gstreamer-audio-sys-0.23.5/.cargo_vcs_info.json0000644000000001610000000000100150250ustar { "git": { "sha1": "8eb8ab921bc627f854bac4e457521e7312403d0a" }, "path_in_vcs": "gstreamer-audio/sys" }gstreamer-audio-sys-0.23.5/CHANGELOG.md000064400000000000000000002533741046102023000154460ustar 00000000000000# Changelog All notable changes to this project will be documented in this file. The format is based on [Keep a Changelog](http://keepachangelog.com/en/1.0.0/) and this project adheres to [Semantic Versioning](http://semver.org/spec/v2.0.0.html), specifically the [variant used by Rust](http://doc.crates.io/manifest.html#the-version-field). ## [0.23.5] - 2025-02-17 ### Fixed - Properly validate `gst::IntRange::with_step()` step size - Fix `gst::Buffer` serde serialization - Forward gap events by default in `gst_utils::StreamProducer`. - Correctly account for alternate interlace mode in `gst_video::VideoMeta::add_full()`. - Return `Result`s instead of `bool`s in new `gst_play` API. ### Added - Support for `TracerImpl::USE_STRUCTURE_PARAMS` with GStreamer < 1.26. - Bindings for `gst_analytics::ODMtd`. - Bindings for `TopSurroundRight` and `TopSurroundLeft` audio channels. - Bindings for AV1 and H266 codec helpers API. - Bindings for `gst_audio::reorder_channels_with_reorder_map()`. ### Changed - Updated GStreamer gir files for latest 1.26 APIs. - Documentation links URL was updated. ## [0.23.4] - 2024-12-21 ### Fixed - `gst_video::VideoFrame::plane_data()` does not return a truncated buffer for the alpha plane in A420 and similar formats anymore. - `FnMut` closures are now correctly passed via a mutable reference to FFI code. - Order of arguments in `gst_video::VideoFormat::from_mask()` was corrected. ### Added - Bindings for 1.26 `gst_analytics::Tensor` API. - `gst::DebugCategory::as_ptr()` and `Hash` impl, `gst::DebugMessage::as_ptr()`. - Support for hex-dumping `&mut [u8]` in addition to `&[u8]`, `gst::Buffer`, etc. - Functions to work with meta `glib::Type`s. ### Changed - Updated GStreamer gir files for latest 1.26 APIs. ## [0.23.3] - 2024-11-01 ### Fixed - Bind `gst::Pad::proxy_query_caps()` to the correct C function. - Update `gst_utils::StreamProducer` appsrc latency upon appsink latency event. - Fix type of `gst_app::AppSinkBuilder::processing_deadline()`. ### Added - Various new `gst::Iterator` constructors for convenience. ### Changed - Updated GStreamer gir files for latest 1.26 APIs. ## [0.23.2] - 2024-09-28 ### Fixed - Lifetime of `gst::TagList::index()` return value is correctly bound to `&self` now. - Don't assume `gst::Structure` name / field names have `'static` lifetime. - Set pad probe data to `NULL` if `HANDLED` is returned and the item is an event, buffer or buffer list. - Don't unnecessarily add `#[link]` attribute to the `extern "C"` sections to allow linking against gstreamer-full and make static linking easier. ### Changed - Add `#[must_use]` to `gst_video::VideoTimeCode::add_interval()`. ### Added - Add API to take events/buffers from a `gst::PadProbeInfo`. - Add `gst::EventViewMut` and `gst::Event::view_mut()`, and a few setters for event fields. - Add `gst::MessageViewMut` and `gst::Message::view_mut()`, and a few setters for message fields. ## [0.23.1] - 2024-08-27 ### Fixed - Support `gst_utils::StreamProducer` API on platforms without 64 bit atomics. - Fix off-by-one in `gst::BufferList::remove()` range end handling. - Pass an empty tag array instead of NULL in `gst::CustomMeta::register_simple()`. - Fix various new clippy warnings. ### Added - Add getters for `gst::format::Percent`. ## [0.23.0] - 2024-07-11 ### Changed - Compatible with gtk-rs-core 0.20 / gtk4-rs 0.9. - Update GStreamer gir files to latest (upcoming) 1.26 APIs. - Minimum support Rust version is updated from 1.70 to 1.71.1. - Move `gst::Meta` tags into separate modules and improve API around them. - Improve `gst::Meta` transform functions and make the API more generic, and as part of that add support the video meta transform. - Pass an immutable instead of mutable output buffer reference to `gst_rtp::RtpHeaderExtension::write()` function. - Make `gst_net::PtpClock::new()` constructor fallible. - Change `gst_rtsp_server::RTSPToken` API to be consistent with `gst::Structure`, specifically add a builder. - Change `from_glib_ptr_borrow()` functions to work with references instead of raw pointers for improved safety. - Improve code generation when building with `panic=abort`. - Change `gst::BufferList` APIs to work with ranges instead of index+length. - Use various `usize` instead of `u32` for various indices in `gst::Buffer`, `gst::Caps`, `gst::Structure` and related APIs. - `gst::Clock` calibration-related API uses plain `u64` instead of `gst::ClockTime` for the clock rate. - `gst::debug!` and related macros use `obj = x` instead of `obj: x` for specifying the target object now. Similar for `imp` and `id`. The old syntax is still supported but deprecated. The new syntax works better with tooling, especially rustfmt. ### Added - Mutable access to the `gst_webrtc::WebRTCSessionDescription` fields. - `gst::StructureBuilder::field_if_some()` and the same for related builders to only set a value if `Some(x)` is provided. - `gst::StructureBuilder::field_from_iter()` and `field_if_not_empty()` for various builders. - `gst::PadBuilder` API for selecting an automatically generated name. - Adapter for the `log` crate around the GStreamer debug log system. This allows the macros from the `log` crate to be output via the GStreamer debug log system. - Bindings for the double click `gst_video::Navigation` event. - Bindings for `gst_pbutils` missing/install plugins API. - Setters for `gst_editing_services::FrameCompositionMeta`. - `ges::UriClipAsset::new()`. ## [0.22.6] - 2024-06-19 ### Fixed - When logging with an id and a formatted log message this would previously panic. - A couple of clippy warnings. ## [0.22.5] - 2024-05-23 ### Fixed - A couple of clippy warnings and compiler warnings about unused imports with latest rustc. - Memory leak in builder for the `SelectStreams` event. - Add parameter validity assertions to various `BufferList` and `Caps` APIs where these assertions were missing to avoid assertions in C. ### Added - `StreamProducer::set_forward_preroll()` API to configure if the preroll buffer should be directly forwarded or not yet. ### Changed - Remove nonsensical gstreamer-video test that fails with latest GStreamer main. - Update to itertools 0.13. ## [0.22.4] - 2024-04-08 ### Added - Implement `From` / `ToValue` for `gst_audio::AudioConverterConfig` and `gst_video::VideoConverterConfig`. ### Changed - Fixed various 1.77 clippy warnings. ## [0.22.3] - 2024-03-19 ### Changed - Change `ges::CompositionMeta` position fields to `f64`s in correspondence with the C API. - Change `gst_analytics::AnalyticsMtdRef::obj_type()` to an `Option` in correspondence with the C API. ### Added - `gst::Fraction::new_raw()` and `from_integer()` const constructors. ## [0.22.2] - 2024-02-26 ### Changed - Update GStreamer gir files and add more new 1.24 API. ### Fixed - Add `gst::Object` as parent class for various `gst_rtp` types. - Handle all already queued messages in `gst::BusStream` instead of just new messages. ### Added - Add `gst::CustomMeta::is_registered()`. ## [0.22.1] - 2024-02-13 ### Changed - Update GStreamer gir files and add more new 1.24 API. ### Fixed - Make `AnalyticsODLocation` struct fields public. - `MetaRefMut::upcast_mut()` returns a mutable reference now. ## [0.22.0] - 2024-02-08 ### Changed - Compatible with gtk-rs-core 0.19 / gtk4-rs 0.8. - Update GStreamer gir files to latest (upcoming) 1.24 APIs. - Various standalone functions were moved to separate modules or methods. - `gst::Rank` is not implemented as an enum but as a struct with associated constants now. - Optimized `gst::Buffer::from_slice()` and `Memory::from_slice()` implementations that have one heap allocation fewer. - Various `gst::Buffer` and `gst::Memory` functions take ranges now instead of offset/size parameters. ### Added - Bindings for `gst_gl::GLContext::thread_add()`, `GLFrameBuffer::draw_to_texture()`. - New `gst_gl::GLVideoFrame` type that replaces `gst_video::VideoFrame` for GL-specific API, and comes with mostly the same interface. - Basic gstreamer-tag bindings. - `gst::Buffer:dump()` and `dump_range()` together with the same API on `gst::Memory` for hex-dumping the whole buffer/memory content. - Implement `Clone` on `gst::MetaRef`. - Bindings for `gst::Buffer::map_range_readable()` and its writable variant. - Array-based accessor for `gst_video::VideoFrame` and `gst_audio::AudioBuffer` plane data. - Support for handling custom authentication in `gstreamer-rtsp-server`. - Accessors for various base class struct fields. - Owned buffer getter for `AudioBuffer` / `VideoFrame`. - `gst_rtp::RTPSourceMeta` bindings. - `gst::macos_main()` bindings. - gstreamer-analytics bindings. ### Fixed - API typo in owned `gst::ReferenceTimestampMeta` reference getter. - Allow variable expansion in `gst::loggable_error!` macro. - `gstreamer-gl-*` crates can build docs again on stable. ### Removed - `gst::Pad::caps()` property getter. Use `current_caps()` instead which does the same thing. - Various deprecated APIs that were deprecated in previous releases. - Getter for a mutable buffer reference from `AudioBuffer` / `VideoFrame` as that allowed invalidating the buffer map. ### Fixed ## [0.21.3] - 2023-12-18 ### Added - Update GStreamer gir files to latest (upcoming) 1.24 APIs. - Add an example for writing subclasses with virtual methods. - Add `gst::ClockTime::absdiff()` and same for similar types. ### Fixed - In `Play` example, set bus to flushing before dropping `Play` instance. - Add missing `docsrs` configuration for correct documentation generation. - Make `gst_pbutils::element_properties` module public. - Add missing `gst_audio::AudioFilterImpl::parent_allowed_caps()`. - Fix assertions in `gst::Memory` copy/share/resize functions. ### Changed - Update to itertool 0.12, pretty-hex 0.4. ## [0.21.2] - 2023-11-11 ### Changed - Update GStreamer gir files to latest (upcoming) 1.24 APIs. - Update to latest gir code generator from the gtk-rs 0.18 branch. ### Fixed - Big endian video format order is correct again. - `gst::MetaRef::has_tags()` and `tags()` API actually works and works based on the tags of the concrete meta instance. - `gst::MetaRef::tags()` returns strings with arbitrary lifetimes now because they're statically stored anyway. - Fix another potential deadlock in `gst_utils::StreamProducer` when sending force-keyunit events. ### Added - Bindings for `gst_video::VBIEncoder` and `VBIParser`. - Accessors for the different `gst::PadProbeData` types on `PadProbeInfo`. - `Default` impl for `gst::AllocationParams`. - `From` / `TryFrom` implementations between formatted types (e.g. `gst::Bytes`) and `usize`. - `gst::MetaRef::copy()` to copy metas from one buffer to another. - `gst::ElementImpl::catch_panic_future()` to wrap a `Future` in such a way that panics are converted to GStreamer error messages and the element is marked as unusable. - `gst_gl::GLDisplay::handle()` to get a raw display handle. ## [0.21.1] - 2023-10-04 ### Changed - Update GStreamer gir files to latest (upcoming) 1.24 APIs. ### Fixed - Use correct media links in the tutorials code. - Fix a couple of new 1.72/1.73 clippy warnings. - Fix description of gstreamer-validate crate. - Copyright/license files for the gstreamer-gl were added. - Ordering of raw video formats follows the rules of latest libgstvideo now. - Fix potential deadlock in `gst_utils::StreamProducer` when sending force-keyunit events. ### Added - `max-time` / `max-bytes` setters to `gst_app::AppSink` builder. - `gst::CustomMeta::register_simple()`. ## [0.21.0] - 2023-08-08 ### Changed - Minimum supported Rust version is updated to 1.70.0. - Compatible with gtk-rs-core 0.18. - `gst::Bin::add_many()`, `remove_many()` and `gst::Element::link_many()`, `unlink_many()` are more generic now. - `gst_base::Aggregator::src_pad()` returns an `AggregatorPad`. - `gst::Bus::add_watch()` now returns a guard value that automatically removes the watch when it goes out of scope. - `gst::Bin`, `Pipeline` and `Pad` constructors don't take the optional name parameter anymore but it can instead be provided via the builder API. - `gst::Pad` and `GhostPad` builders inherit name from the pad template (or target) if possible and no other name is provided explicitly. - The preroll samples and selected sticky events are forwarded to `StreamProducer` consumers. ### Added - Support for the upcoming GStreamer 1.24 APIs. - Support for inline variable names in format strings for error/warning/info messages. - Methods for converting between floating point seconds and `gst::ClockTime`. - Various additions to the gst-validate bindings. - `Display` implementations for error/warning/info messages. - More useful `Debug` implementations for messages, events and queries and `gst_pbutils::DiscovererInfo` related structs. - API for listing/checking `gst::Meta` tags. ## [0.20.7] - 2023-07-05 ### Fixed - Fix `wait-for-eos` property name string in `appsink`. - Fix various memory leaks in `BaseTransform` subclassing bindings. - Mark some GES APIs as `Send+Sync`. ### Added - Implement `DiscovererInfo::debug()` and on related structs. + Add subclassing bindings for `GESFormatter`. ## [0.20.6] - 2023-06-06 ### Added - Getter for the `gst_rtsp_server::RTSPContext` URI field. ### Fixed - `gst_pbutils::DiscovererStreamInfo::stream_id()` can return `NULL`. This is mapped to the empty string for this release to keep backwards compatibility. - `gst_pbutils::DiscovererStreamInfo` iterator methods can be called on any subclass directly now without casting. - Debug logs use the actual function name against instead of the name of a closure generated by the log macros. ### Changed - Minor performance improvements to debug logging. ## [0.20.5] - 2023-04-22 ### Added - `glib::HasParamSpec` impl for miniobjects to allow using them with the properties derive macro. - `Default` impl for `gst_player::Player`. ## [0.20.4] - 2023-04-07 ### Fixed - Work around `gst_webrtc::WebRTCICE::add_candidate()` API breakage in 1.24. ### Changed - Reduce size of `gst_audio::AudioBuffer` and `gst_video::VideoFrame` by a factor of two by not storing an unnecessary copy of the audio/video info. ## [0.20.3] - 2023-03-14 ### Fixed - `gst::ParamSpecArray` uses the correct `glib::Type` now. - Work around accidental ABI breakage in 1.18 gst-rtsp-server `GstRTSPClient`. ### Added - Document `gst_utils::StreamProducer::forward_eos()` default value. ## [0.20.2] - 2023-02-21 ### Added - `glib::HasParamSpec` impl for `gst::ClockTime` - `Default` impl for `gst_play::Play` - Constructors for non-raw `gst_audio::AudioCapsBuilder` / `gst_video::VideoCapsBuilder` ## [0.20.1] - 2023-02-13 ### Fixed - Fix memory leaks when converting a `gst_audio::AudioBuffer` or `gst_video::VideoFrame` to a `gst::Buffer` or FFI type. ## [0.20.0] - 2023-02-10 ### Fixed - Make `gst_gL::GLDisplay::create_context()` `other_context` parameter optional. - Make allocation query caps optional. ### Added - Conversions between `gst::Signed` and `T` and signed integer types. - Bindings for the object lock via `gst::Object::lock()`. - Various `FromIterator`, `Extend` and `From` impls for creating `Caps`, `Structure`, `Buffer`, `BufferList`, `CapsFeatures` and other types. - `PartialEq` impls between owned/borrowed miniobjects/structures. - API for appending items to `gst::Array` and `gst::List`. ### Changed - Compatible with the 0.17 gtk-rs release. - Updated minimum supported Rust version to 1.64. - Require GStreamer 1.22.0 or newer when enabling the `v1_22` feature. - Require the object lock to be taken for various `gst_gl::GLDisplay` methods. - Renamed `gst::TagSetter::add()` to `add_tags()` to avoid name conflict with `Bin::add()`. - Mark various un-extendable enums as exhaustive. - Make use of `glib::GStr` and related API in caps, structure, tags and logging API to reduce temporary string allocations. - Various code optimizations to reduce generated code size and allow more optimal code to be generated. - Reduce size of various types, including reduction of `gst_audio::AudioInfo` from 832 to 320 bytes. - Use actual function name instead of module name in log output. - Change `gst_utils::StreamProducer` API to forward buffers by default and allow temporarily discarding via new `set_discard()` function. ## [0.19.8] - 2023-02-09 ### Changed - Update GStreamer .gir files to 1.22.0 release. ### Fixed - Marked `gst::MessageType` as non-exhaustive. ### Added - Added bindings for `gst::Message::structure_mut()`. - Added subclassing support for `gst_allocators::FdAllocator` and `DmabufAllocator`. ## [0.19.7] - 2023-01-19 ### Fixed - Work around the possibility that the caps in the allocation query can be `NULL` by returning any caps for now. This will be handled properly with a minimal API change in the 0.20 release. ## [0.19.6] - 2023-01-18 ### Fixed - The `AppSrc` and `AppSink` builders now assert that GStreamer is initialized before creating an instance. ## [0.19.5] - 2022-12-27 ### Fixed - Clear video frame values when mapping as GL texture to avoid out of bounds reads when trying to access the GL texture as raw video frame. - Allow returning `Handled` from `BufferList` pad probes. ### Changed - Update GStreamer .gir files to latest 1.21 git. ## [0.19.4] - 2022-12-16 ### Added - Subclassing bindings for `gst_audio::AudioFilter`. ### Fixed - Various new clippy warnings. ### Changed - Update GStreamer .gir files to 1.21.3 release. ## [0.19.3] - 2022-11-28 ### Added - `FromIterator` and `Extend` for `Caps`. - `PartialEq` impls between owned/borrowed miniobjects/structures. ### Fixed - Sticky event ordering for instant-rate-change. ### Changed - Updated GStreamer .gir files to post 1.22.2 release. ## [0.19.2] - 2022-11-13 ### Added - Subclassing support for `gst::Allocator`. - `gst_gl::GLBaseMemory::context()` to retrieve the GL context used by the memory. ### Changed - Updated GStreamer .gir files to 1.22.2 release. ### Fixed - `gst::Allocator::register()` does not cause use-after free with GStreamer < 1.20.5 anymore. - Don't generate version constants in `gstreamer-editing-services-sys` as they are useless and change with every update. ### Changed - Fixed various new clippy warnings. ## [0.19.1] - 2022-10-24 ### Changed - Make it possible to use objects not just as reference in the logging macros. ## [0.19.0] - 2022-10-22 ### Added - Builders for element construction. `gst::ElementFactory::make()` returns a builder now that allows to easily set the name or any other property at construction time. The old API is available as `make_with_name()`. - Builders for `Bin` and `Pipeline` as well as a `Default` trait implementation to simplify object construction. - Builders for `appsrc` and `appsink`, which allow type-safe construction of both elements while also allowing to easily set all their properties at construction time. - Builders for the GStreamer-specific fraction/array param/property specs. - Infrastructure for casting between `gst::Memory` subtypes/supertypes, and make use of it for GL memory. - Bindings for the `gstreamer-allocator` library with support for file descriptor-based and DMABUF memory. - Complete bindings for `gst_video` `Navigation` events. - Constructors for error/warning/info messages with a pre-built `glib::Error`. This also leads to some minor simplification of the existing API. - Accessors for static pads of various base classes for making accessing them cheaper and less error-prone than getting them by name. - Builder for pad templates. - Static PTP clock API for statistics, initialization and deinitialization. - New `gstreamer-utils` crate that currently contains only a `StreamProducer` API. This allows building 1:N bridges between live pipelines via `appsink` / `appsrc` elements. - Bindings for the new `gstreamer-play` library that was added in 1.20. - `gst::Caps::new_empty_simple()` to create caps without fields and just a name. - `gst_audio::AudioCapsBuilder` and `gst_video::VideoCapsBuilder` for building (possibly) unfixed raw audio/videos caps with typed setters for the various fields. This makes it impossible to mix up types and e.g. use an `u32` instead of an `i32` for the width of video caps. - `gst::Buffer::ptr_eq()` to compare buffers by pointer instead of doing a deep comparison, and also `ptr_eq()` on all other miniobject types. - Accessors for `gst_webrtc::WebRTCICECandidateStats` fields. - Bindings for the `gstreamer-validate` API. - Subclassing bindings for `gst_audio::AudioVisualizer` base class for easily writing audio visualization elements. - `gst_pbutils::EncodingProfile` API for element properties. - Support for returning buffer lists from `BaseSrc` / `PushSrc` subclasses. - Support for implementing `gst::Bin::do_latency()`. - Minimal bindings for the `gstreamer-mpegts` library. ### Fixed - Signature for `gst_base::Aggregator::connect_samples_selected()` to remove unnecessary generic parameter and make it straightforward to use. - Various APIs had optional parameters/return types corrected to match the C API more closely. - Logging does not evaluate its arguments anymore if the debug category is not enabled or below the current threshold. - Registering custom metas is now possible without transform function. - `gst::subclass::ElementImpl::request_new_pad()` signature uses a `&str` instead of an owned `String` now. ### Removed - `fragile` dependency and instead use the same functionality from `glib`. - `gst_audio::AudioAggregator` `ignore_inactive_pads` property, which was duplicated from the `Aggregator` base class. ### Changed - Compatible with the 0.16 gtk-rs release. - Updated minimum supported GStreamer version from 1.8 to 1.14. - Updated to the latest GStreamer 1.22 APIs while still supporting up to GStreamer 1.14. Any new 1.22 APIs might still change until the stable 1.22 release. - Updated minimum supported Rust version to 1.63. - In `EventView` / `QueryView`, getters that return references now return references that can outlive the view and are only bound by the lifetime of the corresponding event/query. - In addition `Query`, `Event` and `Message` views are implemented more consistently now, which makes them easier to use and as a side effect allows to pass e.g. more strongly typed queries to functions that only accept a single query type. - Various improvements to `gst::ClockTime`, `gst::format::Bytes`, `gst::format::Signed` and related types and their usage in the API, which should make its use from applications easier and less error-prone. Check the `gst::format` module-level documentation for details. - `gst::StreamsSelected` event builder takes the selected streams as iterator instead of slice. - For consistency with other macros the `gst` prefix of the logging macros was also removed. - Various iterator implementations were added and the existing ones were optimized by implementing more specialized traits and custom implementations for a couple of iterator functions. - GStreamer initialization safety checks were optimized. - `gst::Bus::post()` takes ownership of the passed messages like the C API. - Better and easier to read `Debug` impls for `Caps`, `TagList`, `Structure` and `Promise`. - `ser_de` feature was renamed to `serde`. - `gst::Tracer` implementations get result enums passed as `Result`s now instead of single enums. - `gst::Pad`, `ProxyPad`, `GhostPad` default functions are all associated functions instead of methods now to avoid conflicts between multiple types with the same method. - `Pad` tasks are catching panics from the task function and if the parent of the pad is an element then the panic is converted into an error message and the task is simply stopped. Otherwise the panic is rethrown. ## [0.18.8] - 2022-04-26 ### Added - Bindings for `RTPBasePayload` and `RTPBaseDepayload`. - Accessors for `RTPBuffer` buffer. - Bindings for `RTPBuffer` length calculation API. - More complete `gst::Task` bindings. ### Fixed - Export `gst::subclass::TaskPoolFunction`. ## [0.18.7] - 2022-04-04 ### Added - Bindings for `VideoAggregator` and the `VideoAggregatorPad`s. - Bindings for `AudioAggregator` and the `AudioAggregatorPad`s. - Bindings for `TaskPool`. - Various helper functions for `VideoFormatInfo`, `VideoInfo` and `VideoFrame`. ## [0.18.6] - 2022-03-08 ### Fixed - Require `Send` and not `Sync` for the values of an `gst::Array` / `gst::List`. ### Changed - Simplify and speed up log message string construction ## [0.18.5] - 2022-02-20 ### Changed - Require GStreamer 1.20.0 at least when building with `v1_20`. Earlier versions were already going to fail due to API mismatches before. ### Added - `gst::BufferPool` subclassing support. - `Debug` impl for `gst::MiniObject`. - `gst_rtsp_server::RTSPOnvifServer` and related API, including subclassing support. ### Fixed - Handle empty slices correctly at the FFI layer. - `MiniObjectRef::downcast_ref()` and similar functions return the correct type now. While this is an API change, the previous API would've never worked. ## [0.18.4] - 2022-02-04 ### Changed - Update gir files to GStreamer 1.20.0 release. ### Added - `gst_video::VideoCodecFrame::input_buffer_owned()` for getting an owned reference. ### Fixed - All documentation links in the `README.md`s are correct again. ## [0.18.3] - 2022-01-31 ### Added - `Default` implementation for `gst_video::VideoOverlayComposition` when targeting GStreamer 1.20. - `gst_video::VideoOverlayComposition::add_rectangle()` in addition to the addition of all rectangles via an iterator during construction. - Subclassing support for `gst_rtp::RTPHeaderExtension`. - `gst_webrtc::WebRTCError` for programmatically handling WebRTC errors. ### Fixed - `gst_rtp::RTPHeaderExtension` has `gst::Element` set as parent class now. - Global functions are re-exported from the `gst_rtp` crate root. ### Changed - GIO-style async operations in GES no longer need `Send`-able closures. ### Removed - `fragile` is no longer a dependency and instead the corresponding GLib API is used. ## [0.18.2] - 2022-01-24 ### Added - `glib::FromValue` for mini object references. - Bindings for `gst::DebugCategory::get_line()`. ## [0.18.1] - 2022-01-18 ### Fixed - `Message::view()` also handles the redirect message now. - `Message` and `Query` view variants that return references now borrow again from the underlying query and not the view enum, allowing to use them in a wider scope. ### Changed - All miniobjects, `VideoTimeCode`, `Structure` and `CapsFeatures` are marked as `#[repr(transparent)]` now to ensure that their memory representation is exactly the underlying raw pointer. ## [0.18.0] - 2022-01-16 ### Added - `gst_rtp::RtpHeaderExtension::read()` and `write()`. - `gst::ElementMetadata` has a `const` constructor now. - `gst_rtp::RtpBuffer` API works on buffer references instead of plain buffers for statically enforcing writability and usage in more places. - `gst_video::VideoCodecAlphaMeta` and `gst::CustomMeta`. - `gst::MiniObject` for generically passing around mini objects instead of their concrete types. - `gst_app::AppSink` `new-event` callback and `pull_object()` function. - `gst_pbutils::PbUtilsCapsDescriptionFlags` and `pb_utils_get_caps_description_flags()`. - `gst_rtp::RtpBuffer::remove_extension_data()`. - `gst_video::VideoDecoder` subframe API. - `gst_webrtc::WebRTCSCTPTransport`. - `gst::ElementFactory` `create_with_properties()` / `make_with_properties()`. - `gst_video::VideoContentLightLevel` and `VideoMasteringDisplayInfo` for HDR signalling. - Lots of missing `GES` API. - `gst::AllocationParams` and support in the allocation query. - `propose_allocation()` and `decide_allocation()` support in the various base classes. - `Iterator` implementation for `gst_video::VideoOverlayComposition`. - `Extend`, `IntoIterator` and `FromIterator` implementations for `Buffer`, `Caps`, `BufferList`, `CapsFeatures`, `StreamCollection` and `Structure` for more natural Rust APIs. - `instant-rate-change` events/messages bindings. - Support for arithmetic operations on `Option` and related types. - `gst_video::ColorBalance`. - `gst::MetaFlags`. - `gst_base::Aggregator::set_position()`. - Convenience getters for `gst::ElementFactory` and `gst::DeviceProviderFactory` metadata. - `gst_rtp::RtpBuffer::set_padding()`, `get_padding()` and `payload_mut()`. - `#[must_use]` to many types and functions. - `gst::Event`, `gst::Message` and `gst::Structure` `has_name()`. - `gst_video::Navigation` subclassing support and API improvements. - `gst::Structure` and `gst::Caps` `foreach()`, `map_in_place()` and `filter_map_in_place()`. - `gst_gl::GLBufferPool` and various GL constants and functions. - `gst_pbutils` codec utils APIs. ### Fixed - `gst_base::BaseTransform::prepare_output_buffer()` correctly reflects buffer writability. ### Changed - Compatible with the 0.15 gtk-rs release. - Updated to the latest GStreamer 1.20 APIs while still supporting up to GStreamer 1.8. Any new 1.20 APIs might still change until the stable 1.20 release. - Update all code to the Rust 2021 edition. This causes no user-facing changes. - `gst::Sample::segment()` returns a reference instead of a copy of the segment. - `gst::Object::set_property_from_str()` returns a `Result` now instead of silently failing like the C version. - Allow handling passed in buffers in `gst_base::PushSrc::create`. - Allow passing in `None` in `gst_player::Player::set_uri()`. - Use `[[f32; 4]; 4]` instead of `[f32; 16]` for affine transformation matrix. - `gst::Pad::sticky_event()` statically gets the event of the requested type instead of requiring to match on it afterwards. - Clean up `gst_pbutils` `EncodingProfile` API to be harder to misuse and less confusing. - Various `gst::Array`, `gst::List`, `gst::IntRange` and `gst::Fraction` API improvements that should reduce some friction. - Directly generate `NUL`-terminated C strings in debug log API instead of having multiple allocations per message. - Various functions return `glib::SList` and `glib::List` now to avoid copying into a `Vec` if only iteration is needed. - `gst::ChildProxy` API is more consistent with object property API. - Improved `gst::Buffer::foreach()`, `gst::Pad::sticky_events_foreach()` and `gst::BufferList::foreach()` APIs. - Don't post error messages from `propose_allocation()` and `decide_allocation()`. ## [0.17.4] - 2021-09-13 ### Added - Add constructor for device provider metadata. ## [0.17.3] - 2021-08-23 ### Fixed - `gst::Value::deserialize()` takes the target type as parameter now. This is technically an API change but the function would've never worked previously. ### Added - The release date-time parameter to `gst::plugin_define!` is optional now like in the C version of the macro. - Bindings to `gst::Tracer` and `gst::TracerFactory` for allowing to implement custom tracers in Rust. - Bindings for the new `gst::Value::deserialize_with_psec()` function from GStreamer 1.20. - serde `Serialize`/`Deserialize` impls for `gst::PadDirection`, `gst::PadPresence`, `gst::URIType` and `gst::Rank`. ## [0.17.2] - 2021-08-05 ### Fixed - Various new clippy warnings. - Compilation of `gstreamer-audio` on big-endian platforms. ### Added - Support for 1.20 `Gap` event `GapFlags`. - Support for 1.20 `Structure::serialize()` / `Caps::serialize()`. ## [0.17.1] - 2021-07-13 ### Fixed - Store 1.19 as plugin version when building plugins with `v1_20`. Otherwise plugins fail to load with GStreamer versions below 1.20.0. - Fix documentation for `gst::Element::request_pad_simple()` to actually show up. ## [0.17.0] - 2021-06-28 ### Fixed - Use `#[repr(transparent)]` where it is more correct and remove unneeded `#[repr(C)]` annotations. - Don't provide direct access to the logged object in logging functions as the object might currently be finalized and might be unsafe to access. - Moved X11/EGL/Wayland-specific GL APIs into their own crates instead of having them inside gstreamer-gl and behind feature flags. This simplifies conditional usage of them in applications. - Various nullability issues: parameters and return values that should've been or shouldn't have been nullable were fixed. - Print source object correctly in `gst::Message` `Debug` impl. - `gst_rtsp_server::RTSPServer::attach()` is fallible. - `gst::ElementFactoryListType` is a proper bitflags type now instead of generic `u64`. - `gst::PluginFeature::load()` returns the same type as the one passed in. - Value returned by `gst::PromiseFuture` can no longer be freed while still in scope. - Only assign to `GError**`s in subclassing code if they're not `NULL`. ### Added - Bindings for the GStreamer Controller library and the corresponding core API. - Subclassing support for `gst_player::PlayerVideoRenderer`. - `gst::PARAM_FLAG_CONTROLLABLE` and related bindings. - `gst_video::VideoOrientation` and `VideoOrientationMethod` bindings. - Support for removing pad probes from inside the pad probe callback. - `gst_check::Harness::pull_until_eos()` bindings. - `ges::TransitionClip` and `OperationClip`. - Bindings for `gst_gl::GLMemory` and related APIs. - Subclassing support for `gst_gl::GLFilter` and `gst_gl::BaseSrc`. - `gst::TagList::remove()`. - `gst::CapsFeatures` and `gst::Structure` API based on `glib::Quark`s instead of strings. - Subclassing support for `gst_video::VideoFilter`. - Bindings for various new 1.20 APIs: `gst_app::LeakyType`, `gst_video::VideoDecoderRequestSyncPointFlags`, `gst_rtp::RTPHeaderExtension`, `gst_audio::AudioLevelMeta`, `gst_webrtc::WebRTCKind` and various other new flags/enum types. - Subclassing support for `gst_rtsp_server::RTSPMountPoints`. ### Removed - Deprecated APIs in 0.16. - Don't declare that `gst_app::AppSink` and `AppSrc` inherit from `gst_base::BaseSink` and `BaseSrc` to avoid exposing API that is meant for subclasses to applications. - `gst_app::AppSrc` and `AppSink` signals that are also covered by the callbacks. The callbacks are more flexible and have lower overhead. - Duplicated getters/setters for `gst_base::BaseSink` and `BaseTransform` properties. ### Changed - Compatible with the 0.14 gtk-rs release. - Updated to the new GStreamer 1.20 APIs while still supporting up to GStreamer 1.8. Any new 1.20 APIs might still change until the stable 1.20 release. - FFI and safe high-level bindings are in the same repository now and use the same version numbers. - The .gir files are shared with gtk-rs and the GStreamer-specific ones are in a separate git submodule. - Update all code to the Rust 2018 edition. As part of this, most macros lost their `gst_` prefix. - Re-export dependency crates from the different preludes. - Getter functions don't have a `get_` prefix anymore and GObject property accessors don't include the `_property_` part in the middle of their function names anymore. Applications developers should use [`fix-getters-calls`](https://crates.io/crates/fix-getters-calls) to ease migration of their applications. Use [`fix-getters-def`](https://crates.io/crates/fix-getters-def) if you also want your `get` functions definition to comply with the API standards applied in this release. - Lots of changes to the subclassing API. Check the various elements in [gst-plugins-rs](https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs) for examples. - Major improvements to the documentation infrastructure and generated documentation. - `gst::ClockID` bindings are refactored to use different types for single-shot and periodic clock ids, which makes misuse harder. - `gst::ProxyPad` extension trait uses trait functions instead of associated functions now for usability reasons. - Use `Result` for overriding flow returns from pad probes. - `gst_video::VideoInfo::align()` returns a `Result` instead of a `bool`. - Use actual error types instead of `()` in `gst_sdp` APIs. - `Display` impl for `gst::ClockTime` provides better human-readable strings. - `gst::Element::link_filtered()` and `link_pads_filtered()` takes a non-optional caps now. That's easier to use and for not providing caps the non-filtered variants of the functions exist. - Replace various manual bindings with auto-generated ones. - `gst::Element::get_request_pad()` is replaced by `request_pad_simple()` as a simpler version of `request_pad()` and in accordance with the deprecation in GStreamer 1.20. - `gst::ClockTime` and APIs working on it were changed to make possibility of using `GST_CLOCK_TIME_NONE` expressed in the type system. `Option` can be `None` while `gst::ClockTime` is always a valid time. ## [0.16.7] - 2021-02-13 ### Fixed - Usage of the logging system with a GStreamer library with the logging system compiled out does not crash any longer. - Double-free in `gst_video::VideoTimeCode` API when converting between validated and unvalidated timecodes. ### Added - `gst::Element::get_current_state()` and `get_pending_state()` convenience APIs. - `gst_audio::AudioConverterConfig` for setting the configuration on e.g. the `audiomixer` element. The low-level `AudioConverter` API is still not included in the bindings. ## [0.16.6] - 2020-12-20 ### Fixed - `VideoTimeCodeInterval`'s `Ord` and `PartialEq` implementations compare against the correct fields now. - `SDPMessage::medias_mut()` iterator does not crash any longer. ### Added - `PartialEq` and `Eq` implementations on `VideoAlignment`. - Alignment API for `VideoMeta` and `get_plane_height()` / `get_plane_size()`. - `VideoInfo::align_full()`. ## [0.16.5] - 2020-11-23 ### Fixed - Make sure to use `$crate` in more macros to allow them to work without anything special in scope already. - Update documentation location. - Don't panic if C code stores invalid seqnums in events and the seqnum is used directly or via the `Display` impl. - Fix docs build for some crates on docs.rs. - Fix `Debug` impl for `gst_video::VideoTimeCode` to print the correct type name. - Fix plugin version to be 1.18 instead of 1.17 when compiling a plugin with `v1_18`. ### Added - Event handling support in pad probes, that is returning `PadProbeReturn::Handled` for events. - `EventRef::get_structure_mut()` getter that allows changing the events' structures. ### Changed - Remove unnecessary `PhantomData` markers and use `repr(transparent)` instead of `repr(C)` where it is more correct. ## [0.16.4] - 2020-10-09 ### Fixed - Correctly implement `ExactSizeIterator` on the `AudioFormat` and `VideoFormat` iterators. Previously they returned the overall size instead of the remaining size, and they didn't implement `Iterator::size_hint()`. - Don't implement `ExactSizeIterator` on the buffer `gst::Meta` iterator. The overall length is not known easily and the implementation would've simply panicked in the past. ### Added - `gst::ClockID::wait_async_stream()` for async integration for clock waiting. - `From` / `TryFrom` impls for converting between `gst::ClockTime` and `std::time::Duration`. ## [0.16.3] - 2020-09-08 ### Fixed - Reset vfuncs if calling `BaseTransformClass::configure()` multiple times. - Fix `gst::debug_remove_default_log_function()` to actually remove the default log function. ### Added - Some more new APIs added in 1.18. - API for getting an owned buffer from a readable `gst_video::VideoFrame` / `VideoFrameRef`. ### Changed - Updated bindings to 1.18.0. This stabilized GStreamer 1.18 support and any API behind the "v1_18" feature is considered stable now. - Factor out some common code from `gst::Pad::ProbeInfo` code. This reduces the code generated for each pad probe considerably. - Update paste dependency to 1.0 and pretty-hex to 0.2. ## [0.16.2] - 2020-07-27 ### Fixed - Use correct pointer for the plane data in `gst_audio::AudioBuffer`. ### Added - Add `gst::GhostPad` convenience constructors that take a target pad, similar to the ones that existed in 0.15 and before. - Add `gst::parse_bin_from_description_with_name` that allows setting a name for the created bin without having to use unsafe code in application code. ## [0.16.1] - 2020-07-10 ### Fixed - Allow calling `gst::DebugCategory::new()` before `gst::init()` again. ## [0.16.0] - 2020-07-06 ### Added - Updated bindings to 1.17.2, adding experimental 1.18 support. This can be opted-in via the "v1_18" feature flag but there might still be API changes in the newly added API. - `gst::MemoryRef::dump()` for dumping contents of a memory. - `gst::Bus::stream()` instead of a custom constructor on the `BusStream`. - Use more accurate types for `Seqnum`, `GroupId` and `MetaSeqnum`. These are now proper wrapper types instead of plain integers, which makes misuse harder. - Provide `TryFrom` impls for conversion between `glib::DateTime` and `gst::DateTime`. - Add `get_allocator()` functions to `gst_base::{Aggregator, BaseTransform, BaseSrc}`, and allow overriding `BaseSrc::alloc()`. - Add subclassing bindings for `gst_base::PushSrc`. - Add new `gst::BufferCursor` API that allows to handle a buffer as `Read`, `Write` and `Seek` and accesses the underlying memories of the buffer individually without mapping them all together. - Add `gst::Plugin::get_plugin_name()`. - Support for `gst_video::VideoAFDMeta` and `VideoBarMeta`. - API for getting all / iterating over all `gst_audio::AudioFormat` and `gst_video::VideoFormat`. - Bindings and subclassing bindings for `gst_video::VideoSink`. - `gst::Pad` can be constructed via the builder pattern and `gst::PadBuilder` now, which allows to safely set the pad functions and various other fields during construction. The `PadBuilder` works on any `gst::Pad` subclass and also has special support for `GhostPad`s by allowing to set pad functions of the proxy pad. - `gst::Message`, `gst::Event` and `gst::Query` type constructors are now on the specific target type instead of various `new_XXX()` functions on the basic type. E.g. `gst::message::Eos::new()`. - Support for overriding `gst_audio::AudioSrc/Sink::reset()`. - Support for overriding `gst_base::BaseParse::stop()`. - Support for overriding `gst::Element::post_message()`. - Added bindings for `gst::BufferList::foreach()` and `foreach_mut()`. - Added bindings for `gst::Buffer::foreach_meta()` and `foreach_meta_mut()`. ### Fixed - Allow using any `glib::Object` as target object for logging instead of just `gst::Object`. - Remove restriction API from `gst_pbutils::EncodingContainerProfile`. They are supposed to be used only with the other encoding profiles. - Return `&'static str` for various `gst::StructureRef` functions where the string is backed by a `glib::Quark`. - Fix various `gst::DateTime` functions to actually return `Option`s. - Add support for filling in a buffer passed to the `gst::Pad` getrange function, allow passing one in into `get_range()` and `pull_range()` and provide the corresponding API on `gst_base::BaseSrc` too. - Allocator in audio/video `Decoder` base classes is optional and can return `None`. - `gst_video::ValidVideoTimeCode::add_interval()` always returns a valid timecode again. - Allow resolving a `gst::Promise` with `None` and also handle that correctly in the callback. This is allowed by the API. - Allow calling various debugging related functions before `gst::init()`. - Various enum/function versions were fixed to only show up if the corresponding version feature is enabled. - `gst::Pad` function setters are marked unsafe now as changing the functions is not thread-safe. - Remove `gst::Object::set_name()` as changing the name after construction generally causes problems and is potentially unsafe. - Remove `gst::Pad::set_pad_template()` as changing the pad template after construction is generally unsafe. - `gst::Pad::stream_lock()` borrows the pad now instead of taking a new reference. - Unimplemented `Jitter` and `Buffer` queries were removed from the bindings. These are not implemented in C and only have a type registered. - Various `LAST`, `NONE` variants of enums and flags were removed as these only make sense in C. - Call the parent impl of various vfuncs that were omitted before to not require further subclasses of them to implement them but automatically call the parent ones. ### Changed - Use `NonZeroU64/U32` for various ID types to allow further optimizations. - Use `thiserror` crate for deriving error types. - Switch from `lazy_static` to `once_cell`. - Change various miniobject functions like `gst::Caps::append()` from taking the object by value to modifying it internally. This makes them easier to use and only applies to functions that are defined on the non-reference type and take ownership of the values passed in. - Use `mem::ManuallyDrop` instead of `mem::forget()` everywhere. - Replace most `mem::transmute()` calls with safer alternatives. - `gst:StreamCollection` API was changed to the builder pattern for construction as the collection must not be changed after construction. - `gst::ProxyPad` default functions are plain functions on `ProxyPad` now instead of trait functions to allow easier usage of them. - Use proper error types in various `TryFrom` impls. - `gst_video::VideoMeta::add()` returns a `Result` now instead of panicking. - Various constructors were renamed from `new_with_XXX()` and `new_from_XXX()` to the more idiomatic `with_XXX()` and `from_XXX()`. - Miniobject bindings are simplified now and there is no `gst::GstRc` type anymore, instead everything is directly implemented on the concrete types. As part of this the `gst::MiniObject` trait was also removed as it was unneeded now. ## [0.15.7] - 2020-06-08 ### Fixed - Allow multiple filter types per process with `gst::Iterator::filter()`. - Check that `VideoInfo` is valid when creating a `VideoFrame`. - Don't potentially dereference a `NULL` pointer when getting the format from an invalid `VideoInfo` or `AudioInfo`. - Don't unmap borrowed `VideoFrameRef`s. ### Added - `gst::ProtectionMeta`, `gst_video::VideoAffineTransformationMeta`, `VideoCropMeta` and `VideoRegionOfInterestMeta` bindings. - Various new `gst_rtp::RTPBuffer` methods. - `gst_audio::audio_buffer_truncate()`, `AudioMeta` and `AudioBuffer` bindings. ## [0.15.6] - 2020-05-28 ### Fixed - Assert that the data passed to `VideoCaptionMeta::add()` is not empty. - Don't store strong references to the object in the bus, appsink and appsrc futures `Stream` / `Sink` adapters. This would keep them alive unnecessarily and would prevent the `Stream` / `Sink` to ever "finish" on its own. - Handle receiving a `None` reply in the change function of `gst::Promise`. This is apparently valid. For backwards compatibility reasons this is currently replaced with an empty structure but in 0.16 the API will explicitly handle `None`. ### Added - `gst::Stream::debug()` and `gst::StreamCollection::debug()` for converting into a structured string with the actual contents of each. - `gst::Structure::from_iter()` and `gst::Caps::from_iter()` to create structures/caps from iterators. - `gst::Event` support for getting/setting the `gst::Stream` in the `StreamStart` event. - `gst_video::calculate_display_ratio()` and `::guess_framerate()`. - Various video related `gst::CapsFeatures` in `gst_video`. - `TryFrom`/`From` impls for converting between `gst::Structure` and `gst_video::VideoConverterConfig`. - Various `glib::Value` trait impls for `SDPMessage`, `StructureRef`, `CapsFeatureRef` and all borrowed variants of miniobjects to be able to work with the borrowed, non-owned variants when handling `glib::Value`s. ## [0.15.5] - 2020-05-03 ### Fixed - Revert: Allow logging any `glib::Object` and not just `gst::Object`. This broke API in subtle ways and needs to wait until 0.16 - Replace `%` in log output with `%%` to prevent accidental C formatting - Add missing manual traits to the documentation ### Added - `BufferRef::peek_memory_mut()` to give a mutable reference to a given memory - Different iterators for iterating over the memories of a buffer - Support for `gst_audio::AudioClippingMeta` - `gst::Plugin::get_plugin_name()` was added - `gst::Element::get_current_clock_time()` and `gst::Element::get_current_running_time() helper functions - `gst::State` and `StateChange` API for calculating next/previous state and convert from/to the components of a state change ### Changed - Use `mem::ManuallyDrop` instead of `mem::forget` everywhere ## [0.15.4] - 2020-03-09 ### Fixed - Allow logging any `glib::Object` and not just `gst::Object` - Fix floating reference handling in `RTSPMedia::take_pipeline()` - Hold `GMutex` guards for the remainder of the function and warn if they're directly dropped - Work around empty/any caps handling bugs in `Caps::fixate()` ### Added - Add `BaseTransform::prepare_output_buffer()` subclassing support - `RTSPServer`, `RTSPClient`, `RTSPMedia` and `RTSPMediaFactory` subclassing support - Handle panicking in `appsrc`/`appsink` callbacks by posting an error message instead of killing the process ## [0.15.3] - 2020-02-15 ### Fixed - `UniqueFlowCombiner::clear()` should take a mutable reference. - `AudioStreamAlign` doesn't require mutable references for getters anymore. - Don't use bool return value of `gst_video_info_set_format()` and `gst_video_info_align()` with GStreamer < 1.11.1 as it returned void back then. We'd otherwise use some random value. - Make `VideoInfo::align()` is available since 1.8. - Fix changing/clearing of `AppSrc`, `AppSink` callbacks and `Bus` sync handler. Before 1.16.3 this was not thread-safe and caused crashes. When running with older versions changing them causes a panic now and unsetting the bus sync handler has not effect. With newer versions it works correctly. ### Added - Add `Clone` impls for `BufferPoolConfig` and `PlayerConfig`. - Add `VideoConverter` bindings. - Add `Future`s variant for `gst::Promise` constructor. - Add `Future`s variant for `gst_video::convert_sample_async()`. - Add `submit_input_buffer()`, `generate_output()`, `before_transform()`, `copy_metadata()` and `transform_meta()` virtual method support for `BaseTransform`. - Add `AppSink` `Stream` adapter and `AppSrc` `Sink` adapter for integrating both into Rust async contexts. ### Changed - More generic implementations of `VideoFrame` / `VideoFrameRef` functions to allow usage in more generic contexts. ## [0.15.2] - 2020-01-30 ### Fixed - Fix another race condition in the `gst::Bus` `Stream` that could cause it to not wake up although a message is available. ## [0.15.1] - 2020-01-23 ### Added - Use static inner lifetime for `VideoCodecState` so that it can be stored safely on the heap. - Getters/setters for `BinFlags` on `gst::Bin`. - `gst::Caps::builder_full()` for building caps with multiple structures conveniently. - `gst::Element::call_async_future()` for asynchronously spawning a closure and returning a `Future` for awaiting its return value. ### Fixed - Various clippy warnings. - Getters/setters for `PadFlags` on `gst::Pad` now provide the correct behaviour. - Take mutex before popping messages in the `gst::Bus` `Stream` to close a small race condition that could cause it to not be woken up. - `gst::ChildProxy` implementers do not have to provide `child_added()` and `child_removed()` functions anymore but these are optional now. - Manually implement `Debug` impls for various generic types where to `Debug` impl should not depend on their type parameters also implementing `Debug`. ## [0.15.0] - 2019-12-18 ### Added - `StructureRef::get_optional()` for returning `None` if the field does not exist instead of `Err` - Bindings for `gstreamer-rtp` library, mostly `RTPBuffer` - Support for writing `Preset`, `TagSetter`, `Clock`, `SystemClock` subclasses - Bindings for `Typefind::get_length()` - Bindings for `BaseSrcImpl::get_times()` - Bindings (incl. subclassing) for `AudioSink` and `AudioSrc` - Missing `Send`/`Sync` impl for various types ### Fixed - Cleanup of cargo features/dependencies to improve build times - Serde serialization with optional values. Attention: This changes the format of the serialization! - `VideoEncoder`/`VideoDecoder` `proxy_getcaps()` can't return `None` - Use non-panicking UTF8 conversion in log handler. We don't want to panic just because some C code printed a non-UTF8 string - Re-rexport all traits from the crate level and also ensure that all traits are actually included in the preludes - Actually export `is_video_overlay_prepare_window_handle_message()` function - Use `FnMut` for the `appsink` callbacks instead of `Fn` - `Promise` change function returns the actual reply to the promise now instead of just passing the promise itself - Memory leak in `Iterator::filter()` - `BinImpl::add()` takes ownership of floating references - `DeviceImpl::create_element()` preserves floating flag - `BinImpl::remove()` takes a strong reference of the element now as the last reference might be owned by the bin and otherwise we would potentially have a use-after-free afterwards - `BaseParseFrame` and `VideoCodecFrame` take a `&mut self` now for various functions that actually change the frame ### Changed - Minimum supported Rust version is 1.39 - Allow passing `None` to `VideoEncoder::finish_frame()` - Various `to_string()` methods were moved into the `Display` trait impl and for some types `to_str()` was added to return a `&'static str` - .gir files were updated to 1.16.2 release - `Sample` constructor uses the builder pattern now - `VideoMeta::add_full()` is simplified and requires parameters - `BasetransformImpl::set_caps()` returns a `Result` instead of `bool` - SDP data type getters for strings return an `Option` now as these can be `None` in practice although not allowed by the SDP spec - Various functions returning `Option`s were changed to return `Results` if `None` actually signalled an error instead of just a missing value ### Removed - "subclassing" and "futures" cargo features. These are enabled by default now ## [0.14.5] - 2019-09-17 ### Added - Support subclassing of `gst::Device`, `gst::DeviceProvider`, `gst_audio::AudioDecoder` and `::AudioEncoder` - Support for `Element::set_clock` and `::provide_clock` virtual methods - `ElementClass::add_metadata` was added - `gst_video::VideoDecoder` and `::VideoEncoder` got support for `get_caps`, `negotiate`, `src/sink_query/event` and the `drain` virtual methods - `Element::num_pads`, `::num_src_pads` and `::num_sink_pads` functions - `gst_video::VideoDecoder` and `::VideoEncoder` got `get_allocator` bindings - `gst::Iterator` implements `IntoIterator` now for providing `std::iter::Iterator>` adapter - Error macros for audio/video decoder subclasses to handle decoding errors more gracefully and only actually error out after many consecutive errors ### Fixed - Macros now also work in Rust 2018 edition without `#[macro_use]` but explicit imports - The log handler unit test runs reliable in parallel with other tests - Manually implement `Debug` for `gst::Iterator` to allow it for any `T` instead of `T: Debug` - `Device::create_element` has correct reference count handling now - Return `NotNegotiated` in the video codec base classes if setting the output state fails instead of `Error` ## [0.14.4] - 2019-08-14 ### Added - Bindings for adding/removing custom log functions - Bindings for `calculate_linear_regression()` - Constants for base class custom flow returns ### Fixed - Ownership of pad in `Element::release_pad()` virtual method implementations ## [0.14.3] - 2019-07-16 ### Added - `Buffer::unset_flags()` for unsetting specific buffer flags - `VideoBufferFlags` flags type and `VideoBufferExt::set_video_flags()`, `unset_video_flags()` and `get_video_flags()` for working with video buffer flags from safe code. ### Fixed - Setting buffer flags does not override arbitrary other flags anymore but only sets the flags in question. This is necessary to not override extension buffer flags like `gst_video::VideoBufferFlags`. ## [0.14.2] - 2019-07-15 ### Added - Support for `ReferenceTimestampMeta` ## [0.14.1] - 2019-07-06 ### Added - Various new WebRTC enum types from 1.14.1/1.16.0 ### Fixed - Correctly generate interlaced `VideoInfo` by using `gst_video_info_set_interlaced_format()` instead of the generic function. - serde serialization unit tests for `gst::format` succeed again now. ### Changed - `Debug` impls for `VideoFormatInfo` and `AudioFormatInfo` now print all the details of the format instead of only the name, and the `Debug` impls for `VideoInfo` and `AudioInfo` also print the format now. ## [0.14.0] - 2019-06-24 ### Added - Bindings for `GLSyncMeta`. - Bindings for setting/getting `TagScope` on a `TagList` - Bindings for `GLDisplayWayland` and `GLDisplayX11` in addition to the already existing `GLDisplayEGL` - Bindings for `Bus::pop_filtered()` and related functions - Bindings for getting/setting `Object`, `Element`, `Bin`, `Pipeline` and `Plugin` flags - Bindings for `VideoCaptionMeta` - `Debug` impl of `Buffer` now also shows the metas of the buffers - Expose flow return in `PadProbeInfo` for overriding the return value - Bindings for `VideoDecoder` and `VideoEncoder`, including subclassing support - Bindings for `Memory`, `Allocator` and `VideoBufferPool` - Bindings for `VideoFormatInfo::pack` and `::unpack` for format conversion - Bindings for `BaseParse`, including subclassing support - Various new arithmetic operation impls for fractions, formatted values and `ClockTime` - Bindings for `VideoInfo::align()` ### Changed - The `SDPMessage` and `SDPMedia` bindings were completely rewritten as they were broken before and caused crashes in various usages. As part of this there's also some more convenience API available on these types, like iterators for example, and API to modify the `SDPMedia` contained in a `SDPMessage`. - Update to GStreamer 1.16. - Regenerate with latest gir. - Run all autogenerated code through rustfmt after generation too. - Updated to latest versions of GLib/GIO/etc crates. - Updated to futures 0.3 / `std::future` - `ProxyPad` default functions moved to an extension trait instead of plain functions on `ProxyPad`, making them more in sync with the default `Pad` functions - GStreamer plugins are now exporting the new 1.14+ plugin symbols if they were configured for GStreamer 1.14+ - Arithmetic operations on formatted values and `ClockTime` do overflow checks now and replace the result with the `NONE` value on overflow - `TryFrom`/`TryInto` traits are used in various places now instead of the previous ad-hoc implementations of them. - Registering element/typefind/device monitor factories requires passing a value of `gst::Rank` now instead of an arbitrary `u32` ### Fixed - Use correct type for destroying pad task closure data. This was previously using the wrong type, causing crashes at runtime. - `DeviceAdded`/`DeviceRemoved` message getters are transfer full so we don't need to take an additional reference that would be leaked. - `AppSink` callbacks are correctly marked as `Send` instead of `Send+Sync`, allowing a wider range of closures to be used for them. - Handle `PadProbeReturn::Handled` return values from pad probes more correctly. - `ToOwned::to_owned()` on miniobjects has to create copies instead of only increasing the reference count. Otherwise it was possible to create multiple mutable and immutable references to the same object at the same time. - Various functions take references to owned miniobjects instead of borrowed references as it was otherwise possible to create multiple mutable or immutable references to the same object at the same time. - `URIHandler::set_uri` does not accept `None` anymore as this is not allowed by the C function. - Comparisons and addition of `TypeFindProbability` and `Rank` work correctly now - Various `Display` implementations were fixed to not cause a stack overflow due to infinite recursion anymore - Various `::to_string()` functions don't take ownership of C strings anymore that they do not own, which caused double frees before ### Removed - MIKEY related bindings from the SDP library. The bindings were broken and until someone needs them these are not available anymore. ## [0.13.0] - 2019-02-22 ### Added - Subclassing infrastructure was moved directly into the bindings, making the `gst-plugin` crate deprecated. This involves many API changes but generally cleans up code and makes it more flexible. Take a look at the `gst-plugins-rs` crate for various examples. - Bindings for GStreamer GL library - Bindings for `CapsFeatures` and `Meta` - Bindings for `ParentBufferMeta, `VideoMeta` and `VideoOverlayCompositionMeta` - Bindings for `VideoOverlayComposition` and `VideoOverlayRectangle` - Bindings for `VideoTimeCode` - Bindings for `NetAddressMeta` - Bindings for registering custom tags - `UniqueFlowCombiner` and `UniqueAdapter` wrappers that make use of the Rust compile-time mutability checks and expose more API in a safe way, and as a side-effect implement `Sync` and `Send` now - `Bus::add_watch_local()` and `gst_video::convert_frame_async_local()` that allows to use a closure that does not implement `Send` but can only be called from the thread owning the main context. - More complete bindings for `Allocation` `Query` - `pbutils` functions for codec descriptions - `TagList::iter()` for iterating over all tags while getting a single value per tag. The old `::iter_tag_list()` function was renamed to `::iter_generic()` and still provides access to each value for a tag - `Bus::iter()` and `Bus::iter_timed()` iterators around the corresponding `::pop*()` functions - Getters for `VideoColorimetry` to access its fields - `Debug` impls for various missing types. - serde serialization of `Value` can also handle `Buffer` now - Extensive comments to all examples with explanations - Transmuxing example showing how to use `typefind`, `multiqueue` and dynamic pads - basic-tutorial-12 was ported and added ### Changed - Rust 1.31 is the minimum supported Rust version now - Update to latest gir code generator and glib bindings - Functions returning e.g. `gst::FlowReturn` or other "combined" enums were changed to return split enums like `Result` to allow usage of the standard Rust error handling. - Various functions and callbacks returning `bool` or `Option<_>` were changed to return a `Result<_, glib::BoolError>` or `Result<_, gst::LoggableError>` or `Result<_, gst::ErrorMessage>` for better integration with Rust's error handling infrastructure. - Some infallible functions returning `bool` were changed to return `()`. - `MiniObject` subclasses are now newtype wrappers around the underlying `GstRc` wrapper. This does not change the API in any breaking way for the current usages, but allows `MiniObject`s to also be implemented in other crates and makes sure `rustdoc` places the documentation in the right places. - `BinExt` extension trait was renamed to `GstBinExt` to prevent conflicts with `gtk::Bin` if both are imported - `Buffer::from_slice()` can't possible return `None` ### Fixed - `gst::tag::Album` is the album tag now instead of artist sortname - Return `0` for the channel mask corresponding to negative `AudioChannelPosition`s. - `PartialOrd` and related traits are implemented via pointer equality on `ClockId` instead of using the compare function. Two clock ids with the same timestamp are not necessarily the same. - Various functions that are actually fallible are now returning an `Option<_>`. - Various `clippy` warnings ## [0.12.2] - 2018-11-26 ### Fixed - PTP clock constructor actually creates a PTP instead of NTP clock ### Added - Bindings for GStreamer Editing Services - Bindings for GStreamer Check testing library - Bindings for the encoding profile API (encodebin) - VideoFrame, VideoInfo, AudioInfo, StructureRef implements Send and Sync now - VideoFrame has a function to get the raw FFI pointer - From impls from the Error/Success enums to the combined enums like FlowReturn - Bin-to-dot file functions were added to the Bin trait - gst_base::Adapter implements SendUnique now ### Changed - All references were updated from GitHub to freedesktop.org GitLab - Fix various links in the README.md - Link to the correct location for the documentation - Remove GitLab badge as that only works with gitlab.com currently ## [0.12.1] - 2018-09-21 ### Added - More complete bindings for the gst_video::VideoOverlay interface, especially gst_video::is_video_overlay_prepare_window_handle_message() ## [0.12.0] - 2018-09-08 ### Added - Bindings for the GStreamer SDP and WebRTC libraries - Generic API for working with tags that is based on string tag names and glib::Value for the tag values - Bindings for Aggregator and AggregatorPad - Bindings for BaseTransform/BaseSrc::get_buffer_pool() - Optional serde implementations for the basic GStreamer data flow and metadata types ### Changed - Use ptr::NonNull in various places - Updated to muldiv 0.2, num-rational 0.2 - Bus::create_watch() can't return None - Remove CallbackGuard as unwinding across FFI boundaries is not undefined behaviour anymore but will directly cause a panic - Changed from the futures to the futures-preview crate as an optional dependency - Various Caps operations take a &CapsRef instead of &Caps - "deep-notify" signal takes the whole ParamSpec as parameter instead of only the signal name - Some structs were changed from empty struct to empty enums - Pad probe code does not take an additional reference to the data anymore, potentially passing writable events/buffers into the probe - ValueExt::compare() is implemented around std::cmp::Ordering now instead of a custom enum that was basically the same ### Fixed - Pad::add_probe() can return None if an IDLE probe was already called and removed in the meantime - Various compiler and clippy warnings ### Removed - std::Iterator impl for gst::Iterator. It was awkward to use because the gst::Iterator could fail at each iteration ## [0.11.6] - 2018-08-27 ### Fixed - Build with NLL/two-phase borrows - Explicitly define [bin] section for discoverer example to fix a cargo warning ### Added - Add unsafe gst::deinit() function - Ord/PartialOrd impls on gst::Seqnum - Getter for current pad mode - gst::Pad::sticky_events_foreach() for iterating over all sticky events in a thread-safe way ## [0.11.5] - 2018-07-24 ### Fixed - `gst::Bus`'s sync handler must unref every message if `gst::BusSyncReply::Drop` is returned, otherwise they are all leaked ## [0.11.4] - 2018-07-19 ### Fixed - `gst::Caps::subtract()` does not leak its arguments anymore - `gst::Caps::get_structure()` gracefully returns `None` if the index is out of bounds instead of a `g_return_val_if_fail()` - `gst::Structure::new()` has to give away ownership of the info structure but didn't. For 0.11 we internally copy, in 0.12 it will take the info structure by value - Typefind tests don't fail anymore if the system has typefind factories without caps ### Added - An additional assertion that ensures that miniobjects are actually writable before creating a mutable reference ## [0.11.3] - 2018-06-08 ### Added - `gst::Bus::remove_watch()` is now available to remove a bus watch again - `fmt::Debug` impls for `AudioInfo` and `VideoInfo` were added - `fmt::Debug` impls for mini objects also print the pointer value now to make it easier to track them in debug logs - `PlayerVisualization` has accessors for the name and description fields now, without which there is no sensible way to use them or to set a player visualization ## [0.11.2] - 2018-05-09 ### Fixed - Work-around various floating reference handling changes between 1.12 and 1.14 to be able to run with both versions without memory leaks or other reference count problems. This affects NetTimeProvider, BufferPool, DeviceMonitor, Stream, StreamCollection, and Player, NetClientClock, NetClock, PtpClock which were already previously fixed. ### Changed - Change the appsrc need-data and all appsink callbacks to not require the Sync bound anymore and change from Fn to FnMut. They can only be called from a single thread at a time. This change is only done for the corresponding callbacks, not the signals. ## [0.11.1] - 2018-04-07 ### Fixed - Fix Structure::to_string() to not run into an infinite recursion but call the method on the contained StructureRef instead of on itself ## [0.11.0] - 2018-03-20 ### Changed - Updated everything to GStreamer 1.14.0 - Event, Message and Query types were refactored to improve usability. Especially newly constructed queries allow to directly use the type-specific functions to be used without first creating a view - VideoFrameRef::copy_to_ref() and ::copy_plane_to_ref() are gone now and the original functions work with refs instead of full frames - PadProbeId and NotifyIds are not Copy/Clone anymore and are taken by value - GstPlayer has GstObject as parent class now ### Added - GstPbutils, GstSdp, GstRtsp and GstRtspServer bindings - GstPromise, GstAudioStreamAlign and various other 1.14 API - GstVideoFilter and GstBufferPool bindings - Element::call_async() - Debug impl For Toc and TocEntry - Various new examples (RTP FEC, RTSP server, tag usage, ...) ### Fixed - Memory leak in gst_video::convert_sample_async() ## [0.10.2] - 2018-02-18 ### Fixed - Fix building of messages with custom fields for types that don't have a GstStructure ### Added - VideoFrameRef::copy_to_ref() and ::copy_plane_to_ref(), which work with VideoFrameRefs instead of full VideoFrames - Getters for the BaseSrc/Sink/Transform configured segment - Document the gstreamer-player-1.0 dependency in the README.md ## [0.10.1] - 2018-01-03 ### Fixed - Don't require &mut self for TagSetterExtManual::add() ### Added - A TagSetter example application - Bindings for gst_video::convert_sample() and ::convert_sample_async() - Bindings for gst_video::VideoRectangle - Debug impl for Sample and ::with_buffer_list() constructor - A borrowing version of VideoFrame: VideoFrameRef - Bindings for GstVideoFilter ### Changed - Deprecated Sample::get_info() in favour of ::get_structure() - Player has gst::Object as another parent class now ## [0.10.0] - 2017-12-22 ### Fixed - Various clippy warnings - Memory leak of the tag list in Toc::merge_tags() - Property getters use Values of the correct type - Event::get_structure(), Message::get_structure() and Query::get_structure() can return None for the structure - Various other nullability fixes all over the API, changing functions to accept Option<> or returning Option<>, or only plain types - Functions taking paths/filenames now actually take Paths instead of &strs - Element::remove_pad() is not giving away a new reference to the pad anymore, which caused a memory leak of all pads ever removed - Precision handling in ClockTime's Display impl - Video/AudioInfo are only Send, not Sync ### Added - Various enums now also derive useful traits like Copy, Clone and Hash in addition to PartialEq, Eq and Debug - TagList::merge() and insert() for combining tag lists - EventType gained many useful functions to work with event types and a PartialOrd impl to check expected event order of event types where it matters - MessageRef/EventRef/QueryRef implement ToOwned - Bindings for Registry and PluginFeature - Event::set_running_time_offset() for adjusting the offset while events pass through the pipeline - Event/Message GroupIds and Seqnums now have a newtype wrapper around u32 instead of the plain value, making usage of them slightly more typesafe. Also add an "invalid" value for both, as exists in latest GStreamer now. - FormattedValue, GenericFormattedValue and related types were implemented now, which allows more convenient and type-safe usage of formatted values (time, bytes, etc) - Bindings for force-keyunit and still-frame events were added - MappedBuffer/BufferMap now implement various other useful traits, including AsRef<[u8]>, AsMut, Deref, DerefMut, Debug, PartialEq and Eq - Add VideoMultiviewFramePacking enum, and use it in Player - Bindings for the GStreamer Net library, including PTP/NTP/network client clocks and the GStreamer NetClock provider for network synchronization of pipelines - IteratorError implements std::error:Error - Plugin::add_dependency() and ::add_dependency_simple() was added - Rank and TypeFindProbability implement PartialOrd/Ord now - Bindings for TypeFind, TypeFindFactory and the typefind helpers - StreamCollection::iter() for iterating over all contained streams - ErrorMessage type that can be used e.g. in a Result for passing an error message from somewhere to upper layers to then be posted on an element the same way gst_element_error!() would've done ### Changed - Sample::new(), TagList::add(), Structure::set() and similar functions take the values (ToSendValue impls) by reference instead of value. They were not consumed by the function before. - The Debug impls of various types, including Event/Buffer/Message/Query/Structure were improved to print all the fields, similar to what GST_PTR_FORMAT would do in C - Switched to lazy_static 1.0 - Gap event and Duration tag are using ClockTimes now, as well as various Player signals - Segment is now based on a generic type FormattedSegment that can take any format (time, bytes, etc) or a GenericFormattedValue for more type-safety and convenience. Also functions for "casting" between a generic segment and a segment with a specific format exist on this now - AppSrc and AppSink now have a builder for the callbacks, making it unnecessary to always provide all callbacks even if only one is actually needed - Various functions that returned bool for errors, are now returning a Result - Player configuration is now a custom type with more convenient API - Player VideoInfo uses a Fraction instead of (u32,u32) for the framerate and pixel-aspect-ratio - VideoFrame API has more consistent API between writable and read-only variants - Buffer::copy_into() was added, and ::copy_region() now takes a BufferCopyFlags parameter instead of always using the default flags - ChildProxy::set_child_property() takes a &ToValue now to follow the API of Object::set_property() and improve usability - Proxy/GhostPad default pad functions use the correct specific pad type now instead of a generic Pad - Bus::add_signal_watch_full() takes a Priority for the priority instead of u32 - Clock::(un)adjust_with_calibration() takes no clock parameter anymore ### Removed - FormatValue was removed in favour of GenericFormattedValue and the connected traits and specific format impls ## [0.9.1] - 2017-11-26 ### Fixed - Export `FlowError`/`FlowSuccess`, `ClockError`/`ClockSuccess`, `PadLinkError`/`PadLinkSuccess` too ## [0.9.0] - 2017-11-26 ### Added - Bindings for (outputting to) the GStreamer logging system - Bindings for the GStreamer base library - Bindings for all the `Pad` functions to override pad behaviour, and pad task functions - Bindings for `StaticCaps` and `StaticPadTemplate` - Bindings for `deep-notify` signal on `Object` - Support for directly creating `Error`/`Warning`/`Info` `Messages` and posting them from an element with context information (file, line, module, etc.) similar to the C `GST_ELEMENT_ERROR` macro - Support for setting custom fields in `Messages`/`Events` during construction - Support for creating Buffers out of anything that is `AsRef<[u8]>` or `AsMut<[u8]>` - Support for using the `Read` trait on `Adapter` - Functions for getting all sink/src/all pads of an `Element`, and all children of a `Bin` - Builder for `Caps` and `Structures` in addition to the existing functions - `AppSrc`/`AppSink` implement `BaseSrc`/`BaseSink` and `URIHandler` - Rust ports of the basic tutorials 1 to 8 from https://gstreamer.freedesktop.org/documentation/tutorials/ - "Getting started" and "Installation" sections to the README.md - "dox" feature for generating documentation for all available configurations ### Fixed - `StackTraceFlags` are only available since 1.12 - Worked around macOS requiring a `NSRunLoop` running on the main thread in all examples and tutorials, to be able to show a window or anything else ### Changed - `ClockTime` is now a wrapper around `Option` to handle the `CLOCK_TIME_NONE` case better. This wrapper implements all the arithmetic and other traits as needed and ensures that no accidental calculations with `CLOCK_TIME_NONE` can happen - "Values with format", like in `Duration`/`Position`/`Convert` queries or `Seek` events now return a `FormatValue` type. This contains the actual `Format` together with the value and does any required conversions. This also makes it harder to accidentally mix e.g. values in bytes and time - `PadProbeId` does not implement `Clone`/`Copy` anymore - Property notify watches return a custom type instead of ulong - `Error`/`Warning`/`Info` `Messages` can only be created with specific kinds of `glib::Error` now. Using arbitrary ones does not work - `Iterator` bindings were completely rewritten and provide the item type as a generic type parameter now, greatly simplifying its usage - All `glib::Values` are now `glib::SendValue` instead, e.g. in `Caps` and `Structures`, as their content must be possible to send to different threads safely - `Message::get_src()` can return `None` - Allow `None` as `Caps` in `AppSrc`/`AppSink` - Allow everything implementing `Into>` to be used as a pad name - Moved `copy()` from `GstRc` directly to `MiniObject` - Success/Error enums (like `FlowReturn`, `PadLinkReturn`, `StateChangeReturn`) now implement an `into_result()` function that splits them into a `Result` with the good and bad cases. Also mark them as `#[must_use]` to make it harder to accidentally ignore errors. - Error enums implement the `Error` trait - Many examples use the `failure` crate for error handling now, cleaning up the error handling code quite a bit - Lots of other code cleanup, compiler/clippy warning cleanup, etc. ## [0.8.2] - 2017-11-11 ### Fixed - Implement StaticType of BufferRef instead of Buffer. Buffer aka GstRc already implements StaticType if BufferRef does, and without this it was not possible to use Buffers in GValues. - Free memory of the appsink/appsrc callbacks with the correct type. It was crashing because of using the wrong type before. - Fix documentation URLs in Cargo.toml. ### Added - Installation instructions and links to documentation for getting started to README.md. ## [0.8.1] - 2017-09-15 ### Added - Implement Send+Sync for Query, Message and Event, and their corresponding Ref types. ### Fixed - Constructor for gst_player::Player now works properly with GStreamer 1.12 when passing a video renderer or signal dispatcher. There was a reference counting bug. - Instead of returning &'static references from functions, return references with a generic, unbound lifetime instead. See https://github.com/rust-lang/rust/pull/42417#issue-233404573 - Various "unused external crate" warnings and clippy warnings everywhere. ### Changed - Remove Cargo.lock from GIT, it's not very useful for library crates. - Run everything through latest rustfmt-nightly. - Use while-let (instead of loop and if-let) and CLOCK_TIME_NONE (instead of u64::MAX) in the examples. ## 0.8.0 - 2017-08-31 - Initial release of the autogenerated GStreamer bindings. Older versions (< 0.8.0) of the bindings can be found [here](https://github.com/arturoc/gstreamer1.0-rs). The API of the two is incompatible. [Unreleased]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.23.5...HEAD [0.23.5]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.23.4...0.23.5 [0.23.4]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.23.3...0.23.4 [0.23.3]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.23.2...0.23.3 [0.23.2]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.23.1...0.23.2 [0.23.1]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.23.0...0.23.1 [0.23.0]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.22.6...0.23.0 [0.22.6]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.22.5...0.22.6 [0.22.5]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.22.4...0.22.5 [0.22.4]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.22.3...0.22.4 [0.22.3]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/compare/0.22.2...0.22.3 [0.22.2]: 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licensed only under the terms of the MIT license. Copyrights in the gstreamer-rs project are retained by their contributors. No copyright assignment is required to contribute to the gstreamer-rs project. Some files include explicit copyright notices and/or license notices. For full authorship information, see the version control history. Except as otherwise noted (below and/or in individual files), gstreamer-rs is licensed under the Apache License, Version 2.0 or or the MIT license or , at your option. All the sys crates (e.g. gstreamer/sys and gstreamer-base/sys) are licensed only under the terms of the MIT license. This project provides interoperability with various GStreamer libraries but doesn't distribute any parts of them. Distributing compiled libraries and executables that link to those libraries may be subject to terms of the GNU LGPL or other licenses. 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Cargo will automatically # "normalize" Cargo.toml files for maximal compatibility # with all versions of Cargo and also rewrite `path` dependencies # to registry (e.g., crates.io) dependencies. # # If you are reading this file be aware that the original Cargo.toml # will likely look very different (and much more reasonable). # See Cargo.toml.orig for the original contents. [package] edition = "2021" rust-version = "1.71.1" name = "gstreamer-audio-sys" version = "0.23.5" authors = ["Sebastian Dröge "] build = "build.rs" autolib = false autobins = false autoexamples = false autotests = false autobenches = false description = "FFI bindings to libgstaudio-1.0" homepage = "https://gstreamer.freedesktop.org" documentation = "https://gstreamer.freedesktop.org/documentation/rust/stable/latest/docs/gstreamer_audio_sys/" readme = "README.md" keywords = [ "ffi", "gstreamer", "gnome", "multimedia", ] categories = [ "api-bindings", "multimedia", ] license = "MIT" repository = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs" [package.metadata.docs.rs] all-features = true rustc-args = [ "--cfg", "docsrs", ] rustdoc-args = [ "--cfg", "docsrs", "--generate-link-to-definition", ] [package.metadata.system-deps.gstreamer_audio_1_0] name = "gstreamer-audio-1.0" version = "1.14" [package.metadata.system-deps.gstreamer_audio_1_0.v1_16] version = "1.16" [package.metadata.system-deps.gstreamer_audio_1_0.v1_18] version = "1.18" [package.metadata.system-deps.gstreamer_audio_1_0.v1_20] version = "1.20" [package.metadata.system-deps.gstreamer_audio_1_0.v1_22] version = "1.22" [package.metadata.system-deps.gstreamer_audio_1_0.v1_24] version = "1.24" [package.metadata.system-deps.gstreamer_audio_1_0.v1_26] version = "1.25" [lib] name = "gstreamer_audio_sys" path = "src/lib.rs" [[test]] name = "abi" path = "tests/abi.rs" [dependencies.glib-sys] version = "0.20" [dependencies.gobject-sys] version = "0.20" [dependencies.gstreamer-base-sys] version = "0.23" [dependencies.gstreamer-sys] version = "0.23" [dependencies.libc] version = "0.2" [dev-dependencies.shell-words] version = "1.0.0" [dev-dependencies.tempfile] version = "3" [build-dependencies.system-deps] version = "7" [features] v1_16 = [] v1_18 = ["v1_16"] v1_20 = ["v1_18"] v1_22 = ["v1_20"] v1_24 = ["v1_22"] v1_26 = ["v1_24"] gstreamer-audio-sys-0.23.5/Cargo.toml.orig000064400000000000000000000033321046102023000165070ustar 00000000000000[build-dependencies] system-deps = "7" [dependencies] libc = "0.2" [dependencies.glib-sys] workspace = true [dependencies.gobject-sys] workspace = true [dependencies.gstreamer-sys] workspace = true [dependencies.gstreamer-base-sys] workspace = true [dev-dependencies] shell-words = "1.0.0" tempfile = "3" [features] v1_16 = [] v1_18 = ["v1_16"] v1_20 = ["v1_18"] v1_22 = ["v1_20"] v1_24 = ["v1_22"] v1_26 = ["v1_24"] [lib] name = "gstreamer_audio_sys" [package] authors = ["Sebastian Dröge "] build = "build.rs" description = "FFI bindings to libgstaudio-1.0" documentation = "https://gstreamer.freedesktop.org/documentation/rust/stable/latest/docs/gstreamer_audio_sys/" keywords = ["ffi", "gstreamer", "gnome", "multimedia"] license = "MIT" name = "gstreamer-audio-sys" readme = "README.md" [package.version] workspace = true [package.categories] workspace = true [package.repository] workspace = true [package.homepage] workspace = true [package.edition] workspace = true [package.rust-version] workspace = true [package.metadata.docs.rs] all-features = true rustc-args = ["--cfg", "docsrs"] rustdoc-args = ["--cfg", "docsrs", "--generate-link-to-definition"] [package.metadata.system-deps.gstreamer_audio_1_0] name = "gstreamer-audio-1.0" version = "1.14" [package.metadata.system-deps.gstreamer_audio_1_0.v1_16] version = "1.16" [package.metadata.system-deps.gstreamer_audio_1_0.v1_18] version = "1.18" [package.metadata.system-deps.gstreamer_audio_1_0.v1_20] version = "1.20" [package.metadata.system-deps.gstreamer_audio_1_0.v1_22] version = "1.22" [package.metadata.system-deps.gstreamer_audio_1_0.v1_24] version = "1.24" [package.metadata.system-deps.gstreamer_audio_1_0.v1_26] version = "1.25" gstreamer-audio-sys-0.23.5/Gir.toml000064400000000000000000000031431046102023000152360ustar 00000000000000[options] girs_directories = ["../../gir-files", "../../gst-gir-files"] library = "GstAudio" version = "1.0" min_cfg_version = "1.14" work_mode = "sys" single_version_file = true extra_versions = [ "1.26", "1.24", "1.22", "1.20", "1.18", "1.16", ] external_libraries = [ "GLib", "GObject", ] ignore = [ "GstAudio.AUDIO_FORMATS_ALL", ] [external_libraries] gstreamer="Gst" gstreamer_base="GstBase" [[object]] name = "GstAudio.AudioBuffer" status = "generate" [[object.function]] name = "clip" version = "1.0" [[object.function]] name = "reorder_channels" version = "1.0" [[object]] name = "GstAudio.AudioFormat" status = "generate" [[object.member]] name = "f32" ignore = true [[object.member]] name = "f64" ignore = true [[object.member]] name = "s16" ignore = true [[object.member]] name = "s18" ignore = true [[object.member]] name = "s20" ignore = true [[object.member]] name = "s24" ignore = true [[object.member]] name = "s24_32" ignore = true [[object.member]] name = "s32" ignore = true [[object.member]] name = "u16" ignore = true [[object.member]] name = "u18" ignore = true [[object.member]] name = "u20" ignore = true [[object.member]] name = "u24" ignore = true [[object.member]] name = "u24_32" ignore = true [[object.member]] name = "u32" ignore = true [[object]] name = "GstAudio.*" status = "generate" [[object.function]] name = "audio_meta_api_get_type" version = "1.16" gstreamer-audio-sys-0.23.5/LICENSE000064400000000000000000000017771046102023000146400ustar 00000000000000Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions: The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software. THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. gstreamer-audio-sys-0.23.5/README.md000064400000000000000000000031061046102023000150760ustar 00000000000000# gstreamer-sys [![crates.io](https://img.shields.io/crates/v/gstreamer-audio-sys.svg)](https://crates.io/crates/gstreamer-audio-sys) [![pipeline status](https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/badges/main/pipeline.svg)](https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/commits/main) [GStreamer](https://gstreamer.freedesktop.org/) (Audio library) FFI bindings for Rust. These bindings are providing unsafe FFI API that can be used to interface with GStreamer. Generally they are meant to be used as the building block for higher-level abstractions like: * Bindings for GStreamer applications and plugins: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs * Various GStreamer plugins written in Rust: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs The bindings are autogenerated with [gir](https://github.com/gtk-rs/gir/) based on the [GObject-Introspection](https://wiki.gnome.org/Projects/GObjectIntrospection/) API metadata provided by the GStreamer project. ## LICENSE gstreamer-sys and all crates contained here are licensed under the MIT license ([LICENSE](LICENSE) or http://opensource.org/licenses/MIT). GStreamer itself is licensed under the Lesser General Public License version 2.1 or (at your option) any later version: https://www.gnu.org/licenses/lgpl-2.1.html ## Contribution Any kinds of contributions are welcome as a pull request. Unless you explicitly state otherwise, any contribution intentionally submitted for inclusion in gstreamer-rs by you shall be licensed under the MIT license as above, without any additional terms or conditions. gstreamer-audio-sys-0.23.5/build.rs000064400000000000000000000007761046102023000152760ustar 00000000000000// This file was generated by gir (https://github.com/gtk-rs/gir) // from gir-files (https://github.com/gtk-rs/gir-files) // from gst-gir-files (https://gitlab.freedesktop.org/gstreamer/gir-files-rs.git) // DO NOT EDIT #[cfg(not(docsrs))] use std::process; #[cfg(docsrs)] fn main() {} // prevent linking libraries to avoid documentation failure #[cfg(not(docsrs))] fn main() { if let Err(s) = system_deps::Config::new().probe() { println!("cargo:warning={s}"); process::exit(1); } } gstreamer-audio-sys-0.23.5/src/auto/versions.txt000064400000000000000000000003421046102023000177660ustar 00000000000000Generated by gir (https://github.com/gtk-rs/gir @ 2b05eaddce95) from gir-files (https://github.com/gtk-rs/gir-files @ 5089b7ff80cd) from gst-gir-files (https://gitlab.freedesktop.org/gstreamer/gir-files-rs.git @ 26898eacb093) gstreamer-audio-sys-0.23.5/src/lib.rs000064400000000000000000003110011046102023000155160ustar 00000000000000// This file was generated by gir (https://github.com/gtk-rs/gir) // from gir-files (https://github.com/gtk-rs/gir-files) // from gst-gir-files (https://gitlab.freedesktop.org/gstreamer/gir-files-rs.git) // DO NOT EDIT #![allow(non_camel_case_types, non_upper_case_globals, non_snake_case)] #![allow( clippy::approx_constant, clippy::type_complexity, clippy::unreadable_literal, clippy::upper_case_acronyms )] #![cfg_attr(docsrs, feature(doc_cfg))] use glib_sys as glib; use gobject_sys as gobject; use gstreamer_base_sys as gst_base; use gstreamer_sys as gst; #[cfg(unix)] #[allow(unused_imports)] use libc::{dev_t, gid_t, pid_t, socklen_t, uid_t}; #[allow(unused_imports)] use libc::{intptr_t, off_t, size_t, ssize_t, time_t, uintptr_t, FILE}; #[allow(unused_imports)] use std::ffi::{ c_char, c_double, c_float, c_int, c_long, c_short, c_uchar, c_uint, c_ulong, c_ushort, c_void, }; #[allow(unused_imports)] use glib::{gboolean, gconstpointer, gpointer, GType}; // Enums pub type GstAudioBaseSinkDiscontReason = c_int; pub const GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT: GstAudioBaseSinkDiscontReason = 0; pub const GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS: GstAudioBaseSinkDiscontReason = 1; pub const GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH: GstAudioBaseSinkDiscontReason = 2; pub const GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY: GstAudioBaseSinkDiscontReason = 3; pub const GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT: GstAudioBaseSinkDiscontReason = 4; pub const GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE: GstAudioBaseSinkDiscontReason = 5; pub type GstAudioBaseSinkSlaveMethod = c_int; pub const GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE: GstAudioBaseSinkSlaveMethod = 0; pub const GST_AUDIO_BASE_SINK_SLAVE_SKEW: GstAudioBaseSinkSlaveMethod = 1; pub const GST_AUDIO_BASE_SINK_SLAVE_NONE: GstAudioBaseSinkSlaveMethod = 2; pub const GST_AUDIO_BASE_SINK_SLAVE_CUSTOM: GstAudioBaseSinkSlaveMethod = 3; pub type GstAudioBaseSrcSlaveMethod = c_int; pub const GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: GstAudioBaseSrcSlaveMethod = 0; pub const GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP: GstAudioBaseSrcSlaveMethod = 1; pub const GST_AUDIO_BASE_SRC_SLAVE_SKEW: GstAudioBaseSrcSlaveMethod = 2; pub const GST_AUDIO_BASE_SRC_SLAVE_NONE: GstAudioBaseSrcSlaveMethod = 3; pub type GstAudioCdSrcMode = c_int; pub const GST_AUDIO_CD_SRC_MODE_NORMAL: GstAudioCdSrcMode = 0; pub const GST_AUDIO_CD_SRC_MODE_CONTINUOUS: GstAudioCdSrcMode = 1; pub type GstAudioChannelPosition = c_int; pub const GST_AUDIO_CHANNEL_POSITION_NONE: GstAudioChannelPosition = -3; pub const GST_AUDIO_CHANNEL_POSITION_MONO: GstAudioChannelPosition = -2; pub const GST_AUDIO_CHANNEL_POSITION_INVALID: GstAudioChannelPosition = -1; pub const GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT: GstAudioChannelPosition = 0; pub const GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT: GstAudioChannelPosition = 1; pub const GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER: GstAudioChannelPosition = 2; pub const GST_AUDIO_CHANNEL_POSITION_LFE1: GstAudioChannelPosition = 3; pub const GST_AUDIO_CHANNEL_POSITION_REAR_LEFT: GstAudioChannelPosition = 4; pub const GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT: GstAudioChannelPosition = 5; pub const GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER: GstAudioChannelPosition = 6; pub const GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER: GstAudioChannelPosition = 7; pub const GST_AUDIO_CHANNEL_POSITION_REAR_CENTER: GstAudioChannelPosition = 8; pub const GST_AUDIO_CHANNEL_POSITION_LFE2: GstAudioChannelPosition = 9; pub const GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT: GstAudioChannelPosition = 10; pub const GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT: GstAudioChannelPosition = 11; pub const GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT: GstAudioChannelPosition = 12; pub const GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT: GstAudioChannelPosition = 13; pub const GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER: GstAudioChannelPosition = 14; pub const GST_AUDIO_CHANNEL_POSITION_TOP_CENTER: GstAudioChannelPosition = 15; pub const GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT: GstAudioChannelPosition = 16; pub const GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT: GstAudioChannelPosition = 17; pub const GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT: GstAudioChannelPosition = 18; pub const GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT: GstAudioChannelPosition = 19; pub const GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER: GstAudioChannelPosition = 20; pub const GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER: GstAudioChannelPosition = 21; pub const GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT: GstAudioChannelPosition = 22; pub const GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT: GstAudioChannelPosition = 23; pub const GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT: GstAudioChannelPosition = 24; pub const GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT: GstAudioChannelPosition = 25; pub const GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT: GstAudioChannelPosition = 26; pub const GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT: GstAudioChannelPosition = 27; #[cfg(feature = "v1_26")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_26")))] pub const GST_AUDIO_CHANNEL_POSITION_TOP_SURROUND_LEFT: GstAudioChannelPosition = 28; #[cfg(feature = "v1_26")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_26")))] pub const GST_AUDIO_CHANNEL_POSITION_TOP_SURROUND_RIGHT: GstAudioChannelPosition = 29; pub type GstAudioDitherMethod = c_int; pub const GST_AUDIO_DITHER_NONE: GstAudioDitherMethod = 0; pub const GST_AUDIO_DITHER_RPDF: GstAudioDitherMethod = 1; pub const GST_AUDIO_DITHER_TPDF: GstAudioDitherMethod = 2; pub const GST_AUDIO_DITHER_TPDF_HF: GstAudioDitherMethod = 3; pub type GstAudioFormat = c_int; pub const GST_AUDIO_FORMAT_UNKNOWN: GstAudioFormat = 0; pub const GST_AUDIO_FORMAT_ENCODED: GstAudioFormat = 1; pub const GST_AUDIO_FORMAT_S8: GstAudioFormat = 2; pub const GST_AUDIO_FORMAT_U8: GstAudioFormat = 3; pub const GST_AUDIO_FORMAT_S16LE: GstAudioFormat = 4; pub const GST_AUDIO_FORMAT_S16BE: GstAudioFormat = 5; pub const GST_AUDIO_FORMAT_U16LE: GstAudioFormat = 6; pub const GST_AUDIO_FORMAT_U16BE: GstAudioFormat = 7; pub const GST_AUDIO_FORMAT_S24_32LE: GstAudioFormat = 8; pub const GST_AUDIO_FORMAT_S24_32BE: GstAudioFormat = 9; pub const GST_AUDIO_FORMAT_U24_32LE: GstAudioFormat = 10; pub const GST_AUDIO_FORMAT_U24_32BE: GstAudioFormat = 11; pub const GST_AUDIO_FORMAT_S32LE: GstAudioFormat = 12; pub const GST_AUDIO_FORMAT_S32BE: GstAudioFormat = 13; pub const GST_AUDIO_FORMAT_U32LE: GstAudioFormat = 14; pub const GST_AUDIO_FORMAT_U32BE: GstAudioFormat = 15; pub const GST_AUDIO_FORMAT_S24LE: GstAudioFormat = 16; pub const GST_AUDIO_FORMAT_S24BE: GstAudioFormat = 17; pub const GST_AUDIO_FORMAT_U24LE: GstAudioFormat = 18; pub const GST_AUDIO_FORMAT_U24BE: GstAudioFormat = 19; pub const GST_AUDIO_FORMAT_S20LE: GstAudioFormat = 20; pub const GST_AUDIO_FORMAT_S20BE: GstAudioFormat = 21; pub const GST_AUDIO_FORMAT_U20LE: GstAudioFormat = 22; pub const GST_AUDIO_FORMAT_U20BE: GstAudioFormat = 23; pub const GST_AUDIO_FORMAT_S18LE: GstAudioFormat = 24; pub const GST_AUDIO_FORMAT_S18BE: GstAudioFormat = 25; pub const GST_AUDIO_FORMAT_U18LE: GstAudioFormat = 26; pub const GST_AUDIO_FORMAT_U18BE: GstAudioFormat = 27; pub const GST_AUDIO_FORMAT_F32LE: GstAudioFormat = 28; pub const GST_AUDIO_FORMAT_F32BE: GstAudioFormat = 29; pub const GST_AUDIO_FORMAT_F64LE: GstAudioFormat = 30; pub const GST_AUDIO_FORMAT_F64BE: GstAudioFormat = 31; pub type GstAudioLayout = c_int; pub const GST_AUDIO_LAYOUT_INTERLEAVED: GstAudioLayout = 0; pub const GST_AUDIO_LAYOUT_NON_INTERLEAVED: GstAudioLayout = 1; pub type GstAudioNoiseShapingMethod = c_int; pub const GST_AUDIO_NOISE_SHAPING_NONE: GstAudioNoiseShapingMethod = 0; pub const GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK: GstAudioNoiseShapingMethod = 1; pub const GST_AUDIO_NOISE_SHAPING_SIMPLE: GstAudioNoiseShapingMethod = 2; pub const GST_AUDIO_NOISE_SHAPING_MEDIUM: GstAudioNoiseShapingMethod = 3; pub const GST_AUDIO_NOISE_SHAPING_HIGH: GstAudioNoiseShapingMethod = 4; pub type GstAudioResamplerFilterInterpolation = c_int; pub const GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE: GstAudioResamplerFilterInterpolation = 0; pub const GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: GstAudioResamplerFilterInterpolation = 1; pub const GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC: GstAudioResamplerFilterInterpolation = 2; pub type GstAudioResamplerFilterMode = c_int; pub const GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED: GstAudioResamplerFilterMode = 0; pub const GST_AUDIO_RESAMPLER_FILTER_MODE_FULL: GstAudioResamplerFilterMode = 1; pub const GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO: GstAudioResamplerFilterMode = 2; pub type GstAudioResamplerMethod = c_int; pub const GST_AUDIO_RESAMPLER_METHOD_NEAREST: GstAudioResamplerMethod = 0; pub const GST_AUDIO_RESAMPLER_METHOD_LINEAR: GstAudioResamplerMethod = 1; pub const GST_AUDIO_RESAMPLER_METHOD_CUBIC: GstAudioResamplerMethod = 2; pub const GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL: GstAudioResamplerMethod = 3; pub const GST_AUDIO_RESAMPLER_METHOD_KAISER: GstAudioResamplerMethod = 4; pub type GstAudioRingBufferFormatType = c_int; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW: GstAudioRingBufferFormatType = 0; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW: GstAudioRingBufferFormatType = 1; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW: GstAudioRingBufferFormatType = 2; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM: GstAudioRingBufferFormatType = 3; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG: GstAudioRingBufferFormatType = 4; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM: GstAudioRingBufferFormatType = 5; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958: GstAudioRingBufferFormatType = 6; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3: GstAudioRingBufferFormatType = 7; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3: GstAudioRingBufferFormatType = 8; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS: GstAudioRingBufferFormatType = 9; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC: GstAudioRingBufferFormatType = 10; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC: GstAudioRingBufferFormatType = 11; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW: GstAudioRingBufferFormatType = 12; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW: GstAudioRingBufferFormatType = 13; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC: GstAudioRingBufferFormatType = 14; pub const GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD: GstAudioRingBufferFormatType = 15; pub type GstAudioRingBufferState = c_int; pub const GST_AUDIO_RING_BUFFER_STATE_STOPPED: GstAudioRingBufferState = 0; pub const GST_AUDIO_RING_BUFFER_STATE_PAUSED: GstAudioRingBufferState = 1; pub const GST_AUDIO_RING_BUFFER_STATE_STARTED: GstAudioRingBufferState = 2; pub const GST_AUDIO_RING_BUFFER_STATE_ERROR: GstAudioRingBufferState = 3; pub type GstDsdFormat = c_int; pub const GST_DSD_FORMAT_UNKNOWN: GstDsdFormat = 0; pub const GST_DSD_FORMAT_U8: GstDsdFormat = 1; pub const GST_DSD_FORMAT_U16LE: GstDsdFormat = 2; pub const GST_DSD_FORMAT_U16BE: GstDsdFormat = 3; pub const GST_DSD_FORMAT_U32LE: GstDsdFormat = 4; pub const GST_DSD_FORMAT_U32BE: GstDsdFormat = 5; pub const GST_NUM_DSD_FORMATS: GstDsdFormat = 6; pub const GST_DSD_FORMAT_U16: GstDsdFormat = 2; pub const GST_DSD_FORMAT_U32: GstDsdFormat = 4; pub type GstStreamVolumeFormat = c_int; pub const GST_STREAM_VOLUME_FORMAT_LINEAR: GstStreamVolumeFormat = 0; pub const GST_STREAM_VOLUME_FORMAT_CUBIC: GstStreamVolumeFormat = 1; pub const GST_STREAM_VOLUME_FORMAT_DB: GstStreamVolumeFormat = 2; // Constants pub const GST_AUDIO_CHANNELS_RANGE: &[u8] = b"(int) [ 1, max ]\0"; pub const GST_AUDIO_CONVERTER_OPT_DITHER_METHOD: &[u8] = b"GstAudioConverter.dither-method\0"; pub const GST_AUDIO_CONVERTER_OPT_DITHER_THRESHOLD: &[u8] = b"GstAudioConverter.dither-threshold\0"; pub const GST_AUDIO_CONVERTER_OPT_MIX_MATRIX: &[u8] = b"GstAudioConverter.mix-matrix\0"; pub const GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD: &[u8] = b"GstAudioConverter.noise-shaping-method\0"; pub const GST_AUDIO_CONVERTER_OPT_QUANTIZATION: &[u8] = b"GstAudioConverter.quantization\0"; pub const GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD: &[u8] = b"GstAudioConverter.resampler-method\0"; pub const GST_AUDIO_DECODER_MAX_ERRORS: c_int = -1; pub const GST_AUDIO_DECODER_SINK_NAME: &[u8] = b"sink\0"; pub const GST_AUDIO_DECODER_SRC_NAME: &[u8] = b"src\0"; pub const GST_AUDIO_DEF_CHANNELS: c_int = 2; pub const GST_AUDIO_DEF_FORMAT: &[u8] = b"S16LE\0"; pub const GST_AUDIO_DEF_RATE: c_int = 44100; pub const GST_AUDIO_ENCODER_SINK_NAME: &[u8] = b"sink\0"; pub const GST_AUDIO_ENCODER_SRC_NAME: &[u8] = b"src\0"; pub const GST_AUDIO_FORMAT_LAST: c_int = 32; pub const GST_AUDIO_RATE_RANGE: &[u8] = b"(int) [ 1, max ]\0"; pub const GST_AUDIO_RESAMPLER_OPT_CUBIC_B: &[u8] = b"GstAudioResampler.cubic-b\0"; pub const GST_AUDIO_RESAMPLER_OPT_CUBIC_C: &[u8] = b"GstAudioResampler.cubic-c\0"; pub const GST_AUDIO_RESAMPLER_OPT_CUTOFF: &[u8] = b"GstAudioResampler.cutoff\0"; pub const GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION: &[u8] = b"GstAudioResampler.filter-interpolation\0"; pub const GST_AUDIO_RESAMPLER_OPT_FILTER_MODE: &[u8] = b"GstAudioResampler.filter-mode\0"; pub const GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD: &[u8] = b"GstAudioResampler.filter-mode-threshold\0"; pub const GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE: &[u8] = b"GstAudioResampler.filter-oversample\0"; pub const GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR: &[u8] = b"GstAudioResampler.max-phase-error\0"; pub const GST_AUDIO_RESAMPLER_OPT_N_TAPS: &[u8] = b"GstAudioResampler.n-taps\0"; pub const GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION: &[u8] = b"GstAudioResampler.stop-attenutation\0"; pub const GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH: &[u8] = b"GstAudioResampler.transition-bandwidth\0"; pub const GST_AUDIO_RESAMPLER_QUALITY_DEFAULT: c_int = 4; pub const GST_AUDIO_RESAMPLER_QUALITY_MAX: c_int = 10; pub const GST_AUDIO_RESAMPLER_QUALITY_MIN: c_int = 0; pub const GST_DSD_FORMATS_ALL: &[u8] = b"{ DSDU32BE, DSDU16BE, DSDU8, DSDU32LE, DSDU16LE }\0"; pub const GST_DSD_MEDIA_TYPE: &[u8] = b"audio/x-dsd\0"; pub const GST_DSD_SILENCE_PATTERN_BYTE: c_int = 105; pub const GST_META_TAG_AUDIO_CHANNELS_STR: &[u8] = b"channels\0"; pub const GST_META_TAG_AUDIO_RATE_STR: &[u8] = b"rate\0"; pub const GST_META_TAG_AUDIO_STR: &[u8] = b"audio\0"; pub const GST_META_TAG_DSD_PLANE_OFFSETS_STR: &[u8] = b"dsdplaneoffsets\0"; // Flags pub type GstAudioChannelMixerFlags = c_uint; pub const GST_AUDIO_CHANNEL_MIXER_FLAGS_NONE: GstAudioChannelMixerFlags = 0; pub const GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN: GstAudioChannelMixerFlags = 1; pub const GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT: GstAudioChannelMixerFlags = 2; pub const GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN: GstAudioChannelMixerFlags = 4; pub const GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT: GstAudioChannelMixerFlags = 8; pub type GstAudioConverterFlags = c_uint; pub const GST_AUDIO_CONVERTER_FLAG_NONE: GstAudioConverterFlags = 0; pub const GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE: GstAudioConverterFlags = 1; pub const GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE: GstAudioConverterFlags = 2; pub type GstAudioFlags = c_uint; pub const GST_AUDIO_FLAG_NONE: GstAudioFlags = 0; pub const GST_AUDIO_FLAG_UNPOSITIONED: GstAudioFlags = 1; pub type GstAudioFormatFlags = c_uint; pub const GST_AUDIO_FORMAT_FLAG_INTEGER: GstAudioFormatFlags = 1; pub const GST_AUDIO_FORMAT_FLAG_FLOAT: GstAudioFormatFlags = 2; pub const GST_AUDIO_FORMAT_FLAG_SIGNED: GstAudioFormatFlags = 4; pub const GST_AUDIO_FORMAT_FLAG_COMPLEX: GstAudioFormatFlags = 16; pub const GST_AUDIO_FORMAT_FLAG_UNPACK: GstAudioFormatFlags = 32; pub type GstAudioPackFlags = c_uint; pub const GST_AUDIO_PACK_FLAG_NONE: GstAudioPackFlags = 0; pub const GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE: GstAudioPackFlags = 1; pub type GstAudioQuantizeFlags = c_uint; pub const GST_AUDIO_QUANTIZE_FLAG_NONE: GstAudioQuantizeFlags = 0; pub const GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED: GstAudioQuantizeFlags = 1; pub type GstAudioResamplerFlags = c_uint; pub const GST_AUDIO_RESAMPLER_FLAG_NONE: GstAudioResamplerFlags = 0; pub const GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN: GstAudioResamplerFlags = 1; pub const GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT: GstAudioResamplerFlags = 2; pub const GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE: GstAudioResamplerFlags = 4; // Unions #[derive(Copy, Clone)] #[repr(C)] pub union GstAudioRingBufferSpec_ABI { pub abi: GstAudioRingBufferSpec_ABI_abi, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioRingBufferSpec_ABI { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioRingBufferSpec_ABI @ {self:p}")) .field("abi", unsafe { &self.abi }) .finish() } } // Callbacks pub type GstAudioBaseSinkCustomSlavingCallback = Option< unsafe extern "C" fn( *mut GstAudioBaseSink, gst::GstClockTime, gst::GstClockTime, *mut gst::GstClockTimeDiff, GstAudioBaseSinkDiscontReason, gpointer, ), >; pub type GstAudioClockGetTimeFunc = Option gst::GstClockTime>; pub type GstAudioFormatPack = Option< unsafe extern "C" fn( *const GstAudioFormatInfo, GstAudioPackFlags, gconstpointer, gpointer, c_int, ), >; pub type GstAudioFormatUnpack = Option< unsafe extern "C" fn( *const GstAudioFormatInfo, GstAudioPackFlags, gpointer, gconstpointer, c_int, ), >; pub type GstAudioRingBufferCallback = Option; // Records #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioAggregatorClass { pub parent_class: gst_base::GstAggregatorClass, pub create_output_buffer: Option *mut gst::GstBuffer>, pub aggregate_one_buffer: Option< unsafe extern "C" fn( *mut GstAudioAggregator, *mut GstAudioAggregatorPad, *mut gst::GstBuffer, c_uint, *mut gst::GstBuffer, c_uint, c_uint, ) -> gboolean, >, pub _gst_reserved: [gpointer; 20], } impl ::std::fmt::Debug for GstAudioAggregatorClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioAggregatorClass @ {self:p}")) .field("parent_class", &self.parent_class) .field("create_output_buffer", &self.create_output_buffer) .field("aggregate_one_buffer", &self.aggregate_one_buffer) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioAggregatorConvertPadClass { pub parent_class: GstAudioAggregatorPadClass, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioAggregatorConvertPadClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioAggregatorConvertPadClass @ {self:p}")) .field("parent_class", &self.parent_class) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioAggregatorConvertPadPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioAggregatorConvertPadPrivate = _GstAudioAggregatorConvertPadPrivate; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioAggregatorPadClass { pub parent_class: gst_base::GstAggregatorPadClass, pub convert_buffer: Option< unsafe extern "C" fn( *mut GstAudioAggregatorPad, *mut GstAudioInfo, *mut GstAudioInfo, *mut gst::GstBuffer, ) -> *mut gst::GstBuffer, >, pub update_conversion_info: Option, pub _gst_reserved: [gpointer; 20], } impl ::std::fmt::Debug for GstAudioAggregatorPadClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioAggregatorPadClass @ {self:p}")) .field("parent_class", &self.parent_class) .field("convert_buffer", &self.convert_buffer) .field("update_conversion_info", &self.update_conversion_info) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioAggregatorPadPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioAggregatorPadPrivate = _GstAudioAggregatorPadPrivate; #[repr(C)] #[allow(dead_code)] pub struct _GstAudioAggregatorPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioAggregatorPrivate = _GstAudioAggregatorPrivate; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioBaseSinkClass { pub parent_class: gst_base::GstBaseSinkClass, pub create_ringbuffer: Option *mut GstAudioRingBuffer>, pub payload: Option< unsafe extern "C" fn(*mut GstAudioBaseSink, *mut gst::GstBuffer) -> *mut gst::GstBuffer, >, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioBaseSinkClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioBaseSinkClass @ {self:p}")) .field("parent_class", &self.parent_class) .field("create_ringbuffer", &self.create_ringbuffer) .field("payload", &self.payload) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioBaseSinkPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioBaseSinkPrivate = _GstAudioBaseSinkPrivate; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioBaseSrcClass { pub parent_class: gst_base::GstPushSrcClass, pub create_ringbuffer: Option *mut GstAudioRingBuffer>, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioBaseSrcClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioBaseSrcClass @ {self:p}")) .field("parent_class", &self.parent_class) .field("create_ringbuffer", &self.create_ringbuffer) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioBaseSrcPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioBaseSrcPrivate = _GstAudioBaseSrcPrivate; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioBuffer { pub info: GstAudioInfo, pub n_samples: size_t, pub n_planes: c_int, pub planes: *mut gpointer, pub buffer: *mut gst::GstBuffer, pub map_infos: *mut gst::GstMapInfo, pub priv_planes_arr: [gpointer; 8], pub priv_map_infos_arr: [gst::GstMapInfo; 8], pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioBuffer { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioBuffer @ {self:p}")) .field("info", &self.info) .field("n_samples", &self.n_samples) .field("n_planes", &self.n_planes) .field("planes", &self.planes) .field("buffer", &self.buffer) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioCdSrcClass { pub pushsrc_class: gst_base::GstPushSrcClass, pub open: Option gboolean>, pub close: Option, pub read_sector: Option *mut gst::GstBuffer>, pub _gst_reserved: [gpointer; 20], } impl ::std::fmt::Debug for GstAudioCdSrcClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioCdSrcClass @ {self:p}")) .field("pushsrc_class", &self.pushsrc_class) .field("open", &self.open) .field("close", &self.close) .field("read_sector", &self.read_sector) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioCdSrcPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioCdSrcPrivate = _GstAudioCdSrcPrivate; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioCdSrcTrack { pub is_audio: gboolean, pub num: c_uint, pub start: c_uint, pub end: c_uint, pub tags: *mut gst::GstTagList, pub _gst_reserved1: [c_uint; 2], pub _gst_reserved2: [gpointer; 2], } impl ::std::fmt::Debug for GstAudioCdSrcTrack { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioCdSrcTrack @ {self:p}")) .field("is_audio", &self.is_audio) .field("num", &self.num) .field("start", &self.start) .field("end", &self.end) .field("tags", &self.tags) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioChannelMixer { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioChannelMixer = _GstAudioChannelMixer; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioClippingMeta { pub meta: gst::GstMeta, pub format: gst::GstFormat, pub start: u64, pub end: u64, } impl ::std::fmt::Debug for GstAudioClippingMeta { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioClippingMeta @ {self:p}")) .field("meta", &self.meta) .field("format", &self.format) .field("start", &self.start) .field("end", &self.end) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioClockClass { pub parent_class: gst::GstSystemClockClass, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioClockClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioClockClass @ {self:p}")) .field("parent_class", &self.parent_class) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct GstAudioConverter { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } impl ::std::fmt::Debug for GstAudioConverter { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioConverter @ {self:p}")) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioDecoderClass { pub element_class: gst::GstElementClass, pub start: Option gboolean>, pub stop: Option gboolean>, pub set_format: Option gboolean>, pub parse: Option< unsafe extern "C" fn( *mut GstAudioDecoder, *mut gst_base::GstAdapter, *mut c_int, *mut c_int, ) -> gst::GstFlowReturn, >, pub handle_frame: Option< unsafe extern "C" fn(*mut GstAudioDecoder, *mut *mut gst::GstBuffer) -> gst::GstFlowReturn, >, pub flush: Option, pub pre_push: Option< unsafe extern "C" fn(*mut GstAudioDecoder, *mut *mut gst::GstBuffer) -> gst::GstFlowReturn, >, pub sink_event: Option gboolean>, pub src_event: Option gboolean>, pub open: Option gboolean>, pub close: Option gboolean>, pub negotiate: Option gboolean>, pub decide_allocation: Option gboolean>, pub propose_allocation: Option gboolean>, pub sink_query: Option gboolean>, pub src_query: Option gboolean>, pub getcaps: Option *mut gst::GstCaps>, pub transform_meta: Option< unsafe extern "C" fn( *mut GstAudioDecoder, *mut gst::GstBuffer, *mut gst::GstMeta, *mut gst::GstBuffer, ) -> gboolean, >, pub _gst_reserved: [gpointer; 16], } impl ::std::fmt::Debug for GstAudioDecoderClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioDecoderClass @ {self:p}")) .field("element_class", &self.element_class) .field("start", &self.start) .field("stop", &self.stop) .field("set_format", &self.set_format) .field("parse", &self.parse) .field("handle_frame", &self.handle_frame) .field("flush", &self.flush) .field("pre_push", &self.pre_push) .field("sink_event", &self.sink_event) .field("src_event", &self.src_event) .field("open", &self.open) .field("close", &self.close) .field("negotiate", &self.negotiate) .field("decide_allocation", &self.decide_allocation) .field("propose_allocation", &self.propose_allocation) .field("sink_query", &self.sink_query) .field("src_query", &self.src_query) .field("getcaps", &self.getcaps) .field("transform_meta", &self.transform_meta) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioDecoderPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioDecoderPrivate = _GstAudioDecoderPrivate; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioDownmixMeta { pub meta: gst::GstMeta, pub from_position: *mut GstAudioChannelPosition, pub to_position: *mut GstAudioChannelPosition, pub from_channels: c_int, pub to_channels: c_int, pub matrix: *mut *mut c_float, } impl ::std::fmt::Debug for GstAudioDownmixMeta { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioDownmixMeta @ {self:p}")) .field("meta", &self.meta) .field("from_position", &self.from_position) .field("to_position", &self.to_position) .field("from_channels", &self.from_channels) .field("to_channels", &self.to_channels) .field("matrix", &self.matrix) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioEncoderClass { pub element_class: gst::GstElementClass, pub start: Option gboolean>, pub stop: Option gboolean>, pub set_format: Option gboolean>, pub handle_frame: Option< unsafe extern "C" fn(*mut GstAudioEncoder, *mut *mut gst::GstBuffer) -> gst::GstFlowReturn, >, pub flush: Option, pub pre_push: Option< unsafe extern "C" fn(*mut GstAudioEncoder, *mut *mut gst::GstBuffer) -> gst::GstFlowReturn, >, pub sink_event: Option gboolean>, pub src_event: Option gboolean>, pub getcaps: Option *mut gst::GstCaps>, pub open: Option gboolean>, pub close: Option gboolean>, pub negotiate: Option gboolean>, pub decide_allocation: Option gboolean>, pub propose_allocation: Option gboolean>, pub transform_meta: Option< unsafe extern "C" fn( *mut GstAudioEncoder, *mut gst::GstBuffer, *mut gst::GstMeta, *mut gst::GstBuffer, ) -> gboolean, >, pub sink_query: Option gboolean>, pub src_query: Option gboolean>, pub _gst_reserved: [gpointer; 17], } impl ::std::fmt::Debug for GstAudioEncoderClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioEncoderClass @ {self:p}")) .field("element_class", &self.element_class) .field("start", &self.start) .field("stop", &self.stop) .field("set_format", &self.set_format) .field("handle_frame", &self.handle_frame) .field("flush", &self.flush) .field("pre_push", &self.pre_push) .field("sink_event", &self.sink_event) .field("src_event", &self.src_event) .field("getcaps", &self.getcaps) .field("open", &self.open) .field("close", &self.close) .field("negotiate", &self.negotiate) .field("decide_allocation", &self.decide_allocation) .field("propose_allocation", &self.propose_allocation) .field("transform_meta", &self.transform_meta) .field("sink_query", &self.sink_query) .field("src_query", &self.src_query) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioEncoderPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioEncoderPrivate = _GstAudioEncoderPrivate; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioFilterClass { pub basetransformclass: gst_base::GstBaseTransformClass, pub setup: Option gboolean>, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioFilterClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioFilterClass @ {self:p}")) .field("basetransformclass", &self.basetransformclass) .field("setup", &self.setup) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioFormatInfo { pub format: GstAudioFormat, pub name: *const c_char, pub description: *const c_char, pub flags: GstAudioFormatFlags, pub endianness: c_int, pub width: c_int, pub depth: c_int, pub silence: [u8; 8], pub unpack_format: GstAudioFormat, pub unpack_func: GstAudioFormatUnpack, pub pack_func: GstAudioFormatPack, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioFormatInfo { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioFormatInfo @ {self:p}")) .field("format", &self.format) .field("name", &self.name) .field("description", &self.description) .field("flags", &self.flags) .field("endianness", &self.endianness) .field("width", &self.width) .field("depth", &self.depth) .field("silence", &self.silence) .field("unpack_format", &self.unpack_format) .field("unpack_func", &self.unpack_func) .field("pack_func", &self.pack_func) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioInfo { pub finfo: *const GstAudioFormatInfo, pub flags: GstAudioFlags, pub layout: GstAudioLayout, pub rate: c_int, pub channels: c_int, pub bpf: c_int, pub position: [GstAudioChannelPosition; 64], pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioInfo { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioInfo @ {self:p}")) .field("finfo", &self.finfo) .field("flags", &self.flags) .field("layout", &self.layout) .field("rate", &self.rate) .field("channels", &self.channels) .field("bpf", &self.bpf) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioLevelMeta { pub meta: gst::GstMeta, pub level: u8, pub voice_activity: gboolean, } impl ::std::fmt::Debug for GstAudioLevelMeta { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioLevelMeta @ {self:p}")) .field("meta", &self.meta) .field("level", &self.level) .field("voice_activity", &self.voice_activity) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioMeta { pub meta: gst::GstMeta, pub info: GstAudioInfo, pub samples: size_t, pub offsets: *mut size_t, pub priv_offsets_arr: [size_t; 8], pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioMeta { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioMeta @ {self:p}")) .field("meta", &self.meta) .field("info", &self.info) .field("samples", &self.samples) .field("offsets", &self.offsets) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioQuantize { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioQuantize = _GstAudioQuantize; #[repr(C)] #[allow(dead_code)] pub struct _GstAudioResampler { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioResampler = _GstAudioResampler; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioRingBufferClass { pub parent_class: gst::GstObjectClass, pub open_device: Option gboolean>, pub acquire: Option< unsafe extern "C" fn(*mut GstAudioRingBuffer, *mut GstAudioRingBufferSpec) -> gboolean, >, pub release: Option gboolean>, pub close_device: Option gboolean>, pub start: Option gboolean>, pub pause: Option gboolean>, pub resume: Option gboolean>, pub stop: Option gboolean>, pub delay: Option c_uint>, pub activate: Option gboolean>, pub commit: Option< unsafe extern "C" fn( *mut GstAudioRingBuffer, *mut u64, *mut u8, c_int, c_int, *mut c_int, ) -> c_uint, >, pub clear_all: Option, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioRingBufferClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioRingBufferClass @ {self:p}")) .field("parent_class", &self.parent_class) .field("open_device", &self.open_device) .field("acquire", &self.acquire) .field("release", &self.release) .field("close_device", &self.close_device) .field("start", &self.start) .field("pause", &self.pause) .field("resume", &self.resume) .field("stop", &self.stop) .field("delay", &self.delay) .field("activate", &self.activate) .field("commit", &self.commit) .field("clear_all", &self.clear_all) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct _GstAudioRingBufferPrivate { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } pub type GstAudioRingBufferPrivate = _GstAudioRingBufferPrivate; #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioRingBufferSpec { pub caps: *mut gst::GstCaps, pub type_: GstAudioRingBufferFormatType, pub info: GstAudioInfo, pub latency_time: u64, pub buffer_time: u64, pub segsize: c_int, pub segtotal: c_int, pub seglatency: c_int, pub ABI: GstAudioRingBufferSpec_ABI, } impl ::std::fmt::Debug for GstAudioRingBufferSpec { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioRingBufferSpec @ {self:p}")) .field("caps", &self.caps) .field("type_", &self.type_) .field("info", &self.info) .field("latency_time", &self.latency_time) .field("buffer_time", &self.buffer_time) .field("segsize", &self.segsize) .field("segtotal", &self.segtotal) .field("seglatency", &self.seglatency) .field("ABI", &self.ABI) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioRingBufferSpec_ABI_abi { pub dsd_format: GstDsdFormat, } impl ::std::fmt::Debug for GstAudioRingBufferSpec_ABI_abi { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioRingBufferSpec_ABI_abi @ {self:p}")) .field("dsd_format", &self.dsd_format) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioSinkClass { pub parent_class: GstAudioBaseSinkClass, pub open: Option gboolean>, pub prepare: Option gboolean>, pub unprepare: Option gboolean>, pub close: Option gboolean>, pub write: Option c_int>, pub delay: Option c_uint>, pub reset: Option, pub pause: Option, pub resume: Option, pub stop: Option, pub extension: *mut GstAudioSinkClassExtension, } impl ::std::fmt::Debug for GstAudioSinkClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioSinkClass @ {self:p}")) .field("parent_class", &self.parent_class) .field("open", &self.open) .field("prepare", &self.prepare) .field("unprepare", &self.unprepare) .field("close", &self.close) .field("write", &self.write) .field("delay", &self.delay) .field("reset", &self.reset) .field("pause", &self.pause) .field("resume", &self.resume) .field("stop", &self.stop) .field("extension", &self.extension) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioSinkClassExtension { pub clear_all: Option, } impl ::std::fmt::Debug for GstAudioSinkClassExtension { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioSinkClassExtension @ {self:p}")) .field("clear_all", &self.clear_all) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioSrcClass { pub parent_class: GstAudioBaseSrcClass, pub open: Option gboolean>, pub prepare: Option gboolean>, pub unprepare: Option gboolean>, pub close: Option gboolean>, pub read: Option< unsafe extern "C" fn(*mut GstAudioSrc, gpointer, c_uint, *mut gst::GstClockTime) -> c_uint, >, pub delay: Option c_uint>, pub reset: Option, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioSrcClass { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioSrcClass @ {self:p}")) .field("parent_class", &self.parent_class) .field("open", &self.open) .field("prepare", &self.prepare) .field("unprepare", &self.unprepare) .field("close", &self.close) .field("read", &self.read) .field("delay", &self.delay) .field("reset", &self.reset) .finish() } } #[repr(C)] #[allow(dead_code)] pub struct GstAudioStreamAlign { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } impl ::std::fmt::Debug for GstAudioStreamAlign { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioStreamAlign @ {self:p}")) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstDsdInfo { pub format: GstDsdFormat, pub rate: c_int, pub channels: c_int, pub layout: GstAudioLayout, pub reversed_bytes: gboolean, pub positions: [GstAudioChannelPosition; 64], pub flags: GstAudioFlags, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstDsdInfo { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstDsdInfo @ {self:p}")) .field("format", &self.format) .field("rate", &self.rate) .field("channels", &self.channels) .field("layout", &self.layout) .field("reversed_bytes", &self.reversed_bytes) .field("flags", &self.flags) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstDsdPlaneOffsetMeta { pub meta: gst::GstMeta, pub num_channels: c_int, pub num_bytes_per_channel: size_t, pub offsets: *mut size_t, pub priv_offsets_arr: [size_t; 8], pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstDsdPlaneOffsetMeta { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstDsdPlaneOffsetMeta @ {self:p}")) .field("meta", &self.meta) .field("num_channels", &self.num_channels) .field("num_bytes_per_channel", &self.num_bytes_per_channel) .field("offsets", &self.offsets) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstStreamVolumeInterface { pub iface: gobject::GTypeInterface, } impl ::std::fmt::Debug for GstStreamVolumeInterface { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstStreamVolumeInterface @ {self:p}")) .field("iface", &self.iface) .finish() } } // Classes #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioAggregator { pub parent: gst_base::GstAggregator, pub current_caps: *mut gst::GstCaps, pub priv_: *mut GstAudioAggregatorPrivate, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioAggregator { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioAggregator @ {self:p}")) .field("parent", &self.parent) .field("current_caps", &self.current_caps) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioAggregatorConvertPad { pub parent: GstAudioAggregatorPad, pub priv_: *mut GstAudioAggregatorConvertPadPrivate, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioAggregatorConvertPad { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioAggregatorConvertPad @ {self:p}")) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioAggregatorPad { pub parent: gst_base::GstAggregatorPad, pub info: GstAudioInfo, pub priv_: *mut GstAudioAggregatorPadPrivate, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioAggregatorPad { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioAggregatorPad @ {self:p}")) .field("parent", &self.parent) .field("info", &self.info) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioBaseSink { pub element: gst_base::GstBaseSink, pub ringbuffer: *mut GstAudioRingBuffer, pub buffer_time: u64, pub latency_time: u64, pub next_sample: u64, pub provided_clock: *mut gst::GstClock, pub eos_rendering: gboolean, pub priv_: *mut GstAudioBaseSinkPrivate, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioBaseSink { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioBaseSink @ {self:p}")) .field("element", &self.element) .field("ringbuffer", &self.ringbuffer) .field("buffer_time", &self.buffer_time) .field("latency_time", &self.latency_time) .field("next_sample", &self.next_sample) .field("provided_clock", &self.provided_clock) .field("eos_rendering", &self.eos_rendering) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioBaseSrc { pub element: gst_base::GstPushSrc, pub ringbuffer: *mut GstAudioRingBuffer, pub buffer_time: gst::GstClockTime, pub latency_time: gst::GstClockTime, pub next_sample: u64, pub clock: *mut gst::GstClock, pub priv_: *mut GstAudioBaseSrcPrivate, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioBaseSrc { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioBaseSrc @ {self:p}")) .field("element", &self.element) .field("ringbuffer", &self.ringbuffer) .field("buffer_time", &self.buffer_time) .field("latency_time", &self.latency_time) .field("next_sample", &self.next_sample) .field("clock", &self.clock) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioCdSrc { pub pushsrc: gst_base::GstPushSrc, pub tags: *mut gst::GstTagList, pub priv_: *mut GstAudioCdSrcPrivate, pub _gst_reserved1: [c_uint; 2], pub _gst_reserved2: [gpointer; 2], } impl ::std::fmt::Debug for GstAudioCdSrc { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioCdSrc @ {self:p}")) .field("pushsrc", &self.pushsrc) .field("tags", &self.tags) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioClock { pub clock: gst::GstSystemClock, pub func: GstAudioClockGetTimeFunc, pub user_data: gpointer, pub destroy_notify: glib::GDestroyNotify, pub last_time: gst::GstClockTime, pub time_offset: gst::GstClockTimeDiff, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioClock { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioClock @ {self:p}")) .field("clock", &self.clock) .field("func", &self.func) .field("user_data", &self.user_data) .field("destroy_notify", &self.destroy_notify) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioDecoder { pub element: gst::GstElement, pub sinkpad: *mut gst::GstPad, pub srcpad: *mut gst::GstPad, pub stream_lock: glib::GRecMutex, pub input_segment: gst::GstSegment, pub output_segment: gst::GstSegment, pub priv_: *mut GstAudioDecoderPrivate, pub _gst_reserved: [gpointer; 20], } impl ::std::fmt::Debug for GstAudioDecoder { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioDecoder @ {self:p}")) .field("element", &self.element) .field("sinkpad", &self.sinkpad) .field("srcpad", &self.srcpad) .field("stream_lock", &self.stream_lock) .field("input_segment", &self.input_segment) .field("output_segment", &self.output_segment) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioEncoder { pub element: gst::GstElement, pub sinkpad: *mut gst::GstPad, pub srcpad: *mut gst::GstPad, pub stream_lock: glib::GRecMutex, pub input_segment: gst::GstSegment, pub output_segment: gst::GstSegment, pub priv_: *mut GstAudioEncoderPrivate, pub _gst_reserved: [gpointer; 20], } impl ::std::fmt::Debug for GstAudioEncoder { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioEncoder @ {self:p}")) .field("element", &self.element) .field("sinkpad", &self.sinkpad) .field("srcpad", &self.srcpad) .field("stream_lock", &self.stream_lock) .field("input_segment", &self.input_segment) .field("output_segment", &self.output_segment) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioFilter { pub basetransform: gst_base::GstBaseTransform, pub info: GstAudioInfo, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioFilter { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioFilter @ {self:p}")) .field("basetransform", &self.basetransform) .field("info", &self.info) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioRingBuffer { pub object: gst::GstObject, pub cond: glib::GCond, pub open: gboolean, pub acquired: gboolean, pub memory: *mut u8, pub size: size_t, pub timestamps: *mut gst::GstClockTime, pub spec: GstAudioRingBufferSpec, pub samples_per_seg: c_int, pub empty_seg: *mut u8, pub state: c_int, pub segdone: c_int, pub segbase: c_int, pub waiting: c_int, pub callback: GstAudioRingBufferCallback, pub cb_data: gpointer, pub need_reorder: gboolean, pub channel_reorder_map: [c_int; 64], pub flushing: gboolean, pub may_start: c_int, pub active: gboolean, pub cb_data_notify: glib::GDestroyNotify, pub priv_: *mut GstAudioRingBufferPrivate, pub _gst_reserved: [gpointer; 2], } impl ::std::fmt::Debug for GstAudioRingBuffer { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioRingBuffer @ {self:p}")) .field("object", &self.object) .field("cond", &self.cond) .field("open", &self.open) .field("acquired", &self.acquired) .field("memory", &self.memory) .field("size", &self.size) .field("spec", &self.spec) .field("samples_per_seg", &self.samples_per_seg) .field("empty_seg", &self.empty_seg) .field("state", &self.state) .field("segdone", &self.segdone) .field("segbase", &self.segbase) .field("waiting", &self.waiting) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioSink { pub element: GstAudioBaseSink, pub thread: *mut glib::GThread, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioSink { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioSink @ {self:p}")) .field("element", &self.element) .finish() } } #[derive(Copy, Clone)] #[repr(C)] pub struct GstAudioSrc { pub element: GstAudioBaseSrc, pub thread: *mut glib::GThread, pub _gst_reserved: [gpointer; 4], } impl ::std::fmt::Debug for GstAudioSrc { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { f.debug_struct(&format!("GstAudioSrc @ {self:p}")) .field("element", &self.element) .finish() } } // Interfaces #[repr(C)] #[allow(dead_code)] pub struct GstStreamVolume { _data: [u8; 0], _marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>, } impl ::std::fmt::Debug for GstStreamVolume { fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result { write!(f, "GstStreamVolume @ {self:p}") } } extern "C" { //========================================================================= // GstAudioBaseSinkDiscontReason //========================================================================= pub fn gst_audio_base_sink_discont_reason_get_type() -> GType; //========================================================================= // GstAudioBaseSinkSlaveMethod //========================================================================= pub fn gst_audio_base_sink_slave_method_get_type() -> GType; //========================================================================= // GstAudioBaseSrcSlaveMethod //========================================================================= pub fn gst_audio_base_src_slave_method_get_type() -> GType; //========================================================================= // GstAudioCdSrcMode //========================================================================= pub fn gst_audio_cd_src_mode_get_type() -> GType; //========================================================================= // GstAudioChannelPosition //========================================================================= pub fn gst_audio_channel_position_get_type() -> GType; //========================================================================= // GstAudioDitherMethod //========================================================================= pub fn gst_audio_dither_method_get_type() -> GType; //========================================================================= // GstAudioFormat //========================================================================= pub fn gst_audio_format_get_type() -> GType; pub fn gst_audio_format_build_integer( sign: gboolean, endianness: c_int, width: c_int, depth: c_int, ) -> GstAudioFormat; pub fn gst_audio_format_fill_silence( info: *const GstAudioFormatInfo, dest: gpointer, length: size_t, ); pub fn gst_audio_format_from_string(format: *const c_char) -> GstAudioFormat; pub fn gst_audio_format_get_info(format: GstAudioFormat) -> *const GstAudioFormatInfo; pub fn gst_audio_format_to_string(format: GstAudioFormat) -> *const c_char; //========================================================================= // GstAudioLayout //========================================================================= pub fn gst_audio_layout_get_type() -> GType; //========================================================================= // GstAudioNoiseShapingMethod //========================================================================= pub fn gst_audio_noise_shaping_method_get_type() -> GType; //========================================================================= // GstAudioResamplerFilterInterpolation //========================================================================= pub fn gst_audio_resampler_filter_interpolation_get_type() -> GType; //========================================================================= // GstAudioResamplerFilterMode //========================================================================= pub fn gst_audio_resampler_filter_mode_get_type() -> GType; //========================================================================= // GstAudioResamplerMethod //========================================================================= pub fn gst_audio_resampler_method_get_type() -> GType; //========================================================================= // GstAudioRingBufferFormatType //========================================================================= pub fn gst_audio_ring_buffer_format_type_get_type() -> GType; //========================================================================= // GstAudioRingBufferState //========================================================================= pub fn gst_audio_ring_buffer_state_get_type() -> GType; //========================================================================= // GstDsdFormat //========================================================================= #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_format_get_type() -> GType; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_format_from_string(str: *const c_char) -> GstDsdFormat; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_format_get_width(format: GstDsdFormat) -> c_uint; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_format_to_string(format: GstDsdFormat) -> *const c_char; //========================================================================= // GstAudioChannelMixerFlags //========================================================================= pub fn gst_audio_channel_mixer_flags_get_type() -> GType; //========================================================================= // GstAudioConverterFlags //========================================================================= pub fn gst_audio_converter_flags_get_type() -> GType; //========================================================================= // GstAudioFlags //========================================================================= pub fn gst_audio_flags_get_type() -> GType; //========================================================================= // GstAudioFormatFlags //========================================================================= pub fn gst_audio_format_flags_get_type() -> GType; //========================================================================= // GstAudioPackFlags //========================================================================= pub fn gst_audio_pack_flags_get_type() -> GType; //========================================================================= // GstAudioQuantizeFlags //========================================================================= pub fn gst_audio_quantize_flags_get_type() -> GType; //========================================================================= // GstAudioResamplerFlags //========================================================================= pub fn gst_audio_resampler_flags_get_type() -> GType; //========================================================================= // GstAudioBuffer //========================================================================= #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_audio_buffer_unmap(buffer: *mut GstAudioBuffer); pub fn gst_audio_buffer_clip( buffer: *mut gst::GstBuffer, segment: *const gst::GstSegment, rate: c_int, bpf: c_int, ) -> *mut gst::GstBuffer; #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_audio_buffer_map( buffer: *mut GstAudioBuffer, info: *const GstAudioInfo, gstbuffer: *mut gst::GstBuffer, flags: gst::GstMapFlags, ) -> gboolean; pub fn gst_audio_buffer_reorder_channels( buffer: *mut gst::GstBuffer, format: GstAudioFormat, channels: c_int, from: *const GstAudioChannelPosition, to: *const GstAudioChannelPosition, ) -> gboolean; #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_audio_buffer_truncate( buffer: *mut gst::GstBuffer, bpf: c_int, trim: size_t, samples: size_t, ) -> *mut gst::GstBuffer; //========================================================================= // GstAudioChannelMixer //========================================================================= pub fn gst_audio_channel_mixer_free(mix: *mut GstAudioChannelMixer); pub fn gst_audio_channel_mixer_is_passthrough(mix: *mut GstAudioChannelMixer) -> gboolean; pub fn gst_audio_channel_mixer_samples( mix: *mut GstAudioChannelMixer, in_: *const gpointer, out: *mut gpointer, samples: c_int, ); pub fn gst_audio_channel_mixer_new( flags: GstAudioChannelMixerFlags, format: GstAudioFormat, in_channels: c_int, in_position: *mut GstAudioChannelPosition, out_channels: c_int, out_position: *mut GstAudioChannelPosition, ) -> *mut GstAudioChannelMixer; pub fn gst_audio_channel_mixer_new_with_matrix( flags: GstAudioChannelMixerFlags, format: GstAudioFormat, in_channels: c_int, out_channels: c_int, matrix: *mut *mut c_float, ) -> *mut GstAudioChannelMixer; //========================================================================= // GstAudioClippingMeta //========================================================================= pub fn gst_audio_clipping_meta_get_info() -> *const gst::GstMetaInfo; //========================================================================= // GstAudioConverter //========================================================================= pub fn gst_audio_converter_get_type() -> GType; pub fn gst_audio_converter_new( flags: GstAudioConverterFlags, in_info: *mut GstAudioInfo, out_info: *mut GstAudioInfo, config: *mut gst::GstStructure, ) -> *mut GstAudioConverter; pub fn gst_audio_converter_convert( convert: *mut GstAudioConverter, flags: GstAudioConverterFlags, in_: gpointer, in_size: size_t, out: *mut u8, out_size: *mut size_t, ) -> gboolean; pub fn gst_audio_converter_free(convert: *mut GstAudioConverter); pub fn gst_audio_converter_get_config( convert: *mut GstAudioConverter, in_rate: *mut c_int, out_rate: *mut c_int, ) -> *const gst::GstStructure; pub fn gst_audio_converter_get_in_frames( convert: *mut GstAudioConverter, out_frames: size_t, ) -> size_t; pub fn gst_audio_converter_get_max_latency(convert: *mut GstAudioConverter) -> size_t; pub fn gst_audio_converter_get_out_frames( convert: *mut GstAudioConverter, in_frames: size_t, ) -> size_t; #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_audio_converter_is_passthrough(convert: *mut GstAudioConverter) -> gboolean; pub fn gst_audio_converter_reset(convert: *mut GstAudioConverter); pub fn gst_audio_converter_samples( convert: *mut GstAudioConverter, flags: GstAudioConverterFlags, in_: *mut gpointer, in_frames: size_t, out: *mut gpointer, out_frames: size_t, ) -> gboolean; pub fn gst_audio_converter_supports_inplace(convert: *mut GstAudioConverter) -> gboolean; pub fn gst_audio_converter_update_config( convert: *mut GstAudioConverter, in_rate: c_int, out_rate: c_int, config: *mut gst::GstStructure, ) -> gboolean; //========================================================================= // GstAudioDownmixMeta //========================================================================= pub fn gst_audio_downmix_meta_get_info() -> *const gst::GstMetaInfo; //========================================================================= // GstAudioFilterClass //========================================================================= pub fn gst_audio_filter_class_add_pad_templates( klass: *mut GstAudioFilterClass, allowed_caps: *mut gst::GstCaps, ); //========================================================================= // GstAudioFormatInfo //========================================================================= pub fn gst_audio_format_info_get_type() -> GType; #[cfg(feature = "v1_20")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))] pub fn gst_audio_format_info_fill_silence( info: *const GstAudioFormatInfo, dest: gpointer, length: size_t, ); //========================================================================= // GstAudioInfo //========================================================================= pub fn gst_audio_info_get_type() -> GType; pub fn gst_audio_info_new() -> *mut GstAudioInfo; #[cfg(feature = "v1_20")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))] pub fn gst_audio_info_new_from_caps(caps: *const gst::GstCaps) -> *mut GstAudioInfo; pub fn gst_audio_info_convert( info: *const GstAudioInfo, src_fmt: gst::GstFormat, src_val: i64, dest_fmt: gst::GstFormat, dest_val: *mut i64, ) -> gboolean; pub fn gst_audio_info_copy(info: *const GstAudioInfo) -> *mut GstAudioInfo; pub fn gst_audio_info_free(info: *mut GstAudioInfo); pub fn gst_audio_info_is_equal( info: *const GstAudioInfo, other: *const GstAudioInfo, ) -> gboolean; pub fn gst_audio_info_set_format( info: *mut GstAudioInfo, format: GstAudioFormat, rate: c_int, channels: c_int, position: *const [GstAudioChannelPosition; 64], ); pub fn gst_audio_info_to_caps(info: *const GstAudioInfo) -> *mut gst::GstCaps; pub fn gst_audio_info_from_caps(info: *mut GstAudioInfo, caps: *const gst::GstCaps) -> gboolean; pub fn gst_audio_info_init(info: *mut GstAudioInfo); //========================================================================= // GstAudioLevelMeta //========================================================================= #[cfg(feature = "v1_20")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))] pub fn gst_audio_level_meta_get_info() -> *const gst::GstMetaInfo; //========================================================================= // GstAudioMeta //========================================================================= #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_audio_meta_get_info() -> *const gst::GstMetaInfo; //========================================================================= // GstAudioQuantize //========================================================================= pub fn gst_audio_quantize_free(quant: *mut GstAudioQuantize); pub fn gst_audio_quantize_reset(quant: *mut GstAudioQuantize); pub fn gst_audio_quantize_samples( quant: *mut GstAudioQuantize, in_: *const gpointer, out: *mut gpointer, samples: c_uint, ); pub fn gst_audio_quantize_new( dither: GstAudioDitherMethod, ns: GstAudioNoiseShapingMethod, flags: GstAudioQuantizeFlags, format: GstAudioFormat, channels: c_uint, quantizer: c_uint, ) -> *mut GstAudioQuantize; //========================================================================= // GstAudioResampler //========================================================================= pub fn gst_audio_resampler_free(resampler: *mut GstAudioResampler); pub fn gst_audio_resampler_get_in_frames( resampler: *mut GstAudioResampler, out_frames: size_t, ) -> size_t; pub fn gst_audio_resampler_get_max_latency(resampler: *mut GstAudioResampler) -> size_t; pub fn gst_audio_resampler_get_out_frames( resampler: *mut GstAudioResampler, in_frames: size_t, ) -> size_t; pub fn gst_audio_resampler_resample( resampler: *mut GstAudioResampler, in_: *mut gpointer, in_frames: size_t, out: *mut gpointer, out_frames: size_t, ); pub fn gst_audio_resampler_reset(resampler: *mut GstAudioResampler); pub fn gst_audio_resampler_update( resampler: *mut GstAudioResampler, in_rate: c_int, out_rate: c_int, options: *mut gst::GstStructure, ) -> gboolean; pub fn gst_audio_resampler_new( method: GstAudioResamplerMethod, flags: GstAudioResamplerFlags, format: GstAudioFormat, channels: c_int, in_rate: c_int, out_rate: c_int, options: *mut gst::GstStructure, ) -> *mut GstAudioResampler; pub fn gst_audio_resampler_options_set_quality( method: GstAudioResamplerMethod, quality: c_uint, in_rate: c_int, out_rate: c_int, options: *mut gst::GstStructure, ); //========================================================================= // GstAudioStreamAlign //========================================================================= pub fn gst_audio_stream_align_get_type() -> GType; pub fn gst_audio_stream_align_new( rate: c_int, alignment_threshold: gst::GstClockTime, discont_wait: gst::GstClockTime, ) -> *mut GstAudioStreamAlign; pub fn gst_audio_stream_align_copy( align: *const GstAudioStreamAlign, ) -> *mut GstAudioStreamAlign; pub fn gst_audio_stream_align_free(align: *mut GstAudioStreamAlign); pub fn gst_audio_stream_align_get_alignment_threshold( align: *const GstAudioStreamAlign, ) -> gst::GstClockTime; pub fn gst_audio_stream_align_get_discont_wait( align: *const GstAudioStreamAlign, ) -> gst::GstClockTime; pub fn gst_audio_stream_align_get_rate(align: *const GstAudioStreamAlign) -> c_int; pub fn gst_audio_stream_align_get_samples_since_discont( align: *const GstAudioStreamAlign, ) -> u64; pub fn gst_audio_stream_align_get_timestamp_at_discont( align: *const GstAudioStreamAlign, ) -> gst::GstClockTime; pub fn gst_audio_stream_align_mark_discont(align: *mut GstAudioStreamAlign); pub fn gst_audio_stream_align_process( align: *mut GstAudioStreamAlign, discont: gboolean, timestamp: gst::GstClockTime, n_samples: c_uint, out_timestamp: *mut gst::GstClockTime, out_duration: *mut gst::GstClockTime, out_sample_position: *mut u64, ) -> gboolean; pub fn gst_audio_stream_align_set_alignment_threshold( align: *mut GstAudioStreamAlign, alignment_threshold: gst::GstClockTime, ); pub fn gst_audio_stream_align_set_discont_wait( align: *mut GstAudioStreamAlign, discont_wait: gst::GstClockTime, ); pub fn gst_audio_stream_align_set_rate(align: *mut GstAudioStreamAlign, rate: c_int); //========================================================================= // GstDsdInfo //========================================================================= #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_get_type() -> GType; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_new() -> *mut GstDsdInfo; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_new_from_caps(caps: *const gst::GstCaps) -> *mut GstDsdInfo; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_copy(info: *const GstDsdInfo) -> *mut GstDsdInfo; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_free(info: *mut GstDsdInfo); #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_is_equal(info: *const GstDsdInfo, other: *const GstDsdInfo) -> gboolean; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_set_format( info: *mut GstDsdInfo, format: GstDsdFormat, rate: c_int, channels: c_int, positions: *const [GstAudioChannelPosition; 64], ); #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_to_caps(info: *const GstDsdInfo) -> *mut gst::GstCaps; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_from_caps(info: *mut GstDsdInfo, caps: *const gst::GstCaps) -> gboolean; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_info_init(info: *mut GstDsdInfo); //========================================================================= // GstDsdPlaneOffsetMeta //========================================================================= #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_plane_offset_meta_get_info() -> *const gst::GstMetaInfo; //========================================================================= // GstAudioAggregator //========================================================================= pub fn gst_audio_aggregator_get_type() -> GType; pub fn gst_audio_aggregator_set_sink_caps( aagg: *mut GstAudioAggregator, pad: *mut GstAudioAggregatorPad, caps: *mut gst::GstCaps, ); //========================================================================= // GstAudioAggregatorConvertPad //========================================================================= pub fn gst_audio_aggregator_convert_pad_get_type() -> GType; //========================================================================= // GstAudioAggregatorPad //========================================================================= pub fn gst_audio_aggregator_pad_get_type() -> GType; //========================================================================= // GstAudioBaseSink //========================================================================= pub fn gst_audio_base_sink_get_type() -> GType; pub fn gst_audio_base_sink_create_ringbuffer( sink: *mut GstAudioBaseSink, ) -> *mut GstAudioRingBuffer; pub fn gst_audio_base_sink_get_alignment_threshold( sink: *mut GstAudioBaseSink, ) -> gst::GstClockTime; pub fn gst_audio_base_sink_get_discont_wait(sink: *mut GstAudioBaseSink) -> gst::GstClockTime; pub fn gst_audio_base_sink_get_drift_tolerance(sink: *mut GstAudioBaseSink) -> i64; pub fn gst_audio_base_sink_get_provide_clock(sink: *mut GstAudioBaseSink) -> gboolean; pub fn gst_audio_base_sink_get_slave_method( sink: *mut GstAudioBaseSink, ) -> GstAudioBaseSinkSlaveMethod; pub fn gst_audio_base_sink_report_device_failure(sink: *mut GstAudioBaseSink); pub fn gst_audio_base_sink_set_alignment_threshold( sink: *mut GstAudioBaseSink, alignment_threshold: gst::GstClockTime, ); pub fn gst_audio_base_sink_set_custom_slaving_callback( sink: *mut GstAudioBaseSink, callback: GstAudioBaseSinkCustomSlavingCallback, user_data: gpointer, notify: glib::GDestroyNotify, ); pub fn gst_audio_base_sink_set_discont_wait( sink: *mut GstAudioBaseSink, discont_wait: gst::GstClockTime, ); pub fn gst_audio_base_sink_set_drift_tolerance( sink: *mut GstAudioBaseSink, drift_tolerance: i64, ); pub fn gst_audio_base_sink_set_provide_clock(sink: *mut GstAudioBaseSink, provide: gboolean); pub fn gst_audio_base_sink_set_slave_method( sink: *mut GstAudioBaseSink, method: GstAudioBaseSinkSlaveMethod, ); //========================================================================= // GstAudioBaseSrc //========================================================================= pub fn gst_audio_base_src_get_type() -> GType; pub fn gst_audio_base_src_create_ringbuffer( src: *mut GstAudioBaseSrc, ) -> *mut GstAudioRingBuffer; pub fn gst_audio_base_src_get_provide_clock(src: *mut GstAudioBaseSrc) -> gboolean; pub fn gst_audio_base_src_get_slave_method( src: *mut GstAudioBaseSrc, ) -> GstAudioBaseSrcSlaveMethod; pub fn gst_audio_base_src_set_provide_clock(src: *mut GstAudioBaseSrc, provide: gboolean); pub fn gst_audio_base_src_set_slave_method( src: *mut GstAudioBaseSrc, method: GstAudioBaseSrcSlaveMethod, ); //========================================================================= // GstAudioCdSrc //========================================================================= pub fn gst_audio_cd_src_get_type() -> GType; pub fn gst_audio_cd_src_add_track( src: *mut GstAudioCdSrc, track: *mut GstAudioCdSrcTrack, ) -> gboolean; //========================================================================= // GstAudioClock //========================================================================= pub fn gst_audio_clock_get_type() -> GType; pub fn gst_audio_clock_new( name: *const c_char, func: GstAudioClockGetTimeFunc, user_data: gpointer, destroy_notify: glib::GDestroyNotify, ) -> *mut gst::GstClock; pub fn gst_audio_clock_adjust( clock: *mut GstAudioClock, time: gst::GstClockTime, ) -> gst::GstClockTime; pub fn gst_audio_clock_get_time(clock: *mut GstAudioClock) -> gst::GstClockTime; pub fn gst_audio_clock_invalidate(clock: *mut GstAudioClock); pub fn gst_audio_clock_reset(clock: *mut GstAudioClock, time: gst::GstClockTime); //========================================================================= // GstAudioDecoder //========================================================================= pub fn gst_audio_decoder_get_type() -> GType; pub fn gst_audio_decoder_allocate_output_buffer( dec: *mut GstAudioDecoder, size: size_t, ) -> *mut gst::GstBuffer; pub fn gst_audio_decoder_finish_frame( dec: *mut GstAudioDecoder, buf: *mut gst::GstBuffer, frames: c_int, ) -> gst::GstFlowReturn; #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_audio_decoder_finish_subframe( dec: *mut GstAudioDecoder, buf: *mut gst::GstBuffer, ) -> gst::GstFlowReturn; pub fn gst_audio_decoder_get_allocator( dec: *mut GstAudioDecoder, allocator: *mut *mut gst::GstAllocator, params: *mut gst::GstAllocationParams, ); pub fn gst_audio_decoder_get_audio_info(dec: *mut GstAudioDecoder) -> *mut GstAudioInfo; pub fn gst_audio_decoder_get_delay(dec: *mut GstAudioDecoder) -> c_int; pub fn gst_audio_decoder_get_drainable(dec: *mut GstAudioDecoder) -> gboolean; pub fn gst_audio_decoder_get_estimate_rate(dec: *mut GstAudioDecoder) -> c_int; pub fn gst_audio_decoder_get_latency( dec: *mut GstAudioDecoder, min: *mut gst::GstClockTime, max: *mut gst::GstClockTime, ); pub fn gst_audio_decoder_get_max_errors(dec: *mut GstAudioDecoder) -> c_int; pub fn gst_audio_decoder_get_min_latency(dec: *mut GstAudioDecoder) -> gst::GstClockTime; pub fn gst_audio_decoder_get_needs_format(dec: *mut GstAudioDecoder) -> gboolean; pub fn gst_audio_decoder_get_parse_state( dec: *mut GstAudioDecoder, sync: *mut gboolean, eos: *mut gboolean, ); pub fn gst_audio_decoder_get_plc(dec: *mut GstAudioDecoder) -> gboolean; pub fn gst_audio_decoder_get_plc_aware(dec: *mut GstAudioDecoder) -> c_int; pub fn gst_audio_decoder_get_tolerance(dec: *mut GstAudioDecoder) -> gst::GstClockTime; pub fn gst_audio_decoder_merge_tags( dec: *mut GstAudioDecoder, tags: *const gst::GstTagList, mode: gst::GstTagMergeMode, ); pub fn gst_audio_decoder_negotiate(dec: *mut GstAudioDecoder) -> gboolean; pub fn gst_audio_decoder_proxy_getcaps( decoder: *mut GstAudioDecoder, caps: *mut gst::GstCaps, filter: *mut gst::GstCaps, ) -> *mut gst::GstCaps; pub fn gst_audio_decoder_set_allocation_caps( dec: *mut GstAudioDecoder, allocation_caps: *mut gst::GstCaps, ); pub fn gst_audio_decoder_set_drainable(dec: *mut GstAudioDecoder, enabled: gboolean); pub fn gst_audio_decoder_set_estimate_rate(dec: *mut GstAudioDecoder, enabled: gboolean); pub fn gst_audio_decoder_set_latency( dec: *mut GstAudioDecoder, min: gst::GstClockTime, max: gst::GstClockTime, ); pub fn gst_audio_decoder_set_max_errors(dec: *mut GstAudioDecoder, num: c_int); pub fn gst_audio_decoder_set_min_latency(dec: *mut GstAudioDecoder, num: gst::GstClockTime); pub fn gst_audio_decoder_set_needs_format(dec: *mut GstAudioDecoder, enabled: gboolean); #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_audio_decoder_set_output_caps( dec: *mut GstAudioDecoder, caps: *mut gst::GstCaps, ) -> gboolean; pub fn gst_audio_decoder_set_output_format( dec: *mut GstAudioDecoder, info: *const GstAudioInfo, ) -> gboolean; pub fn gst_audio_decoder_set_plc(dec: *mut GstAudioDecoder, enabled: gboolean); pub fn gst_audio_decoder_set_plc_aware(dec: *mut GstAudioDecoder, plc: gboolean); pub fn gst_audio_decoder_set_tolerance(dec: *mut GstAudioDecoder, tolerance: gst::GstClockTime); pub fn gst_audio_decoder_set_use_default_pad_acceptcaps( decoder: *mut GstAudioDecoder, use_: gboolean, ); //========================================================================= // GstAudioEncoder //========================================================================= pub fn gst_audio_encoder_get_type() -> GType; pub fn gst_audio_encoder_allocate_output_buffer( enc: *mut GstAudioEncoder, size: size_t, ) -> *mut gst::GstBuffer; pub fn gst_audio_encoder_finish_frame( enc: *mut GstAudioEncoder, buffer: *mut gst::GstBuffer, samples: c_int, ) -> gst::GstFlowReturn; pub fn gst_audio_encoder_get_allocator( enc: *mut GstAudioEncoder, allocator: *mut *mut gst::GstAllocator, params: *mut gst::GstAllocationParams, ); pub fn gst_audio_encoder_get_audio_info(enc: *mut GstAudioEncoder) -> *mut GstAudioInfo; pub fn gst_audio_encoder_get_drainable(enc: *mut GstAudioEncoder) -> gboolean; pub fn gst_audio_encoder_get_frame_max(enc: *mut GstAudioEncoder) -> c_int; pub fn gst_audio_encoder_get_frame_samples_max(enc: *mut GstAudioEncoder) -> c_int; pub fn gst_audio_encoder_get_frame_samples_min(enc: *mut GstAudioEncoder) -> c_int; pub fn gst_audio_encoder_get_hard_min(enc: *mut GstAudioEncoder) -> gboolean; pub fn gst_audio_encoder_get_hard_resync(enc: *mut GstAudioEncoder) -> gboolean; pub fn gst_audio_encoder_get_latency( enc: *mut GstAudioEncoder, min: *mut gst::GstClockTime, max: *mut gst::GstClockTime, ); pub fn gst_audio_encoder_get_lookahead(enc: *mut GstAudioEncoder) -> c_int; pub fn gst_audio_encoder_get_mark_granule(enc: *mut GstAudioEncoder) -> gboolean; pub fn gst_audio_encoder_get_perfect_timestamp(enc: *mut GstAudioEncoder) -> gboolean; pub fn gst_audio_encoder_get_tolerance(enc: *mut GstAudioEncoder) -> gst::GstClockTime; pub fn gst_audio_encoder_merge_tags( enc: *mut GstAudioEncoder, tags: *const gst::GstTagList, mode: gst::GstTagMergeMode, ); pub fn gst_audio_encoder_negotiate(enc: *mut GstAudioEncoder) -> gboolean; pub fn gst_audio_encoder_proxy_getcaps( enc: *mut GstAudioEncoder, caps: *mut gst::GstCaps, filter: *mut gst::GstCaps, ) -> *mut gst::GstCaps; pub fn gst_audio_encoder_set_allocation_caps( enc: *mut GstAudioEncoder, allocation_caps: *mut gst::GstCaps, ); pub fn gst_audio_encoder_set_drainable(enc: *mut GstAudioEncoder, enabled: gboolean); pub fn gst_audio_encoder_set_frame_max(enc: *mut GstAudioEncoder, num: c_int); pub fn gst_audio_encoder_set_frame_samples_max(enc: *mut GstAudioEncoder, num: c_int); pub fn gst_audio_encoder_set_frame_samples_min(enc: *mut GstAudioEncoder, num: c_int); pub fn gst_audio_encoder_set_hard_min(enc: *mut GstAudioEncoder, enabled: gboolean); pub fn gst_audio_encoder_set_hard_resync(enc: *mut GstAudioEncoder, enabled: gboolean); pub fn gst_audio_encoder_set_headers(enc: *mut GstAudioEncoder, headers: *mut glib::GList); pub fn gst_audio_encoder_set_latency( enc: *mut GstAudioEncoder, min: gst::GstClockTime, max: gst::GstClockTime, ); pub fn gst_audio_encoder_set_lookahead(enc: *mut GstAudioEncoder, num: c_int); pub fn gst_audio_encoder_set_mark_granule(enc: *mut GstAudioEncoder, enabled: gboolean); pub fn gst_audio_encoder_set_output_format( enc: *mut GstAudioEncoder, caps: *mut gst::GstCaps, ) -> gboolean; pub fn gst_audio_encoder_set_perfect_timestamp(enc: *mut GstAudioEncoder, enabled: gboolean); pub fn gst_audio_encoder_set_tolerance(enc: *mut GstAudioEncoder, tolerance: gst::GstClockTime); //========================================================================= // GstAudioFilter //========================================================================= pub fn gst_audio_filter_get_type() -> GType; //========================================================================= // GstAudioRingBuffer //========================================================================= pub fn gst_audio_ring_buffer_get_type() -> GType; pub fn gst_audio_ring_buffer_debug_spec_buff(spec: *mut GstAudioRingBufferSpec); pub fn gst_audio_ring_buffer_debug_spec_caps(spec: *mut GstAudioRingBufferSpec); pub fn gst_audio_ring_buffer_parse_caps( spec: *mut GstAudioRingBufferSpec, caps: *mut gst::GstCaps, ) -> gboolean; pub fn gst_audio_ring_buffer_acquire( buf: *mut GstAudioRingBuffer, spec: *mut GstAudioRingBufferSpec, ) -> gboolean; pub fn gst_audio_ring_buffer_activate( buf: *mut GstAudioRingBuffer, active: gboolean, ) -> gboolean; pub fn gst_audio_ring_buffer_advance(buf: *mut GstAudioRingBuffer, advance: c_uint); pub fn gst_audio_ring_buffer_clear(buf: *mut GstAudioRingBuffer, segment: c_int); pub fn gst_audio_ring_buffer_clear_all(buf: *mut GstAudioRingBuffer); pub fn gst_audio_ring_buffer_close_device(buf: *mut GstAudioRingBuffer) -> gboolean; pub fn gst_audio_ring_buffer_commit( buf: *mut GstAudioRingBuffer, sample: *mut u64, data: *mut u8, in_samples: c_int, out_samples: c_int, accum: *mut c_int, ) -> c_uint; pub fn gst_audio_ring_buffer_convert( buf: *mut GstAudioRingBuffer, src_fmt: gst::GstFormat, src_val: i64, dest_fmt: gst::GstFormat, dest_val: *mut i64, ) -> gboolean; pub fn gst_audio_ring_buffer_delay(buf: *mut GstAudioRingBuffer) -> c_uint; pub fn gst_audio_ring_buffer_device_is_open(buf: *mut GstAudioRingBuffer) -> gboolean; #[cfg(feature = "v1_26")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_26")))] pub fn gst_audio_ring_buffer_get_segbase(buf: *mut GstAudioRingBuffer) -> u64; #[cfg(feature = "v1_26")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_26")))] pub fn gst_audio_ring_buffer_get_segdone(buf: *mut GstAudioRingBuffer) -> u64; pub fn gst_audio_ring_buffer_is_acquired(buf: *mut GstAudioRingBuffer) -> gboolean; pub fn gst_audio_ring_buffer_is_active(buf: *mut GstAudioRingBuffer) -> gboolean; pub fn gst_audio_ring_buffer_is_flushing(buf: *mut GstAudioRingBuffer) -> gboolean; pub fn gst_audio_ring_buffer_may_start(buf: *mut GstAudioRingBuffer, allowed: gboolean); pub fn gst_audio_ring_buffer_open_device(buf: *mut GstAudioRingBuffer) -> gboolean; pub fn gst_audio_ring_buffer_pause(buf: *mut GstAudioRingBuffer) -> gboolean; pub fn gst_audio_ring_buffer_prepare_read( buf: *mut GstAudioRingBuffer, segment: *mut c_int, readptr: *mut *mut u8, len: *mut c_int, ) -> gboolean; pub fn gst_audio_ring_buffer_read( buf: *mut GstAudioRingBuffer, sample: u64, data: *mut u8, len: c_uint, timestamp: *mut gst::GstClockTime, ) -> c_uint; pub fn gst_audio_ring_buffer_release(buf: *mut GstAudioRingBuffer) -> gboolean; pub fn gst_audio_ring_buffer_samples_done(buf: *mut GstAudioRingBuffer) -> u64; pub fn gst_audio_ring_buffer_set_callback( buf: *mut GstAudioRingBuffer, cb: GstAudioRingBufferCallback, user_data: gpointer, ); pub fn gst_audio_ring_buffer_set_callback_full( buf: *mut GstAudioRingBuffer, cb: GstAudioRingBufferCallback, user_data: gpointer, notify: glib::GDestroyNotify, ); pub fn gst_audio_ring_buffer_set_channel_positions( buf: *mut GstAudioRingBuffer, position: *const GstAudioChannelPosition, ); #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_audio_ring_buffer_set_errored(buf: *mut GstAudioRingBuffer); pub fn gst_audio_ring_buffer_set_flushing(buf: *mut GstAudioRingBuffer, flushing: gboolean); pub fn gst_audio_ring_buffer_set_sample(buf: *mut GstAudioRingBuffer, sample: u64); #[cfg(feature = "v1_26")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_26")))] pub fn gst_audio_ring_buffer_set_segdone(buf: *mut GstAudioRingBuffer, segdone: u64); pub fn gst_audio_ring_buffer_set_timestamp( buf: *mut GstAudioRingBuffer, readseg: c_int, timestamp: gst::GstClockTime, ); pub fn gst_audio_ring_buffer_start(buf: *mut GstAudioRingBuffer) -> gboolean; pub fn gst_audio_ring_buffer_stop(buf: *mut GstAudioRingBuffer) -> gboolean; //========================================================================= // GstAudioSink //========================================================================= pub fn gst_audio_sink_get_type() -> GType; //========================================================================= // GstAudioSrc //========================================================================= pub fn gst_audio_src_get_type() -> GType; //========================================================================= // GstStreamVolume //========================================================================= pub fn gst_stream_volume_get_type() -> GType; pub fn gst_stream_volume_convert_volume( from: GstStreamVolumeFormat, to: GstStreamVolumeFormat, val: c_double, ) -> c_double; pub fn gst_stream_volume_get_mute(volume: *mut GstStreamVolume) -> gboolean; pub fn gst_stream_volume_get_volume( volume: *mut GstStreamVolume, format: GstStreamVolumeFormat, ) -> c_double; pub fn gst_stream_volume_set_mute(volume: *mut GstStreamVolume, mute: gboolean); pub fn gst_stream_volume_set_volume( volume: *mut GstStreamVolume, format: GstStreamVolumeFormat, val: c_double, ); //========================================================================= // Other functions //========================================================================= pub fn gst_audio_channel_get_fallback_mask(channels: c_int) -> u64; pub fn gst_audio_channel_positions_from_mask( channels: c_int, channel_mask: u64, position: *mut GstAudioChannelPosition, ) -> gboolean; pub fn gst_audio_channel_positions_to_mask( position: *const GstAudioChannelPosition, channels: c_int, force_order: gboolean, channel_mask: *mut u64, ) -> gboolean; pub fn gst_audio_channel_positions_to_string( position: *const GstAudioChannelPosition, channels: c_int, ) -> *mut c_char; pub fn gst_audio_channel_positions_to_valid_order( position: *mut GstAudioChannelPosition, channels: c_int, ) -> gboolean; pub fn gst_audio_check_valid_channel_positions( position: *const GstAudioChannelPosition, channels: c_int, force_order: gboolean, ) -> gboolean; pub fn gst_audio_clipping_meta_api_get_type() -> GType; pub fn gst_audio_downmix_meta_api_get_type() -> GType; #[cfg(feature = "v1_18")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_18")))] pub fn gst_audio_formats_raw(len: *mut c_uint) -> *const GstAudioFormat; pub fn gst_audio_get_channel_reorder_map( channels: c_int, from: *const GstAudioChannelPosition, to: *const GstAudioChannelPosition, reorder_map: *mut c_int, ) -> gboolean; pub fn gst_audio_iec61937_frame_size(spec: *const GstAudioRingBufferSpec) -> c_uint; pub fn gst_audio_iec61937_payload( src: *const u8, src_n: c_uint, dst: *mut u8, dst_n: c_uint, spec: *const GstAudioRingBufferSpec, endianness: c_int, ) -> gboolean; #[cfg(feature = "v1_20")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))] pub fn gst_audio_level_meta_api_get_type() -> GType; #[cfg(feature = "v1_18")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_18")))] pub fn gst_audio_make_raw_caps( formats: *const GstAudioFormat, len: c_uint, layout: GstAudioLayout, ) -> *mut gst::GstCaps; #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_audio_meta_api_get_type() -> GType; pub fn gst_audio_reorder_channels( data: gpointer, size: size_t, format: GstAudioFormat, channels: c_int, from: *const GstAudioChannelPosition, to: *const GstAudioChannelPosition, ) -> gboolean; #[cfg(feature = "v1_26")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_26")))] pub fn gst_audio_reorder_channels_with_reorder_map( data: gpointer, size: size_t, bps: c_int, channels: c_int, reorder_map: *const c_int, ); pub fn gst_buffer_add_audio_clipping_meta( buffer: *mut gst::GstBuffer, format: gst::GstFormat, start: u64, end: u64, ) -> *mut GstAudioClippingMeta; pub fn gst_buffer_add_audio_downmix_meta( buffer: *mut gst::GstBuffer, from_position: *const GstAudioChannelPosition, from_channels: c_int, to_position: *const GstAudioChannelPosition, to_channels: c_int, matrix: *mut *const c_float, ) -> *mut GstAudioDownmixMeta; #[cfg(feature = "v1_20")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))] pub fn gst_buffer_add_audio_level_meta( buffer: *mut gst::GstBuffer, level: u8, voice_activity: gboolean, ) -> *mut GstAudioLevelMeta; #[cfg(feature = "v1_16")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))] pub fn gst_buffer_add_audio_meta( buffer: *mut gst::GstBuffer, info: *const GstAudioInfo, samples: size_t, offsets: *mut size_t, ) -> *mut GstAudioMeta; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_buffer_add_dsd_plane_offset_meta( buffer: *mut gst::GstBuffer, num_channels: c_int, num_bytes_per_channel: size_t, offsets: *mut size_t, ) -> *mut GstDsdPlaneOffsetMeta; pub fn gst_buffer_get_audio_downmix_meta_for_channels( buffer: *mut gst::GstBuffer, to_position: *const GstAudioChannelPosition, to_channels: c_int, ) -> *mut GstAudioDownmixMeta; #[cfg(feature = "v1_20")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))] pub fn gst_buffer_get_audio_level_meta(buffer: *mut gst::GstBuffer) -> *mut GstAudioLevelMeta; #[cfg(feature = "v1_24")] #[cfg_attr(docsrs, doc(cfg(feature = "v1_24")))] pub fn gst_dsd_convert( input_data: *const u8, output_data: *mut u8, input_format: GstDsdFormat, output_format: GstDsdFormat, input_layout: GstAudioLayout, output_layout: GstAudioLayout, input_plane_offsets: *const size_t, output_plane_offsets: *const size_t, num_dsd_bytes: size_t, num_channels: c_int, reverse_byte_bits: gboolean, ); pub fn gst_dsd_plane_offset_meta_api_get_type() -> GType; } gstreamer-audio-sys-0.23.5/tests/abi.rs000064400000000000000000000714661046102023000161000ustar 00000000000000// This file was generated by gir (https://github.com/gtk-rs/gir) // from gir-files (https://github.com/gtk-rs/gir-files) // from gst-gir-files (https://gitlab.freedesktop.org/gstreamer/gir-files-rs.git) // DO NOT EDIT #![cfg(unix)] use gstreamer_audio_sys::*; use std::env; use std::error::Error; use std::ffi::OsString; use std::mem::{align_of, size_of}; use std::path::Path; use std::process::{Command, Stdio}; use std::str; use tempfile::Builder; static PACKAGES: &[&str] = &["gstreamer-audio-1.0"]; #[derive(Clone, Debug)] struct Compiler { pub args: Vec, } impl Compiler { pub fn new() -> Result> { let mut args = get_var("CC", "cc")?; args.push("-Wno-deprecated-declarations".to_owned()); // For _Generic args.push("-std=c11".to_owned()); // For %z support in printf when using MinGW. args.push("-D__USE_MINGW_ANSI_STDIO".to_owned()); args.extend(get_var("CFLAGS", "")?); args.extend(get_var("CPPFLAGS", "")?); args.extend(pkg_config_cflags(PACKAGES)?); Ok(Self { args }) } pub fn compile(&self, src: &Path, out: &Path) -> Result<(), Box> { let mut cmd = self.to_command(); cmd.arg(src); cmd.arg("-o"); cmd.arg(out); let status = cmd.spawn()?.wait()?; if !status.success() { return Err(format!("compilation command {cmd:?} failed, {status}").into()); } Ok(()) } fn to_command(&self) -> Command { let mut cmd = Command::new(&self.args[0]); cmd.args(&self.args[1..]); cmd } } fn get_var(name: &str, default: &str) -> Result, Box> { match env::var(name) { Ok(value) => Ok(shell_words::split(&value)?), Err(env::VarError::NotPresent) => Ok(shell_words::split(default)?), Err(err) => Err(format!("{name} {err}").into()), } } fn pkg_config_cflags(packages: &[&str]) -> Result, Box> { if packages.is_empty() { return Ok(Vec::new()); } let pkg_config = env::var_os("PKG_CONFIG").unwrap_or_else(|| OsString::from("pkg-config")); let mut cmd = Command::new(pkg_config); cmd.arg("--cflags"); cmd.args(packages); cmd.stderr(Stdio::inherit()); let out = cmd.output()?; if !out.status.success() { let (status, stdout) = (out.status, String::from_utf8_lossy(&out.stdout)); return Err(format!("command {cmd:?} failed, {status:?}\nstdout: {stdout}").into()); } let stdout = str::from_utf8(&out.stdout)?; Ok(shell_words::split(stdout.trim())?) } #[derive(Copy, Clone, Debug, Eq, PartialEq)] struct Layout { size: usize, alignment: usize, } #[derive(Copy, Clone, Debug, Default, Eq, PartialEq)] struct Results { /// Number of successfully completed tests. passed: usize, /// Total number of failed tests (including those that failed to compile). failed: usize, } impl Results { fn record_passed(&mut self) { self.passed += 1; } fn record_failed(&mut self) { self.failed += 1; } fn summary(&self) -> String { format!("{} passed; {} failed", self.passed, self.failed) } fn expect_total_success(&self) { if self.failed == 0 { println!("OK: {}", self.summary()); } else { panic!("FAILED: {}", self.summary()); }; } } #[test] fn cross_validate_constants_with_c() { let mut c_constants: Vec<(String, String)> = Vec::new(); for l in get_c_output("constant").unwrap().lines() { let (name, value) = l.split_once(';').expect("Missing ';' separator"); c_constants.push((name.to_owned(), value.to_owned())); } let mut results = Results::default(); for ((rust_name, rust_value), (c_name, c_value)) in RUST_CONSTANTS.iter().zip(c_constants.iter()) { if rust_name != c_name { results.record_failed(); eprintln!("Name mismatch:\nRust: {rust_name:?}\nC: {c_name:?}"); continue; } if rust_value != c_value { results.record_failed(); eprintln!( "Constant value mismatch for {rust_name}\nRust: {rust_value:?}\nC: {c_value:?}", ); continue; } results.record_passed(); } results.expect_total_success(); } #[test] fn cross_validate_layout_with_c() { let mut c_layouts = Vec::new(); for l in get_c_output("layout").unwrap().lines() { let (name, value) = l.split_once(';').expect("Missing first ';' separator"); let (size, alignment) = value.split_once(';').expect("Missing second ';' separator"); let size = size.parse().expect("Failed to parse size"); let alignment = alignment.parse().expect("Failed to parse alignment"); c_layouts.push((name.to_owned(), Layout { size, alignment })); } let mut results = Results::default(); for ((rust_name, rust_layout), (c_name, c_layout)) in RUST_LAYOUTS.iter().zip(c_layouts.iter()) { if rust_name != c_name { results.record_failed(); eprintln!("Name mismatch:\nRust: {rust_name:?}\nC: {c_name:?}"); continue; } if rust_layout != c_layout { results.record_failed(); eprintln!("Layout mismatch for {rust_name}\nRust: {rust_layout:?}\nC: {c_layout:?}",); continue; } results.record_passed(); } results.expect_total_success(); } fn get_c_output(name: &str) -> Result> { let tmpdir = Builder::new().prefix("abi").tempdir()?; let exe = tmpdir.path().join(name); let c_file = Path::new("tests").join(name).with_extension("c"); let cc = Compiler::new().expect("configured compiler"); cc.compile(&c_file, &exe)?; let mut cmd = Command::new(exe); cmd.stderr(Stdio::inherit()); let out = cmd.output()?; if !out.status.success() { let (status, stdout) = (out.status, String::from_utf8_lossy(&out.stdout)); return Err(format!("command {cmd:?} failed, {status:?}\nstdout: {stdout}").into()); } Ok(String::from_utf8(out.stdout)?) } const RUST_LAYOUTS: &[(&str, Layout)] = &[ ( "GstAudioAggregator", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioAggregatorClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioAggregatorConvertPad", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioAggregatorConvertPadClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioAggregatorPad", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioAggregatorPadClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioBaseSink", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioBaseSinkClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioBaseSinkDiscontReason", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioBaseSinkSlaveMethod", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioBaseSrc", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioBaseSrcClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioBaseSrcSlaveMethod", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioBuffer", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioCdSrc", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioCdSrcClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioCdSrcMode", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioCdSrcTrack", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioChannelMixerFlags", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioChannelPosition", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioClippingMeta", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioClock", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioClockClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioConverterFlags", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioDecoder", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioDecoderClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioDitherMethod", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioDownmixMeta", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioEncoder", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioEncoderClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioFilter", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioFilterClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioFlags", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioFormat", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioFormatFlags", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioFormatInfo", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioInfo", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioLayout", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioLevelMeta", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioMeta", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioNoiseShapingMethod", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioPackFlags", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioQuantizeFlags", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioResamplerFilterInterpolation", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioResamplerFilterMode", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioResamplerFlags", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioResamplerMethod", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioRingBuffer", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioRingBufferClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioRingBufferFormatType", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioRingBufferSpec", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioRingBufferState", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioSink", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioSinkClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioSinkClassExtension", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioSrc", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstAudioSrcClass", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstDsdFormat", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstDsdInfo", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstDsdPlaneOffsetMeta", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstStreamVolumeFormat", Layout { size: size_of::(), alignment: align_of::(), }, ), ( "GstStreamVolumeInterface", Layout { size: size_of::(), alignment: align_of::(), }, ), ]; const RUST_CONSTANTS: &[(&str, &str)] = &[ ("(gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT", "4"), ( "(gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE", "5", ), ("(gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH", "2"), ("(gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS", "1"), ("(gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT", "0"), ( "(gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY", "3", ), ("(gint) GST_AUDIO_BASE_SINK_SLAVE_CUSTOM", "3"), ("(gint) GST_AUDIO_BASE_SINK_SLAVE_NONE", "2"), ("(gint) GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE", "0"), ("(gint) GST_AUDIO_BASE_SINK_SLAVE_SKEW", "1"), ("(gint) GST_AUDIO_BASE_SRC_SLAVE_NONE", "3"), ("(gint) GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE", "0"), ("(gint) GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP", "1"), ("(gint) GST_AUDIO_BASE_SRC_SLAVE_SKEW", "2"), ("(gint) GST_AUDIO_CD_SRC_MODE_CONTINUOUS", "1"), ("(gint) GST_AUDIO_CD_SRC_MODE_NORMAL", "0"), ("GST_AUDIO_CHANNELS_RANGE", "(int) [ 1, max ]"), ("(guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_NONE", "0"), ( "(guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN", "1", ), ( "(guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT", "2", ), ("(guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN", "4"), ( "(guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT", "8", ), ( "(gint) GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER", "21", ), ("(gint) GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT", "22"), ("(gint) GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT", "23"), ("(gint) GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER", "2"), ("(gint) GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT", "0"), ( "(gint) GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER", "6", ), ("(gint) GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT", "1"), ( "(gint) GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER", "7", ), ("(gint) GST_AUDIO_CHANNEL_POSITION_INVALID", "-1"), ("(gint) GST_AUDIO_CHANNEL_POSITION_LFE1", "3"), ("(gint) GST_AUDIO_CHANNEL_POSITION_LFE2", "9"), ("(gint) GST_AUDIO_CHANNEL_POSITION_MONO", "-2"), ("(gint) GST_AUDIO_CHANNEL_POSITION_NONE", "-3"), ("(gint) GST_AUDIO_CHANNEL_POSITION_REAR_CENTER", "8"), ("(gint) GST_AUDIO_CHANNEL_POSITION_REAR_LEFT", "4"), ("(gint) GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT", "5"), ("(gint) GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT", "10"), ("(gint) GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT", "11"), ("(gint) GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT", "26"), ("(gint) GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT", "27"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_CENTER", "15"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER", "14"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT", "12"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT", "13"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER", "20"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT", "16"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT", "17"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT", "18"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT", "19"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_SURROUND_LEFT", "28"), ("(gint) GST_AUDIO_CHANNEL_POSITION_TOP_SURROUND_RIGHT", "29"), ("(gint) GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT", "24"), ("(gint) GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT", "25"), ("(guint) GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE", "1"), ("(guint) GST_AUDIO_CONVERTER_FLAG_NONE", "0"), ("(guint) GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE", "2"), ( "GST_AUDIO_CONVERTER_OPT_DITHER_METHOD", "GstAudioConverter.dither-method", ), ( "GST_AUDIO_CONVERTER_OPT_DITHER_THRESHOLD", "GstAudioConverter.dither-threshold", ), ( "GST_AUDIO_CONVERTER_OPT_MIX_MATRIX", "GstAudioConverter.mix-matrix", ), ( "GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD", "GstAudioConverter.noise-shaping-method", ), ( "GST_AUDIO_CONVERTER_OPT_QUANTIZATION", "GstAudioConverter.quantization", ), ( "GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD", "GstAudioConverter.resampler-method", ), ("GST_AUDIO_DECODER_MAX_ERRORS", "-1"), ("GST_AUDIO_DECODER_SINK_NAME", "sink"), ("GST_AUDIO_DECODER_SRC_NAME", "src"), ("GST_AUDIO_DEF_CHANNELS", "2"), ("GST_AUDIO_DEF_FORMAT", "S16LE"), ("GST_AUDIO_DEF_RATE", "44100"), ("(gint) GST_AUDIO_DITHER_NONE", "0"), ("(gint) GST_AUDIO_DITHER_RPDF", "1"), ("(gint) GST_AUDIO_DITHER_TPDF", "2"), ("(gint) GST_AUDIO_DITHER_TPDF_HF", "3"), ("GST_AUDIO_ENCODER_SINK_NAME", "sink"), ("GST_AUDIO_ENCODER_SRC_NAME", "src"), ("(guint) GST_AUDIO_FLAG_NONE", "0"), ("(guint) GST_AUDIO_FLAG_UNPOSITIONED", "1"), ("(gint) GST_AUDIO_FORMAT_ENCODED", "1"), ("(gint) GST_AUDIO_FORMAT_F32BE", "29"), ("(gint) GST_AUDIO_FORMAT_F32LE", "28"), ("(gint) GST_AUDIO_FORMAT_F64BE", "31"), ("(gint) GST_AUDIO_FORMAT_F64LE", "30"), ("(guint) GST_AUDIO_FORMAT_FLAG_COMPLEX", "16"), ("(guint) GST_AUDIO_FORMAT_FLAG_FLOAT", "2"), ("(guint) GST_AUDIO_FORMAT_FLAG_INTEGER", "1"), ("(guint) GST_AUDIO_FORMAT_FLAG_SIGNED", "4"), ("(guint) GST_AUDIO_FORMAT_FLAG_UNPACK", "32"), ("GST_AUDIO_FORMAT_LAST", "32"), ("(gint) GST_AUDIO_FORMAT_S16BE", "5"), ("(gint) GST_AUDIO_FORMAT_S16LE", "4"), ("(gint) GST_AUDIO_FORMAT_S18BE", "25"), ("(gint) GST_AUDIO_FORMAT_S18LE", "24"), ("(gint) GST_AUDIO_FORMAT_S20BE", "21"), ("(gint) GST_AUDIO_FORMAT_S20LE", "20"), ("(gint) GST_AUDIO_FORMAT_S24BE", "17"), ("(gint) GST_AUDIO_FORMAT_S24LE", "16"), ("(gint) GST_AUDIO_FORMAT_S24_32BE", "9"), ("(gint) GST_AUDIO_FORMAT_S24_32LE", "8"), ("(gint) GST_AUDIO_FORMAT_S32BE", "13"), ("(gint) GST_AUDIO_FORMAT_S32LE", "12"), ("(gint) GST_AUDIO_FORMAT_S8", "2"), ("(gint) GST_AUDIO_FORMAT_U16BE", "7"), ("(gint) GST_AUDIO_FORMAT_U16LE", "6"), ("(gint) GST_AUDIO_FORMAT_U18BE", "27"), ("(gint) GST_AUDIO_FORMAT_U18LE", "26"), ("(gint) GST_AUDIO_FORMAT_U20BE", "23"), ("(gint) GST_AUDIO_FORMAT_U20LE", "22"), ("(gint) GST_AUDIO_FORMAT_U24BE", "19"), ("(gint) GST_AUDIO_FORMAT_U24LE", "18"), ("(gint) GST_AUDIO_FORMAT_U24_32BE", "11"), ("(gint) GST_AUDIO_FORMAT_U24_32LE", "10"), ("(gint) GST_AUDIO_FORMAT_U32BE", "15"), ("(gint) GST_AUDIO_FORMAT_U32LE", "14"), ("(gint) GST_AUDIO_FORMAT_U8", "3"), ("(gint) GST_AUDIO_FORMAT_UNKNOWN", "0"), ("(gint) GST_AUDIO_LAYOUT_INTERLEAVED", "0"), ("(gint) GST_AUDIO_LAYOUT_NON_INTERLEAVED", "1"), ("(gint) GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK", "1"), ("(gint) GST_AUDIO_NOISE_SHAPING_HIGH", "4"), ("(gint) GST_AUDIO_NOISE_SHAPING_MEDIUM", "3"), ("(gint) GST_AUDIO_NOISE_SHAPING_NONE", "0"), ("(gint) GST_AUDIO_NOISE_SHAPING_SIMPLE", "2"), ("(guint) GST_AUDIO_PACK_FLAG_NONE", "0"), ("(guint) GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE", "1"), ("(guint) GST_AUDIO_QUANTIZE_FLAG_NONE", "0"), ("(guint) GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED", "1"), ("GST_AUDIO_RATE_RANGE", "(int) [ 1, max ]"), ("(gint) GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC", "2"), ( "(gint) GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR", "1", ), ("(gint) GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE", "0"), ("(gint) GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO", "2"), ("(gint) GST_AUDIO_RESAMPLER_FILTER_MODE_FULL", "1"), ("(gint) GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED", "0"), ("(guint) GST_AUDIO_RESAMPLER_FLAG_NONE", "0"), ("(guint) GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN", "1"), ("(guint) GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT", "2"), ("(guint) GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE", "4"), ("(gint) GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL", "3"), ("(gint) GST_AUDIO_RESAMPLER_METHOD_CUBIC", "2"), ("(gint) GST_AUDIO_RESAMPLER_METHOD_KAISER", "4"), ("(gint) GST_AUDIO_RESAMPLER_METHOD_LINEAR", "1"), ("(gint) GST_AUDIO_RESAMPLER_METHOD_NEAREST", "0"), ( "GST_AUDIO_RESAMPLER_OPT_CUBIC_B", "GstAudioResampler.cubic-b", ), ( "GST_AUDIO_RESAMPLER_OPT_CUBIC_C", "GstAudioResampler.cubic-c", ), ("GST_AUDIO_RESAMPLER_OPT_CUTOFF", "GstAudioResampler.cutoff"), ( "GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION", "GstAudioResampler.filter-interpolation", ), ( "GST_AUDIO_RESAMPLER_OPT_FILTER_MODE", "GstAudioResampler.filter-mode", ), ( "GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD", "GstAudioResampler.filter-mode-threshold", ), ( "GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE", "GstAudioResampler.filter-oversample", ), ( "GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR", "GstAudioResampler.max-phase-error", ), ("GST_AUDIO_RESAMPLER_OPT_N_TAPS", "GstAudioResampler.n-taps"), ( "GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION", "GstAudioResampler.stop-attenutation", ), ( "GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH", "GstAudioResampler.transition-bandwidth", ), ("GST_AUDIO_RESAMPLER_QUALITY_DEFAULT", "4"), ("GST_AUDIO_RESAMPLER_QUALITY_MAX", "10"), ("GST_AUDIO_RESAMPLER_QUALITY_MIN", "0"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3", "7"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW", "2"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD", "15"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS", "9"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3", "8"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC", "14"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM", "5"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958", "6"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM", "3"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG", "4"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC", "10"), ( "(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW", "12", ), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC", "11"), ( "(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW", "13", ), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW", "1"), ("(gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW", "0"), ("(gint) GST_AUDIO_RING_BUFFER_STATE_ERROR", "3"), ("(gint) GST_AUDIO_RING_BUFFER_STATE_PAUSED", "1"), ("(gint) GST_AUDIO_RING_BUFFER_STATE_STARTED", "2"), ("(gint) GST_AUDIO_RING_BUFFER_STATE_STOPPED", "0"), ( "GST_DSD_FORMATS_ALL", "{ DSDU32BE, DSDU16BE, DSDU8, DSDU32LE, DSDU16LE }", ), ("(gint) GST_DSD_FORMAT_U16", "2"), ("(gint) GST_DSD_FORMAT_U16BE", "3"), ("(gint) GST_DSD_FORMAT_U16LE", "2"), ("(gint) GST_DSD_FORMAT_U32", "4"), ("(gint) GST_DSD_FORMAT_U32BE", "5"), ("(gint) GST_DSD_FORMAT_U32LE", "4"), ("(gint) GST_DSD_FORMAT_U8", "1"), ("(gint) GST_DSD_FORMAT_UNKNOWN", "0"), ("GST_DSD_MEDIA_TYPE", "audio/x-dsd"), ("GST_DSD_SILENCE_PATTERN_BYTE", "105"), ("GST_META_TAG_AUDIO_CHANNELS_STR", "channels"), ("GST_META_TAG_AUDIO_RATE_STR", "rate"), ("GST_META_TAG_AUDIO_STR", "audio"), ("GST_META_TAG_DSD_PLANE_OFFSETS_STR", "dsdplaneoffsets"), ("(gint) GST_NUM_DSD_FORMATS", "6"), ("(gint) GST_STREAM_VOLUME_FORMAT_CUBIC", "1"), ("(gint) GST_STREAM_VOLUME_FORMAT_DB", "2"), ("(gint) GST_STREAM_VOLUME_FORMAT_LINEAR", "0"), ]; gstreamer-audio-sys-0.23.5/tests/constant.c000064400000000000000000000311071046102023000167600ustar 00000000000000// This file was generated by gir (https://github.com/gtk-rs/gir) // from gir-files (https://github.com/gtk-rs/gir-files) // from gst-gir-files (https://gitlab.freedesktop.org/gstreamer/gir-files-rs.git) // DO NOT EDIT #include "manual.h" #include #define PRINT_CONSTANT(CONSTANT_NAME) \ printf("%s;", #CONSTANT_NAME); \ printf(_Generic((CONSTANT_NAME), \ char *: "%s", \ const char *: "%s", \ char: "%c", \ signed char: "%hhd", \ unsigned char: "%hhu", \ short int: "%hd", \ unsigned short int: "%hu", \ int: "%d", \ unsigned int: "%u", \ long: "%ld", \ unsigned long: "%lu", \ long long: "%lld", \ unsigned long long: "%llu", \ float: "%f", \ double: "%f", \ long double: "%ld"), \ CONSTANT_NAME); \ printf("\n"); int main() { PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_SLAVE_CUSTOM); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_SLAVE_NONE); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SINK_SLAVE_SKEW); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SRC_SLAVE_NONE); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP); PRINT_CONSTANT((gint) GST_AUDIO_BASE_SRC_SLAVE_SKEW); PRINT_CONSTANT((gint) GST_AUDIO_CD_SRC_MODE_CONTINUOUS); PRINT_CONSTANT((gint) GST_AUDIO_CD_SRC_MODE_NORMAL); PRINT_CONSTANT(GST_AUDIO_CHANNELS_RANGE); PRINT_CONSTANT((guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_NONE); PRINT_CONSTANT((guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN); PRINT_CONSTANT((guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT); PRINT_CONSTANT((guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN); PRINT_CONSTANT((guint) GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_INVALID); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_LFE1); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_LFE2); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_MONO); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_NONE); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_REAR_CENTER); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_REAR_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_CENTER); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_SURROUND_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_TOP_SURROUND_RIGHT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT); PRINT_CONSTANT((gint) GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT); PRINT_CONSTANT((guint) GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE); PRINT_CONSTANT((guint) GST_AUDIO_CONVERTER_FLAG_NONE); PRINT_CONSTANT((guint) GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE); PRINT_CONSTANT(GST_AUDIO_CONVERTER_OPT_DITHER_METHOD); PRINT_CONSTANT(GST_AUDIO_CONVERTER_OPT_DITHER_THRESHOLD); PRINT_CONSTANT(GST_AUDIO_CONVERTER_OPT_MIX_MATRIX); PRINT_CONSTANT(GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD); PRINT_CONSTANT(GST_AUDIO_CONVERTER_OPT_QUANTIZATION); PRINT_CONSTANT(GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD); PRINT_CONSTANT(GST_AUDIO_DECODER_MAX_ERRORS); PRINT_CONSTANT(GST_AUDIO_DECODER_SINK_NAME); PRINT_CONSTANT(GST_AUDIO_DECODER_SRC_NAME); PRINT_CONSTANT(GST_AUDIO_DEF_CHANNELS); PRINT_CONSTANT(GST_AUDIO_DEF_FORMAT); PRINT_CONSTANT(GST_AUDIO_DEF_RATE); PRINT_CONSTANT((gint) GST_AUDIO_DITHER_NONE); PRINT_CONSTANT((gint) GST_AUDIO_DITHER_RPDF); PRINT_CONSTANT((gint) GST_AUDIO_DITHER_TPDF); PRINT_CONSTANT((gint) GST_AUDIO_DITHER_TPDF_HF); PRINT_CONSTANT(GST_AUDIO_ENCODER_SINK_NAME); PRINT_CONSTANT(GST_AUDIO_ENCODER_SRC_NAME); PRINT_CONSTANT((guint) GST_AUDIO_FLAG_NONE); PRINT_CONSTANT((guint) GST_AUDIO_FLAG_UNPOSITIONED); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_ENCODED); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_F32BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_F32LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_F64BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_F64LE); PRINT_CONSTANT((guint) GST_AUDIO_FORMAT_FLAG_COMPLEX); PRINT_CONSTANT((guint) GST_AUDIO_FORMAT_FLAG_FLOAT); PRINT_CONSTANT((guint) GST_AUDIO_FORMAT_FLAG_INTEGER); PRINT_CONSTANT((guint) GST_AUDIO_FORMAT_FLAG_SIGNED); PRINT_CONSTANT((guint) GST_AUDIO_FORMAT_FLAG_UNPACK); PRINT_CONSTANT(GST_AUDIO_FORMAT_LAST); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S16BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S16LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S18BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S18LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S20BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S20LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S24BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S24LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S24_32BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S24_32LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S32BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S32LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_S8); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U16BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U16LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U18BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U18LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U20BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U20LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U24BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U24LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U24_32BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U24_32LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U32BE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U32LE); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_U8); PRINT_CONSTANT((gint) GST_AUDIO_FORMAT_UNKNOWN); PRINT_CONSTANT((gint) GST_AUDIO_LAYOUT_INTERLEAVED); PRINT_CONSTANT((gint) GST_AUDIO_LAYOUT_NON_INTERLEAVED); PRINT_CONSTANT((gint) GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK); PRINT_CONSTANT((gint) GST_AUDIO_NOISE_SHAPING_HIGH); PRINT_CONSTANT((gint) GST_AUDIO_NOISE_SHAPING_MEDIUM); PRINT_CONSTANT((gint) GST_AUDIO_NOISE_SHAPING_NONE); PRINT_CONSTANT((gint) GST_AUDIO_NOISE_SHAPING_SIMPLE); PRINT_CONSTANT((guint) GST_AUDIO_PACK_FLAG_NONE); PRINT_CONSTANT((guint) GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE); PRINT_CONSTANT((guint) GST_AUDIO_QUANTIZE_FLAG_NONE); PRINT_CONSTANT((guint) GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED); PRINT_CONSTANT(GST_AUDIO_RATE_RANGE); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_FILTER_MODE_FULL); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED); PRINT_CONSTANT((guint) GST_AUDIO_RESAMPLER_FLAG_NONE); PRINT_CONSTANT((guint) GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN); PRINT_CONSTANT((guint) GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT); PRINT_CONSTANT((guint) GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_METHOD_CUBIC); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_METHOD_KAISER); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_METHOD_LINEAR); PRINT_CONSTANT((gint) GST_AUDIO_RESAMPLER_METHOD_NEAREST); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_CUBIC_B); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_CUBIC_C); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_CUTOFF); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_FILTER_MODE); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_N_TAPS); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_QUALITY_DEFAULT); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_QUALITY_MAX); PRINT_CONSTANT(GST_AUDIO_RESAMPLER_QUALITY_MIN); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_STATE_ERROR); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_STATE_PAUSED); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_STATE_STARTED); PRINT_CONSTANT((gint) GST_AUDIO_RING_BUFFER_STATE_STOPPED); PRINT_CONSTANT(GST_DSD_FORMATS_ALL); PRINT_CONSTANT((gint) GST_DSD_FORMAT_U16); PRINT_CONSTANT((gint) GST_DSD_FORMAT_U16BE); PRINT_CONSTANT((gint) GST_DSD_FORMAT_U16LE); PRINT_CONSTANT((gint) GST_DSD_FORMAT_U32); PRINT_CONSTANT((gint) GST_DSD_FORMAT_U32BE); PRINT_CONSTANT((gint) GST_DSD_FORMAT_U32LE); PRINT_CONSTANT((gint) GST_DSD_FORMAT_U8); PRINT_CONSTANT((gint) GST_DSD_FORMAT_UNKNOWN); PRINT_CONSTANT(GST_DSD_MEDIA_TYPE); PRINT_CONSTANT(GST_DSD_SILENCE_PATTERN_BYTE); PRINT_CONSTANT(GST_META_TAG_AUDIO_CHANNELS_STR); PRINT_CONSTANT(GST_META_TAG_AUDIO_RATE_STR); PRINT_CONSTANT(GST_META_TAG_AUDIO_STR); PRINT_CONSTANT(GST_META_TAG_DSD_PLANE_OFFSETS_STR); PRINT_CONSTANT((gint) GST_NUM_DSD_FORMATS); PRINT_CONSTANT((gint) GST_STREAM_VOLUME_FORMAT_CUBIC); PRINT_CONSTANT((gint) GST_STREAM_VOLUME_FORMAT_DB); PRINT_CONSTANT((gint) GST_STREAM_VOLUME_FORMAT_LINEAR); return 0; } gstreamer-audio-sys-0.23.5/tests/layout.c000064400000000000000000000161341046102023000164470ustar 00000000000000// This file was generated by gir (https://github.com/gtk-rs/gir) // from gir-files (https://github.com/gtk-rs/gir-files) // from gst-gir-files (https://gitlab.freedesktop.org/gstreamer/gir-files-rs.git) // DO NOT EDIT #include "manual.h" #include #include int main() { printf("%s;%zu;%zu\n", "GstAudioAggregator", sizeof(GstAudioAggregator), alignof(GstAudioAggregator)); printf("%s;%zu;%zu\n", "GstAudioAggregatorClass", sizeof(GstAudioAggregatorClass), alignof(GstAudioAggregatorClass)); printf("%s;%zu;%zu\n", "GstAudioAggregatorConvertPad", sizeof(GstAudioAggregatorConvertPad), alignof(GstAudioAggregatorConvertPad)); printf("%s;%zu;%zu\n", "GstAudioAggregatorConvertPadClass", sizeof(GstAudioAggregatorConvertPadClass), alignof(GstAudioAggregatorConvertPadClass)); printf("%s;%zu;%zu\n", "GstAudioAggregatorPad", sizeof(GstAudioAggregatorPad), alignof(GstAudioAggregatorPad)); printf("%s;%zu;%zu\n", "GstAudioAggregatorPadClass", sizeof(GstAudioAggregatorPadClass), alignof(GstAudioAggregatorPadClass)); printf("%s;%zu;%zu\n", "GstAudioBaseSink", sizeof(GstAudioBaseSink), alignof(GstAudioBaseSink)); printf("%s;%zu;%zu\n", "GstAudioBaseSinkClass", sizeof(GstAudioBaseSinkClass), alignof(GstAudioBaseSinkClass)); printf("%s;%zu;%zu\n", "GstAudioBaseSinkDiscontReason", sizeof(GstAudioBaseSinkDiscontReason), alignof(GstAudioBaseSinkDiscontReason)); printf("%s;%zu;%zu\n", "GstAudioBaseSinkSlaveMethod", sizeof(GstAudioBaseSinkSlaveMethod), alignof(GstAudioBaseSinkSlaveMethod)); printf("%s;%zu;%zu\n", "GstAudioBaseSrc", sizeof(GstAudioBaseSrc), alignof(GstAudioBaseSrc)); printf("%s;%zu;%zu\n", "GstAudioBaseSrcClass", sizeof(GstAudioBaseSrcClass), alignof(GstAudioBaseSrcClass)); printf("%s;%zu;%zu\n", "GstAudioBaseSrcSlaveMethod", sizeof(GstAudioBaseSrcSlaveMethod), alignof(GstAudioBaseSrcSlaveMethod)); printf("%s;%zu;%zu\n", "GstAudioBuffer", sizeof(GstAudioBuffer), alignof(GstAudioBuffer)); printf("%s;%zu;%zu\n", "GstAudioCdSrc", sizeof(GstAudioCdSrc), alignof(GstAudioCdSrc)); printf("%s;%zu;%zu\n", "GstAudioCdSrcClass", sizeof(GstAudioCdSrcClass), alignof(GstAudioCdSrcClass)); printf("%s;%zu;%zu\n", "GstAudioCdSrcMode", sizeof(GstAudioCdSrcMode), alignof(GstAudioCdSrcMode)); printf("%s;%zu;%zu\n", "GstAudioCdSrcTrack", sizeof(GstAudioCdSrcTrack), alignof(GstAudioCdSrcTrack)); printf("%s;%zu;%zu\n", "GstAudioChannelMixerFlags", sizeof(GstAudioChannelMixerFlags), alignof(GstAudioChannelMixerFlags)); printf("%s;%zu;%zu\n", "GstAudioChannelPosition", sizeof(GstAudioChannelPosition), alignof(GstAudioChannelPosition)); printf("%s;%zu;%zu\n", "GstAudioClippingMeta", sizeof(GstAudioClippingMeta), alignof(GstAudioClippingMeta)); printf("%s;%zu;%zu\n", "GstAudioClock", sizeof(GstAudioClock), alignof(GstAudioClock)); printf("%s;%zu;%zu\n", "GstAudioClockClass", sizeof(GstAudioClockClass), alignof(GstAudioClockClass)); printf("%s;%zu;%zu\n", "GstAudioConverterFlags", sizeof(GstAudioConverterFlags), alignof(GstAudioConverterFlags)); printf("%s;%zu;%zu\n", "GstAudioDecoder", sizeof(GstAudioDecoder), alignof(GstAudioDecoder)); printf("%s;%zu;%zu\n", "GstAudioDecoderClass", sizeof(GstAudioDecoderClass), alignof(GstAudioDecoderClass)); printf("%s;%zu;%zu\n", "GstAudioDitherMethod", sizeof(GstAudioDitherMethod), alignof(GstAudioDitherMethod)); printf("%s;%zu;%zu\n", "GstAudioDownmixMeta", sizeof(GstAudioDownmixMeta), alignof(GstAudioDownmixMeta)); printf("%s;%zu;%zu\n", "GstAudioEncoder", sizeof(GstAudioEncoder), alignof(GstAudioEncoder)); printf("%s;%zu;%zu\n", "GstAudioEncoderClass", sizeof(GstAudioEncoderClass), alignof(GstAudioEncoderClass)); printf("%s;%zu;%zu\n", "GstAudioFilter", sizeof(GstAudioFilter), alignof(GstAudioFilter)); printf("%s;%zu;%zu\n", "GstAudioFilterClass", sizeof(GstAudioFilterClass), alignof(GstAudioFilterClass)); printf("%s;%zu;%zu\n", "GstAudioFlags", sizeof(GstAudioFlags), alignof(GstAudioFlags)); printf("%s;%zu;%zu\n", "GstAudioFormat", sizeof(GstAudioFormat), alignof(GstAudioFormat)); printf("%s;%zu;%zu\n", "GstAudioFormatFlags", sizeof(GstAudioFormatFlags), alignof(GstAudioFormatFlags)); printf("%s;%zu;%zu\n", "GstAudioFormatInfo", sizeof(GstAudioFormatInfo), alignof(GstAudioFormatInfo)); printf("%s;%zu;%zu\n", "GstAudioInfo", sizeof(GstAudioInfo), alignof(GstAudioInfo)); printf("%s;%zu;%zu\n", "GstAudioLayout", sizeof(GstAudioLayout), alignof(GstAudioLayout)); printf("%s;%zu;%zu\n", "GstAudioLevelMeta", sizeof(GstAudioLevelMeta), alignof(GstAudioLevelMeta)); printf("%s;%zu;%zu\n", "GstAudioMeta", sizeof(GstAudioMeta), alignof(GstAudioMeta)); printf("%s;%zu;%zu\n", "GstAudioNoiseShapingMethod", sizeof(GstAudioNoiseShapingMethod), alignof(GstAudioNoiseShapingMethod)); printf("%s;%zu;%zu\n", "GstAudioPackFlags", sizeof(GstAudioPackFlags), alignof(GstAudioPackFlags)); printf("%s;%zu;%zu\n", "GstAudioQuantizeFlags", sizeof(GstAudioQuantizeFlags), alignof(GstAudioQuantizeFlags)); printf("%s;%zu;%zu\n", "GstAudioResamplerFilterInterpolation", sizeof(GstAudioResamplerFilterInterpolation), alignof(GstAudioResamplerFilterInterpolation)); printf("%s;%zu;%zu\n", "GstAudioResamplerFilterMode", sizeof(GstAudioResamplerFilterMode), alignof(GstAudioResamplerFilterMode)); printf("%s;%zu;%zu\n", "GstAudioResamplerFlags", sizeof(GstAudioResamplerFlags), alignof(GstAudioResamplerFlags)); printf("%s;%zu;%zu\n", "GstAudioResamplerMethod", sizeof(GstAudioResamplerMethod), alignof(GstAudioResamplerMethod)); printf("%s;%zu;%zu\n", "GstAudioRingBuffer", sizeof(GstAudioRingBuffer), alignof(GstAudioRingBuffer)); printf("%s;%zu;%zu\n", "GstAudioRingBufferClass", sizeof(GstAudioRingBufferClass), alignof(GstAudioRingBufferClass)); printf("%s;%zu;%zu\n", "GstAudioRingBufferFormatType", sizeof(GstAudioRingBufferFormatType), alignof(GstAudioRingBufferFormatType)); printf("%s;%zu;%zu\n", "GstAudioRingBufferSpec", sizeof(GstAudioRingBufferSpec), alignof(GstAudioRingBufferSpec)); printf("%s;%zu;%zu\n", "GstAudioRingBufferState", sizeof(GstAudioRingBufferState), alignof(GstAudioRingBufferState)); printf("%s;%zu;%zu\n", "GstAudioSink", sizeof(GstAudioSink), alignof(GstAudioSink)); printf("%s;%zu;%zu\n", "GstAudioSinkClass", sizeof(GstAudioSinkClass), alignof(GstAudioSinkClass)); printf("%s;%zu;%zu\n", "GstAudioSinkClassExtension", sizeof(GstAudioSinkClassExtension), alignof(GstAudioSinkClassExtension)); printf("%s;%zu;%zu\n", "GstAudioSrc", sizeof(GstAudioSrc), alignof(GstAudioSrc)); printf("%s;%zu;%zu\n", "GstAudioSrcClass", sizeof(GstAudioSrcClass), alignof(GstAudioSrcClass)); printf("%s;%zu;%zu\n", "GstDsdFormat", sizeof(GstDsdFormat), alignof(GstDsdFormat)); printf("%s;%zu;%zu\n", "GstDsdInfo", sizeof(GstDsdInfo), alignof(GstDsdInfo)); printf("%s;%zu;%zu\n", "GstDsdPlaneOffsetMeta", sizeof(GstDsdPlaneOffsetMeta), alignof(GstDsdPlaneOffsetMeta)); printf("%s;%zu;%zu\n", "GstStreamVolumeFormat", sizeof(GstStreamVolumeFormat), alignof(GstStreamVolumeFormat)); printf("%s;%zu;%zu\n", "GstStreamVolumeInterface", sizeof(GstStreamVolumeInterface), alignof(GstStreamVolumeInterface)); return 0; } gstreamer-audio-sys-0.23.5/tests/manual.h000064400000000000000000000001671046102023000164130ustar 00000000000000// Feel free to edit this file, it won't be regenerated by gir generator unless removed. #include