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310 lines
12 KiB
ObjectPascal
310 lines
12 KiB
ObjectPascal
(*
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* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
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*
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* This file is part of libswresample
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*
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* libswresample is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* libswresample is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with libswresample; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*)
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unit swresample;
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{$include ffmpeg.inc}
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interface
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(*
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* @defgroup lswr Libswresample
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* @{
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*
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* Libswresample (lswr) is a library that handles audio resampling, sample
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* format conversion and mixing.
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*
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* Interaction with lswr is done through SwrContext, which is
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* allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
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* must be set with the @ref avoptions API.
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*
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* For example the following code will setup conversion from planar float sample
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* format to interleaved signed 16-bit integer, downsampling from 48kHz to
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* 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
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* matrix):
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* @code
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* SwrContext *swr = swr_alloc();
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* av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
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* av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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* av_opt_set_int(swr, "in_sample_rate", 48000, 0);
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* av_opt_set_int(swr, "out_sample_rate", 44100, 0);
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* av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
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* av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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* @endcode
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*
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* Once all values have been set, it must be initialized with swr_init(). If
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* you need to change the conversion parameters, you can change the parameters
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* as described above, or by using swr_alloc_set_opts(), then call swr_init()
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* again.
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*
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* The conversion itself is done by repeatedly calling swr_convert().
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* Note that the samples may get buffered in swr if you provide insufficient
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* output space or if sample rate conversion is done, which requires "future"
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* samples. Samples that do not require future input can be retrieved at any
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* time by using swr_convert() (in_count can be set to 0).
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* At the end of conversion the resampling buffer can be flushed by calling
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* swr_convert() with NULL in and 0 in_count.
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*
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* The delay between input and output, can at any time be found by using
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* swr_get_delay().
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*
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* The following code demonstrates the conversion loop assuming the parameters
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* from above and caller-defined functions get_input() and handle_output():
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* @code
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* uint8_t **input;
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* int in_samples;
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*
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* while (get_input(&input, &in_samples)) {
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* uint8_t *output;
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* int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
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* in_samples, 44100, 48000, AV_ROUND_UP);
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* av_samples_alloc(&output, NULL, 2, out_samples,
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* AV_SAMPLE_FMT_S16, 0);
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* out_samples = swr_convert(swr, &output, out_samples,
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* input, in_samples);
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* handle_output(output, out_samples);
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* av_freep(&output);
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* }
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* @endcode
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*
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* When the conversion is finished, the conversion
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* context and everything associated with it must be freed with swr_free().
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* There will be no memory leak if the data is not completely flushed before
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* swr_free().
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*)
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//#if LIBSWRESAMPLE_VERSION_MAJOR < 1
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//#define SWR_CH_MAX 32 ///< Maximum number of channels
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//#endif
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//#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
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//TODO use int resample ?
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//long term TODO can we enable this dynamically?
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//enum SwrDitherType {
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// SWR_DITHER_NONE = 0,
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// SWR_DITHER_RECTANGULAR,
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// SWR_DITHER_TRIANGULAR,
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// SWR_DITHER_TRIANGULAR_HIGHPASS,
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//
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// SWR_DITHER_NS = 64, ///< not part of API/ABI
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// SWR_DITHER_NS_LIPSHITZ,
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// SWR_DITHER_NS_F_WEIGHTED,
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// SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
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// SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
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// SWR_DITHER_NS_SHIBATA,
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// SWR_DITHER_NS_LOW_SHIBATA,
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// SWR_DITHER_NS_HIGH_SHIBATA,
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// SWR_DITHER_NB, ///< not part of API/ABI
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//};
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//
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//(* Resampling Engines *)
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//enum SwrEngine {
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// SWR_ENGINE_SWR, (*< SW Resampler *)
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// SWR_ENGINE_SOXR, (*< SoX Resampler *)
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// SWR_ENGINE_NB, ///< not part of API/ABI
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//};
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//
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//(* Resampling Filter Types *)
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//enum SwrFilterType {
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// SWR_FILTER_TYPE_CUBIC, (*< Cubic *)
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// SWR_FILTER_TYPE_BLACKMAN_NUTTALL, (*< Blackman Nuttall Windowed Sinc *)
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// SWR_FILTER_TYPE_KAISER, (*< Kaiser Windowed Sinc *)
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//};
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Type
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//typedef struct SwrContext SwrContext;
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pSwrContext = ^TSwrContext;
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TSwrContext = packed record
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end;
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(*
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* Get the AVClass for swrContext. It can be used in combination with
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* AV_OPT_SEARCH_FAKE_OBJ for examining options.
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*
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* @see av_opt_find().
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*)
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//const AVClass *swr_get_class(void);
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(*
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* Allocate SwrContext.
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*
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* If you use this function you will need to set the parameters (manually or
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* with swr_alloc_set_opts()) before calling swr_init().
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*
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* @see swr_alloc_set_opts(), swr_init(), swr_free()
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* @return NULL on error, allocated context otherwise
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*)
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//struct SwrContext *swr_alloc(void);
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(*
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* Initialize context after user parameters have been set.
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*
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* @return AVERROR error code in case of failure.
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*)
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//int swr_init(struct SwrContext *s);
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(*
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* Allocate SwrContext if needed and set/reset common parameters.
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*
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* This function does not require s to be allocated with swr_alloc(). On the
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* other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
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* on the allocated context.
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*
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* @param s Swr context, can be NULL
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* @param out_ch_layout output channel layout (AV_CH_LAYOUT_* )
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* @param out_sample_fmt output sample format (AV_SAMPLE_FMT_* ).
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* @param out_sample_rate output sample rate (frequency in Hz)
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* @param in_ch_layout input channel layout (AV_CH_LAYOUT_* )
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* @param in_sample_fmt input sample format (AV_SAMPLE_FMT_* ).
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* @param in_sample_rate input sample rate (frequency in Hz)
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* @param log_offset logging level offset
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* @param log_ctx parent logging context, can be NULL
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*
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* @see swr_init(), swr_free()
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* @return NULL on error, allocated context otherwise
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*)
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//struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
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// int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
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// int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
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// int log_offset, void *log_ctx);
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(*
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* Free the given SwrContext and set the pointer to NULL.
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*)
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//void swr_free(struct SwrContext **s);
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(*
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* Convert audio.
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*
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* in and in_count can be set to 0 to flush the last few samples out at the
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* end.
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*
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* If more input is provided than output space then the input will be buffered.
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* You can avoid this buffering by providing more output space than input.
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* Convertion will run directly without copying whenever possible.
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*
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* @param s allocated Swr context, with parameters set
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* @param out output buffers, only the first one need be set in case of packed audio
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* @param out_count amount of space available for output in samples per channel
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* @param in input buffers, only the first one need to be set in case of packed audio
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* @param in_count number of input samples available in one channel
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*
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* @return number of samples output per channel, negative value on error
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*)
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//int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
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// const uint8_t **in , int in_count);
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function swr_convert(s:pSwrContext; Var out_:PByte; out_count:Integer;
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const in_:pByte;in_count:Integer):Integer;cdecl;
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(*
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* Convert the next timestamp from input to output
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* timestamps are in 1/(in_sample_rate * out_sample_rate) units.
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*
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* @note There are 2 slightly differently behaving modes.
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* First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
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* in this case timestamps will be passed through with delays compensated
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* Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX)
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* in this case the output timestamps will match output sample numbers
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*
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* @param pts timestamp for the next input sample, INT64_MIN if unknown
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* @return the output timestamp for the next output sample
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*)
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//int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
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(*
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* Activate resampling compensation.
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*)
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//int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
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(*
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* Set a customized input channel mapping.
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*
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* @param s allocated Swr context, not yet initialized
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* @param channel_map customized input channel mapping (array of channel
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* indexes, -1 for a muted channel)
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* @return AVERROR error code in case of failure.
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*)
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//int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
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(*
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* Set a customized remix matrix.
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*
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* @param s allocated Swr context, not yet initialized
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* @param matrix remix coefficients; matrix[i + stride * o] is
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* the weight of input channel i in output channel o
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* @param stride offset between lines of the matrix
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* @return AVERROR error code in case of failure.
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*)
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//int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
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(*
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* Drops the specified number of output samples.
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*)
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//int swr_drop_output(struct SwrContext *s, int count);
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(*
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* Injects the specified number of silence samples.
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*)
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//int swr_inject_silence(struct SwrContext *s, int count);
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(*
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* Gets the delay the next input sample will experience relative to the next output sample.
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*
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* Swresample can buffer data if more input has been provided than available
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* output space, also converting between sample rates needs a delay.
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* This function returns the sum of all such delays.
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* The exact delay is not necessarily an integer value in either input or
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* output sample rate. Especially when downsampling by a large value, the
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* output sample rate may be a poor choice to represent the delay, similarly
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* for upsampling and the input sample rate.
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*
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* @param s swr context
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* @param base timebase in which the returned delay will be
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* if its set to 1 the returned delay is in seconds
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* if its set to 1000 the returned delay is in milli seconds
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* if its set to the input sample rate then the returned delay is in input samples
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* if its set to the output sample rate then the returned delay is in output samples
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* an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate)
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* @returns the delay in 1/base units.
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*)
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//int64_t swr_get_delay(struct SwrContext *s, int64_t base);
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(*
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* Return the LIBSWRESAMPLE_VERSION_INT constant.
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*)
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//unsigned swresample_version(void);
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(*
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* Return the swr build-time configuration.
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*)
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//const char *swresample_configuration(void);
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(*
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* Return the swr license.
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*)
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//const char *swresample_license(void);
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implementation
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uses ffmpeglib;
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function swr_convert; external swscale_dll;
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end.
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