23 #include <opus_multistream.h> 43 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 59 0, 1, 1, 2, 2, 2, 2, 3
70 { 0, 4, 1, 2, 3, 5, 6 },
71 { 0, 6, 1, 2, 3, 4, 5, 7 },
82 { 0, 1, 5, 6, 2, 4, 3 },
83 { 0, 1, 6, 7, 4, 5, 2, 3 },
87 int coupled_stream_count,
95 bytestream_put_byte(&p, 1);
96 bytestream_put_byte(&p, channels);
99 bytestream_put_le16(&p, 0);
102 bytestream_put_byte(&p, mapping_family);
103 if (mapping_family != 0) {
104 bytestream_put_byte(&p, stream_count);
105 bytestream_put_byte(&p, coupled_stream_count);
117 "Quality-based encoding not supported, " 118 "please specify a bitrate and VBR setting.\n");
122 ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->
bit_rate));
123 if (ret != OPUS_OK) {
125 "Failed to set bitrate: %s\n", opus_strerror(ret));
129 ret = opus_multistream_encoder_ctl(enc,
133 "Unable to set complexity: %s\n", opus_strerror(ret));
135 ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->
vbr));
138 "Unable to set VBR: %s\n", opus_strerror(ret));
140 ret = opus_multistream_encoder_ctl(enc,
141 OPUS_SET_VBR_CONSTRAINT(opts->
vbr == 2));
144 "Unable to set constrained VBR: %s\n", opus_strerror(ret));
146 ret = opus_multistream_encoder_ctl(enc,
150 "Unable to set expected packet loss percentage: %s\n",
153 ret = opus_multistream_encoder_ctl(enc,
154 OPUS_SET_INBAND_FEC(opts->
fec));
157 "Unable to set inband FEC: %s\n",
161 ret = opus_multistream_encoder_ctl(enc,
165 "Unable to set maximum bandwidth: %s\n", opus_strerror(ret));
168 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 169 ret = opus_multistream_encoder_ctl(enc,
170 OPUS_SET_PHASE_INVERSION_DISABLED(!opts->apply_phase_inv));
173 "Unable to set phase inversion: %s\n",
181 if (avctx->
channels > max_channels) {
195 "No channel layout specified. Opus encoder will use Vorbis " 196 "channel layout for %d channels.\n", avctx->
channels);
202 "Invalid channel layout %s for specified mapping family %d.\n",
203 name, mapping_family);
214 const uint8_t ** channel_map_result)
219 switch (mapping_family) {
247 "Unknown channel mapping family %d. Output channel layout may be invalid.\n",
252 *channel_map_result = channel_map;
260 uint8_t libopus_channel_mapping[255];
263 int coupled_stream_count, header_size,
frame_size;
267 switch (frame_size) {
272 "LPC mode cannot be used with a frame duration of less " 273 "than 10ms. Enabling restricted low-delay mode.\n" 274 "Use a longer frame duration if this is not what you want.\n");
282 #ifdef OPUS_FRAMESIZE_120_MS 292 "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40" 293 #ifdef OPUS_FRAMESIZE_120_MS
294 ", 60, 80, 100 or 120.\n",
304 "Compression level must be in the range 0 to 10. " 305 "Defaulting to 10.\n");
330 "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n" 331 "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",
349 mapping_family = avctx->
channels > 2 ? 1 : 0;
352 memcpy(libopus_channel_mapping,
354 avctx->
channels *
sizeof(*libopus_channel_mapping));
356 enc = opus_multistream_encoder_create(
358 coupled_stream_count,
366 enc = opus_multistream_surround_encoder_create(
368 &opus->
stream_count, &coupled_stream_count, libopus_channel_mapping,
372 if (ret != OPUS_OK) {
374 "Failed to create encoder: %s\n", opus_strerror(ret));
381 32000 * coupled_stream_count;
383 "No bit rate set. Defaulting to %"PRId64
" bps.\n", avctx->
bit_rate);
388 "Please choose a value between 500 and %d.\n", avctx->
bit_rate,
395 if (ret != OPUS_OK) {
401 header_size = 19 + (mapping_family == 0 ? 0 : 2 + avctx->
channels);
418 ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->
initial_padding));
421 "Unable to get number of lookahead samples: %s\n",
425 mapping_family, libopus_channel_mapping);
434 opus_multistream_encoder_destroy(enc);
441 int nb_channels,
int nb_samples,
int bytes_per_sample) {
443 for (sample = 0; sample < nb_samples; ++
sample) {
445 const size_t src_pos = bytes_per_sample * (nb_channels * sample +
channel);
446 const size_t dst_pos = bytes_per_sample * (nb_channels * sample + channel_map[
channel]);
448 memcpy(&dst[dst_pos], &src[src_pos], bytes_per_sample);
458 const int sample_size = avctx->
channels * bytes_per_sample;
476 audio = frame->
data[0];
491 ret = opus_multistream_encode_float(opus->
enc, (
float *)audio,
495 ret = opus_multistream_encode(opus->
enc, (opus_int16 *)audio,
501 "Error encoding frame: %s\n", opus_strerror(ret));
512 if ((discard_padding < opus->
opts.packet_size) != (avpkt->
duration > 0)) {
516 if (discard_padding > 0) {
524 AV_WL32(side_data + 4, discard_padding);
536 opus_multistream_encoder_destroy(opus->
enc);
546 #define OFFSET(x) offsetof(LibopusEncContext, opts.x) 547 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM 549 {
"application",
"Intended application type",
OFFSET(
application),
AV_OPT_TYPE_INT, { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY,
FLAGS,
"application" },
550 {
"voip",
"Favor improved speech intelligibility", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP }, 0, 0,
FLAGS,
"application" },
551 {
"audio",
"Favor faithfulness to the input", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO }, 0, 0,
FLAGS,
"application" },
552 {
"lowdelay",
"Restrict to only the lowest delay modes", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0,
FLAGS,
"application" },
561 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 562 {
"apply_phase_inv",
"Apply intensity stereo phase inversion",
OFFSET(apply_phase_inv),
AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1,
FLAGS },
576 {
"compression_level",
"10" },
581 48000, 24000, 16000, 12000, 8000, 0,
600 .wrapper_name =
"libopus",
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
This structure describes decoded (raw) audio or video data.
static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc, LibopusEncOpts *opts)
static int libopus_check_vorbis_layout(AVCodecContext *avctx, int mapping_family)
#define AV_LOG_WARNING
Something somehow does not look correct.
int64_t bit_rate
the average bitrate
#define LIBAVUTIL_VERSION_INT
void av_shrink_packet(AVPacket *pkt, int size)
Reduce packet size, correctly zeroing padding.
static av_cold int init(AVCodecContext *avctx)
static int libopus_validate_layout_and_get_channel_map(AVCodecContext *avctx, int mapping_family, const uint8_t **channel_map_result)
static void libopus_write_header(AVCodecContext *avctx, int stream_count, int coupled_stream_count, int mapping_family, const uint8_t *channel_mapping)
const char * av_default_item_name(void *ptr)
Return the context name.
#define FF_ARRAY_ELEMS(a)
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
int ff_opus_error_to_averror(int err)
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
enum AVSampleFormat sample_fmt
audio sample format
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
static const AVClass libopus_class
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
const uint8_t * encoder_channel_map
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int libopus_check_max_channels(AVCodecContext *avctx, int max_channels)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static const uint8_t libavcodec_libopus_channel_map[8][8]
int initial_padding
Audio only.
const char * name
Name of the codec implementation.
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
uint64_t channel_layout
Audio channel layout.
static const uint8_t opus_coupled_streams[8]
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
static const int libopus_sample_rates[]
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
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
static void libopus_copy_samples_with_channel_map(uint8_t *dst, const uint8_t *src, const uint8_t *channel_map, int nb_channels, int nb_samples, int bytes_per_sample)
int frame_size
Number of samples per channel in an audio frame.
static av_cold int libopus_encode_init(AVCodecContext *avctx)
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
int sample_rate
samples per second
main external API structure.
static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
const uint64_t ff_vorbis_channel_layouts[9]
AVCodec ff_libopus_encoder
Describe the class of an AVClass context structure.
Recommmends skipping the specified number of samples.
static av_cold int libopus_encode_close(AVCodecContext *avctx)
int global_quality
Global quality for codecs which cannot change it per frame.
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.
static const uint8_t opus_vorbis_channel_map[8][8]
common internal api header.
channel
Use these values when setting the channel map with ebur128_set_channel().
static const AVCodecDefault libopus_defaults[]
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
int cutoff
Audio cutoff bandwidth (0 means "automatic")
int channels
number of audio channels
static const AVOption libopus_options[]
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
static enum AVSampleFormat sample_fmts[]
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
const uint8_t ff_vorbis_channel_layout_offsets[8][8]
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
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
void * av_mallocz_array(size_t nmemb, size_t size)