| /dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 21 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
22 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
27 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
28 BufferLevelFilter filter; in TEST()
55 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
56 BufferLevelFilter filter; in TEST()
98 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
99 BufferLevelFilter filter; in TEST()
131 TEST(BufferLevelFilter, TimeStretchedSamplesNegativeUnevenFrames) { in TEST() argument
132 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 17 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
21 void BufferLevelFilter::Reset() { in Reset()
26 void BufferLevelFilter::Update(size_t buffer_size_packets, in Update()
49 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) { in SetTargetBufferLevel()
61 int BufferLevelFilter::filtered_current_level() const { in filtered_current_level()
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| H A D | buffer_level_filter.h | 20 class BufferLevelFilter {
22 BufferLevelFilter();
23 virtual ~BufferLevelFilter() {} in ~BufferLevelFilter()
45 RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
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| /dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 21 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
22 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
27 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
28 BufferLevelFilter filter; in TEST()
55 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
56 BufferLevelFilter filter; in TEST()
98 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
99 BufferLevelFilter filter; in TEST()
131 TEST(BufferLevelFilter, TimeStretchedSamplesNegativeUnevenFrames) { in TEST() argument
132 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 17 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
21 void BufferLevelFilter::Reset() { in Reset()
26 void BufferLevelFilter::Update(size_t buffer_size_packets, in Update()
49 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) { in SetTargetBufferLevel()
61 int BufferLevelFilter::filtered_current_level() const { in filtered_current_level()
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| /dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 21 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
22 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
27 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
28 BufferLevelFilter filter; in TEST()
55 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
56 BufferLevelFilter filter; in TEST()
98 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
99 BufferLevelFilter filter; in TEST()
131 TEST(BufferLevelFilter, TimeStretchedSamplesNegativeUnevenFrames) { in TEST() argument
132 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 17 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
21 void BufferLevelFilter::Reset() { in Reset()
26 void BufferLevelFilter::Update(int buffer_size_packets, in Update()
49 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) { in SetTargetBufferLevel()
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| H A D | buffer_level_filter.h | 18 class BufferLevelFilter {
20 BufferLevelFilter();
21 virtual ~BufferLevelFilter() {} in ~BufferLevelFilter()
43 DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
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| /dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 21 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
22 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
27 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
28 BufferLevelFilter filter; in TEST()
55 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
56 BufferLevelFilter filter; in TEST()
98 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
99 BufferLevelFilter filter; in TEST()
131 TEST(BufferLevelFilter, TimeStretchedSamplesNegativeUnevenFrames) { in TEST() argument
132 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 17 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
21 void BufferLevelFilter::Reset() { in Reset()
26 void BufferLevelFilter::Update(size_t buffer_size_packets, in Update()
49 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) { in SetTargetBufferLevel()
61 int BufferLevelFilter::filtered_current_level() const { in filtered_current_level()
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| /dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 21 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
22 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
27 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
28 BufferLevelFilter filter; in TEST()
55 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
56 BufferLevelFilter filter; in TEST()
98 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
99 BufferLevelFilter filter; in TEST()
131 TEST(BufferLevelFilter, TimeStretchedSamplesNegativeUnevenFrames) { in TEST() argument
132 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 17 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
21 void BufferLevelFilter::Reset() { in Reset()
26 void BufferLevelFilter::Update(size_t buffer_size_packets, in Update()
49 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) { in SetTargetBufferLevel()
61 int BufferLevelFilter::filtered_current_level() const { in filtered_current_level()
|
| /dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/webrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 21 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
22 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
27 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
28 BufferLevelFilter filter; in TEST()
55 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
56 BufferLevelFilter filter; in TEST()
98 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
99 BufferLevelFilter filter; in TEST()
131 TEST(BufferLevelFilter, TimeStretchedSamplesNegativeUnevenFrames) { in TEST() argument
132 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 17 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
21 void BufferLevelFilter::Reset() { in Reset()
26 void BufferLevelFilter::Update(size_t buffer_size_packets, in Update()
49 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) { in SetTargetBufferLevel()
61 int BufferLevelFilter::filtered_current_level() const { in filtered_current_level()
|
| /dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 22 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
23 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
28 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
29 BufferLevelFilter filter; in TEST()
54 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
55 BufferLevelFilter filter; in TEST()
90 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
91 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 21 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
25 void BufferLevelFilter::Reset() { in Reset()
30 void BufferLevelFilter::Update(size_t buffer_size_samples, in Update()
48 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) { in SetTargetBufferLevel()
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| H A D | buffer_level_filter.h | 20 class BufferLevelFilter {
22 BufferLevelFilter();
23 virtual ~BufferLevelFilter() {} in ~BufferLevelFilter()
46 RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
|
| /dports/net-im/tg_owt/tg_owt-d578c76/src/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 22 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
23 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
28 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
29 BufferLevelFilter filter; in TEST()
54 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
55 BufferLevelFilter filter; in TEST()
90 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
91 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 21 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
25 void BufferLevelFilter::Reset() { in Reset()
30 void BufferLevelFilter::Update(size_t buffer_size_samples, in Update()
47 void BufferLevelFilter::SetFilteredBufferLevel(int buffer_size_samples) { in SetFilteredBufferLevel()
51 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level_ms) { in SetTargetBufferLevel()
|
| /dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 22 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
23 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
28 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
29 BufferLevelFilter filter; in TEST()
54 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
55 BufferLevelFilter filter; in TEST()
90 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
91 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 21 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
25 void BufferLevelFilter::Reset() { in Reset()
30 void BufferLevelFilter::Update(size_t buffer_size_samples, in Update()
48 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level_ms) { in SetTargetBufferLevel()
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| H A D | buffer_level_filter.h | 20 class BufferLevelFilter {
22 BufferLevelFilter();
23 virtual ~BufferLevelFilter() {} in ~BufferLevelFilter()
44 RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
|
| /dports/www/firefox/firefox-99.0/third_party/libwebrtc/modules/audio_coding/neteq/ |
| H A D | buffer_level_filter_unittest.cc | 22 TEST(BufferLevelFilter, CreateAndDestroy) { in TEST() argument
23 BufferLevelFilter* filter = new BufferLevelFilter(); in TEST()
28 TEST(BufferLevelFilter, ConvergenceTest) { in TEST() argument
29 BufferLevelFilter filter; in TEST()
54 TEST(BufferLevelFilter, FilterFactor) { in TEST() argument
55 BufferLevelFilter filter; in TEST()
90 TEST(BufferLevelFilter, TimeStretchedSamples) { in TEST() argument
91 BufferLevelFilter filter; in TEST()
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| H A D | buffer_level_filter.cc | 21 BufferLevelFilter::BufferLevelFilter() { in BufferLevelFilter() function in webrtc::BufferLevelFilter
25 void BufferLevelFilter::Reset() { in Reset()
30 void BufferLevelFilter::Update(size_t buffer_size_samples, in Update()
48 void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) { in SetTargetBufferLevel()
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| H A D | buffer_level_filter.h | 20 class BufferLevelFilter {
22 BufferLevelFilter();
23 virtual ~BufferLevelFilter() {} in ~BufferLevelFilter()
46 RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
|