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}
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index afc9082..80f2d52 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -29,6 +29,8 @@
"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_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 b103bc6..1a4fd77 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -38,7 +38,6 @@
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 @@
// 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 0cbdf7e..f696d5a 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 @@
// 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 0000000..22f575d
--- /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 0000000..97cd2c7
--- /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_
