188 "resample_out_buffer");
313 }
else if (converted) {
331 uint8_t **output,
int out_plane_size,
332 int out_samples,
uint8_t **input,
333 int in_plane_size,
int in_samples)
338 int ret, direct_output;
377 current_buffer = &input_buffer;
389 current_buffer = &output_buffer;
391 (!direct_output || out_samples < in_samples)) {
436 current_buffer =
NULL;
443 resample_out = &output_buffer;
447 current_buffer ? current_buffer->
name :
"null",
460 current_buffer = resample_out;
471 if (current_buffer == &output_buffer) {
477 if (direct_output && out_samples >= current_buffer->
nb_samples) {
554 int out_linesize = 0, in_linesize = 0;
555 int out_nb_samples = 0, in_nb_samples = 0;
571 in_data, in_linesize,
590 if (!bytes_per_sample)
593 samples = out->
linesize[0] / bytes_per_sample;
598 return samples / channels;
639 int in_channels, out_channels, i, o;
658 for (o = 0; o < out_channels; o++)
659 for (i = 0; i < in_channels; i++)
660 matrix[o * stride + i] = avr->
mix_matrix[o * in_channels + i];
668 int in_channels, out_channels, i, o;
689 for (o = 0; o < out_channels; o++)
690 for (i = 0; i < in_channels; i++)
691 avr->
mix_matrix[o * in_channels + i] = matrix[o * stride + i];
697 const int *channel_map)
700 int in_channels, ch, i;
708 memset(info, 0,
sizeof(*info));
711 for (ch = 0; ch < in_channels; ch++) {
712 if (channel_map[ch] >= in_channels) {
716 if (channel_map[ch] < 0) {
720 }
else if (info->
input_map[channel_map[ch]] >= 0) {
733 for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) {
734 while (ch < in_channels && info->input_map[ch] >= 0)
736 while (i < in_channels && info->channel_map[i] >= 0)
738 if (ch >= in_channels || i >= in_channels)
765 if (samples > INT_MAX)
785 #define LICENSE_PREFIX "libavresample license: "
791 return FFMPEG_CONFIGURATION;
int in_channels
number of input channels
AudioConvert * ac_in
input sample format conversion context
const char * name
name for debug logging
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
int ff_audio_data_realloc(AudioData *a, int nb_samples)
Reallocate AudioData.
This structure describes decoded (raw) audio or video data.
#define LIBAVRESAMPLE_VERSION_INT
int input_map[AVRESAMPLE_MAX_CHANNELS]
dest index of each input channel
AudioData * out_buffer
buffer for converted output
Audio buffer used for intermediate storage between conversion phases.
int64_t av_rescale_rnd(int64_t a, int64_t b, int64_t c, enum AVRounding rnd)
Rescale a 64-bit integer with specified rounding.
static int config_changed(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
int avresample_convert_frame(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
Convert the samples in the input AVFrame and write them to the output AVFrame.
memory handling functions
int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, int nb_samples)
Add samples in AudioData to an AVAudioFifo.
int do_zero
zeroing needed
AudioData * ff_audio_data_alloc(int channels, int nb_samples, enum AVSampleFormat sample_fmt, const char *name)
Allocate AudioData.
static int convert_frame(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
Read samples from the output FIFO.
double * mix_matrix
mix matrix only used if avresample_set_matrix() is called before avresample_open() ...
uint64_t out_channel_layout
output channel layout
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
Convert audio data from one sample format to another.
int avresample_set_channel_mapping(AVAudioResampleContext *avr, const int *channel_map)
Set a customized input channel mapping.
int ff_audio_mix(AudioMix *am, AudioData *src)
Apply channel mixing to audio data using the current mixing matrix.
int channel_zero[AVRESAMPLE_MAX_CHANNELS]
dest index to zero
void avresample_free(AVAudioResampleContext **avr)
Free AVAudioResampleContext and associated AVOption values.
int nb_samples
current number of samples
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
AudioData * in_buffer
buffer for converted input
int allocated_channels
allocated channel count
int out_channels
number of output channels
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
static int handle_buffered_output(AVAudioResampleContext *avr, AudioData *output, AudioData *converted)
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
Set the current mixing matrix.
int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels)
const char * avresample_license(void)
Return the libavresample license.
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
AudioConvert * ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map)
Allocate and initialize AudioConvert context for sample format conversion.
void avresample_close(AVAudioResampleContext *avr)
Close AVAudioResampleContext.
AudioMix * am
channel mixing context
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
int ff_audio_data_set_channels(AudioData *a, int channels)
int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
Resample audio data.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int out_convert_needed
output sample format conversion is needed
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
void ff_audio_convert_free(AudioConvert **ac)
Free AudioConvert.
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
AudioConvert * ac_out
output sample format conversion context
int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
Read samples from an AVAudioFifo to AudioData.
reference-counted frame API
int channel_copy[AVRESAMPLE_MAX_CHANNELS]
dest index to copy from
uint64_t channel_layout
Channel layout of the audio data.
int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
Configure or reconfigure the AVAudioResampleContext using the information provided by the AVFrames...
int upmix_needed
upmixing is needed
ResampleContext * resample
resampling context
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
enum RemapPoint remap_point
#define AVERROR_INPUT_CHANGED
Input changed between calls. Reconfiguration is required. (can be OR-ed with AVERROR_OUTPUT_CHANGED) ...
int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels, int nb_samples, enum AVSampleFormat sample_fmt, int read_only, const char *name)
Initialize AudioData using a given source.
ChannelMapInfo ch_map_info
uint64_t in_channel_layout
input channel layout
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, int stride)
Set channel mixing matrix.
int in_sample_rate
input sample rate
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
int avresample_get_delay(AVAudioResampleContext *avr)
Return the number of samples currently in the resampling delay buffer.
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix, int stride)
Get the current channel mixing matrix.
int avresample_available(AVAudioResampleContext *avr)
Return the number of available samples in the output FIFO.
void ff_audio_resample_free(ResampleContext **c)
Free a ResampleContext.
AVSampleFormat
Audio sample formats.
AVAudioFifo * out_fifo
FIFO for output samples.
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
#define AVRESAMPLE_MAX_CHANNELS
int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
Provide the upper bound on the number of samples the configured conversion would output.
enum AVSampleFormat internal_sample_fmt
internal sample format
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
int force_resampling
force resampling
void ff_audio_mix_free(AudioMix **am_p)
Free an AudioMix context.
int in_copy_needed
input data copy is needed
const char * avresample_configuration(void)
Return the libavresample build-time configuration.
int sample_rate
Sample rate of the audio data.
ResampleContext * ff_audio_resample_init(AVAudioResampleContext *avr)
Allocate and initialize a ResampleContext.
enum AVSampleFormat in_sample_fmt
input sample format
int attribute_align_arg 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 in_convert_needed
input sample format conversion is needed
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
int channel_map[AVRESAMPLE_MAX_CHANNELS]
source index of each output channel, -1 if not remapped
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
#define AVERROR_OUTPUT_CHANGED
Output changed between calls. Reconfiguration is required. (can be OR-ed with AVERROR_INPUT_CHANGED) ...
enum AVSampleFormat out_sample_fmt
output sample format
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map)
Copy data from one AudioData to another.
GLint GLenum GLboolean GLsizei stride
void av_opt_free(void *obj)
Free all allocated objects in obj.
common internal and external API header
int resample_channels
number of channels used for resampling
AudioData * resample_out_buffer
buffer for output from resampler
int resample_needed
resampling is needed
AudioMix * ff_audio_mix_alloc(AVAudioResampleContext *avr)
Allocate and initialize an AudioMix context.
int avresample_is_open(AVAudioResampleContext *avr)
Check whether an AVAudioResampleContext is open or closed.
unsigned avresample_version(void)
Return the LIBAVRESAMPLE_VERSION_INT constant.
int allocated_samples
number of samples the buffer can hold
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
int out_sample_rate
output sample rate
void ff_audio_data_free(AudioData **a)
Free AudioData.
int downmix_needed
downmixing is needed
static int available_samples(AVFrame *out)
uint8_t ** extended_data
pointers to the data planes/channels.
int avresample_open(AVAudioResampleContext *avr)
Initialize AVAudioResampleContext.
int nb_samples
number of audio samples (per channel) described by this frame
int mixing_needed
either upmixing or downmixing is needed
int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
Get the current mixing matrix.