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32 #if CONFIG_LIBOPENCORE_AMRNB_DECODER || CONFIG_LIBOPENCORE_AMRWB_DECODER
52 #if CONFIG_LIBOPENCORE_AMRNB
54 #include <opencore-amrnb/interf_dec.h>
55 #include <opencore-amrnb/interf_enc.h>
68 #if CONFIG_LIBOPENCORE_AMRNB_DECODER
74 if ((
ret = amr_decode_fix_avctx(avctx)) < 0)
77 s->dec_state = Decoder_Interface_init();
90 Decoder_Interface_exit(
s->dec_state);
100 int buf_size = avpkt->
size;
102 static const uint8_t block_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };
104 int packet_size,
ret;
106 ff_dlog(avctx,
"amr_decode_frame buf=%p buf_size=%d frame_count=%d!!\n",
110 frame->nb_samples = 160;
114 dec_mode = (buf[0] >> 3) & 0x000F;
115 packet_size = block_size[dec_mode] + 1;
117 if (packet_size > buf_size) {
119 buf_size, packet_size);
123 ff_dlog(avctx,
"packet_size=%d buf= 0x%"PRIx8
" %"PRIx8
" %"PRIx8
" %"PRIx8
"\n",
124 packet_size, buf[0], buf[1], buf[2], buf[3]);
126 Decoder_Interface_Decode(
s->dec_state, buf, (
short *)
frame->data[0], 0);
134 .
name =
"libopencore_amrnb",
139 .
init = amr_nb_decode_init,
140 .close = amr_nb_decode_close,
141 .
decode = amr_nb_decode_frame,
146 #if CONFIG_LIBOPENCORE_AMRNB_ENCODER
148 typedef struct AMR_bitrates {
154 static int get_bitrate_mode(
int bitrate,
void *log_ctx)
157 static const AMR_bitrates
rates[] = {
158 { 4750, MR475 }, { 5150, MR515 }, { 5900, MR59 }, { 6700, MR67 },
159 { 7400, MR74 }, { 7950, MR795 }, { 10200, MR102 }, { 12200, MR122 }
161 int i, best = -1, min_diff = 0;
164 for (
i = 0;
i < 8;
i++) {
173 snprintf(log_buf,
sizeof(log_buf),
"bitrate not supported: use one of ");
174 for (
i = 0;
i < 8;
i++)
187 static const AVClass amrnb_class = {
212 s->enc_state = Encoder_Interface_init(
s->enc_dtx);
218 s->enc_mode = get_bitrate_mode(avctx->
bit_rate, avctx);
228 Encoder_Interface_exit(
s->enc_state);
238 int16_t *flush_buf =
NULL;
242 s->enc_mode = get_bitrate_mode(avctx->
bit_rate, avctx);
254 memcpy(flush_buf,
samples,
frame->nb_samples *
sizeof(*flush_buf));
257 s->enc_last_frame = -1;
264 if (
s->enc_last_frame < 0)
270 s->enc_last_frame = -1;
273 written = Encoder_Interface_Encode(
s->enc_state,
s->enc_mode,
samples,
275 ff_dlog(avctx,
"amr_nb_encode_frame encoded %u bytes, bitrate %u, first byte was %#02x\n",
276 written,
s->enc_mode, avpkt->
data[0]);
282 avpkt->
size = written;
289 .
name =
"libopencore_amrnb",
294 .
init = amr_nb_encode_init,
295 .encode2 = amr_nb_encode_frame,
296 .close = amr_nb_encode_close,
300 .priv_class = &amrnb_class,
307 #if CONFIG_LIBOPENCORE_AMRWB_DECODER
309 #include <opencore-amrwb/dec_if.h>
310 #include <opencore-amrwb/if_rom.h>
321 if ((
ret = amr_decode_fix_avctx(avctx)) < 0)
324 s->state = D_IF_init();
330 int *got_frame_ptr,
AVPacket *avpkt)
334 int buf_size = avpkt->
size;
338 static const uint8_t block_size[16] = {18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1};
341 frame->nb_samples = 320;
345 mode = (buf[0] >> 3) & 0x000F;
346 packet_size = block_size[
mode];
348 if (packet_size > buf_size) {
350 buf_size, packet_size + 1);
358 D_IF_decode(
s->state, buf, (
short *)
frame->data[0], _good_frame);
374 .
name =
"libopencore_amrwb",
379 .
init = amr_wb_decode_init,
380 .close = amr_wb_decode_close,
381 .
decode = amr_wb_decode_frame,
383 .wrapper_name =
"libopencore_amrwb",
int frame_size
Number of samples per channel in an audio frame.
#define AV_LOG_WARNING
Something somehow does not look correct.
static av_cold int init(AVCodecContext *avctx)
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
uint64_t channel_layout
Audio channel layout.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
int sample_rate
samples per second
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
static enum AVSampleFormat sample_fmts[]
#define AV_CH_LAYOUT_MONO
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
This structure describes decoded (raw) audio or video data.
void * av_mallocz_array(size_t nmemb, size_t size)
Allocate a memory block for an array with av_mallocz().
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
#define FF_COMPLIANCE_UNOFFICIAL
Allow unofficial extensions.
int initial_padding
Audio only.
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Mode
Frame type (Table 1a in 3GPP TS 26.101)
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
static const AVClass av_class
#define AV_OPT_FLAG_AUDIO_PARAM
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
int64_t bit_rate
the average bitrate
const char * av_default_item_name(void *ptr)
Return the context name.
const OptionDef options[]
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() 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
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
AVCodec ff_libopencore_amrnb_decoder
int channels
number of audio channels
AVCodec ff_libopencore_amrwb_decoder
#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.
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
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
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
main external API structure.
#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 frame_number
Frame counter, set by libavcodec.
AVCodec ff_libopencore_amrnb_encoder
This structure stores compressed data.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.