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66 av_log(avctx,
AV_LOG_ERROR,
"Too big bitrate: reduce sample rate, bitdepth or channels.\n");
71 s->samples_per_block = 1;
81 s->samples_per_block = 4 / avctx->
channels;
82 s->groups_per_block = 1;
87 s->samples_per_block = 1;
88 s->groups_per_block = 2;
94 s->samples_per_block = 4;
103 s->header[1] = (
quant << 6) | (freq << 4) | (avctx->
channels - 1);
117 int64_t pkt_size = (
frame->nb_samples /
s->samples_per_block) *
s->block_size + 3;
118 int blocks = (pkt_size - 3) /
s->block_size;
119 const int16_t *src16;
127 memcpy(avpkt->
data,
s->header, 3);
129 src16 = (
const int16_t *)
frame->data[0];
137 bytestream2_put_be16(&pb, *src16++);
143 for (
int i = 2;
i;
i--) {
144 bytestream2_put_be16(&pb, src32[0] >> 16);
145 bytestream2_put_be16(&pb, src32[1] >> 16);
146 bytestream2_put_byte(&pb, (*src32++) >> 24);
147 bytestream2_put_byte(&pb, (*src32++) >> 24);
152 for (
int i =
s->groups_per_block;
i;
i--) {
153 bytestream2_put_be16(&pb, src32[0] >> 16);
154 bytestream2_put_be16(&pb, src32[1] >> 16);
155 bytestream2_put_be16(&pb, src32[2] >> 16);
156 bytestream2_put_be16(&pb, src32[3] >> 16);
157 bytestream2_put_byte(&pb, (*src32++) >> 24);
158 bytestream2_put_byte(&pb, (*src32++) >> 24);
159 bytestream2_put_byte(&pb, (*src32++) >> 24);
160 bytestream2_put_byte(&pb, (*src32++) >> 24);
183 .supported_samplerates = (
const int[]) { 48000, 96000, 0},
int frame_size
Number of samples per channel in an audio frame.
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
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
int sample_rate
samples per second
#define AV_CH_LAYOUT_MONO
This structure describes decoded (raw) audio or video data.
static av_cold int pcm_dvd_encode_init(AVCodecContext *avctx)
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
#define AV_CH_LAYOUT_STEREO
static int quant(float coef, const float Q, const float rounding)
Quantize one coefficient.
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static av_always_inline void bytestream2_init_writer(PutByteContext *p, uint8_t *buf, int buf_size)
int64_t bit_rate
the average bitrate
#define AV_CH_LAYOUT_5POINT1
#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
int channels
number of audio channels
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
#define i(width, name, range_min, range_max)
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
AVSampleFormat
Audio sample formats.
#define AV_CH_LAYOUT_7POINT1
@ 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
main external API structure.
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
Filter the word “frame” indicates either a video frame or a group of audio samples
const AVCodec ff_pcm_dvd_encoder
static int pcm_dvd_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
This structure stores compressed data.
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
@ AV_SAMPLE_FMT_S32
signed 32 bits