21 #include <speex/speex.h> 22 #include <speex/speex_header.h> 23 #include <speex/speex_stereo.h> 24 #include <speex/speex_callbacks.h> 43 const SpeexMode *
mode;
48 header = speex_packet_to_header(avctx->
extradata,
66 s->
pktsize = ((
const int[]){5,10,15,20,20,28,28,38,38,46,62})[quality];
71 avctx->
channels = header->nb_channels;
72 spx_mode = header->mode;
73 speex_header_free(header);
76 case 8000: spx_mode = 0;
break;
77 case 16000: spx_mode = 1;
break;
78 case 32000: spx_mode = 2;
break;
82 "Decoding as 32kHz ultra-wideband\n",
88 mode = speex_lib_get_mode(spx_mode);
100 "Decoding as stereo.\n", avctx->
channels);
106 speex_bits_init(&s->
bits);
115 callback.callback_id = SPEEX_INBAND_STEREO;
116 callback.func = speex_std_stereo_request_handler;
117 callback.data = &s->
stereo;
118 s->
stereo = (SpeexStereoState)SPEEX_STEREO_STATE_INIT;
119 speex_decoder_ctl(s->
dec_state, SPEEX_SET_HANDLER, &callback);
126 int *got_frame_ptr,
AVPacket *avpkt)
129 int buf_size = avpkt->
size;
133 int ret, consumed = 0;
140 output = (int16_t *)frame->
data[0];
146 if (speex_bits_remaining(&s->
bits) < 5 ||
147 speex_bits_peek_unsigned(&s->
bits, 5) == 0xF) {
149 if (!buf || !buf_size) {
153 if (s->
pktsize && buf_size == 62)
156 speex_bits_read_from(&s->
bits, buf, buf_size);
157 consumed = avpkt->
size;
180 speex_bits_destroy(&s->
bits);
189 speex_bits_reset(&s->
bits);
203 .wrapper_name =
"libspeex",
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
AVCodec ff_libspeex_decoder
ptrdiff_t const GLvoid * data
static void flush(AVCodecContext *avctx)
#define AV_LOG_WARNING
Something somehow does not look correct.
int64_t bit_rate
the average bitrate
static av_cold int init(AVCodecContext *avctx)
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
#define AV_CH_LAYOUT_STEREO
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
enum AVSampleFormat sample_fmt
audio sample format
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce output
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static const uint8_t header[24]
static av_cold void libspeex_decode_flush(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. ...
const char * name
Name of the codec implementation.
uint64_t channel_layout
Audio channel layout.
static void callback(void *priv_data, int index, uint8_t *buf, int buf_size, int64_t time, enum dshowDeviceType devtype)
audio channel layout utility functions
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But a word about quality
static av_cold int libspeex_decode_init(AVCodecContext *avctx)
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
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Libavcodec external API header.
int sample_rate
samples per second
main external API structure.
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static av_cold int libspeex_decode_close(AVCodecContext *avctx)
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time...
static int libspeex_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
common internal api header.
common internal and external API header
int channels
number of audio channels
#define AV_CH_LAYOUT_MONO
#define MKTAG(a, b, c, d)
This structure stores compressed data.
mode
Use these values in ebur128_init (or'ed).
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators...