NullAudioPoller functionality into AudioState class.
>
> 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}
TBR=saza@webrtc.org,srte@webrtc.org
Change-Id: I4c77259f7b6477fc1cb79350f2d47817f106770d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168046
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30431}
-rw-r--r-- | audio/BUILD.gn | 3 | ||||
-rw-r--r-- | audio/audio_state.cc | 28 | ||||
-rw-r--r-- | audio/audio_state.h | 4 | ||||
-rw-r--r-- | audio/null_audio_poller.cc | 71 | ||||
-rw-r--r-- | audio/null_audio_poller.h | 40 |
5 files changed, 118 insertions, 28 deletions
diff --git a/audio/BUILD.gn b/audio/BUILD.gn index afc9082321..80f2d523e1 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -29,6 +29,8 @@ 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", ] @@ -80,7 +82,6 @@ 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", diff --git a/audio/audio_state.cc b/audio/audio_state.cc index b103bc6964..1a4fd77ed2 100644 --- a/audio/audio_state.cc +++ b/audio/audio_state.cc @@ -38,7 +38,6 @@ 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() { @@ -177,31 +176,10 @@ void AudioState::UpdateNullAudioPollerState() { // Run NullAudioPoller when there are receiving streams and playout is // disabled. if (!receiving_streams_.empty() && !playout_enabled_) { - 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; - }); - } + if (!null_audio_poller_) + null_audio_poller_ = std::make_unique<NullAudioPoller>(&audio_transport_); } else { - null_audio_poller_.Stop(); + null_audio_poller_.reset(); } } } // namespace internal diff --git a/audio/audio_state.h b/audio/audio_state.h index 0cbdf7e40f..f696d5a8fe 100644 --- a/audio/audio_state.h +++ b/audio/audio_state.h @@ -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. - RepeatingTaskHandle null_audio_poller_; + std::unique_ptr<NullAudioPoller> null_audio_poller_; std::unordered_set<webrtc::AudioReceiveStream*> receiving_streams_; struct StreamProperties { diff --git a/audio/null_audio_poller.cc b/audio/null_audio_poller.cc new file mode 100644 index 0000000000..22f575d8bb --- /dev/null +++ b/audio/null_audio_poller.cc @@ -0,0 +1,71 @@ +/* + * 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 diff --git a/audio/null_audio_poller.h b/audio/null_audio_poller.h new file mode 100644 index 0000000000..97cd2c7e6c --- /dev/null +++ b/audio/null_audio_poller.h @@ -0,0 +1,40 @@ +/* + * 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_ |