
Reason for revert: Fix gyp build Original issue's description: > Revert of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #2 id:20001 of https://codereview.webrtc.org/2463313002/ ) > > Reason for revert: > Broke dependent project because the .gn changes weren't accompanied by corresponding .gyp changes. > > Original issue's description: > > Add a webrtc{en,de}coderfactory implementation for VideoToolbox > > > > This CL removes the coupling of the VideoToolbox h264 implementation > > to the generic h264 code. The files have been moved into sdb/obj/Framework > > and all dependency on them has been removed from the rest of WebRTC. > > We now add it as an external encoder via a factory supplied to the > > CreatePeerConnectionFactory call. This also brings the iOS implementation > > closer to what we do on Android for MediaCodec. > > > > BUG=webrtc:6619 > > > > Committed: https://crrev.com/6a5047dad31f14f53dd9f8bc1ecde19e1dede2e4 > > Cr-Commit-Position: refs/heads/master@{#14953} > > TBR=magjed@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org > BUG=webrtc:6619 > > Committed: https://crrev.com/d69ad84420d9c0e1c11450c352f6c92e7c9583f1 > Cr-Commit-Position: refs/heads/master@{#14985} R=magjed@webrtc.org TBR=kwiberg@webrtc.org, magjed@webrtc.org, stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6619 Review URL: https://codereview.webrtc.org/2487723004 . Cr-Commit-Position: refs/heads/master@{#14992}
138 lines
5.0 KiB
Plaintext
138 lines
5.0 KiB
Plaintext
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import "RTCPeerConnectionFactory+Private.h"
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#import "NSString+StdString.h"
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#import "RTCAVFoundationVideoSource+Private.h"
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#import "RTCAudioSource+Private.h"
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#import "RTCAudioTrack+Private.h"
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#import "RTCMediaConstraints+Private.h"
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#import "RTCMediaStream+Private.h"
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#import "RTCPeerConnection+Private.h"
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#import "RTCVideoSource+Private.h"
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#import "RTCVideoTrack+Private.h"
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#import "WebRTC/RTCLogging.h"
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#include "videotoolboxvideocodecfactory.h"
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@implementation RTCPeerConnectionFactory {
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std::unique_ptr<rtc::Thread> _networkThread;
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std::unique_ptr<rtc::Thread> _workerThread;
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std::unique_ptr<rtc::Thread> _signalingThread;
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BOOL _hasStartedAecDump;
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}
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@synthesize nativeFactory = _nativeFactory;
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- (instancetype)init {
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if ((self = [super init])) {
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_networkThread = rtc::Thread::CreateWithSocketServer();
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BOOL result = _networkThread->Start();
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NSAssert(result, @"Failed to start network thread.");
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_workerThread = rtc::Thread::Create();
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result = _workerThread->Start();
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NSAssert(result, @"Failed to start worker thread.");
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_signalingThread = rtc::Thread::Create();
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result = _signalingThread->Start();
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NSAssert(result, @"Failed to start signaling thread.");
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const auto encoder_factory = new webrtc::VideoToolboxVideoEncoderFactory();
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const auto decoder_factory = new webrtc::VideoToolboxVideoDecoderFactory();
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// Ownership of encoder/decoder factories is passed on to the
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// peerconnectionfactory, that handles deleting them.
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_nativeFactory = webrtc::CreatePeerConnectionFactory(
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_networkThread.get(), _workerThread.get(), _signalingThread.get(),
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nullptr, encoder_factory, decoder_factory);
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NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
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}
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return self;
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}
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- (RTCAudioSource *)audioSourceWithConstraints:(nullable RTCMediaConstraints *)constraints {
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std::unique_ptr<webrtc::MediaConstraints> nativeConstraints;
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if (constraints) {
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nativeConstraints = constraints.nativeConstraints;
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}
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rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
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_nativeFactory->CreateAudioSource(nativeConstraints.get());
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return [[RTCAudioSource alloc] initWithNativeAudioSource:source];
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}
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- (RTCAudioTrack *)audioTrackWithTrackId:(NSString *)trackId {
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RTCAudioSource *audioSource = [self audioSourceWithConstraints:nil];
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return [self audioTrackWithSource:audioSource trackId:trackId];
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}
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- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source
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trackId:(NSString *)trackId {
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return [[RTCAudioTrack alloc] initWithFactory:self
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source:source
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trackId:trackId];
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}
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- (RTCAVFoundationVideoSource *)avFoundationVideoSourceWithConstraints:
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(nullable RTCMediaConstraints *)constraints {
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return [[RTCAVFoundationVideoSource alloc] initWithFactory:self
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constraints:constraints];
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}
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- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source
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trackId:(NSString *)trackId {
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return [[RTCVideoTrack alloc] initWithFactory:self
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source:source
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trackId:trackId];
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}
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- (RTCMediaStream *)mediaStreamWithStreamId:(NSString *)streamId {
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return [[RTCMediaStream alloc] initWithFactory:self
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streamId:streamId];
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}
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- (RTCPeerConnection *)peerConnectionWithConfiguration:
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(RTCConfiguration *)configuration
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constraints:
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(RTCMediaConstraints *)constraints
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delegate:
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(nullable id<RTCPeerConnectionDelegate>)delegate {
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return [[RTCPeerConnection alloc] initWithFactory:self
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configuration:configuration
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constraints:constraints
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delegate:delegate];
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}
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- (BOOL)startAecDumpWithFilePath:(NSString *)filePath
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maxSizeInBytes:(int64_t)maxSizeInBytes {
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RTC_DCHECK(filePath.length);
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RTC_DCHECK_GT(maxSizeInBytes, 0);
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if (_hasStartedAecDump) {
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RTCLogError(@"Aec dump already started.");
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return NO;
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}
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int fd = open(filePath.UTF8String, O_WRONLY | O_CREAT | O_TRUNC, S_IRUSR | S_IWUSR);
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if (fd < 0) {
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RTCLogError(@"Error opening file: %@. Error: %d", filePath, errno);
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return NO;
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}
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_hasStartedAecDump = _nativeFactory->StartAecDump(fd, maxSizeInBytes);
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return _hasStartedAecDump;
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}
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- (void)stopAecDump {
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_nativeFactory->StopAecDump();
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_hasStartedAecDump = NO;
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}
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@end
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