haikuwebkit/LayoutTests/webrtc/audio-peer-connection-webau...


PASS Basic audio playback through a peer connection
[WebRTC] Fix remote audio rendering https://bugs.webkit.org/show_bug.cgi?id=168898 Reviewed by Eric Carlson. Source/WebCore: Test: webrtc/audio-peer-connection-webaudio.html Fix MediaStreamAudioSourceNode by not bailing out early if the input sample rate doesn't match the AudioContext's sample rate; there's code in setFormat() to do the sample rate conversion correctly. * Modules/webaudio/MediaStreamAudioSourceNode.cpp: (WebCore::MediaStreamAudioSourceNode::setFormat): Fix AudioSampleBufferList by making the AudioConverter input proc a free function, and passing its refCon a struct containing only the information it needs to perform its task. Because the conversion may result in a different number of output samples than input ones, just ask to generate the entire capacity of the scratch buffer, and signal that the input buffer was fully converted with a special return value. * platform/audio/mac/AudioSampleBufferList.cpp: (WebCore::audioConverterFromABLCallback): (WebCore::AudioSampleBufferList::copyFrom): (WebCore::AudioSampleBufferList::convertInput): Deleted. (WebCore::AudioSampleBufferList::audioConverterCallback): Deleted. * platform/audio/mac/AudioSampleBufferList.h: Fix AudioSampleDataSource by updating both the sampleCount and the sampleTime after doing a sample rate conversion to take into account that both the number of samples may have changed, as well as the timeScale of the sampleTime. This may result in small off-by-one rounding errors due to the sample rate conversion of sampleTime, so remember what the next expected sampleTime should be, and correct sampleTime if it is indeed off-by-one. If the pull operation has gotten ahead of the push operation, delay the next pull by the empty amount by rolling back the m_outputSampleOffset. Introduce the same offset behavior during pull operations. * platform/audio/mac/AudioSampleDataSource.h: * platform/audio/mac/AudioSampleDataSource.mm: (WebCore::AudioSampleDataSource::pushSamplesInternal): (WebCore::AudioSampleDataSource::pullSamplesInternal): (WebCore::AudioSampleDataSource::pullAvalaibleSamplesAsChunks): Fix MediaPlayerPrivateMediaStreamAVFObjC by obeying the m_muted property. * platform/graphics/avfoundation/objc/MediaPlayerPrivateMediaStreamAVFObjC.mm: (WebCore::MediaPlayerPrivateMediaStreamAVFObjC::setVolume): (WebCore::MediaPlayerPrivateMediaStreamAVFObjC::setMuted): Fix LibWebRTCAudioModule by sleeping for the correct amount after emitting frames. Previously, LibWebRTCAudioModule would sleep for a fixed amount of time, which meant it would get slowly out of sync when emitting frames took a non-zero amount of time. Now, the amount of time before the next cycle starts is correctly calculated, and then LibWebRTCAudioModule sleeps for a dynamic amount of time in order to wake up correctly at the beginning of the next cycle. * platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp: (WebCore::LibWebRTCAudioModule::StartPlayoutOnAudioThread): Fix AudioTrackPrivateMediaStreamCocoa by just using the output unit's preferred format description (with the current system sample rate), rather than whatever is the current input description. * platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.cpp: (WebCore::AudioTrackPrivateMediaStreamCocoa::createAudioUnit): (WebCore::AudioTrackPrivateMediaStreamCocoa::audioSamplesAvailable): * platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.h: Fix RealtimeIncomingAudioSource by actually creating an AudioSourceProvider when asked. * platform/mediastream/mac/RealtimeIncomingAudioSource.cpp: (WebCore::RealtimeIncomingAudioSource::OnData): (WebCore::RealtimeIncomingAudioSource::audioSourceProvider): * platform/mediastream/mac/RealtimeIncomingAudioSource.h: Fix RealtimeOutgoingAudioSource by using the outgoing format description rather than the incoming one to determine the sample rate, channel count, sample byte size, etc., to use when delivering data upstream to libWebRTC. * platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp: (WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable): (WebCore::RealtimeOutgoingAudioSource::pullAudioData): * platform/mediastream/mac/RealtimeOutgoingAudioSource.h: Fix WebAudioSourceProviderAVFObjC by using a AudioSampleDataSource to do format and sample rate conversion rather than trying to duplicate all that code and use a CARingBuffer and AudioConverter directly. * platform/mediastream/mac/WebAudioSourceProviderAVFObjC.h: * platform/mediastream/mac/WebAudioSourceProviderAVFObjC.mm: (WebCore::WebAudioSourceProviderAVFObjC::~WebAudioSourceProviderAVFObjC): (WebCore::WebAudioSourceProviderAVFObjC::provideInput): (WebCore::WebAudioSourceProviderAVFObjC::prepare): (WebCore::WebAudioSourceProviderAVFObjC::unprepare): (WebCore::WebAudioSourceProviderAVFObjC::audioSamplesAvailable): Fix the MockLibWebRTCAudioTrack by passing along the AddSink() sink to its AudioSourceInterface, allowing the RealtimeOutgoingAudioSource to push data into the libWebRTC network stack. Also, make sure m_enabled is initialized to a good value. * testing/MockLibWebRTCPeerConnection.h: LayoutTests: * webrtc/audio-peer-connection-webaudio-expected.txt: Added. * webrtc/audio-peer-connection-webaudio.html: Added. Canonical link: https://commits.webkit.org/185912@main git-svn-id: https://svn.webkit.org/repository/webkit/trunk@213080 268f45cc-cd09-0410-ab3c-d52691b4dbfc 2017-02-27 18:22:49 +00:00
Remove trailing spaces from expected.txt files (excluding WPT for now since that contains too many) https://bugs.webkit.org/show_bug.cgi?id=217016 Reviewed by Zalan Bujtas. Done with this command: find LayoutTests -name '*-expected.txt' \| grep -v imported \| xargs awk '/^CONSOLE MESSAGE:/ { next }; /^<\?xml/ { nextfile }; { print FILENAME; nextfile }' \| xargs perl -i -pe 's/ +$//g' This affects many files, and it's not helpful to clog the change log with a list of them. Canonical link: https://commits.webkit.org/229803@main git-svn-id: https://svn.webkit.org/repository/webkit/trunk@267644 268f45cc-cd09-0410-ab3c-d52691b4dbfc 2020-09-27 02:15:19 +00:00			`PASS Basic audio playback through a peer connection`
[WebRTC] Fix remote audio rendering https://bugs.webkit.org/show_bug.cgi?id=168898 Reviewed by Eric Carlson. Source/WebCore: Test: webrtc/audio-peer-connection-webaudio.html Fix MediaStreamAudioSourceNode by not bailing out early if the input sample rate doesn't match the AudioContext's sample rate; there's code in setFormat() to do the sample rate conversion correctly. * Modules/webaudio/MediaStreamAudioSourceNode.cpp: (WebCore::MediaStreamAudioSourceNode::setFormat): Fix AudioSampleBufferList by making the AudioConverter input proc a free function, and passing its refCon a struct containing only the information it needs to perform its task. Because the conversion may result in a different number of output samples than input ones, just ask to generate the entire capacity of the scratch buffer, and signal that the input buffer was fully converted with a special return value. * platform/audio/mac/AudioSampleBufferList.cpp: (WebCore::audioConverterFromABLCallback): (WebCore::AudioSampleBufferList::copyFrom): (WebCore::AudioSampleBufferList::convertInput): Deleted. (WebCore::AudioSampleBufferList::audioConverterCallback): Deleted. * platform/audio/mac/AudioSampleBufferList.h: Fix AudioSampleDataSource by updating both the sampleCount and the sampleTime after doing a sample rate conversion to take into account that both the number of samples may have changed, as well as the timeScale of the sampleTime. This may result in small off-by-one rounding errors due to the sample rate conversion of sampleTime, so remember what the next expected sampleTime should be, and correct sampleTime if it is indeed off-by-one. If the pull operation has gotten ahead of the push operation, delay the next pull by the empty amount by rolling back the m_outputSampleOffset. Introduce the same offset behavior during pull operations. * platform/audio/mac/AudioSampleDataSource.h: * platform/audio/mac/AudioSampleDataSource.mm: (WebCore::AudioSampleDataSource::pushSamplesInternal): (WebCore::AudioSampleDataSource::pullSamplesInternal): (WebCore::AudioSampleDataSource::pullAvalaibleSamplesAsChunks): Fix MediaPlayerPrivateMediaStreamAVFObjC by obeying the m_muted property. * platform/graphics/avfoundation/objc/MediaPlayerPrivateMediaStreamAVFObjC.mm: (WebCore::MediaPlayerPrivateMediaStreamAVFObjC::setVolume): (WebCore::MediaPlayerPrivateMediaStreamAVFObjC::setMuted): Fix LibWebRTCAudioModule by sleeping for the correct amount after emitting frames. Previously, LibWebRTCAudioModule would sleep for a fixed amount of time, which meant it would get slowly out of sync when emitting frames took a non-zero amount of time. Now, the amount of time before the next cycle starts is correctly calculated, and then LibWebRTCAudioModule sleeps for a dynamic amount of time in order to wake up correctly at the beginning of the next cycle. * platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp: (WebCore::LibWebRTCAudioModule::StartPlayoutOnAudioThread): Fix AudioTrackPrivateMediaStreamCocoa by just using the output unit's preferred format description (with the current system sample rate), rather than whatever is the current input description. * platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.cpp: (WebCore::AudioTrackPrivateMediaStreamCocoa::createAudioUnit): (WebCore::AudioTrackPrivateMediaStreamCocoa::audioSamplesAvailable): * platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.h: Fix RealtimeIncomingAudioSource by actually creating an AudioSourceProvider when asked. * platform/mediastream/mac/RealtimeIncomingAudioSource.cpp: (WebCore::RealtimeIncomingAudioSource::OnData): (WebCore::RealtimeIncomingAudioSource::audioSourceProvider): * platform/mediastream/mac/RealtimeIncomingAudioSource.h: Fix RealtimeOutgoingAudioSource by using the outgoing format description rather than the incoming one to determine the sample rate, channel count, sample byte size, etc., to use when delivering data upstream to libWebRTC. * platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp: (WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable): (WebCore::RealtimeOutgoingAudioSource::pullAudioData): * platform/mediastream/mac/RealtimeOutgoingAudioSource.h: Fix WebAudioSourceProviderAVFObjC by using a AudioSampleDataSource to do format and sample rate conversion rather than trying to duplicate all that code and use a CARingBuffer and AudioConverter directly. * platform/mediastream/mac/WebAudioSourceProviderAVFObjC.h: * platform/mediastream/mac/WebAudioSourceProviderAVFObjC.mm: (WebCore::WebAudioSourceProviderAVFObjC::~WebAudioSourceProviderAVFObjC): (WebCore::WebAudioSourceProviderAVFObjC::provideInput): (WebCore::WebAudioSourceProviderAVFObjC::prepare): (WebCore::WebAudioSourceProviderAVFObjC::unprepare): (WebCore::WebAudioSourceProviderAVFObjC::audioSamplesAvailable): Fix the MockLibWebRTCAudioTrack by passing along the AddSink() sink to its AudioSourceInterface, allowing the RealtimeOutgoingAudioSource to push data into the libWebRTC network stack. Also, make sure m_enabled is initialized to a good value. * testing/MockLibWebRTCPeerConnection.h: LayoutTests: * webrtc/audio-peer-connection-webaudio-expected.txt: Added. * webrtc/audio-peer-connection-webaudio.html: Added. Canonical link: https://commits.webkit.org/185912@main git-svn-id: https://svn.webkit.org/repository/webkit/trunk@213080 268f45cc-cd09-0410-ab3c-d52691b4dbfc 2017-02-27 18:22:49 +00:00