57 s->
state = gsm_create();
66 gsm_option(s->
state, GSM_OPT_WAV49, &one);
78 gsm_destroy(s->
state);
90 int buf_size = avpkt->
size;
93 if (buf_size < avctx->block_align) {
102 samples = (int16_t *)frame->
data[0];
105 if ((ret = gsm_decode(s->
state, buf, samples)) < 0)
120 gsm_destroy(s->
state);
121 s->
state = gsm_create();
123 gsm_option(s->
state, GSM_OPT_WAV49, &one);
126 #if CONFIG_LIBGSM_DECODER 138 .wrapper_name =
"libgsm",
141 #if CONFIG_LIBGSM_MS_DECODER 153 .wrapper_name =
"libgsm",
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
ptrdiff_t const GLvoid * data
static void flush(AVCodecContext *avctx)
static av_cold int init(AVCodecContext *avctx)
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
static int libgsm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
enum AVSampleFormat sample_fmt
audio sample format
#define GSM_MS_BLOCK_SIZE
static av_cold int libgsm_decode_init(AVCodecContext *avctx)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
AVCodec ff_libgsm_ms_decoder
const char * name
Name of the codec implementation.
static void libgsm_flush(AVCodecContext *avctx)
uint64_t channel_layout
Audio channel layout.
audio channel layout utility functions
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 the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
int frame_size
Number of samples per channel in an audio frame.
Libavcodec external API header.
int sample_rate
samples per second
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static av_cold int libgsm_decode_close(AVCodecContext *avctx)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
common internal api header.
common internal and external API header
AVCodec ff_libgsm_decoder
as in Berlin toast format
int channels
number of audio channels
Filter the word “frame” indicates either a video frame or a group of audio samples
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.