Audio sample formats.

• The data described by the sample format is always in native-endian order. Sample values can be expressed by native C types, hence the lack of a signed 24-bit sample format even though it is a common raw audio data format.
• The floating-point formats are based on full volume being in the range [-1.0, 1.0]. Any values outside this range are beyond full volume level.
• The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg (such as AVFrame in libavcodec) is as follows:

For planar sample formats, each audio channel is in a separate data plane, and linesize is the buffer size, in bytes, for a single plane. All data planes must be the same size. For packed sample formats, only the first data plane is used, and samples for each channel are interleaved. In this case, linesize is the buffer size, in bytes, for the 1 plane.

Definition at line 59 of file samplefmt.h.

Return the name of sample_fmt, or NULL if sample_fmt is not recognized.

Definition at line 47 of file samplefmt.c.

Referenced by audio_decode_frame(), audio_encode_example(), audio_frame_cksum(), audio_thread(), av_audio_resample_init(), av_opt_get(), avcodec_open2(), avcodec_string(), avresample_open(), choose_sample_fmt(), config_output(), config_props(), configure_input_audio_filter(), decode_audio(), dprint_options(), ff_all_formats(), ff_audio_convert_set_func(), ff_audio_mix_alloc(), ff_audio_mix_set_func(), ff_audio_resample_init(), ff_dither_alloc(), ff_tlog_link(), filter_frame(), get_format_from_sample_fmt(), init_audio(), init_filter(), init_filter_graph(), init_filters(), lavfi_read_header(), main(), mix_function_init(), opt_list(), pick_format(), print_digraph(), print_link_prop(), show_frame(), show_stream(), and swr_init().

Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE on error.

Definition at line 54 of file samplefmt.c.

Referenced by ff_parse_sample_format(), init(), new_audio_stream(), and set_string_sample_fmt().

Return the planar<->packed alternative form of the given sample format, or AV_SAMPLE_FMT_NONE on error.

If the passed sample_fmt is already in the requested planar/packed format, the format returned is the same as the input.

Definition at line 64 of file samplefmt.c.

Referenced by get(), set(), and shift().

Get the packed alternative form of the given sample format.

If the passed sample_fmt is already in packed format, the format returned is the same as the input.

Returns

the packed alternative form of the given sample format or AV_SAMPLE_FMT_NONE on error.

Definition at line 73 of file samplefmt.c.

Referenced by auto_matrix(), cmp_audio_fmts(), ff_audio_convert_set_func(), ff_dither_alloc(), ff_volume_init_x86(), filter_frame(), get_fmt_score(), main(), PUT_FUNC(), swap_sample_fmts_on_filter(), swri_dither_init(), and volume_init().

Get the planar alternative form of the given sample format.

If the passed sample_fmt is already in planar format, the format returned is the same as the input.

Returns

the planar alternative form of the given sample format or AV_SAMPLE_FMT_NONE on error.

Definition at line 82 of file samplefmt.c.

Referenced by avcodec_open2(), avresample_open(), config_output(), swap_sample_fmts_on_filter(), and swr_init().

Generate a string corresponding to the sample format with sample_fmt, or a header if sample_fmt is negative.

Parameters

buf the buffer where to write the string

buf_size the size of buf

sample_fmt the number of the sample format to print the corresponding info string, or a negative value to print the corresponding header.

Returns

the pointer to the filled buffer or NULL if sample_fmt is unknown or in case of other errors

Definition at line 91 of file samplefmt.c.

Referenced by show_sample_fmts().

Return number of bytes per sample.

Parameters

sample_fmt the sample format

Returns

number of bytes per sample or zero if unknown for the given sample format

Definition at line 104 of file samplefmt.c.

Referenced by audio_decode_example(), audio_decode_frame(), audio_resample(), av_audio_resample_init(), av_samples_copy(), av_samples_get_buffer_size(), av_samples_set_silence(), available_samples(), avcodec_string(), avresample_open(), buffer_offset(), config_input(), config_output(), copy_samples(), decode_frame(), decode_init(), decode_packet(), fdk_aac_decode_frame(), ff_audio_data_alloc(), ff_audio_data_init(), ff_audio_resample(), ff_audio_resample_init(), ff_dither_alloc(), ff_parse_sample_format(), ff_planar_sample_fmts(), filter_frame(), flac_set_bps(), get_fmt_score(), init_sample_buffers(), lavfi_read_packet(), libopus_encode(), libopus_encode_init(), mkv_write_track(), process_output(), pulse_write_frame(), pulse_write_header(), pulse_write_packet(), resample_init(), set_audiodata_fmt(), setup_array(), shift(), swap_sample_fmts_on_filter(), swri_dither_init(), swri_rematrix(), vmdaudio_decode_init(), wavpack_decode_block(), wv_unpack_mono(), wv_unpack_stereo(), and yae_reset().

Check if the sample format is planar.

Parameters

sample_fmt the sample format to inspect

Returns

1 if the sample format is planar, 0 if it is interleaved

Definition at line 110 of file samplefmt.c.

Referenced by audio_frame_cksum(), audio_write_frame(), av_audio_fifo_alloc(), av_frame_get_plane_buffer(), av_samples_alloc_array_and_samples(), av_samples_copy(), av_samples_fill_arrays(), av_samples_get_buffer_size(), av_samples_set_silence(), available_samples(), avcodec_encode_audio2(), avcodec_fill_audio_frame(), buffer_offset(), calc_ptr_alignment(), config_output(), ff_planar_sample_fmts(), ff_sample_fmt_is_planar(), filter_frame(), flac_set_bps(), frame_copy_audio(), get(), get_audio_buffer(), get_fmt_score(), main(), process_output(), pulse_write_frame(), set(), set_audiodata_fmt(), setup_array(), shift(), and update_frame_pool().

Get the required buffer size for the given audio parameters.

Parameters

[out] linesize calculated linesize, may be NULL

nb_channels the number of channels

nb_samples the number of samples in a single channel

sample_fmt the sample format

align buffer size alignment (0 = default, 1 = no alignment)

Returns

required buffer size, or negative error code on failure

Definition at line 117 of file samplefmt.c.

Referenced by allocate_buffers(), audio_decode_frame(), audio_encode_example(), audio_open(), av_audio_fifo_alloc(), av_audio_fifo_realloc(), av_samples_alloc(), av_samples_fill_arrays(), avcodec_fill_audio_frame(), ff_audio_data_realloc(), get_audio_buffer(), main(), tak_decode_frame(), and update_frame_pool().