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71 const uint8_t *buf = avpkt->
data;
72 int buf_size = avpkt->
size;
75 if (buf_size < avctx->block_align) {
110 memset(
s, 0,
sizeof(*
s));
113 #if CONFIG_GSM_DECODER
127 #if CONFIG_GSM_MS_DECODER
int frame_size
Number of samples per channel in an audio frame.
#define AV_LOG_WARNING
Something somehow does not look correct.
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
static av_cold int gsm_init(AVCodecContext *avctx)
uint64_t channel_layout
Audio channel layout.
const AVCodec ff_gsm_ms_decoder
int sample_rate
samples per second
#define AV_CH_LAYOUT_MONO
This structure describes decoded (raw) audio or video data.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
int ff_msgsm_decode_block(AVCodecContext *avctx, int16_t *samples, const uint8_t *buf, int mode)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static int gsm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
#define GSM_MS_BLOCK_SIZE
static void flush(AVCodecContext *avctx)
@ AV_CODEC_ID_GSM
as in Berlin toast format
#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.
enum AVSampleFormat sample_fmt
audio sample format
#define MSN_MIN_BLOCK_SIZE
const AVCodec ff_gsm_decoder
int channels
number of audio channels
static int gsm_decode_block(AVCodecContext *avctx, int16_t *samples, GetBitContext *gb, int mode)
@ 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 void gsm_flush(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.