(*
* copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
* This is a part of Pascal porting of ffmpeg.
* - Originally by Victor Zinetz for Delphi and Free Pascal on Windows.
* - For Mac OS X, some modifications were made by The Creative CAT, denoted as CAT
* in the source codes.
* - Changes and updates by the UltraStar Deluxe Team
*
* Conversion of
*
* libswresample/swresample.h:
* version: 0.18.100
*
*)
unit swresample;
{$IFDEF FPC}
{$MODE DELPHI}
{$PACKENUM 4} (* use 4-byte enums *)
{$PACKRECORDS C} (* C/C++-compatible record packing *)
{$ELSE}
{$MINENUMSIZE 4} (* use 4-byte enums *)
{$ENDIF}
{$IFDEF DARWIN}
{$linklib swresample}
{$ENDIF}
interface
uses
ctypes,
rational,
{$IFDEF UNIX}
BaseUnix,
{$ENDIF}
UConfig;
const
(*
* IMPORTANT: The official FFmpeg C headers change very quickly. Often some
* of the data structures are changed so that they become incompatible with
* older header files. The Pascal headers have to be adjusted to those changes,
* otherwise the application might crash randomly or strange bugs (not
* necessarily related to video or audio due to buffer overflows etc.) might
* occur.
*
* In the past users reported problems with USDX that took hours to fix and
* the problem was an unsupported version of FFmpeg. So we decided to disable
* support for future versions of FFmpeg until the headers are revised by us
* for that version as they otherwise most probably will break USDX.
*
* If the headers do not yet support your FFmpeg version you may want to
* adjust the max. version numbers manually but please note: it may work but
* in many cases it does not. The USDX team does NOT PROVIDE ANY SUPPORT
* for the game if the MAX. VERSION WAS CHANGED.
*
* The only safe way to support new versions of FFmpeg is to add the changes
* of the FFmpeg git repository C headers to the Pascal headers.
* You can accelerate this process by posting a patch with the git changes
* translated to Pascal to our bug tracker (please join our IRC chat before
* you start working on it). Simply adjusting the max. versions is NOT a valid
* fix.
*)
(* Supported version by this header *)
LIBSWRESAMPLE_MAX_VERSION_MAJOR = 0;
LIBSWRESAMPLE_MAX_VERSION_MINOR = 18;
LIBSWRESAMPLE_MAX_VERSION_RELEASE = 100;
LIBSWRESAMPLE_MAX_VERSION = (LIBSWRESAMPLE_MAX_VERSION_MAJOR * VERSION_MAJOR) +
(LIBSWRESAMPLE_MAX_VERSION_MINOR * VERSION_MINOR) +
(LIBSWRESAMPLE_VERSION_RELEASE * VERSION_RELEASE);
(* Min. supported version by this header *)
LIBSWRESAMPLE_MIN_VERSION_MAJOR = 0;
LIBSWRESAMPLE_MIN_VERSION_MINOR = 18;
LIBSWRESAMPLE_MIN_VERSION_RELEASE = 100;
LIBSWRESAMPLE_MIN_VERSION = (LIBSWRESAMPLE_MIN_VERSION_MAJOR * VERSION_MAJOR) +
(LIBSWRESAMPLE_MIN_VERSION_MINOR * VERSION_MINOR) +
(LIBSWRESAMPLE_MIN_VERSION_RELEASE * VERSION_RELEASE);
(* Check if linked versions are supported *)
{$IF (LIBSWRESAMPLE_VERSION < LIBSWRESAMPLE_MIN_VERSION)}
{$MESSAGE Error 'Linked version of libswresample is too old!'}
{$IFEND}
(* Check if linked version is supported *)
{$IF (LIBSWRESAMPLE_VERSION > LIBSWRESAMPLE_MAX_VERSION)}
{$MESSAGE Error 'Linked version of libswresample is not yet supported!'}
{$IFEND}
{$IF LIBRESAMPLE_VERSION_MAJOR < 1}
SWR_CH_MAX = 32; (* < Maximum number of channels *)
{$ENDIF}
SWR_FLAG_RESAMPLE = 1; (* < Force resampling even if equal sample rate *)
type
TSwrDitherType = (
SWR_DITHER_NONE = 0,
SWR_DITHER_RECTANGULAR,
SWR_DITHER_TRIANGULAR,
SWR_DITHER_TRIANGULAR_HIGHPASS,
SWR_DITHER_NS = 64, (* < not part of API/ABI *)
SWR_DITHER_NS_LIPSHITZ,
SWR_DITHER_NS_F_WEIGHTED,
SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
SWR_DITHER_NS_SHIBATA,
SWR_DITHER_NS_LOW_SHIBATA,
SWR_DITHER_NS_HIGH_SHIBATA,
SWR_DITHER_NB (* < not part of API/ABI *)
);
TSwrEngine = (
SWR_ENGINE_SWR, (* < SW Resampler *)
SWR_ENGINE_SOXR, (* < SoX Resampler *)
SWR_ENGINE_NB (* < not part of API/ABI *)
);
TSwrFilterType = (
SWR_FILTER_TYPE_CUBIC, (* < Cubic *)
SWR_FILTER_TYPE_BLACKMAN_NUTTALL, (* < Blackman Nuttall Windowed Sinc *)
SWR_FILTER_TYPE_KAISER (* < Kaiser Windowed Sinc *)
);
PPSwrContext= ^PSwrContext;
PSwrContext = ^TSwrContext;
TSwrContext = record
end;
(**
* Get the AVClass for swrContext. It can be used in combination with
* AV_OPT_SEARCH_FAKE_OBJ for examining options.
*
* @see av_opt_find().
*)
function swr_get_class(): PAVClass;
cdecl; external swresample;
(**
* Allocate SwrContext.
*
* If you use this function you will need to set the parameters (manually or
* with swr_alloc_set_opts()) before calling swr_init().
*
* @see swr_alloc_set_opts(), swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*)
function swr_alloc(): PSwrContext;
cdecl; external swresample;
(**
* Initialize context after user parameters have been set.
*
* @return AVERROR error code in case of failure.
*)
function swr_init(s: PSwrContext): cint;
cdecl; external swresample;
(**
* Check whether an swr context has been initialized or not.
*
* @return positive if it has been initialized, 0 if not initialized
*)
function swr_is_initialized(s: PSwrContext): cint;
cdecl; external swresample;
(**
* Allocate SwrContext if needed and set/reset common parameters.
*
* This function does not require s to be allocated with swr_alloc(). On the
* other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
* on the allocated context.
*
* @param s Swr context, can be NULL
* @param out_ch_layout output channel layout (AV_CH_LAYOUT_* )
* @param out_sample_fmt output sample format (AV_SAMPLE_FMT_* ).
* @param out_sample_rate output sample rate (frequency in Hz)
* @param in_ch_layout input channel layout (AV_CH_LAYOUT_* )
* @param in_sample_fmt input sample format (AV_SAMPLE_FMT_* ).
* @param in_sample_rate input sample rate (frequency in Hz)
* @param log_offset logging level offset
* @param log_ctx parent logging context, can be NULL
*
* @see swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*)
function swr_alloc_set_opts(s: PSwrContext;
out_ch_layout: cint64; out_sample_fmt: TAVSampleFormat; out_sample_rate: cint;
in_ch_layout: cint64; in_sample_fmt: TAVSampleFormat; in_sample_rate: cint;
log_offset: cint; log_ctx: pointer): PSwrContext;
cdecl; external swresample;
(**
* Free the given SwrContext and set the pointer to NULL.
*)
procedure swr_free(s: PPSwrContext);
cdecl; external swresample;
(**
* Convert audio.
*
* in and in_count can be set to 0 to flush the last few samples out at the
* end.
*
* If more input is provided than output space then the input will be buffered.
* You can avoid this buffering by providing more output space than input.
* Convertion will run directly without copying whenever possible.
*
* @param s allocated Swr context, with parameters set
* @param out output buffers, only the first one need be set in case of packed audio
* @param out_count amount of space available for output in samples per channel
* @param in input buffers, only the first one need to be set in case of packed audio
* @param in_count number of input samples available in one channel
*
* @return number of samples output per channel, negative value on error
*)
function swr_convert(s: PSwrContext; out_: PByte; out_count: cint;
{const} in_: PByte; in_count: cint): cint;
cdecl; external swresample;
(**
* Convert the next timestamp from input to output
* timestamps are in 1/(in_sample_rate * out_sample_rate) units.
*
* @note There are 2 slightly differently behaving modes.
* First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
* in this case timestamps will be passed through with delays compensated
* Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX)
* in this case the output timestamps will match output sample numbers
*
* @param pts timestamp for the next input sample, INT64_MIN if unknown
* @return the output timestamp for the next output sample
*)
function swr_next_pts(s: PSwrContext; pts: cint64): cint64;
cdecl; external swresample;
(**
* Activate resampling compensation.
*)
function swr_set_compensation(s: PSwrContext; sample_delta: cint; compensation_distance: cint): cint;
cdecl; external swresample;
(**
* Set a customized input channel mapping.
*
* @param s allocated Swr context, not yet initialized
* @param channel_map customized input channel mapping (array of channel
* indexes, -1 for a muted channel)
* @return AVERROR error code in case of failure.
*)
function swr_set_channel_mapping(s: PSwrContext; {const} channel_map: pcint): cint;
cdecl; external swresample;
(**
* Set a customized remix matrix.
*
* @param s allocated Swr context, not yet initialized
* @param matrix remix coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o
* @param stride offset between lines of the matrix
* @return AVERROR error code in case of failure.
*)
function swr_set_matrix(s: PSwrContext; {const} matrix: pcdouble; stride: cint): cint;
cdecl; external swresample;
(**
* Drops the specified number of output samples.
*)
function swr_drop_output(s: PSwrContext; count: cint): cint;
cdecl; external swresample;
(**
* Injects the specified number of silence samples.
*)
function swr_inject_silence(s: PSwrContext; count: cint): cint;
cdecl; external swresample;
(**
* Gets the delay the next input sample will experience relative to the next output sample.
*
* Swresample can buffer data if more input has been provided than available
* output space, also converting between sample rates needs a delay.
* This function returns the sum of all such delays.
* The exact delay is not necessarily an integer value in either input or
* output sample rate. Especially when downsampling by a large value, the
* output sample rate may be a poor choice to represent the delay, similarly
* for upsampling and the input sample rate.
*
* @param s swr context
* @param base timebase in which the returned delay will be
* if its set to 1 the returned delay is in seconds
* if its set to 1000 the returned delay is in milli seconds
* if its set to the input sample rate then the returned delay is in input samples
* if its set to the output sample rate then the returned delay is in output samples
* an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate)
* @returns the delay in 1/base units.
*)
function swr_get_delay(s: PSwrContext; base: cint64): cint64;
cdecl; external swresample;
(**
* Return the LIBSWRESAMPLE_VERSION_INT constant.
*)
function swresample_version(): cuint;
cdecl; external swresample;
(**
* Return the swr build-time configuration.
*)
function swresample_configuration(): cuchar;
cdecl; external swresample;
(**
* Return the swr license.
*)
function swresample_license(): cuchar;
cdecl; external swresample;
(**
* @
*)
implementation
end.
