46 #define BLOCK_TYPE_AUDIO 1 47 #define BLOCK_TYPE_INITIAL 2 48 #define BLOCK_TYPE_SILENCE 3 56 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
57 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
58 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
59 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
60 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
61 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
62 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
63 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
64 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
65 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
66 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
67 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
68 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
98 "block align = %d, sample rate = %d\n",
109 const uint8_t *buf_end = buf + buf_size;
111 int st = channels - 1;
115 predictor[ch] = (int16_t)
AV_RL16(buf);
117 *out++ = predictor[ch];
122 while (buf < buf_end) {
128 predictor[ch] = av_clip_int16(predictor[ch]);
129 *out++ = predictor[ch];
135 int *got_frame_ptr,
AVPacket *avpkt)
140 int buf_size = avpkt->
size;
142 int block_type, silent_chunks, audio_chunks;
145 int16_t *output_samples_s16;
170 silent_chunks = av_popcount(flags);
173 }
else if (block_type == BLOCK_TYPE_SILENCE) {
184 if (silent_chunks + audio_chunks >= INT_MAX / avctx->
block_align)
192 output_samples_u8 = frame->
data[0];
193 output_samples_s16 = (int16_t *)frame->
data[0];
196 if (silent_chunks > 0) {
197 int silent_size = avctx->
block_align * silent_chunks;
201 memset(output_samples_s16, 0x00, silent_size * 2);
202 output_samples_s16 += silent_size;
204 memset(output_samples_u8, 0x80, silent_size);
205 output_samples_u8 += silent_size;
210 if (audio_chunks > 0) {
211 buf_end = buf + buf_size;
219 memcpy(output_samples_u8, buf, s->
chunk_size);
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
ptrdiff_t const GLvoid * data
#define AV_LOG_WARNING
Something somehow does not look correct.
static av_cold int init(AVCodecContext *avctx)
#define AV_CH_LAYOUT_STEREO
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
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)
#define av_assert0(cond)
assert() equivalent, that is always enabled.
enum AVSampleFormat sample_fmt
audio sample format
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
static av_cold int vmdaudio_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. ...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
simple assert() macros that are a bit more flexible than ISO C assert().
const char * name
Name of the codec implementation.
static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size, int channels)
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
#define BLOCK_TYPE_SILENCE
static const uint16_t vmdaudio_table[128]
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.
#define flags(name, subs,...)
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.
#define BLOCK_TYPE_INITIAL
common internal api header.
common internal and external API header
AVCodec ff_vmdaudio_decoder
int channels
number of audio channels
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
#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.
static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)