FFmpeg
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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 | |
file | version.h |
Libavresample version macros. | |
Macros | |
#define | AVRESAMPLE_MAX_CHANNELS 32 |
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. | |
const char * | avresample_configuration (void) |
Return the libavresample build-time configuration. | |
const char * | avresample_license (void) |
Return the libavresample license. | |
const AVClass * | avresample_get_class (void) |
Get the AVClass for AVAudioResampleContext. | |
AVAudioResampleContext * | avresample_alloc_context (void) |
Allocate AVAudioResampleContext and set options. | |
int | avresample_open (AVAudioResampleContext *avr) |
Initialize AVAudioResampleContext. | |
int | avresample_is_open (AVAudioResampleContext *avr) |
Check whether an AVAudioResampleContext is open or closed. | |
void | avresample_close (AVAudioResampleContext *avr) |
Close AVAudioResampleContext. | |
void | avresample_free (AVAudioResampleContext **avr) |
Free AVAudioResampleContext and associated AVOption values. | |
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. | |
int | avresample_get_matrix (AVAudioResampleContext *avr, double *matrix, int stride) |
Get the current channel mixing matrix. | |
int | avresample_set_matrix (AVAudioResampleContext *avr, const double *matrix, int stride) |
Set channel mixing matrix. | |
int | avresample_set_channel_mapping (AVAudioResampleContext *avr, const int *channel_map) |
Set a customized input channel mapping. | |
int | avresample_set_compensation (AVAudioResampleContext *avr, int sample_delta, int compensation_distance) |
Set compensation for resampling. | |
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. | |
int | avresample_get_delay (AVAudioResampleContext *avr) |
Return the number of samples currently in the resampling delay buffer. | |
int | avresample_available (AVAudioResampleContext *avr) |
Return the number of available samples in the output FIFO. | |
int | avresample_read (AVAudioResampleContext *avr, uint8_t **output, int nb_samples) |
Read samples from the output FIFO. | |
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().
#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(), and main().
enum AVMixCoeffType |
Mixing Coefficient Types.
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.
enum AVResampleFilterType |
Resampling Filter Types.
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.
Definition at line 122 of file avresample.h.
unsigned avresample_version | ( | void | ) |
const char* avresample_configuration | ( | void | ) |
const char* avresample_license | ( | void | ) |
Get the AVClass for AVAudioResampleContext.
Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options without allocating a context.
Definition at line 110 of file options.c.
Referenced by init(), opt_default(), and resample_child_class_next().
AVAudioResampleContext* avresample_alloc_context | ( | void | ) |
Allocate AVAudioResampleContext and set options.
Definition at line 96 of file options.c.
Referenced by config_output(), config_props(), and main().
int avresample_open | ( | AVAudioResampleContext * | avr | ) |
Initialize AVAudioResampleContext.
avr | audio resample context |
Definition at line 35 of file utils.c.
Referenced by avresample_set_compensation(), config_output(), config_props(), and main().
int avresample_is_open | ( | AVAudioResampleContext * | avr | ) |
Check whether an AVAudioResampleContext is open or closed.
avr | AVAudioResampleContext to check |
Definition at line 262 of file utils.c.
Referenced by avresample_open().
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.
avr | audio resample context |
Definition at line 267 of file utils.c.
Referenced by avresample_free(), avresample_open(), avresample_set_compensation(), config_output(), main(), and uninit().
void avresample_free | ( | AVAudioResampleContext ** | avr | ) |
Free AVAudioResampleContext and associated AVOption values.
This also calls avresample_close() before freeing.
avr | audio resample context |
Definition at line 283 of file utils.c.
Referenced by config_output(), main(), and uninit().
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.
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) |
Definition at line 87 of file audio_mix_matrix.c.
Referenced by ff_audio_mix_alloc().
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.
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 |
Definition at line 509 of file utils.c.
Referenced by avresample_set_compensation().
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.
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 |
Definition at line 538 of file utils.c.
Referenced by avresample_set_compensation().
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 FFmpeg 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 }
avr | audio resample context |
channel_map | customized input channel mapping |
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.
avr | audio resample context |
sample_delta | compensation delta, in samples |
compensation_distance | compensation distance, in samples |
Definition at line 247 of file resample.c.
Referenced by filter_frame().
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().
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 |
Definition at line 335 of file utils.c.
Referenced by filter_frame(), main(), request_frame(), wrap(), and write_to_fifo().
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.
avr | audio resample context |
Definition at line 463 of file resample.c.
Referenced by filter_frame(), get_delay(), main(), and request_frame().
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().
avr | audio resample context |
Definition at line 620 of file utils.c.
Referenced by filter_frame(), get_delay(), handle_trimming(), 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.
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 |
Definition at line 625 of file utils.c.
Referenced by filter_frame(), and handle_trimming().