Files
platform-external-webrtc/pc/remote_audio_source.cc
Mirko Bonadei 317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00

171 lines
5.4 KiB
C++

/*
* Copyright 2014 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 "pc/remote_audio_source.h"
#include <stddef.h>
#include <memory>
#include <string>
#include "absl/algorithm/container.h"
#include "api/scoped_refptr.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
// This proxy is passed to the underlying media engine to receive audio data as
// they come in. The data will then be passed back up to the RemoteAudioSource
// which will fan it out to all the sinks that have been added to it.
class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
public:
explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) {
RTC_DCHECK(source);
}
~AudioDataProxy() override { source_->OnAudioChannelGone(); }
// AudioSinkInterface implementation.
void OnData(const AudioSinkInterface::Data& audio) override {
source_->OnData(audio);
}
private:
const rtc::scoped_refptr<RemoteAudioSource> source_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioDataProxy);
};
RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)
: main_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
state_(MediaSourceInterface::kLive) {
RTC_DCHECK(main_thread_);
RTC_DCHECK(worker_thread_);
}
RemoteAudioSource::~RemoteAudioSource() {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(audio_observers_.empty());
RTC_DCHECK(sinks_.empty());
}
void RemoteAudioSource::Start(cricket::VoiceMediaChannel* media_channel,
uint32_t ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
// Register for callbacks immediately before AddSink so that we always get
// notified when a channel goes out of scope (signaled when "AudioDataProxy"
// is destroyed).
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
media_channel->SetRawAudioSink(ssrc,
std::make_unique<AudioDataProxy>(this));
});
}
void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
uint32_t ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
worker_thread_->Invoke<void>(
RTC_FROM_HERE, [&] { media_channel->SetRawAudioSink(ssrc, nullptr); });
}
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
RTC_DCHECK(main_thread_->IsCurrent());
return state_;
}
bool RemoteAudioSource::remote() const {
RTC_DCHECK(main_thread_->IsCurrent());
return true;
}
void RemoteAudioSource::SetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
for (auto* observer : audio_observers_) {
observer->OnSetVolume(volume);
}
}
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer));
audio_observers_.push_back(observer);
}
void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
audio_observers_.remove(observer);
}
void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
if (state_ != MediaSourceInterface::kLive) {
RTC_LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
return;
}
rtc::CritScope lock(&sink_lock_);
RTC_DCHECK(!absl::c_linear_search(sinks_, sink));
sinks_.push_back(sink);
}
void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
rtc::CritScope lock(&sink_lock_);
sinks_.remove(sink);
}
void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
// Called on the externally-owned audio callback thread, via/from webrtc.
rtc::CritScope lock(&sink_lock_);
for (auto* sink : sinks_) {
sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
audio.samples_per_channel);
}
}
void RemoteAudioSource::OnAudioChannelGone() {
// Called when the audio channel is deleted. It may be the worker thread
// in libjingle or may be a different worker thread.
// This object needs to live long enough for the cleanup logic in OnMessage to
// run, so take a reference to it as the data. Sometimes the message may not
// be processed (because the thread was destroyed shortly after this call),
// but that is fine because the thread destructor will take care of destroying
// the message data which will release the reference on RemoteAudioSource.
main_thread_->Post(RTC_FROM_HERE, this, 0,
new rtc::ScopedRefMessageData<RemoteAudioSource>(this));
}
void RemoteAudioSource::OnMessage(rtc::Message* msg) {
RTC_DCHECK(main_thread_->IsCurrent());
sinks_.clear();
state_ = MediaSourceInterface::kEnded;
FireOnChanged();
// Will possibly delete this RemoteAudioSource since it is reference counted
// in the message.
delete msg->pdata;
}
} // namespace webrtc