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Unified Diff: talk/app/webrtc/audiotrack.cc

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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Index: talk/app/webrtc/audiotrack.cc
diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
deleted file mode 100644
index b3223cd29fbb8bd0d72e4e0792824d479c188db5..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/audiotrack.cc
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2011 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/app/webrtc/audiotrack.h"
-
-#include "webrtc/base/checks.h"
-
-using rtc::scoped_refptr;
-
-namespace webrtc {
-
-const char MediaStreamTrackInterface::kAudioKind[] = "audio";
-
-// static
-scoped_refptr<AudioTrack> AudioTrack::Create(
- const std::string& id,
- const scoped_refptr<AudioSourceInterface>& source) {
- return new rtc::RefCountedObject<AudioTrack>(id, source);
-}
-
-AudioTrack::AudioTrack(const std::string& label,
- const scoped_refptr<AudioSourceInterface>& source)
- : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
- if (audio_source_) {
- audio_source_->RegisterObserver(this);
- OnChanged();
- }
-}
-
-AudioTrack::~AudioTrack() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- set_state(MediaStreamTrackInterface::kEnded);
- if (audio_source_)
- audio_source_->UnregisterObserver(this);
-}
-
-std::string AudioTrack::kind() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- return kAudioKind;
-}
-
-AudioSourceInterface* AudioTrack::GetSource() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- return audio_source_.get();
-}
-
-void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (audio_source_)
- audio_source_->AddSink(sink);
-}
-
-void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (audio_source_)
- audio_source_->RemoveSink(sink);
-}
-
-void AudioTrack::OnChanged() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (state() == kFailed)
- return; // We can't recover from this state (do we ever set it?).
-
- TrackState new_state = kInitializing;
-
- // |audio_source_| must be non-null if we ever get here.
- switch (audio_source_->state()) {
- case MediaSourceInterface::kLive:
- case MediaSourceInterface::kMuted:
- new_state = kLive;
- break;
- case MediaSourceInterface::kEnded:
- new_state = kEnded;
- break;
- case MediaSourceInterface::kInitializing:
- default:
- // use kInitializing.
- break;
- }
-
- set_state(new_state);
-}
-
-} // namespace webrtc
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