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31 #include "config_components.h"
59 s->state = gsm_create();
68 gsm_option(
s->state, GSM_OPT_WAV49, &one);
80 gsm_destroy(
s->state);
90 uint8_t *buf = avpkt->
data;
91 int buf_size = avpkt->
size;
94 if (buf_size < avctx->block_align) {
106 if ((
ret = gsm_decode(
s->state, buf,
samples)) < 0)
121 gsm_destroy(
s->state);
122 s->state = gsm_create();
124 gsm_option(
s->state, GSM_OPT_WAV49, &one);
127 #if CONFIG_LIBGSM_DECODER
134 .p.wrapper_name =
"libgsm",
142 #if CONFIG_LIBGSM_MS_DECODER
144 .
p.
name =
"libgsm_ms",
149 .p.wrapper_name =
"libgsm",
int frame_size
Number of samples per channel in an audio frame.
int sample_rate
samples per second
This structure describes decoded (raw) audio or video data.
#define AV_CHANNEL_LAYOUT_MONO
AVCodec p
The public AVCodec.
AVChannelLayout ch_layout
Audio channel layout.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
const FFCodec ff_libgsm_decoder
#define FF_CODEC_DECODE_CB(func)
#define GSM_MS_BLOCK_SIZE
const FFCodec ff_libgsm_ms_decoder
@ AV_CODEC_ID_GSM
as in Berlin toast format
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
An AVChannelLayout holds information about the channel layout of audio data.
enum AVSampleFormat sample_fmt
audio sample format
static void libgsm_flush(AVCodecContext *avctx)
#define i(width, name, range_min, range_max)
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
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
static av_cold int libgsm_decode_init(AVCodecContext *avctx)
static av_cold int libgsm_decode_close(AVCodecContext *avctx)
main external API structure.
Filter the word “frame” indicates either a video frame or a group of audio samples
This structure stores compressed data.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static int libgsm_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)