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Libavresample (lavr) is a library that handles audio resampling, sample format conversion and mixing. More...
Files | |
file | avresample.h |
external API header | |
Macros | |
#define | AVRESAMPLE_MAX_CHANNELS 32 |
Typedefs | |
typedef struct AVAudioResampleContext | AVAudioResampleContext |
Enumerations | |
enum | AVMixCoeffType { AV_MIX_COEFF_TYPE_Q8, AV_MIX_COEFF_TYPE_Q15, AV_MIX_COEFF_TYPE_FLT, AV_MIX_COEFF_TYPE_NB } |
Mixing Coefficient Types. More... | |
enum | AVResampleFilterType { AV_RESAMPLE_FILTER_TYPE_CUBIC, AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, AV_RESAMPLE_FILTER_TYPE_KAISER } |
Resampling Filter Types. More... | |
enum | AVResampleDitherMethod { AV_RESAMPLE_DITHER_NONE, AV_RESAMPLE_DITHER_RECTANGULAR, AV_RESAMPLE_DITHER_TRIANGULAR, AV_RESAMPLE_DITHER_TRIANGULAR_HP, AV_RESAMPLE_DITHER_TRIANGULAR_NS, AV_RESAMPLE_DITHER_NB } |
Functions | |
unsigned | avresample_version (void) |
Return the LIBAVRESAMPLE_VERSION_INT constant. More... | |
const char * | avresample_configuration (void) |
Return the libavresample build-time configuration. More... | |
const char * | avresample_license (void) |
Return the libavresample license. More... | |
const AVClass * | avresample_get_class (void) |
Get the AVClass for AVAudioResampleContext. More... | |
AVAudioResampleContext * | avresample_alloc_context (void) |
Allocate AVAudioResampleContext and set options. More... | |
int | avresample_open (AVAudioResampleContext *avr) |
Initialize AVAudioResampleContext. More... | |
void | avresample_close (AVAudioResampleContext *avr) |
Close AVAudioResampleContext. More... | |
void | avresample_free (AVAudioResampleContext **avr) |
Free AVAudioResampleContext and associated AVOption values. More... | |
int | avresample_build_matrix (uint64_t in_layout, uint64_t out_layout, double center_mix_level, double surround_mix_level, double lfe_mix_level, int normalize, double *matrix, int stride, enum AVMatrixEncoding matrix_encoding) |
Generate a channel mixing matrix. More... | |
int | avresample_get_matrix (AVAudioResampleContext *avr, double *matrix, int stride) |
Get the current channel mixing matrix. More... | |
int | avresample_set_matrix (AVAudioResampleContext *avr, const double *matrix, int stride) |
Set channel mixing matrix. More... | |
int | avresample_set_channel_mapping (AVAudioResampleContext *avr, const int *channel_map) |
Set a customized input channel mapping. More... | |
int | avresample_set_compensation (AVAudioResampleContext *avr, int sample_delta, int compensation_distance) |
Set compensation for resampling. More... | |
int | avresample_convert (AVAudioResampleContext *avr, uint8_t **output, int out_plane_size, int out_samples, uint8_t **input, int in_plane_size, int in_samples) |
Convert input samples and write them to the output FIFO. More... | |
int | avresample_get_delay (AVAudioResampleContext *avr) |
Return the number of samples currently in the resampling delay buffer. More... | |
int | avresample_available (AVAudioResampleContext *avr) |
Return the number of available samples in the output FIFO. More... | |
int | avresample_read (AVAudioResampleContext *avr, uint8_t **output, int nb_samples) |
Read samples from the output FIFO. More... | |
Detailed Description
Libavresample (lavr) is a library that handles audio resampling, sample format conversion and mixing.
Interaction with lavr is done through AVAudioResampleContext, which is allocated with avresample_alloc_context(). It is opaque, so all parameters must be set with the AVOptions API.
For example the following code will setup conversion from planar float sample format to interleaved signed 16-bit integer, downsampling from 48kHz to 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing matrix):
Once the context is initialized, it must be opened with avresample_open(). If you need to change the conversion parameters, you must close the context with avresample_close(), change the parameters as described above, then reopen it again.
The conversion itself is done by repeatedly calling avresample_convert(). Note that the samples may get buffered in two places in lavr. The first one is the output FIFO, where the samples end up if the output buffer is not large enough. The data stored in there may be retrieved at any time with avresample_read(). The second place is the resampling delay buffer, applicable only when resampling is done. The samples in it require more input before they can be processed. Their current amount is returned by avresample_get_delay(). At the end of conversion the resampling buffer can be flushed by calling avresample_convert() with NULL input.
The following code demonstrates the conversion loop assuming the parameters from above and caller-defined functions get_input() and handle_output():
When the conversion is finished and the FIFOs are flushed if required, the conversion context and everything associated with it must be freed with avresample_free().
Macro Definition Documentation
#define AVRESAMPLE_MAX_CHANNELS 32 |
Definition at line 103 of file avresample.h.
Referenced by audiogen(), avresample_build_matrix(), avresample_get_matrix(), avresample_open(), avresample_set_channel_mapping(), avresample_set_compensation(), avresample_set_matrix(), ff_audio_data_add_to_fifo(), ff_audio_data_alloc(), ff_audio_data_init(), ff_audio_data_realloc(), ff_audio_data_set_channels(), ff_audio_mix(), ff_audio_mix_get_matrix(), ff_audio_mix_set_matrix(), if(), and main().
Typedef Documentation
typedef struct AVAudioResampleContext AVAudioResampleContext |
Definition at line 105 of file avresample.h.
Enumeration Type Documentation
enum AVMixCoeffType |
Mixing Coefficient Types.
Enumerator | |
---|---|
AV_MIX_COEFF_TYPE_Q8 | |
AV_MIX_COEFF_TYPE_Q15 |
16-bit 8.8 fixed-point |
AV_MIX_COEFF_TYPE_FLT |
32-bit 17.15 fixed-point |
AV_MIX_COEFF_TYPE_NB |
floating-point |
Definition at line 108 of file avresample.h.
Definition at line 122 of file avresample.h.
enum AVResampleFilterType |
Resampling Filter Types.
Enumerator | |
---|---|
AV_RESAMPLE_FILTER_TYPE_CUBIC |
Cubic. |
AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL |
Blackman Nuttall Windowed Sinc. |
AV_RESAMPLE_FILTER_TYPE_KAISER |
Kaiser Windowed Sinc. |
Definition at line 116 of file avresample.h.
Function Documentation
AVAudioResampleContext* avresample_alloc_context | ( | void | ) |
Allocate AVAudioResampleContext and set options.
- Returns
- allocated audio resample context, or NULL on failure
Definition at line 94 of file libavresample/options.c.
Referenced by config_output(), config_props(), and main().
int avresample_available | ( | AVAudioResampleContext * | avr | ) |
Return the number of available samples in the output FIFO.
During conversion, if the user does not specify an output buffer or specifies an output buffer that is smaller than what is needed, remaining samples that are not written to the output are stored to an internal FIFO buffer. The samples in the FIFO can be read with avresample_read() or avresample_convert().
- Parameters
-
avr audio resample context
- Returns
- number of samples available for reading
Definition at line 609 of file libavresample/utils.c.
Referenced by filter_frame(), get_delay(), handle_trimming(), and main().
int avresample_build_matrix | ( | uint64_t | in_layout, |
uint64_t | out_layout, | ||
double | center_mix_level, | ||
double | surround_mix_level, | ||
double | lfe_mix_level, | ||
int | normalize, | ||
double * | matrix, | ||
int | stride, | ||
enum AVMatrixEncoding | matrix_encoding | ||
) |
Generate a channel mixing matrix.
This function is the one used internally by libavresample for building the default mixing matrix. It is made public just as a utility function for building custom matrices.
- Parameters
-
in_layout input channel layout out_layout output channel layout center_mix_level mix level for the center channel surround_mix_level mix level for the surround channel(s) lfe_mix_level mix level for the low-frequency effects channel normalize if 1, coefficients will be normalized to prevent overflow. if 0, coefficients will not be normalized. [out] matrix mixing coefficients; matrix[i + stride * o] is the weight of input channel i in output channel o. stride distance between adjacent input channels in the matrix array matrix_encoding matrixed stereo downmix mode (e.g. dplii)
- Returns
- 0 on success, negative AVERROR code on failure
Definition at line 87 of file audio_mix_matrix.c.
Referenced by ff_audio_mix_alloc().
void avresample_close | ( | AVAudioResampleContext * | avr | ) |
Close AVAudioResampleContext.
This closes the context, but it does not change the parameters. The context can be reopened with avresample_open(). It does, however, clear the output FIFO and any remaining leftover samples in the resampling delay buffer. If there was a custom matrix being used, that is also cleared.
- Parameters
-
avr audio resample context
Definition at line 257 of file libavresample/utils.c.
Referenced by avresample_free(), avresample_open(), avresample_set_compensation(), config_output(), main(), and uninit().
const char* avresample_configuration | ( | void | ) |
Return the libavresample build-time configuration.
- Returns
- configure string
Definition at line 632 of file libavresample/utils.c.
int avresample_convert | ( | AVAudioResampleContext * | avr, |
uint8_t ** | output, | ||
int | out_plane_size, | ||
int | out_samples, | ||
uint8_t ** | input, | ||
int | in_plane_size, | ||
int | in_samples | ||
) |
Convert input samples and write them to the output FIFO.
The upper bound on the number of output samples is given by avresample_available() + (avresample_get_delay() + number of input samples) * output sample rate / input sample rate.
The output data can be NULL or have fewer allocated samples than required. In this case, any remaining samples not written to the output will be added to an internal FIFO buffer, to be returned at the next call to this function or to avresample_read().
If converting sample rate, there may be data remaining in the internal resampling delay buffer. avresample_get_delay() tells the number of remaining samples. To get this data as output, call avresample_convert() with NULL input.
At the end of the conversion process, there may be data remaining in the internal FIFO buffer. avresample_available() tells the number of remaining samples. To get this data as output, either call avresample_convert() with NULL input or call avresample_read().
- Parameters
-
avr audio resample context output output data pointers out_plane_size output plane size, in bytes. This can be 0 if unknown, but that will lead to optimized functions not being used directly on the output, which could slow down some conversions. out_samples maximum number of samples that the output buffer can hold input input data pointers in_plane_size input plane size, in bytes This can be 0 if unknown, but that will lead to optimized functions not being used directly on the input, which could slow down some conversions. in_samples number of input samples to convert
- Returns
- number of samples written to the output buffer, not including converted samples added to the internal output FIFO
Definition at line 325 of file libavresample/utils.c.
Referenced by filter_frame(), main(), request_frame(), and write_to_fifo().
void avresample_free | ( | AVAudioResampleContext ** | avr | ) |
Free AVAudioResampleContext and associated AVOption values.
This also calls avresample_close() before freeing.
- Parameters
-
avr audio resample context
Definition at line 273 of file libavresample/utils.c.
Referenced by config_output(), main(), and uninit().
Get the AVClass for AVAudioResampleContext.
Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options without allocating a context.
- See also
- av_opt_find().
- Returns
- AVClass for AVAudioResampleContext
Definition at line 108 of file libavresample/options.c.
Referenced by init(), opt_default(), and resample_child_class_next().
int avresample_get_delay | ( | AVAudioResampleContext * | avr | ) |
Return the number of samples currently in the resampling delay buffer.
When resampling, there may be a delay between the input and output. Any unconverted samples in each call are stored internally in a delay buffer. This function allows the user to determine the current number of samples in the delay buffer, which can be useful for synchronization.
- See also
- avresample_convert()
- Parameters
-
avr audio resample context
- Returns
- number of samples currently in the resampling delay buffer
Definition at line 463 of file libavresample/resample.c.
Referenced by filter_frame(), get_delay(), main(), and request_frame().
int avresample_get_matrix | ( | AVAudioResampleContext * | avr, |
double * | matrix, | ||
int | stride | ||
) |
Get the current channel mixing matrix.
If no custom matrix has been previously set or the AVAudioResampleContext is not open, an error is returned.
- Parameters
-
avr audio resample context matrix mixing coefficients; matrix[i + stride * o] is the weight of input channel i in output channel o. stride distance between adjacent input channels in the matrix array
- Returns
- 0 on success, negative AVERROR code on failure
Definition at line 498 of file libavresample/utils.c.
Referenced by avresample_set_compensation().
const char* avresample_license | ( | void | ) |
Return the libavresample license.
Definition at line 626 of file libavresample/utils.c.
int avresample_open | ( | AVAudioResampleContext * | avr | ) |
Initialize AVAudioResampleContext.
- Parameters
-
avr audio resample context
- Returns
- 0 on success, negative AVERROR code on failure
Definition at line 35 of file libavresample/utils.c.
Referenced by avresample_set_compensation(), config_output(), config_props(), and main().
int avresample_read | ( | AVAudioResampleContext * | avr, |
uint8_t ** | output, | ||
int | nb_samples | ||
) |
Read samples from the output FIFO.
During conversion, if the user does not specify an output buffer or specifies an output buffer that is smaller than what is needed, remaining samples that are not written to the output are stored to an internal FIFO buffer. This function can be used to read samples from that internal FIFO.
- Parameters
-
avr audio resample context output output data pointers. May be NULL, in which case nb_samples of data is discarded from output FIFO. nb_samples number of samples to read from the FIFO
- Returns
- the number of samples written to output
Definition at line 614 of file libavresample/utils.c.
Referenced by filter_frame(), and handle_trimming().
int avresample_set_channel_mapping | ( | AVAudioResampleContext * | avr, |
const int * | channel_map | ||
) |
Set a customized input channel mapping.
This function can only be called when the allocated context is not open. Also, the input channel layout must have already been set.
Calling avresample_close() on the context will clear the channel mapping.
The map for each input channel specifies the channel index in the source to use for that particular channel, or -1 to mute the channel. Source channels can be duplicated by using the same index for multiple input channels.
Examples:
Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to Libav order (L,R,C,LFE,Ls,Rs): { 1, 2, 0, 5, 3, 4 }
Muting the 3rd channel in 4-channel input: { 0, 1, -1, 3 }
Duplicating the left channel of stereo input: { 0, 0 }
- Parameters
-
avr audio resample context channel_map customized input channel mapping
- Returns
- 0 on success, negative AVERROR code on failure
Definition at line 558 of file libavresample/utils.c.
int avresample_set_compensation | ( | AVAudioResampleContext * | avr, |
int | sample_delta, | ||
int | compensation_distance | ||
) |
Set compensation for resampling.
This can be called anytime after avresample_open(). If resampling is not automatically enabled because of a sample rate conversion, the "force_resampling" option must have been set to 1 when opening the context in order to use resampling compensation.
- Parameters
-
avr audio resample context sample_delta compensation delta, in samples compensation_distance compensation distance, in samples
- Returns
- 0 on success, negative AVERROR code on failure
Definition at line 247 of file libavresample/resample.c.
Referenced by filter_frame().
int avresample_set_matrix | ( | AVAudioResampleContext * | avr, |
const double * | matrix, | ||
int | stride | ||
) |
Set channel mixing matrix.
Allows for setting a custom mixing matrix, overriding the default matrix generated internally during avresample_open(). This function can be called anytime on an allocated context, either before or after calling avresample_open(), as long as the channel layouts have been set. avresample_convert() always uses the current matrix. Calling avresample_close() on the context will clear the current matrix.
- See also
- avresample_close()
- Parameters
-
avr audio resample context matrix mixing coefficients; matrix[i + stride * o] is the weight of input channel i in output channel o. stride distance between adjacent input channels in the matrix array
- Returns
- 0 on success, negative AVERROR code on failure
Definition at line 527 of file libavresample/utils.c.
Referenced by avresample_set_compensation().
unsigned avresample_version | ( | void | ) |
Return the LIBAVRESAMPLE_VERSION_INT constant.
Definition at line 621 of file libavresample/utils.c.
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