43 "fixed",
"float",
"double" 49 "nb_consumed_samples",
62 #define OFFSET(x) offsetof(VolumeContext, x) 63 #define A AV_OPT_FLAG_AUDIO_PARAM 64 #define F AV_OPT_FLAG_FILTERING_PARAM 65 #define T AV_OPT_FLAG_RUNTIME_PARAM 68 {
"volume",
"set volume adjustment expression",
70 {
"precision",
"select mathematical precision",
78 {
"replaygain",
"Apply replaygain side data when present",
84 {
"replaygain_preamp",
"Apply replaygain pre-amplification",
86 {
"replaygain_noclip",
"Apply replaygain clipping prevention",
104 "Error when evaluating the volume expression '%s'\n", expr);
181 int nb_samples,
int volume)
184 for (i = 0; i < nb_samples; i++)
185 dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
189 int nb_samples,
int volume)
192 for (i = 0; i < nb_samples; i++)
193 dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
197 int nb_samples,
int volume)
200 int16_t *smp_dst = (int16_t *)dst;
201 const int16_t *smp_src = (
const int16_t *)src;
202 for (i = 0; i < nb_samples; i++)
203 smp_dst[i] = av_clip_int16(((int64_t)smp_src[
i] * volume + 128) >> 8);
207 int nb_samples,
int volume)
210 int16_t *smp_dst = (int16_t *)dst;
211 const int16_t *smp_src = (
const int16_t *)src;
212 for (i = 0; i < nb_samples; i++)
213 smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
217 int nb_samples,
int volume)
222 for (i = 0; i < nb_samples; i++)
223 smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
323 char *res,
int res_len,
int flags)
328 if (!strcmp(cmd,
"volume")) {
353 uint32_t peak = 100000;
362 }
else if (replaygain->
album_gain != INT32_MIN) {
369 "values are unknown.\n");
371 g = gain / 100000.0f;
372 p = peak / 100000.0f;
375 "Using gain %f dB from replaygain side data.\n", g);
424 int p, plane_samples;
432 for (p = 0; p < vol->
planes; p++) {
438 for (p = 0; p < vol->
planes; p++) {
441 vol->
volume, plane_samples);
444 for (p = 0; p < vol->
planes; p++) {
447 vol->
volume, plane_samples);
485 .priv_class = &volume_class,
488 .
inputs = avfilter_af_volume_inputs,
489 .
outputs = avfilter_af_volume_outputs,
This structure describes decoded (raw) audio or video data.
int64_t pkt_pos
reordered pos from the last AVPacket that has been input into the decoder
#define AV_LOG_WARNING
Something somehow does not look correct.
Main libavfilter public API header.
static av_cold void uninit(AVFilterContext *ctx)
static const char *const var_names[]
static av_cold void volume_init(VolumeContext *vol)
static av_cold int init(AVFilterContext *ctx)
static int set_volume(AVFilterContext *ctx)
int av_expr_parse(AVExpr **expr, const char *s, const char *const *const_names, const char *const *func1_names, double(*const *funcs1)(void *, double), const char *const *func2_names, double(*const *funcs2)(void *, double, double), int log_offset, void *log_ctx)
Parse an expression.
double var_values[VAR_VARS_NB]
uint32_t track_peak
Peak track amplitude, with 100000 representing full scale (but values may overflow).
AVFrameSideData * av_frame_get_side_data(const AVFrame *frame, enum AVFrameSideDataType type)
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static av_cold int end(AVCodecContext *avctx)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Structure to hold side data for an AVFrame.
static double av_q2d(AVRational a)
Convert an AVRational to a double.
#define AV_LOG_VERBOSE
Detailed information.
int32_t album_gain
Same as track_gain, but for the whole album.
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
static void scale_samples_s32(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
A filter pad used for either input or output.
A link between two filters.
void(* vector_dmul_scalar)(double *dst, const double *src, double mul, int len)
Multiply a vector of double by a scalar double.
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
int sample_rate
samples per second
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void * priv
private data for use by the filter
AVFILTER_DEFINE_CLASS(volume)
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
static int query_formats(AVFilterContext *ctx)
static void scale_samples_s16(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
audio channel layout utility functions
static void scale_samples_u8_small(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
static const AVFilterPad avfilter_af_volume_outputs[]
AVFilterContext * src
source filter
static void scale_samples_u8(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
static const AVFilterPad outputs[]
int format
agreed upon media format
static const char *const precision_str[]
A list of supported channel layouts.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
enum AVSampleFormat sample_fmt
AVSampleFormat
Audio sample formats.
void av_expr_free(AVExpr *e)
Free a parsed expression previously created with av_expr_parse().
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
void av_frame_remove_side_data(AVFrame *frame, enum AVFrameSideDataType type)
Remove and free all side data instances of the given type.
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
const char * name
Filter name.
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static const AVOption volume_options[]
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
static void scale_samples_s16_small(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
#define flags(name, subs,...)
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
internal math functions header
void av_opt_free(void *obj)
Free all allocated objects in obj.
void ff_volume_init_x86(VolumeContext *vol)
static const AVFilterPad avfilter_af_volume_inputs[]
common internal and external API header
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
Get the packed alternative form of the given sample format.
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
int channels
Number of channels.
static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
uint32_t album_peak
Same as track_peak, but for the whole album,.
double av_expr_eval(AVExpr *e, const double *const_values, void *opaque)
Evaluate a previously parsed expression.
AVFilterContext * dst
dest filter
static enum AVSampleFormat sample_fmts[]
int32_t track_gain
Track replay gain in microbels (divide by 100000 to get the value in dB).
ReplayGain information in the form of the AVReplayGain struct.
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
uint8_t ** extended_data
pointers to the data planes/channels.
ReplayGain information (see http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1.0_specification).
int nb_samples
number of audio samples (per channel) described by this frame
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
void(* scale_samples)(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
simple arithmetic expression evaluator
static int config_output(AVFilterLink *outlink)