50 #define OFFSET(x) offsetof(AMergeContext, x)
51 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
54 {
"inputs",
"specify the number of inputs",
OFFSET(nb_inputs),
81 int i, overlap = 0, nb_ch = 0;
87 "No channel layout for input %d\n", i + 1);
97 if (outlayout & inlayout[i])
99 outlayout |= inlayout[i];
108 "Input channel layouts overlap: "
109 "output layout will be determined by the number of distinct input channels\n");
110 for (i = 0; i < nb_ch; i++)
113 if (!outlayout && nb_ch)
114 outlayout = 0xFFFFFFFFFFFFFFFFULL >> (64 - nb_ch);
117 int c, out_ch_number = 0;
121 route[i] = route[i - 1] + s->
in[i - 1].
nb_ch;
122 for (c = 0; c < 64; c++)
124 if ((inlayout[i] >> c) & 1)
125 *(route[i]++) = out_ch_number++;
151 "Inputs must have the same sample rate "
152 "%d for in%d vs %d\n",
163 av_bprintf(&bp,
"%sin%d:", i ?
" + " :
"", i);
209 for (i = 0; i < nb_inputs; i++)
210 nb_ch += in[i].nb_ch;
213 for (i = 0; i < nb_inputs; i++) {
214 for (c = 0; c < in[i].nb_ch; c++) {
215 memcpy((*outs) + bps * *(route_cur++), ins[i], bps);
219 *outs += nb_ch *
bps;
229 int nb_samples, ns, i;
233 for (input_number = 0; input_number < s->
nb_inputs; input_number++)
234 if (inlink == ctx->
inputs[input_number])
253 outs = outbuf->
data[0];
256 ins[i] = inbuf[i]->
data[0] +
273 ns =
FFMIN(ns, inbuf[i]->nb_samples - s->
in[i].
pos);
300 ins[i] = inbuf[i] ? inbuf[i]->
data[0] :
NULL;
342 "a single multi-channel stream."),
349 .priv_class = &amerge_class,
void av_frame_set_channels(AVFrame *frame, int val)
static AVFrame * ff_bufqueue_get(struct FFBufQueue *queue)
Get the first buffer from the queue and remove it.
struct AMergeContext::amerge_input * in
void av_bprintf(AVBPrint *buf, const char *fmt,...)
int nb_ch
number of channels for the input
This structure describes decoded (raw) audio or video data.
#define AV_LOG_WARNING
Something somehow does not look correct.
static const AVFilterPad outputs[]
Main libavfilter public API header.
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
static enum AVSampleFormat formats[]
AVFILTER_DEFINE_CLASS(amerge)
static AVRational av_make_q(int num, int den)
Create a rational.
struct AVFilterChannelLayouts * in_channel_layouts
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
Structure holding the queue.
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 enum AVSampleFormat ff_packed_sample_fmts_array[]
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static av_cold int init(AVFilterContext *ctx)
#define AV_LOG_VERBOSE
Detailed information.
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
static av_cold void uninit(AVFilterContext *ctx)
A filter pad used for either input or output.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
A link between two filters.
static const AVFilterPad amerge_outputs[]
AVFilterPad * input_pads
array of input pads
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const AVOption amerge_options[]
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 av_bprint_init(AVBPrint *buf, unsigned size_init, unsigned size_max)
void * priv
private data for use by the filter
uint64_t * channel_layouts
list of channel layouts
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
struct AVFilterChannelLayouts * out_channel_layouts
uint64_t channel_layout
Channel layout of the audio data.
char * av_asprintf(const char *fmt,...)
AVFrame * queue[FF_BUFQUEUE_SIZE]
static int request_frame(AVFilterLink *outlink)
static int config_output(AVFilterLink *outlink)
audio channel layout utility functions
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
static int ff_bufqueue_is_full(struct FFBufQueue *queue)
Test if a buffer queue is full.
AVFilterContext * src
source filter
static void ff_bufqueue_discard_all(struct FFBufQueue *queue)
Unref and remove all buffers from the queue.
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
int format
agreed upon media format
A list of supported channel layouts.
void av_bprint_channel_layout(struct AVBPrint *bp, int nb_channels, uint64_t channel_layout)
Append a description of a channel layout to a bprint buffer.
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
static int query_formats(AVFilterContext *ctx)
#define AV_LOG_INFO
Standard information.
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
Describe the class of an AVClass context structure.
static const AVFilterPad inputs[]
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
void * av_calloc(size_t nmemb, size_t size)
Allocate a block of nmemb * size bytes with alignment suitable for all memory accesses (including vec...
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
int nb_channel_layouts
number of channel layouts
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
int channels
Number of channels.
int route[SWR_CH_MAX]
channels routing, see copy_samples
AVFilterContext * dst
dest filter
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
static void copy_samples(int nb_inputs, struct amerge_input in[], int *route, uint8_t *ins[], uint8_t **outs, int ns, int bps)
Copy samples from several input streams to one output stream.
static void ff_bufqueue_add(void *log, struct FFBufQueue *queue, AVFrame *buf)
Add a buffer to the queue.
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
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.
#define AV_NOPTS_VALUE
Undefined timestamp value.
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
static AVFrame * ff_bufqueue_peek(struct FFBufQueue *queue, unsigned index)
Get a buffer from the queue without altering it.