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author | Matt A. Tobin <mattatobin@localhost.localdomain> | 2018-02-02 04:16:08 -0500 |
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committer | Matt A. Tobin <mattatobin@localhost.localdomain> | 2018-02-02 04:16:08 -0500 |
commit | 5f8de423f190bbb79a62f804151bc24824fa32d8 (patch) | |
tree | 10027f336435511475e392454359edea8e25895d /dom/media/directshow/AudioSinkInputPin.cpp | |
parent | 49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff) | |
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Add m-esr52 at 52.6.0
Diffstat (limited to 'dom/media/directshow/AudioSinkInputPin.cpp')
-rw-r--r-- | dom/media/directshow/AudioSinkInputPin.cpp | 195 |
1 files changed, 195 insertions, 0 deletions
diff --git a/dom/media/directshow/AudioSinkInputPin.cpp b/dom/media/directshow/AudioSinkInputPin.cpp new file mode 100644 index 000000000..85a6e3da3 --- /dev/null +++ b/dom/media/directshow/AudioSinkInputPin.cpp @@ -0,0 +1,195 @@ +/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ +/* vim:set ts=2 sw=2 sts=2 et cindent: */ +/* This Source Code Form is subject to the terms of the Mozilla Public + * License, v. 2.0. If a copy of the MPL was not distributed with this + * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ + +#include "AudioSinkInputPin.h" +#include "AudioSinkFilter.h" +#include "SampleSink.h" +#include "mozilla/Logging.h" + +#include <wmsdkidl.h> + +using namespace mozilla::media; + +namespace mozilla { + +static LazyLogModule gDirectShowLog("DirectShowDecoder"); +#define LOG(...) MOZ_LOG(gDirectShowLog, mozilla::LogLevel::Debug, (__VA_ARGS__)) + +AudioSinkInputPin::AudioSinkInputPin(wchar_t* aObjectName, + AudioSinkFilter* aFilter, + mozilla::CriticalSection* aLock, + HRESULT* aOutResult) + : BaseInputPin(aObjectName, aFilter, aLock, aOutResult, aObjectName), + mSegmentStartTime(0) +{ + MOZ_COUNT_CTOR(AudioSinkInputPin); + mSampleSink = new SampleSink(); +} + +AudioSinkInputPin::~AudioSinkInputPin() +{ + MOZ_COUNT_DTOR(AudioSinkInputPin); +} + +HRESULT +AudioSinkInputPin::GetMediaType(int aPosition, MediaType* aOutMediaType) +{ + NS_ENSURE_TRUE(aPosition >= 0, E_INVALIDARG); + NS_ENSURE_TRUE(aOutMediaType, E_POINTER); + + if (aPosition > 0) { + return S_FALSE; + } + + // Note: We set output as PCM, as IEEE_FLOAT only works when using the + // MP3 decoder as an MFT, and we can't do that while using DirectShow. + aOutMediaType->SetType(&MEDIATYPE_Audio); + aOutMediaType->SetSubtype(&MEDIASUBTYPE_PCM); + aOutMediaType->SetType(&FORMAT_WaveFormatEx); + aOutMediaType->SetTemporalCompression(FALSE); + + return S_OK; +} + +HRESULT +AudioSinkInputPin::CheckMediaType(const MediaType* aMediaType) +{ + if (!aMediaType) { + return E_INVALIDARG; + } + + GUID majorType = *aMediaType->Type(); + if (majorType != MEDIATYPE_Audio && majorType != WMMEDIATYPE_Audio) { + return E_INVALIDARG; + } + + if (*aMediaType->Subtype() != MEDIASUBTYPE_PCM) { + return E_INVALIDARG; + } + + if (*aMediaType->FormatType() != FORMAT_WaveFormatEx) { + return E_INVALIDARG; + } + + // We accept the media type, stash its layout format! + WAVEFORMATEX* wfx = (WAVEFORMATEX*)(aMediaType->pbFormat); + GetSampleSink()->SetAudioFormat(wfx); + + return S_OK; +} + +AudioSinkFilter* +AudioSinkInputPin::GetAudioSinkFilter() +{ + return reinterpret_cast<AudioSinkFilter*>(mFilter); +} + +SampleSink* +AudioSinkInputPin::GetSampleSink() +{ + return mSampleSink; +} + +HRESULT +AudioSinkInputPin::SetAbsoluteMediaTime(IMediaSample* aSample) +{ + HRESULT hr; + REFERENCE_TIME start = 0, end = 0; + hr = aSample->GetTime(&start, &end); + NS_ENSURE_TRUE(SUCCEEDED(hr), E_FAIL); + { + CriticalSectionAutoEnter lock(*mLock); + start += mSegmentStartTime; + end += mSegmentStartTime; + } + hr = aSample->SetMediaTime(&start, &end); + NS_ENSURE_TRUE(SUCCEEDED(hr), E_FAIL); + return S_OK; +} + +HRESULT +AudioSinkInputPin::Receive(IMediaSample* aSample ) +{ + HRESULT hr; + NS_ENSURE_TRUE(aSample, E_POINTER); + + hr = BaseInputPin::Receive(aSample); + if (SUCCEEDED(hr) && hr != S_FALSE) { // S_FALSE == flushing + // Set the timestamp of the sample after being adjusted for + // seeking/segments in the "media time" attribute. When we seek, + // DirectShow starts a new "segment", and starts labeling samples + // from time=0 again, so we need to correct for this to get the + // actual timestamps after seeking. + hr = SetAbsoluteMediaTime(aSample); + NS_ENSURE_TRUE(SUCCEEDED(hr), hr); + hr = GetSampleSink()->Receive(aSample); + NS_ENSURE_TRUE(SUCCEEDED(hr), hr); + } + return S_OK; +} + +already_AddRefed<IMediaSeeking> +AudioSinkInputPin::GetConnectedPinSeeking() +{ + RefPtr<IPin> peer = GetConnected(); + if (!peer) + return nullptr; + RefPtr<IMediaSeeking> seeking; + peer->QueryInterface(static_cast<IMediaSeeking**>(getter_AddRefs(seeking))); + return seeking.forget(); +} + +HRESULT +AudioSinkInputPin::BeginFlush() +{ + HRESULT hr = media::BaseInputPin::BeginFlush(); + NS_ENSURE_TRUE(SUCCEEDED(hr), hr); + + GetSampleSink()->Flush(); + + return S_OK; +} + +HRESULT +AudioSinkInputPin::EndFlush() +{ + HRESULT hr = media::BaseInputPin::EndFlush(); + NS_ENSURE_TRUE(SUCCEEDED(hr), hr); + + // Reset the EOS flag, so that if we're called after a seek we still work. + GetSampleSink()->Reset(); + + return S_OK; +} + +HRESULT +AudioSinkInputPin::EndOfStream(void) +{ + HRESULT hr = media::BaseInputPin::EndOfStream(); + if (FAILED(hr) || hr == S_FALSE) { + // Pin is stil flushing. + return hr; + } + GetSampleSink()->SetEOS(); + + return S_OK; +} + + +HRESULT +AudioSinkInputPin::NewSegment(REFERENCE_TIME tStart, + REFERENCE_TIME tStop, + double dRate) +{ + CriticalSectionAutoEnter lock(*mLock); + // Record the start time of the new segment, so that we can store the + // correct absolute timestamp in the "media time" each incoming sample. + mSegmentStartTime = tStart; + return S_OK; +} + +} // namespace mozilla + |