if (!*got_output) {
/* no picture yet */
av_freep(&decoded_frame);
- return 0;
+ return ret;
}
- ist->next_pts = ist->pts = guess_correct_pts(&ist->pts_ctx, decoded_frame->pkt_pts,
- decoded_frame->pkt_dts);
+ ist->next_pts = ist->pts = decoded_frame->best_effort_timestamp;
if (ist->st->codec->time_base.num != 0) {
int ticks = ist->st->parser ? ist->st->parser->repeat_pict + 1 :
ist->st->codec->ticks_per_frame;
API changes, most recent first:
+2011-11-03 - 96949da - lavu 51.23.0
+ Add av_strcasecmp() and av_strncasecmp() to avstring.h.
+
+2011-10-20 - b35e9e1 - lavu 51.22.0
+ Add av_strtok() to avstring.h.
+
+ 2011-xx-xx - xxxxxxx - lavu 51.18.0
+ Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and
+ av_samples_alloc(), to samplefmt.h.
+
+ 2011-xx-xx - xxxxxxx - lavu 51.17.0
+ Add planar sample formats and av_sample_fmt_is_planar() to samplefmt.h.
+
2011-xx-xx - xxxxxxx - lavc 53.21.0
Move some AVCodecContext fields to a new private struct, AVCodecInternal,
which is accessed from a new field, AVCodecContext.internal.
*frame_size = 0;
- if(video_sync_method){
- double vdelta = sync_ipts - ost->sync_opts;
+ format_video_sync = video_sync_method;
+ if (format_video_sync < 0)
- format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? 2 : 1;
++ format_video_sync = (s->oformat->flags & AVFMT_NOTIMESTAMPS) ? 0 :
++ (s->oformat->flags & AVFMT_VARIABLE_FPS) ? 2 : 1;
+
+ if (format_video_sync) {
+ double vdelta = sync_ipts - ost->sync_opts + duration;
//FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
if (vdelta < -1.1)
nb_frames = 0;
memset(buf, fill_char, size);
}
-/* pkt = NULL means EOF (needed to flush decoder buffers) */
-static int output_packet(InputStream *ist, int ist_index,
- OutputStream **ost_table, int nb_ostreams,
- const AVPacket *pkt)
+static void flush_encoders(OutputStream *ost_table, int nb_ostreams)
{
- AVFormatContext *os;
- OutputStream *ost;
- int ret, i;
- int got_output;
- AVFrame picture;
- void *buffer_to_free = NULL;
- static unsigned int samples_size= 0;
- AVSubtitle subtitle, *subtitle_to_free;
- int64_t pkt_pts = AV_NOPTS_VALUE;
-#if CONFIG_AVFILTER
- int frame_available;
-#endif
- float quality;
-
- AVPacket avpkt;
- int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
+ int i, ret;
- if(ist->next_pts == AV_NOPTS_VALUE)
- ist->next_pts= ist->pts;
+ for (i = 0; i < nb_ostreams; i++) {
+ OutputStream *ost = &ost_table[i];
+ AVCodecContext *enc = ost->st->codec;
+ AVFormatContext *os = output_files[ost->file_index].ctx;
- if (pkt == NULL) {
- /* EOF handling */
- av_init_packet(&avpkt);
- avpkt.data = NULL;
- avpkt.size = 0;
- goto handle_eof;
- } else {
- avpkt = *pkt;
- }
+ if (!ost->encoding_needed)
+ continue;
- if(pkt->dts != AV_NOPTS_VALUE)
- ist->next_pts = ist->pts = av_rescale_q(pkt->dts, ist->st->time_base, AV_TIME_BASE_Q);
- if(pkt->pts != AV_NOPTS_VALUE)
- pkt_pts = av_rescale_q(pkt->pts, ist->st->time_base, AV_TIME_BASE_Q);
+ if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <=1)
+ continue;
+ if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE))
+ continue;
- //while we have more to decode or while the decoder did output something on EOF
- while (avpkt.size > 0 || (!pkt && got_output)) {
- uint8_t *data_buf, *decoded_data_buf;
- int data_size, decoded_data_size;
- handle_eof:
- ist->pts= ist->next_pts;
+ for(;;) {
+ AVPacket pkt;
+ int fifo_bytes;
+ av_init_packet(&pkt);
+ pkt.stream_index= ost->index;
- if(avpkt.size && avpkt.size != pkt->size &&
- ((!ist->showed_multi_packet_warning && verbose>0) || verbose>1)){
- fprintf(stderr, "Multiple frames in a packet from stream %d\n", pkt->stream_index);
- ist->showed_multi_packet_warning=1;
- }
+ switch (ost->st->codec->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ fifo_bytes = av_fifo_size(ost->fifo);
+ ret = 0;
+ /* encode any samples remaining in fifo */
+ if (fifo_bytes > 0) {
+ int osize = av_get_bytes_per_sample(enc->sample_fmt);
+ int fs_tmp = enc->frame_size;
+
+ av_fifo_generic_read(ost->fifo, audio_buf, fifo_bytes, NULL);
+ if (enc->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) {
+ enc->frame_size = fifo_bytes / (osize * enc->channels);
+ } else { /* pad */
+ int frame_bytes = enc->frame_size*osize*enc->channels;
+ if (allocated_audio_buf_size < frame_bytes)
+ exit_program(1);
+ generate_silence(audio_buf+fifo_bytes, enc->sample_fmt, frame_bytes - fifo_bytes);
+ }
- /* decode the packet if needed */
- decoded_data_buf = NULL; /* fail safe */
- decoded_data_size= 0;
- data_buf = avpkt.data;
- data_size = avpkt.size;
- subtitle_to_free = NULL;
- if (ist->decoding_needed) {
- switch(ist->st->codec->codec_type) {
- case AVMEDIA_TYPE_AUDIO:{
- if(pkt && samples_size < FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
- samples_size = FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE);
- av_free(samples);
- samples= av_malloc(samples_size);
+ ret = avcodec_encode_audio(enc, bit_buffer, bit_buffer_size, (short *)audio_buf);
+ pkt.duration = av_rescale((int64_t)enc->frame_size*ost->st->time_base.den,
+ ost->st->time_base.num, enc->sample_rate);
+ enc->frame_size = fs_tmp;
}
- decoded_data_size= samples_size;
- /* XXX: could avoid copy if PCM 16 bits with same
- endianness as CPU */
- ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size,
- &avpkt);
- if (ret < 0)
- return ret;
- avpkt.data += ret;
- avpkt.size -= ret;
- data_size = ret;
- got_output = decoded_data_size > 0;
- /* Some bug in mpeg audio decoder gives */
- /* decoded_data_size < 0, it seems they are overflows */
- if (!got_output) {
- /* no audio frame */
- continue;
+ if (ret <= 0) {
+ ret = avcodec_encode_audio(enc, bit_buffer, bit_buffer_size, NULL);
}
- decoded_data_buf = (uint8_t *)samples;
- ist->next_pts += ((int64_t)AV_TIME_BASE/bps * decoded_data_size) /
- (ist->st->codec->sample_rate * ist->st->codec->channels);
- break;}
- case AVMEDIA_TYPE_VIDEO:
- decoded_data_size = (ist->st->codec->width * ist->st->codec->height * 3) / 2;
- /* XXX: allocate picture correctly */
- avcodec_get_frame_defaults(&picture);
- avpkt.pts = pkt_pts;
- avpkt.dts = ist->pts;
- pkt_pts = AV_NOPTS_VALUE;
-
- ret = avcodec_decode_video2(ist->st->codec,
- &picture, &got_output, &avpkt);
- quality = same_quality ? picture.quality : 0;
- if (ret < 0)
- return ret;
- if (!got_output) {
- /* no picture yet */
- goto discard_packet;
- }
- ist->next_pts = ist->pts = guess_correct_pts(&ist->pts_ctx, picture.pkt_pts, picture.pkt_dts);
- if (ist->st->codec->time_base.num != 0) {
- int ticks= ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->st->codec->ticks_per_frame;
- ist->next_pts += ((int64_t)AV_TIME_BASE *
- ist->st->codec->time_base.num * ticks) /
- ist->st->codec->time_base.den;
- }
- avpkt.size = 0;
- buffer_to_free = NULL;
- pre_process_video_frame(ist, (AVPicture *)&picture, &buffer_to_free);
- break;
- case AVMEDIA_TYPE_SUBTITLE:
- ret = avcodec_decode_subtitle2(ist->st->codec,
- &subtitle, &got_output, &avpkt);
- if (ret < 0)
- return ret;
- if (!got_output) {
- goto discard_packet;
+ if (ret < 0) {
+ av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
+ exit_program(1);
}
- subtitle_to_free = &subtitle;
- avpkt.size = 0;
- break;
- default:
- return -1;
- }
- } else {
- switch(ist->st->codec->codec_type) {
- case AVMEDIA_TYPE_AUDIO:
- ist->next_pts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) /
- ist->st->codec->sample_rate;
+ audio_size += ret;
+ pkt.flags |= AV_PKT_FLAG_KEY;
break;
case AVMEDIA_TYPE_VIDEO:
- if (ist->st->codec->time_base.num != 0) {
- int ticks= ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->st->codec->ticks_per_frame;
- ist->next_pts += ((int64_t)AV_TIME_BASE *
- ist->st->codec->time_base.num * ticks) /
- ist->st->codec->time_base.den;
+ ret = avcodec_encode_video(enc, bit_buffer, bit_buffer_size, NULL);
+ if (ret < 0) {
+ av_log(NULL, AV_LOG_FATAL, "Video encoding failed\n");
+ exit_program(1);
+ }
+ video_size += ret;
+ if(enc->coded_frame && enc->coded_frame->key_frame)
+ pkt.flags |= AV_PKT_FLAG_KEY;
+ if (ost->logfile && enc->stats_out) {
+ fprintf(ost->logfile, "%s", enc->stats_out);
}
break;
+ default:
+ ret=-1;
}
- ret = avpkt.size;
- avpkt.size = 0;
+
+ if (ret <= 0)
+ break;
+ pkt.data = bit_buffer;
+ pkt.size = ret;
+ if (enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
+ pkt.pts= av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
+ write_frame(os, &pkt, ost->st->codec, ost->bitstream_filters);
}
+ }
+}
-#if CONFIG_AVFILTER
- if (ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
- for (i = 0; i < nb_ostreams; i++) {
- ost = ost_table[i];
- if (ost->input_video_filter && ost->source_index == ist_index) {
- AVRational sar;
- if (ist->st->sample_aspect_ratio.num)
- sar = ist->st->sample_aspect_ratio;
- else
- sar = ist->st->codec->sample_aspect_ratio;
- // add it to be filtered
- av_vsrc_buffer_add_frame(ost->input_video_filter, &picture,
- ist->pts,
- sar);
- }
+/*
+ * Check whether a packet from ist should be written into ost at this time
+ */
+static int check_output_constraints(InputStream *ist, OutputStream *ost)
+{
+ OutputFile *of = &output_files[ost->file_index];
+ int ist_index = ist - input_streams;
+
+ if (ost->source_index != ist_index)
+ return 0;
+
+ if (of->start_time && ist->pts < of->start_time)
+ return 0;
+
+ if (of->recording_time != INT64_MAX &&
+ av_compare_ts(ist->pts, AV_TIME_BASE_Q, of->recording_time + of->start_time,
+ (AVRational){1, 1000000}) >= 0) {
+ ost->is_past_recording_time = 1;
+ return 0;
+ }
+
+ return 1;
+}
+
+static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *pkt)
+{
+ OutputFile *of = &output_files[ost->file_index];
+ int64_t ost_tb_start_time = av_rescale_q(of->start_time, AV_TIME_BASE_Q, ost->st->time_base);
+ AVPicture pict;
+ AVPacket opkt;
+
+ av_init_packet(&opkt);
+
+ if ((!ost->frame_number && !(pkt->flags & AV_PKT_FLAG_KEY)) &&
+ !ost->copy_initial_nonkeyframes)
+ return;
+
+ /* force the input stream PTS */
+ if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
+ audio_size += pkt->size;
+ else if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
+ video_size += pkt->size;
+ ost->sync_opts++;
+ }
+
+ opkt.stream_index = ost->index;
+ if (pkt->pts != AV_NOPTS_VALUE)
+ opkt.pts = av_rescale_q(pkt->pts, ist->st->time_base, ost->st->time_base) - ost_tb_start_time;
+ else
+ opkt.pts = AV_NOPTS_VALUE;
+
+ if (pkt->dts == AV_NOPTS_VALUE)
+ opkt.dts = av_rescale_q(ist->pts, AV_TIME_BASE_Q, ost->st->time_base);
+ else
+ opkt.dts = av_rescale_q(pkt->dts, ist->st->time_base, ost->st->time_base);
+ opkt.dts -= ost_tb_start_time;
+
+ opkt.duration = av_rescale_q(pkt->duration, ist->st->time_base, ost->st->time_base);
+ opkt.flags = pkt->flags;
+
+ //FIXME remove the following 2 lines they shall be replaced by the bitstream filters
+ if( ost->st->codec->codec_id != CODEC_ID_H264
+ && ost->st->codec->codec_id != CODEC_ID_MPEG1VIDEO
+ && ost->st->codec->codec_id != CODEC_ID_MPEG2VIDEO
+ ) {
+ if (av_parser_change(ist->st->parser, ost->st->codec, &opkt.data, &opkt.size, pkt->data, pkt->size, pkt->flags & AV_PKT_FLAG_KEY))
+ opkt.destruct = av_destruct_packet;
+ } else {
+ opkt.data = pkt->data;
+ opkt.size = pkt->size;
+ }
+ if (of->ctx->oformat->flags & AVFMT_RAWPICTURE) {
+ /* store AVPicture in AVPacket, as expected by the output format */
+ avpicture_fill(&pict, opkt.data, ost->st->codec->pix_fmt, ost->st->codec->width, ost->st->codec->height);
+ opkt.data = (uint8_t *)&pict;
+ opkt.size = sizeof(AVPicture);
+ opkt.flags |= AV_PKT_FLAG_KEY;
+ }
+
+ write_frame(of->ctx, &opkt, ost->st->codec, ost->bitstream_filters);
+ ost->st->codec->frame_number++;
+ ost->frame_number++;
+ av_free_packet(&opkt);
+}
+
+static void rate_emu_sleep(InputStream *ist)
+{
+ if (input_files[ist->file_index].rate_emu) {
+ int64_t pts = av_rescale(ist->pts, 1000000, AV_TIME_BASE);
+ int64_t now = av_gettime() - ist->start;
+ if (pts > now)
+ usleep(pts - now);
+ }
+}
+
+static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
+{
+ static unsigned int samples_size = 0;
+ int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
+ uint8_t *decoded_data_buf = NULL;
+ int decoded_data_size = 0;
+ int i, ret;
+
+ if (pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
+ av_free(samples);
+ samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);
+ samples = av_malloc(samples_size);
+ }
+ decoded_data_size = samples_size;
+
+ ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size,
+ pkt);
+ if (ret < 0)
+ return ret;
- pkt->data += ret;
- pkt->size -= ret;
+ *got_output = decoded_data_size > 0;
+
+ /* Some bug in mpeg audio decoder gives */
+ /* decoded_data_size < 0, it seems they are overflows */
+ if (!*got_output) {
+ /* no audio frame */
- return 0;
++ return ret;
+ }
+
+ decoded_data_buf = (uint8_t *)samples;
+ ist->next_pts += ((int64_t)AV_TIME_BASE/bps * decoded_data_size) /
+ (ist->st->codec->sample_rate * ist->st->codec->channels);
+
+ // preprocess audio (volume)
+ if (audio_volume != 256) {
+ switch (ist->st->codec->sample_fmt) {
+ case AV_SAMPLE_FMT_U8:
+ {
+ uint8_t *volp = samples;
+ for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
+ int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;
+ *volp++ = av_clip_uint8(v);
}
+ break;
}
-#endif
-
- // preprocess audio (volume)
- if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
- if (audio_volume != 256) {
- short *volp;
- volp = samples;
- for(i=0;i<(decoded_data_size / sizeof(short));i++) {
- int v = ((*volp) * audio_volume + 128) >> 8;
- if (v < -32768) v = -32768;
- if (v > 32767) v = 32767;
- *volp++ = v;
- }
+ case AV_SAMPLE_FMT_S16:
+ {
+ int16_t *volp = samples;
+ for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
+ int v = ((*volp) * audio_volume + 128) >> 8;
+ *volp++ = av_clip_int16(v);
}
+ break;
}
-
- /* frame rate emulation */
- if (rate_emu) {
- int64_t pts = av_rescale(ist->pts, 1000000, AV_TIME_BASE);
- int64_t now = av_gettime() - ist->start;
- if (pts > now)
- usleep(pts - now);
+ case AV_SAMPLE_FMT_S32:
+ {
+ int32_t *volp = samples;
+ for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
+ int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);
+ *volp++ = av_clipl_int32(v);
+ }
+ break;
}
- /* if output time reached then transcode raw format,
- encode packets and output them */
- if (start_time == 0 || ist->pts >= start_time)
- for(i=0;i<nb_ostreams;i++) {
- int frame_size;
-
- ost = ost_table[i];
- if (ost->source_index == ist_index) {
-#if CONFIG_AVFILTER
- frame_available = ist->st->codec->codec_type != AVMEDIA_TYPE_VIDEO ||
- !ost->output_video_filter || avfilter_poll_frame(ost->output_video_filter->inputs[0]);
- while (frame_available) {
- AVRational ist_pts_tb;
- if (ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && ost->output_video_filter)
- get_filtered_video_frame(ost->output_video_filter, &picture, &ost->picref, &ist_pts_tb);
- if (ost->picref)
- ist->pts = av_rescale_q(ost->picref->pts, ist_pts_tb, AV_TIME_BASE_Q);
-#endif
- os = output_files[ost->file_index];
-
- /* set the input output pts pairs */
- //ost->sync_ipts = (double)(ist->pts + input_files[ist->file_index].ts_offset - start_time)/ AV_TIME_BASE;
-
- if (ost->encoding_needed) {
- av_assert0(ist->decoding_needed);
- switch(ost->st->codec->codec_type) {
- case AVMEDIA_TYPE_AUDIO:
- do_audio_out(os, ost, ist, decoded_data_buf, decoded_data_size);
- break;
- case AVMEDIA_TYPE_VIDEO:
+ case AV_SAMPLE_FMT_FLT:
+ {
+ float *volp = samples;
+ float scale = audio_volume / 256.f;
+ for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
+ *volp++ *= scale;
+ }
+ break;
+ }
+ case AV_SAMPLE_FMT_DBL:
+ {
+ double *volp = samples;
+ double scale = audio_volume / 256.;
+ for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
+ *volp++ *= scale;
+ }
+ break;
+ }
+ default:
+ av_log(NULL, AV_LOG_FATAL,
+ "Audio volume adjustment on sample format %s is not supported.\n",
+ av_get_sample_fmt_name(ist->st->codec->sample_fmt));
+ exit_program(1);
+ }
+ }
+
+ rate_emu_sleep(ist);
+
+ for (i = 0; i < nb_output_streams; i++) {
+ OutputStream *ost = &output_streams[i];
+
+ if (!check_output_constraints(ist, ost) || !ost->encoding_needed)
+ continue;
+ do_audio_out(output_files[ost->file_index].ctx, ost, ist,
+ decoded_data_buf, decoded_data_size);
+ }
- return 0;
++ return ret;
+}
+
+static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int64_t *pkt_pts, int64_t *pkt_dts)
+{
+ AVFrame *decoded_frame, *filtered_frame = NULL;
+ void *buffer_to_free = NULL;
+ int i, ret = 0;
+ float quality = 0;
#if CONFIG_AVFILTER
- if (ost->picref->video && !ost->frame_aspect_ratio)
- ost->st->codec->sample_aspect_ratio = ost->picref->video->pixel_aspect;
+ int frame_available = 1;
#endif
- do_video_out(os, ost, ist, &picture, &frame_size,
- same_quality ? quality : ost->st->codec->global_quality);
- if (vstats_filename && frame_size)
- do_video_stats(os, ost, frame_size);
- break;
- case AVMEDIA_TYPE_SUBTITLE:
- do_subtitle_out(os, ost, ist, &subtitle,
- pkt->pts);
- break;
- default:
- abort();
- }
- } else {
- AVFrame avframe; //FIXME/XXX remove this
- AVPacket opkt;
- int64_t ost_tb_start_time= av_rescale_q(start_time, AV_TIME_BASE_Q, ost->st->time_base);
- av_init_packet(&opkt);
+ if (!(decoded_frame = avcodec_alloc_frame()))
+ return AVERROR(ENOMEM);
+ pkt->pts = *pkt_pts;
+ pkt->dts = *pkt_dts;
+ *pkt_pts = AV_NOPTS_VALUE;
+
+ if(*pkt_dts != AV_NOPTS_VALUE && ist->st->codec->time_base.num != 0) {
+ int ticks= ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->st->codec->ticks_per_frame;
+ *pkt_dts += ((int64_t)AV_TIME_BASE *
+ ist->st->codec->time_base.num * ticks) /
+ ist->st->codec->time_base.den;
+ }else
+ *pkt_dts = AV_NOPTS_VALUE;
- if ((!ost->frame_number && !(pkt->flags & AV_PKT_FLAG_KEY)) && !copy_initial_nonkeyframes)
-#if !CONFIG_AVFILTER
- continue;
-#else
- goto cont;
-#endif
+ ret = avcodec_decode_video2(ist->st->codec,
+ decoded_frame, got_output, pkt);
+ if (ret < 0)
+ goto fail;
- /* no reencoding needed : output the packet directly */
- /* force the input stream PTS */
+ quality = same_quant ? decoded_frame->quality : 0;
+ if (!*got_output) {
+ /* no picture yet */
+ av_freep(&decoded_frame);
- return 0;
++ return ret;
+ }
- avcodec_get_frame_defaults(&avframe);
- ost->st->codec->coded_frame= &avframe;
- avframe.key_frame = pkt->flags & AV_PKT_FLAG_KEY;
+ if(decoded_frame->best_effort_timestamp != AV_NOPTS_VALUE)
+ ist->next_pts = ist->pts = decoded_frame->best_effort_timestamp;
- if(ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
- audio_size += data_size;
- else if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
- video_size += data_size;
- ost->sync_opts++;
- }
+ if (ist->st->codec->time_base.num != 0) {
+ int ticks = ist->st->parser ? ist->st->parser->repeat_pict + 1 :
+ ist->st->codec->ticks_per_frame;
+ ist->next_pts += ((int64_t)AV_TIME_BASE *
+ ist->st->codec->time_base.num * ticks) /
+ ist->st->codec->time_base.den;
+ }
+ pkt->size = 0;
- opkt.stream_index= ost->index;
- if(pkt->pts != AV_NOPTS_VALUE)
- opkt.pts= av_rescale_q(pkt->pts, ist->st->time_base, ost->st->time_base) - ost_tb_start_time;
- else
- opkt.pts= AV_NOPTS_VALUE;
-
- if (pkt->dts == AV_NOPTS_VALUE)
- opkt.dts = av_rescale_q(ist->pts, AV_TIME_BASE_Q, ost->st->time_base);
- else
- opkt.dts = av_rescale_q(pkt->dts, ist->st->time_base, ost->st->time_base);
- opkt.dts -= ost_tb_start_time;
-
- opkt.duration = av_rescale_q(pkt->duration, ist->st->time_base, ost->st->time_base);
- opkt.flags= pkt->flags;
-
- //FIXME remove the following 2 lines they shall be replaced by the bitstream filters
- if( ost->st->codec->codec_id != CODEC_ID_H264
- && ost->st->codec->codec_id != CODEC_ID_MPEG1VIDEO
- && ost->st->codec->codec_id != CODEC_ID_MPEG2VIDEO
- ) {
- if(av_parser_change(ist->st->parser, ost->st->codec, &opkt.data, &opkt.size, data_buf, data_size, pkt->flags & AV_PKT_FLAG_KEY))
- opkt.destruct= av_destruct_packet;
- } else {
- opkt.data = data_buf;
- opkt.size = data_size;
- }
+ pre_process_video_frame(ist, (AVPicture *)decoded_frame, &buffer_to_free);
- write_frame(os, &opkt, ost->st->codec, ost->bitstream_filters);
- ost->st->codec->frame_number++;
- ost->frame_number++;
- av_free_packet(&opkt);
- }
#if CONFIG_AVFILTER
- cont:
- frame_available = (ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) &&
- ost->output_video_filter && avfilter_poll_frame(ost->output_video_filter->inputs[0]);
- if (ost->picref)
- avfilter_unref_buffer(ost->picref);
- }
-#endif
- }
- }
-
- av_free(buffer_to_free);
- /* XXX: allocate the subtitles in the codec ? */
- if (subtitle_to_free) {
- avsubtitle_free(subtitle_to_free);
- subtitle_to_free = NULL;
+ for(i=0;i<nb_output_streams;i++) {
+ OutputStream *ost = ost = &output_streams[i];
+ if(check_output_constraints(ist, ost)){
+ if (!decoded_frame->sample_aspect_ratio.num)
+ decoded_frame->sample_aspect_ratio = ist->st->sample_aspect_ratio;
+ decoded_frame->pts = ist->pts;
+
+ av_vsrc_buffer_add_frame(ost->input_video_filter, decoded_frame, AV_VSRC_BUF_FLAG_OVERWRITE);
}
}
- discard_packet:
- if (pkt == NULL) {
- /* EOF handling */
+#endif
- for(i=0;i<nb_ostreams;i++) {
- ost = ost_table[i];
- if (ost->source_index == ist_index) {
- AVCodecContext *enc= ost->st->codec;
- os = output_files[ost->file_index];
-
- if(ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <=1)
- continue;
- if(ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE))
- continue;
-
- if (ost->encoding_needed) {
- for(;;) {
- AVPacket pkt;
- int fifo_bytes;
- av_init_packet(&pkt);
- pkt.stream_index= ost->index;
-
- switch(ost->st->codec->codec_type) {
- case AVMEDIA_TYPE_AUDIO:
- fifo_bytes = av_fifo_size(ost->fifo);
- ret = 0;
- /* encode any samples remaining in fifo */
- if (fifo_bytes > 0) {
- int osize = av_get_bytes_per_sample(enc->sample_fmt);
- int fs_tmp = enc->frame_size;
-
- av_fifo_generic_read(ost->fifo, audio_buf, fifo_bytes, NULL);
- if (enc->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) {
- enc->frame_size = fifo_bytes / (osize * enc->channels);
- } else { /* pad */
- int frame_bytes = enc->frame_size*osize*enc->channels;
- if (allocated_audio_buf_size < frame_bytes)
- exit_program(1);
- generate_silence(audio_buf+fifo_bytes, enc->sample_fmt, frame_bytes - fifo_bytes);
- }
-
- ret = avcodec_encode_audio(enc, bit_buffer, bit_buffer_size, (short *)audio_buf);
- pkt.duration = av_rescale((int64_t)enc->frame_size*ost->st->time_base.den,
- ost->st->time_base.num, enc->sample_rate);
- enc->frame_size = fs_tmp;
- }
- if(ret <= 0) {
- ret = avcodec_encode_audio(enc, bit_buffer, bit_buffer_size, NULL);
- }
- if (ret < 0) {
- fprintf(stderr, "Audio encoding failed\n");
- exit_program(1);
- }
- audio_size += ret;
- pkt.flags |= AV_PKT_FLAG_KEY;
- break;
- case AVMEDIA_TYPE_VIDEO:
- ret = avcodec_encode_video(enc, bit_buffer, bit_buffer_size, NULL);
- if (ret < 0) {
- fprintf(stderr, "Video encoding failed\n");
- exit_program(1);
- }
- video_size += ret;
- if(enc->coded_frame && enc->coded_frame->key_frame)
- pkt.flags |= AV_PKT_FLAG_KEY;
- if (ost->logfile && enc->stats_out) {
- fprintf(ost->logfile, "%s", enc->stats_out);
- }
- break;
- default:
- ret=-1;
- }
+ rate_emu_sleep(ist);
- if(ret<=0)
- break;
- pkt.data= bit_buffer;
- pkt.size= ret;
- if(enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
- pkt.pts= av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
- write_frame(os, &pkt, ost->st->codec, ost->bitstream_filters);
- }
+ for (i = 0; i < nb_output_streams; i++) {
+ OutputStream *ost = &output_streams[i];
+ int frame_size;
+
+ if (!check_output_constraints(ist, ost) || !ost->encoding_needed)
+ continue;
+
+#if CONFIG_AVFILTER
+ if (ost->input_video_filter) {
+ frame_available = avfilter_poll_frame(ost->output_video_filter->inputs[0]);
+ }
+ while (frame_available) {
+ if (ost->output_video_filter) {
+ AVRational ist_pts_tb = ost->output_video_filter->inputs[0]->time_base;
+ if (av_buffersink_get_buffer_ref(ost->output_video_filter, &ost->picref, 0) < 0)
+ goto cont;
+ if (!filtered_frame && !(filtered_frame = avcodec_alloc_frame())) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ *filtered_frame= *decoded_frame; //for me_threshold
+ if (ost->picref) {
+ avfilter_fill_frame_from_video_buffer_ref(filtered_frame, ost->picref);
+ ist->pts = av_rescale_q(ost->picref->pts, ist_pts_tb, AV_TIME_BASE_Q);
}
}
+ if (ost->picref->video && !ost->frame_aspect_ratio)
+ ost->st->codec->sample_aspect_ratio = ost->picref->video->sample_aspect_ratio;
+#else
+ filtered_frame = decoded_frame;
+#endif
+
+ do_video_out(output_files[ost->file_index].ctx, ost, ist, filtered_frame, &frame_size,
+ same_quant ? quality : ost->st->codec->global_quality);
+ if (vstats_filename && frame_size)
+ do_video_stats(output_files[ost->file_index].ctx, ost, frame_size);
+#if CONFIG_AVFILTER
+ cont:
+ frame_available = ost->output_video_filter && avfilter_poll_frame(ost->output_video_filter->inputs[0]);
+ avfilter_unref_buffer(ost->picref);
}
+ av_freep(&filtered_frame);
+#endif
}
- return 0;
+fail:
+ av_free(buffer_to_free);
+ av_freep(&decoded_frame);
+ return ret;
}
-static void print_sdp(AVFormatContext **avc, int n)
+static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output)
{
- char sdp[2048];
+ AVSubtitle subtitle;
+ int i, ret = avcodec_decode_subtitle2(ist->st->codec,
+ &subtitle, got_output, pkt);
+ if (ret < 0)
+ return ret;
+ if (!*got_output)
- return 0;
-
- pkt->size = 0;
++ return ret;
- av_sdp_create(avc, n, sdp, sizeof(sdp));
- printf("SDP:\n%s\n", sdp);
- fflush(stdout);
+ rate_emu_sleep(ist);
+
+ for (i = 0; i < nb_output_streams; i++) {
+ OutputStream *ost = &output_streams[i];
+
+ if (!check_output_constraints(ist, ost) || !ost->encoding_needed)
+ continue;
+
+ do_subtitle_out(output_files[ost->file_index].ctx, ost, ist, &subtitle, pkt->pts);
+ }
+
+ avsubtitle_free(&subtitle);
- return 0;
++ return ret;
}
-static int copy_chapters(int infile, int outfile)
+/* pkt = NULL means EOF (needed to flush decoder buffers) */
- static int output_packet(InputStream *ist, int ist_index,
++static int output_packet(InputStream *ist,
+ OutputStream *ost_table, int nb_ostreams,
+ const AVPacket *pkt)
{
- OutputStream *ost;
- AVFormatContext *is = input_files[infile].ctx;
- AVFormatContext *os = output_files[outfile];
- int i;
+ int ret = 0, i;
+ int got_output;
+ int64_t pkt_dts = AV_NOPTS_VALUE;
+ int64_t pkt_pts = AV_NOPTS_VALUE;
- for (i = 0; i < is->nb_chapters; i++) {
- AVChapter *in_ch = is->chapters[i], *out_ch;
- int64_t ts_off = av_rescale_q(start_time - input_files[infile].ts_offset,
- AV_TIME_BASE_Q, in_ch->time_base);
- int64_t rt = (recording_time == INT64_MAX) ? INT64_MAX :
- av_rescale_q(recording_time, AV_TIME_BASE_Q, in_ch->time_base);
+ AVPacket avpkt;
- if(ist->next_pts == AV_NOPTS_VALUE)
- ist->next_pts= ist->pts;
++ if (ist->next_pts == AV_NOPTS_VALUE)
++ ist->next_pts = ist->pts;
- if (in_ch->end < ts_off)
- continue;
- if (rt != INT64_MAX && in_ch->start > rt + ts_off)
+ if (pkt == NULL) {
+ /* EOF handling */
+ av_init_packet(&avpkt);
+ avpkt.data = NULL;
+ avpkt.size = 0;
+ goto handle_eof;
+ } else {
+ avpkt = *pkt;
+ }
+
+ if(pkt->dts != AV_NOPTS_VALUE){
+ if(ist->st->codec->codec_type != AVMEDIA_TYPE_VIDEO || !ist->decoding_needed)
+ ist->next_pts = ist->pts = av_rescale_q(pkt->dts, ist->st->time_base, AV_TIME_BASE_Q);
+ pkt_dts = av_rescale_q(pkt->dts, ist->st->time_base, AV_TIME_BASE_Q);
+ }
+ if(pkt->pts != AV_NOPTS_VALUE)
+ pkt_pts = av_rescale_q(pkt->pts, ist->st->time_base, AV_TIME_BASE_Q);
+
+ //while we have more to decode or while the decoder did output something on EOF
+ while (ist->decoding_needed && (avpkt.size > 0 || (!pkt && got_output))) {
+ handle_eof:
- ist->pts= ist->next_pts;
+
- if(avpkt.size && avpkt.size != pkt->size)
++ ist->pts = ist->next_pts;
++
++ if (avpkt.size && avpkt.size != pkt->size) {
+ av_log(NULL, ist->showed_multi_packet_warning ? AV_LOG_VERBOSE : AV_LOG_WARNING,
+ "Multiple frames in a packet from stream %d\n", pkt->stream_index);
- ist->showed_multi_packet_warning=1;
++ ist->showed_multi_packet_warning = 1;
++ }
+
+ switch(ist->st->codec->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ ret = transcode_audio (ist, &avpkt, &got_output);
break;
+ case AVMEDIA_TYPE_VIDEO:
+ ret = transcode_video (ist, &avpkt, &got_output, &pkt_pts, &pkt_dts);
+ break;
+ case AVMEDIA_TYPE_SUBTITLE:
+ ret = transcode_subtitles(ist, &avpkt, &got_output);
+ break;
+ default:
+ return -1;
+ }
- out_ch = av_mallocz(sizeof(AVChapter));
- if (!out_ch)
- return AVERROR(ENOMEM);
+ if (ret < 0)
+ return ret;
++ // touch data and size only if not EOF
++ if (pkt) {
++ if(ist->st->codec->codec_type != AVMEDIA_TYPE_AUDIO)
++ ret = avpkt.size;
++ avpkt.data += ret;
++ avpkt.size -= ret;
++ }
+ if (!got_output) {
- if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
- continue;
- goto discard_packet;
++ continue;
+ }
+ }
- discard_packet:
- out_ch->id = in_ch->id;
- out_ch->time_base = in_ch->time_base;
- out_ch->start = FFMAX(0, in_ch->start - ts_off);
- out_ch->end = FFMIN(rt, in_ch->end - ts_off);
+ /* handle stream copy */
+ if (!ist->decoding_needed) {
+ rate_emu_sleep(ist);
+ switch (ist->st->codec->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ ist->next_pts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) /
+ ist->st->codec->sample_rate;
+ break;
+ case AVMEDIA_TYPE_VIDEO:
+ if (ist->st->codec->time_base.num != 0) {
+ int ticks = ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->st->codec->ticks_per_frame;
+ ist->next_pts += ((int64_t)AV_TIME_BASE *
+ ist->st->codec->time_base.num * ticks) /
+ ist->st->codec->time_base.den;
+ }
+ break;
+ }
+ }
+ for (i = 0; pkt && i < nb_ostreams; i++) {
- ost = &ost_table[i];
++ OutputStream *ost = &ost_table[i];
- if (metadata_chapters_autocopy)
- av_dict_copy(&out_ch->metadata, in_ch->metadata, 0);
+ if (!check_output_constraints(ist, ost) || ost->encoding_needed)
+ continue;
- os->nb_chapters++;
- os->chapters = av_realloc(os->chapters, sizeof(AVChapter)*os->nb_chapters);
- if (!os->chapters)
- return AVERROR(ENOMEM);
- os->chapters[os->nb_chapters - 1] = out_ch;
+ do_streamcopy(ist, ost, pkt);
}
+
return 0;
}
}
}
- /* finish if recording time exhausted */
- if (recording_time != INT64_MAX &&
- av_compare_ts(pkt.pts, ist->st->time_base, recording_time + start_time, (AVRational){1, 1000000}) >= 0) {
- ist->is_past_recording_time = 1;
- goto discard_packet;
- }
-
//fprintf(stderr,"read #%d.%d size=%d\n", ist->file_index, ist->st->index, pkt.size);
- if (output_packet(ist, ist_index, output_streams, nb_output_streams, &pkt) < 0) {
- if (output_packet(ist, ist_index, ost_table, nb_ostreams, &pkt) < 0) {
++ if (output_packet(ist, output_streams, nb_output_streams, &pkt) < 0) {
- if (verbose >= 0)
- fprintf(stderr, "Error while decoding stream #%d.%d\n",
- ist->file_index, ist->st->index);
+ av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d\n",
+ ist->file_index, ist->st->index);
if (exit_on_error)
exit_program(1);
av_free_packet(&pkt);
for (i = 0; i < nb_input_streams; i++) {
ist = &input_streams[i];
if (ist->decoding_needed) {
- output_packet(ist, i, output_streams, nb_output_streams, NULL);
- output_packet(ist, i, ost_table, nb_ostreams, NULL);
++ output_packet(ist, output_streams, nb_output_streams, NULL);
}
}
+ flush_encoders(output_streams, nb_output_streams);
term_exit();
{
uint32_t * palette = (uint32_t *)frame->data[1];
int a;
+
+ if (buf_size < 256*3)
+ return AVERROR_INVALIDDATA;
+
for(a = 0; a < 256; a++){
- palette[a] = AV_RB24(&palette_buffer[a * 3]) * 4;
+ palette[a] = 0xFF << 24 | AV_RB24(&palette_buffer[a * 3]) * 4;
+ palette[a] |= palette[a] >> 6 & 0x30303;
}
frame->palette_has_changed = 1;
+ return 256*3;
}
static int bethsoftvid_decode_frame(AVCodecContext *avctx,
return ret;
}
- static const AVOption options[]={
- {"dpi", "set the image resolution (in dpi)", offsetof(TiffEncoderContext, dpi), AV_OPT_TYPE_INT, {.dbl = 72}, 1, 0x10000, AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_ENCODING_PARAM},
- {NULL}
+ #define OFFSET(x) offsetof(TiffEncoderContext, x)
+ #define VE AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
+ static const AVOption options[] = {
++ {"dpi", "set the image resolution (in dpi)", OFFSET(dpi), AV_OPT_TYPE_INT, {.dbl = 72}, 1, 0x10000, AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_ENCODING_PARAM},
+ { "compression_algo", NULL, OFFSET(compr), AV_OPT_TYPE_INT, {TIFF_PACKBITS}, TIFF_RAW, TIFF_DEFLATE, VE, "compression_algo" },
+ { "packbits", NULL, 0, AV_OPT_TYPE_CONST, {TIFF_PACKBITS}, 0, 0, VE, "compression_algo" },
+ { "raw", NULL, 0, AV_OPT_TYPE_CONST, {TIFF_RAW}, 0, 0, VE, "compression_algo" },
+ { "lzw", NULL, 0, AV_OPT_TYPE_CONST, {TIFF_LZW}, 0, 0, VE, "compression_algo" },
+ #if CONFIG_ZLIB
+ { "deflate", NULL, 0, AV_OPT_TYPE_CONST, {TIFF_DEFLATE}, 0, 0, VE, "compression_algo" },
+ #endif
+ { NULL },
+ };
+
+ static const AVClass tiffenc_class = {
+ .class_name = "TIFF encoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
};
- static const AVClass class = { "tiff", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
AVCodec ff_tiff_encoder = {
.name = "tiff",
PIX_FMT_MONOBLACK, PIX_FMT_MONOWHITE,
PIX_FMT_YUV420P, PIX_FMT_YUV422P,
PIX_FMT_YUV444P, PIX_FMT_YUV410P,
- PIX_FMT_YUV411P,
+ PIX_FMT_YUV411P, PIX_FMT_RGB48LE,
PIX_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("TIFF image"),
- .priv_class= &class,
+ .priv_class = &tiffenc_class,
};
--- /dev/null
- inlink->format, inlink->planar, 16) < 0)
+/*
+ * Copyright (c) 2010 S.N. Hemanth Meenakshisundaram <smeenaks@ucsd.edu>
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2011 Mina Nagy Zaki
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * sample format and channel layout conversion audio filter
+ * based on code in libavcodec/resample.c by Fabrice Bellard and
+ * libavcodec/audioconvert.c by Michael Niedermayer
+ */
+
+#include "libavutil/audioconvert.h"
+#include "libavutil/avstring.h"
+#include "libavcodec/audioconvert.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct {
+ enum AVSampleFormat out_sample_fmt, in_sample_fmt; ///< in/out sample formats
+ int64_t out_chlayout, in_chlayout; ///< in/out channel layout
+ int out_nb_channels, in_nb_channels; ///< number of in/output channels
+ enum AVFilterPacking out_packing_fmt, in_packing_fmt; ///< output packing format
+
+ int max_nb_samples; ///< maximum number of buffered samples
+ AVFilterBufferRef *mix_samplesref; ///< rematrixed buffer
+ AVFilterBufferRef *out_samplesref; ///< output buffer after required conversions
+
+ uint8_t *in_mix[8], *out_mix[8]; ///< input/output for rematrixing functions
+ uint8_t *packed_data[8]; ///< pointers for packing conversion
+ int out_strides[8], in_strides[8]; ///< input/output strides for av_audio_convert
+ uint8_t **in_conv, **out_conv; ///< input/output for av_audio_convert
+
+ AVAudioConvert *audioconvert_ctx; ///< context for conversion to output sample format
+
+ void (*convert_chlayout)(); ///< function to do the requested rematrixing
+} AConvertContext;
+
+#define REMATRIX_FUNC_SIG(NAME) static void REMATRIX_FUNC_NAME(NAME) \
+ (FMT_TYPE *outp[], FMT_TYPE *inp[], int nb_samples, AConvertContext *aconvert)
+
+#define FMT_TYPE uint8_t
+#define REMATRIX_FUNC_NAME(NAME) NAME ## _u8
+#include "af_aconvert_rematrix.c"
+
+#define FMT_TYPE int16_t
+#define REMATRIX_FUNC_NAME(NAME) NAME ## _s16
+#include "af_aconvert_rematrix.c"
+
+#define FMT_TYPE int32_t
+#define REMATRIX_FUNC_NAME(NAME) NAME ## _s32
+#include "af_aconvert_rematrix.c"
+
+#define FLOATING
+
+#define FMT_TYPE float
+#define REMATRIX_FUNC_NAME(NAME) NAME ## _flt
+#include "af_aconvert_rematrix.c"
+
+#define FMT_TYPE double
+#define REMATRIX_FUNC_NAME(NAME) NAME ## _dbl
+#include "af_aconvert_rematrix.c"
+
+#define FMT_TYPE uint8_t
+#define REMATRIX_FUNC_NAME(NAME) NAME
+REMATRIX_FUNC_SIG(stereo_remix_planar)
+{
+ int size = av_get_bytes_per_sample(aconvert->in_sample_fmt) * nb_samples;
+
+ memcpy(outp[0], inp[0], size);
+ memcpy(outp[1], inp[aconvert->in_nb_channels == 1 ? 0 : 1], size);
+}
+
+#define REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC, PACKING) \
+ {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_U8, FUNC##_u8}, \
+ {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S16, FUNC##_s16}, \
+ {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S32, FUNC##_s32}, \
+ {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_FLT, FUNC##_flt}, \
+ {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_DBL, FUNC##_dbl},
+
+#define REGISTER_FUNC(INCHLAYOUT, OUTCHLAYOUT, FUNC) \
+ REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_packed, AVFILTER_PACKED) \
+ REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_planar, AVFILTER_PLANAR)
+
+static const struct RematrixFunctionInfo {
+ int64_t in_chlayout, out_chlayout;
+ int planar, sfmt;
+ void (*func)();
+} rematrix_funcs[] = {
+ REGISTER_FUNC (AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_5POINT1, stereo_to_surround_5p1)
+ REGISTER_FUNC (AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_STEREO, surround_5p1_to_stereo)
+ REGISTER_FUNC_PACKING(AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_MONO, stereo_to_mono_packed, AVFILTER_PACKED)
+ REGISTER_FUNC_PACKING(AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, mono_to_stereo_packed, AVFILTER_PACKED)
+ REGISTER_FUNC (0, AV_CH_LAYOUT_MONO, mono_downmix)
+ REGISTER_FUNC_PACKING(0, AV_CH_LAYOUT_STEREO, stereo_downmix_packed, AVFILTER_PACKED)
+
+ // This function works for all sample formats
+ {0, AV_CH_LAYOUT_STEREO, AVFILTER_PLANAR, -1, stereo_remix_planar}
+};
+
+static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
+{
+ AConvertContext *aconvert = ctx->priv;
+ char *arg, *ptr = NULL;
+ int ret = 0;
+ char *args = av_strdup(args0);
+
+ aconvert->out_sample_fmt = AV_SAMPLE_FMT_NONE;
+ aconvert->out_chlayout = 0;
+ aconvert->out_packing_fmt = -1;
+
+ if ((arg = av_strtok(args, ":", &ptr)) && strcmp(arg, "auto")) {
+ if ((ret = ff_parse_sample_format(&aconvert->out_sample_fmt, arg, ctx)) < 0)
+ goto end;
+ }
+ if ((arg = av_strtok(NULL, ":", &ptr)) && strcmp(arg, "auto")) {
+ if ((ret = ff_parse_channel_layout(&aconvert->out_chlayout, arg, ctx)) < 0)
+ goto end;
+ }
+ if ((arg = av_strtok(NULL, ":", &ptr)) && strcmp(arg, "auto")) {
+ if ((ret = ff_parse_packing_format((int *)&aconvert->out_packing_fmt, arg, ctx)) < 0)
+ goto end;
+ }
+
+end:
+ av_freep(&args);
+ return ret;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AConvertContext *aconvert = ctx->priv;
+ avfilter_unref_buffer(aconvert->mix_samplesref);
+ avfilter_unref_buffer(aconvert->out_samplesref);
+ if (aconvert->audioconvert_ctx)
+ av_audio_convert_free(aconvert->audioconvert_ctx);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AConvertContext *aconvert = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+
+ avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
+ &inlink->out_formats);
+ if (aconvert->out_sample_fmt != AV_SAMPLE_FMT_NONE) {
+ formats = NULL;
+ avfilter_add_format(&formats, aconvert->out_sample_fmt);
+ avfilter_formats_ref(formats, &outlink->in_formats);
+ } else
+ avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
+ &outlink->in_formats);
+
+ avfilter_formats_ref(avfilter_make_all_channel_layouts(),
+ &inlink->out_chlayouts);
+ if (aconvert->out_chlayout != 0) {
+ formats = NULL;
+ avfilter_add_format(&formats, aconvert->out_chlayout);
+ avfilter_formats_ref(formats, &outlink->in_chlayouts);
+ } else
+ avfilter_formats_ref(avfilter_make_all_channel_layouts(),
+ &outlink->in_chlayouts);
+
+ avfilter_formats_ref(avfilter_make_all_packing_formats(),
+ &inlink->out_packing);
+ if (aconvert->out_packing_fmt != -1) {
+ formats = NULL;
+ avfilter_add_format(&formats, aconvert->out_packing_fmt);
+ avfilter_formats_ref(formats, &outlink->in_packing);
+ } else
+ avfilter_formats_ref(avfilter_make_all_packing_formats(),
+ &outlink->in_packing);
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterLink *inlink = outlink->src->inputs[0];
+ AConvertContext *aconvert = outlink->src->priv;
+ char buf1[64], buf2[64];
+
+ aconvert->in_sample_fmt = inlink->format;
+ aconvert->in_packing_fmt = inlink->planar;
+ if (aconvert->out_packing_fmt == -1)
+ aconvert->out_packing_fmt = outlink->planar;
+ aconvert->in_chlayout = inlink->channel_layout;
+ aconvert->in_nb_channels =
+ av_get_channel_layout_nb_channels(inlink->channel_layout);
+
+ /* if not specified in args, use the format and layout of the output */
+ if (aconvert->out_sample_fmt == AV_SAMPLE_FMT_NONE)
+ aconvert->out_sample_fmt = outlink->format;
+ if (aconvert->out_chlayout == 0)
+ aconvert->out_chlayout = outlink->channel_layout;
+ aconvert->out_nb_channels =
+ av_get_channel_layout_nb_channels(outlink->channel_layout);
+
+ av_get_channel_layout_string(buf1, sizeof(buf1),
+ -1, inlink ->channel_layout);
+ av_get_channel_layout_string(buf2, sizeof(buf2),
+ -1, outlink->channel_layout);
+ av_log(outlink->src, AV_LOG_INFO,
+ "fmt:%s cl:%s planar:%i -> fmt:%s cl:%s planar:%i\n",
+ av_get_sample_fmt_name(inlink ->format), buf1, inlink ->planar,
+ av_get_sample_fmt_name(outlink->format), buf2, outlink->planar);
+
+ /* compute which channel layout conversion to use */
+ if (inlink->channel_layout != outlink->channel_layout) {
+ int i;
+ for (i = 0; i < sizeof(rematrix_funcs); i++) {
+ const struct RematrixFunctionInfo *f = &rematrix_funcs[i];
+ if ((f->in_chlayout == 0 || f->in_chlayout == inlink ->channel_layout) &&
+ (f->out_chlayout == 0 || f->out_chlayout == outlink->channel_layout) &&
+ (f->planar == -1 || f->planar == inlink->planar) &&
+ (f->sfmt == -1 || f->sfmt == inlink->format)
+ ) {
+ aconvert->convert_chlayout = f->func;
+ break;
+ }
+ }
+ if (!aconvert->convert_chlayout) {
+ av_log(outlink->src, AV_LOG_ERROR,
+ "Unsupported channel layout conversion '%s -> %s' requested!\n",
+ buf1, buf2);
+ return AVERROR(EINVAL);
+ }
+ }
+
+ return 0;
+}
+
+static int init_buffers(AVFilterLink *inlink, int nb_samples)
+{
+ AConvertContext *aconvert = inlink->dst->priv;
+ AVFilterLink * const outlink = inlink->dst->outputs[0];
+ int i, packed_stride = 0;
+ const unsigned
+ packing_conv = inlink->planar != outlink->planar &&
+ aconvert->out_nb_channels != 1,
+ format_conv = inlink->format != outlink->format;
+ int nb_channels = aconvert->out_nb_channels;
+
+ uninit(inlink->dst);
+ aconvert->max_nb_samples = nb_samples;
+
+ if (aconvert->convert_chlayout) {
+ /* allocate buffer for storing intermediary mixing samplesref */
+ uint8_t *data[8];
+ int linesize[8];
+ int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
+
+ if (av_samples_alloc(data, linesize, nb_channels, nb_samples,
++ inlink->format, 16) < 0)
+ goto fail_no_mem;
+ aconvert->mix_samplesref =
+ avfilter_get_audio_buffer_ref_from_arrays(data, linesize, AV_PERM_WRITE,
+ nb_samples, inlink->format,
+ outlink->channel_layout,
+ inlink->planar);
+ if (!aconvert->mix_samplesref)
+ goto fail_no_mem;
+ }
+
+ // if there's a format/packing conversion we need an audio_convert context
+ if (format_conv || packing_conv) {
+ aconvert->out_samplesref =
+ avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+ if (!aconvert->out_samplesref)
+ goto fail_no_mem;
+
+ aconvert->in_strides [0] = av_get_bytes_per_sample(inlink ->format);
+ aconvert->out_strides[0] = av_get_bytes_per_sample(outlink->format);
+
+ aconvert->out_conv = aconvert->out_samplesref->data;
+ if (aconvert->mix_samplesref)
+ aconvert->in_conv = aconvert->mix_samplesref->data;
+
+ if (packing_conv) {
+ // packed -> planar
+ if (outlink->planar == AVFILTER_PLANAR) {
+ if (aconvert->mix_samplesref)
+ aconvert->packed_data[0] = aconvert->mix_samplesref->data[0];
+ aconvert->in_conv = aconvert->packed_data;
+ packed_stride = aconvert->in_strides[0];
+ aconvert->in_strides[0] *= nb_channels;
+ // planar -> packed
+ } else {
+ aconvert->packed_data[0] = aconvert->out_samplesref->data[0];
+ aconvert->out_conv = aconvert->packed_data;
+ packed_stride = aconvert->out_strides[0];
+ aconvert->out_strides[0] *= nb_channels;
+ }
+ } else if (outlink->planar == AVFILTER_PACKED) {
+ /* If there's no packing conversion, and the stream is packed
+ * then we treat the entire stream as one big channel
+ */
+ nb_channels = 1;
+ }
+
+ for (i = 1; i < nb_channels; i++) {
+ aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride;
+ aconvert->in_strides[i] = aconvert->in_strides[0];
+ aconvert->out_strides[i] = aconvert->out_strides[0];
+ }
+
+ aconvert->audioconvert_ctx =
+ av_audio_convert_alloc(outlink->format, nb_channels,
+ inlink->format, nb_channels, NULL, 0);
+ if (!aconvert->audioconvert_ctx)
+ goto fail_no_mem;
+ }
+
+ return 0;
+
+fail_no_mem:
+ av_log(inlink->dst, AV_LOG_ERROR, "Could not allocate memory.\n");
+ return AVERROR(ENOMEM);
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+{
+ AConvertContext *aconvert = inlink->dst->priv;
+ AVFilterBufferRef *curbuf = insamplesref;
+ AVFilterLink * const outlink = inlink->dst->outputs[0];
+ int chan_mult;
+
+ /* in/reinint the internal buffers if this is the first buffer
+ * provided or it is needed to use a bigger one */
+ if (!aconvert->max_nb_samples ||
+ (curbuf->audio->nb_samples > aconvert->max_nb_samples))
+ if (init_buffers(inlink, curbuf->audio->nb_samples) < 0) {
+ av_log(inlink->dst, AV_LOG_ERROR, "Could not initialize buffers.\n");
+ return;
+ }
+
+ /* if channel mixing is required */
+ if (aconvert->mix_samplesref) {
+ memcpy(aconvert->in_mix, curbuf->data, sizeof(aconvert->in_mix));
+ memcpy(aconvert->out_mix, aconvert->mix_samplesref->data, sizeof(aconvert->out_mix));
+ aconvert->convert_chlayout(aconvert->out_mix,
+ aconvert->in_mix,
+ curbuf->audio->nb_samples,
+ aconvert);
+ curbuf = aconvert->mix_samplesref;
+ }
+
+ if (aconvert->audioconvert_ctx) {
+ if (!aconvert->mix_samplesref) {
+ if (aconvert->in_conv == aconvert->packed_data) {
+ int i, packed_stride = av_get_bytes_per_sample(inlink->format);
+ aconvert->packed_data[0] = curbuf->data[0];
+ for (i = 1; i < aconvert->out_nb_channels; i++)
+ aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride;
+ } else {
+ aconvert->in_conv = curbuf->data;
+ }
+ }
+
+ chan_mult = inlink->planar == outlink->planar && inlink->planar == 0 ?
+ aconvert->out_nb_channels : 1;
+
+ av_audio_convert(aconvert->audioconvert_ctx,
+ (void * const *) aconvert->out_conv,
+ aconvert->out_strides,
+ (const void * const *) aconvert->in_conv,
+ aconvert->in_strides,
+ curbuf->audio->nb_samples * chan_mult);
+
+ curbuf = aconvert->out_samplesref;
+ }
+
+ avfilter_copy_buffer_ref_props(curbuf, insamplesref);
+ curbuf->audio->channel_layout = outlink->channel_layout;
+ curbuf->audio->planar = outlink->planar;
+
+ avfilter_filter_samples(inlink->dst->outputs[0],
+ avfilter_ref_buffer(curbuf, ~0));
+ avfilter_unref_buffer(insamplesref);
+}
+
+AVFilter avfilter_af_aconvert = {
+ .name = "aconvert",
+ .description = NULL_IF_CONFIG_SMALL("Convert the input audio to sample_fmt:channel_layout:packed_fmt."),
+ .priv_size = sizeof(AConvertContext),
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+
+ .inputs = (const AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ, },
+ { .name = NULL}},
+ .outputs = (const AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output, },
+ { .name = NULL}},
+};
--- /dev/null
- sample_fmt, planar, 16);
+/*
+ * Copyright (c) 2010 S.N. Hemanth Meenakshisundaram
+ * Copyright (c) 2011 Mina Nagy Zaki
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * memory buffer source for audio
+ */
+
+#include "libavutil/audioconvert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/fifo.h"
+#include "asrc_abuffer.h"
+#include "internal.h"
+
+typedef struct {
+ // Audio format of incoming buffers
+ int sample_rate;
+ unsigned int sample_format;
+ int64_t channel_layout;
+ int packing_format;
+
+ // FIFO buffer of audio buffer ref pointers
+ AVFifoBuffer *fifo;
+
+ // Normalization filters
+ AVFilterContext *aconvert;
+ AVFilterContext *aresample;
+} ABufferSourceContext;
+
+#define FIFO_SIZE 8
+
+static void buf_free(AVFilterBuffer *ptr)
+{
+ av_free(ptr);
+ return;
+}
+
+static void set_link_source(AVFilterContext *src, AVFilterLink *link)
+{
+ link->src = src;
+ link->srcpad = &(src->output_pads[0]);
+ src->outputs[0] = link;
+}
+
+static int reconfigure_filter(ABufferSourceContext *abuffer, AVFilterContext *filt_ctx)
+{
+ int ret;
+ AVFilterLink * const inlink = filt_ctx->inputs[0];
+ AVFilterLink * const outlink = filt_ctx->outputs[0];
+
+ inlink->format = abuffer->sample_format;
+ inlink->channel_layout = abuffer->channel_layout;
+ inlink->planar = abuffer->packing_format;
+ inlink->sample_rate = abuffer->sample_rate;
+
+ filt_ctx->filter->uninit(filt_ctx);
+ memset(filt_ctx->priv, 0, filt_ctx->filter->priv_size);
+ if ((ret = filt_ctx->filter->init(filt_ctx, NULL , NULL)) < 0)
+ return ret;
+ if ((ret = inlink->srcpad->config_props(inlink)) < 0)
+ return ret;
+ return outlink->srcpad->config_props(outlink);
+}
+
+static int insert_filter(ABufferSourceContext *abuffer,
+ AVFilterLink *link, AVFilterContext **filt_ctx,
+ const char *filt_name)
+{
+ int ret;
+
+ if ((ret = avfilter_open(filt_ctx, avfilter_get_by_name(filt_name), NULL)) < 0)
+ return ret;
+
+ link->src->outputs[0] = NULL;
+ if ((ret = avfilter_link(link->src, 0, *filt_ctx, 0)) < 0) {
+ link->src->outputs[0] = link;
+ return ret;
+ }
+
+ set_link_source(*filt_ctx, link);
+
+ if ((ret = reconfigure_filter(abuffer, *filt_ctx)) < 0) {
+ avfilter_free(*filt_ctx);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void remove_filter(AVFilterContext **filt_ctx)
+{
+ AVFilterLink *outlink = (*filt_ctx)->outputs[0];
+ AVFilterContext *src = (*filt_ctx)->inputs[0]->src;
+
+ (*filt_ctx)->outputs[0] = NULL;
+ avfilter_free(*filt_ctx);
+ *filt_ctx = NULL;
+
+ set_link_source(src, outlink);
+}
+
+static inline void log_input_change(void *ctx, AVFilterLink *link, AVFilterBufferRef *ref)
+{
+ char old_layout_str[16], new_layout_str[16];
+ av_get_channel_layout_string(old_layout_str, sizeof(old_layout_str),
+ -1, link->channel_layout);
+ av_get_channel_layout_string(new_layout_str, sizeof(new_layout_str),
+ -1, ref->audio->channel_layout);
+ av_log(ctx, AV_LOG_INFO,
+ "Audio input format changed: "
+ "%s:%s:%d -> %s:%s:%d, normalizing\n",
+ av_get_sample_fmt_name(link->format),
+ old_layout_str, (int)link->sample_rate,
+ av_get_sample_fmt_name(ref->format),
+ new_layout_str, ref->audio->sample_rate);
+}
+
+int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *ctx,
+ AVFilterBufferRef *samplesref,
+ int av_unused flags)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ AVFilterLink *link;
+ int ret, logged = 0;
+
+ if (av_fifo_space(abuffer->fifo) < sizeof(samplesref)) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Buffering limit reached. Please consume some available frames "
+ "before adding new ones.\n");
+ return AVERROR(EINVAL);
+ }
+
+ // Normalize input
+
+ link = ctx->outputs[0];
+ if (samplesref->audio->sample_rate != link->sample_rate) {
+
+ log_input_change(ctx, link, samplesref);
+ logged = 1;
+
+ abuffer->sample_rate = samplesref->audio->sample_rate;
+
+ if (!abuffer->aresample) {
+ ret = insert_filter(abuffer, link, &abuffer->aresample, "aresample");
+ if (ret < 0) return ret;
+ } else {
+ link = abuffer->aresample->outputs[0];
+ if (samplesref->audio->sample_rate == link->sample_rate)
+ remove_filter(&abuffer->aresample);
+ else
+ if ((ret = reconfigure_filter(abuffer, abuffer->aresample)) < 0)
+ return ret;
+ }
+ }
+
+ link = ctx->outputs[0];
+ if (samplesref->format != link->format ||
+ samplesref->audio->channel_layout != link->channel_layout ||
+ samplesref->audio->planar != link->planar) {
+
+ if (!logged) log_input_change(ctx, link, samplesref);
+
+ abuffer->sample_format = samplesref->format;
+ abuffer->channel_layout = samplesref->audio->channel_layout;
+ abuffer->packing_format = samplesref->audio->planar;
+
+ if (!abuffer->aconvert) {
+ ret = insert_filter(abuffer, link, &abuffer->aconvert, "aconvert");
+ if (ret < 0) return ret;
+ } else {
+ link = abuffer->aconvert->outputs[0];
+ if (samplesref->format == link->format &&
+ samplesref->audio->channel_layout == link->channel_layout &&
+ samplesref->audio->planar == link->planar
+ )
+ remove_filter(&abuffer->aconvert);
+ else
+ if ((ret = reconfigure_filter(abuffer, abuffer->aconvert)) < 0)
+ return ret;
+ }
+ }
+
+ if (sizeof(samplesref) != av_fifo_generic_write(abuffer->fifo, &samplesref,
+ sizeof(samplesref), NULL)) {
+ av_log(ctx, AV_LOG_ERROR, "Error while writing to FIFO\n");
+ return AVERROR(EINVAL);
+ }
+
+ return 0;
+}
+
+int av_asrc_buffer_add_samples(AVFilterContext *ctx,
+ uint8_t *data[8], int linesize[8],
+ int nb_samples, int sample_rate,
+ int sample_fmt, int64_t channel_layout, int planar,
+ int64_t pts, int av_unused flags)
+{
+ AVFilterBufferRef *samplesref;
+
+ samplesref = avfilter_get_audio_buffer_ref_from_arrays(
+ data, linesize, AV_PERM_WRITE,
+ nb_samples,
+ sample_fmt, channel_layout, planar);
+ if (!samplesref)
+ return AVERROR(ENOMEM);
+
+ samplesref->buf->free = buf_free;
+ samplesref->pts = pts;
+ samplesref->audio->sample_rate = sample_rate;
+
+ return av_asrc_buffer_add_audio_buffer_ref(ctx, samplesref, 0);
+}
+
+int av_asrc_buffer_add_buffer(AVFilterContext *ctx,
+ uint8_t *buf, int buf_size, int sample_rate,
+ int sample_fmt, int64_t channel_layout, int planar,
+ int64_t pts, int av_unused flags)
+{
+ uint8_t *data[8];
+ int linesize[8];
+ int nb_channels = av_get_channel_layout_nb_channels(channel_layout),
+ nb_samples = buf_size / nb_channels / av_get_bytes_per_sample(sample_fmt);
+
+ av_samples_fill_arrays(data, linesize,
+ buf, nb_channels, nb_samples,
++ sample_fmt, 16);
+
+ return av_asrc_buffer_add_samples(ctx,
+ data, linesize, nb_samples,
+ sample_rate,
+ sample_fmt, channel_layout, planar,
+ pts, flags);
+}
+
+static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ char *arg = NULL, *ptr, chlayout_str[16];
+ char *args = av_strdup(args0);
+ int ret;
+
+ arg = av_strtok(args, ":", &ptr);
+
+#define ADD_FORMAT(fmt_name) \
+ if (!arg) \
+ goto arg_fail; \
+ if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0) { \
+ av_freep(&args); \
+ return ret; \
+ } \
+ if (*args) \
+ arg = av_strtok(NULL, ":", &ptr)
+
+ ADD_FORMAT(sample_rate);
+ ADD_FORMAT(sample_format);
+ ADD_FORMAT(channel_layout);
+ ADD_FORMAT(packing_format);
+
+ abuffer->fifo = av_fifo_alloc(FIFO_SIZE*sizeof(AVFilterBufferRef*));
+ if (!abuffer->fifo) {
+ av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo, filter init failed.\n");
+ return AVERROR(ENOMEM);
+ }
+
+ av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str),
+ -1, abuffer->channel_layout);
+ av_log(ctx, AV_LOG_INFO, "format:%s layout:%s rate:%d\n",
+ av_get_sample_fmt_name(abuffer->sample_format), chlayout_str,
+ abuffer->sample_rate);
+ av_freep(&args);
+
+ return 0;
+
+arg_fail:
+ av_log(ctx, AV_LOG_ERROR, "Invalid arguments, must be of the form "
+ "sample_rate:sample_fmt:channel_layout:packing\n");
+ av_freep(&args);
+ return AVERROR(EINVAL);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ av_fifo_free(abuffer->fifo);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ AVFilterFormats *formats;
+
+ formats = NULL;
+ avfilter_add_format(&formats, abuffer->sample_format);
+ avfilter_set_common_sample_formats(ctx, formats);
+
+ formats = NULL;
+ avfilter_add_format(&formats, abuffer->channel_layout);
+ avfilter_set_common_channel_layouts(ctx, formats);
+
+ formats = NULL;
+ avfilter_add_format(&formats, abuffer->packing_format);
+ avfilter_set_common_packing_formats(ctx, formats);
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ ABufferSourceContext *abuffer = outlink->src->priv;
+ outlink->sample_rate = abuffer->sample_rate;
+ return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ ABufferSourceContext *abuffer = outlink->src->priv;
+ AVFilterBufferRef *samplesref;
+
+ if (!av_fifo_size(abuffer->fifo)) {
+ av_log(outlink->src, AV_LOG_ERROR,
+ "request_frame() called with no available frames!\n");
+ return AVERROR(EINVAL);
+ }
+
+ av_fifo_generic_read(abuffer->fifo, &samplesref, sizeof(samplesref), NULL);
+ avfilter_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0));
+ avfilter_unref_buffer(samplesref);
+
+ return 0;
+}
+
+static int poll_frame(AVFilterLink *outlink)
+{
+ ABufferSourceContext *abuffer = outlink->src->priv;
+ return av_fifo_size(abuffer->fifo)/sizeof(AVFilterBufferRef*);
+}
+
+AVFilter avfilter_asrc_abuffer = {
+ .name = "abuffer",
+ .description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them accessible to the filterchain."),
+ .priv_size = sizeof(ABufferSourceContext),
+ .query_formats = query_formats,
+
+ .init = init,
+ .uninit = uninit,
+
+ .inputs = (const AVFilterPad[]) {{ .name = NULL }},
+ .outputs = (const AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .poll_frame = poll_frame,
+ .config_props = config_output, },
+ { .name = NULL}},
+};
}
AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int perms,
- enum AVSampleFormat sample_fmt, int size,
- int64_t channel_layout, int planar)
+ int nb_samples)
{
- AVFilterBuffer *samples = av_mallocz(sizeof(AVFilterBuffer));
- AVFilterBufferRef *ref = NULL;
- int i, sample_size, chans_nb, bufsize, per_channel_size, step_size = 0;
- char *buf;
-
- if (!samples || !(ref = av_mallocz(sizeof(AVFilterBufferRef))))
- goto fail;
-
- ref->buf = samples;
- ref->format = sample_fmt;
-
- ref->audio = av_mallocz(sizeof(AVFilterBufferRefAudioProps));
- if (!ref->audio)
- goto fail;
-
- ref->audio->channel_layout = channel_layout;
- ref->audio->size = size;
- ref->audio->planar = planar;
-
- /* make sure the buffer gets read permission or it's useless for output */
- ref->perms = perms | AV_PERM_READ;
-
- samples->refcount = 1;
- samples->free = ff_avfilter_default_free_buffer;
-
- sample_size = av_get_bytes_per_sample(sample_fmt);
- chans_nb = av_get_channel_layout_nb_channels(channel_layout);
-
- per_channel_size = size/chans_nb;
- ref->audio->nb_samples = per_channel_size/sample_size;
-
- /* Set the number of bytes to traverse to reach next sample of a particular channel:
- * For planar, this is simply the sample size.
- * For packed, this is the number of samples * sample_size.
- */
- for (i = 0; i < chans_nb; i++)
- samples->linesize[i] = planar > 0 ? per_channel_size : sample_size;
- memset(&samples->linesize[chans_nb], 0, (8-chans_nb) * sizeof(samples->linesize[0]));
+ AVFilterBufferRef *samplesref = NULL;
+ int linesize[8];
+ uint8_t *data[8];
+ int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
/* Calculate total buffer size, round to multiple of 16 to be SIMD friendly */
- bufsize = (size + 15)&~15;
- buf = av_malloc(bufsize);
- if (!buf)
- goto fail;
-
- /* For planar, set the start point of each channel's data within the buffer
- * For packed, set the start point of the entire buffer only
- */
- samples->data[0] = buf;
- if (buf && planar) {
- for (i = 1; i < chans_nb; i++) {
- step_size += per_channel_size;
- samples->data[i] = buf + step_size;
- }
- } else {
- for (i = 1; i < chans_nb; i++)
- samples->data[i] = buf;
- }
-
- memset(&samples->data[chans_nb], 0, (8-chans_nb) * sizeof(samples->data[0]));
-
- memcpy(ref->data, samples->data, sizeof(ref->data));
- memcpy(ref->linesize, samples->linesize, sizeof(ref->linesize));
+ if (av_samples_alloc(data, linesize,
+ nb_channels, nb_samples, link->format,
- link->planar, 16) < 0)
++ 16) < 0)
+ return NULL;
- return ref;
+ samplesref =
+ avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
+ nb_samples, link->format,
+ link->channel_layout, link->planar);
+ if (!samplesref) {
+ av_free(data[0]);
+ return NULL;
+ }
-fail:
- if (ref)
- av_free(ref->audio);
- av_free(ref);
- av_free(samples);
- return NULL;
+ return samplesref;
}
void avfilter_default_start_frame(AVFilterLink *inlink, AVFilterBufferRef *picref)
{
int ret;
- ret = av_get_packet(s->pb, pkt, CDG_PACKET_SIZE);
+ while (1) {
+ ret = av_get_packet(s->pb, pkt, CDG_PACKET_SIZE);
+ if (ret < 1 || (pkt->data[0] & CDG_MASK) == CDG_COMMAND)
+ break;
+ av_free_packet(pkt);
+ }
pkt->stream_index = 0;
+ pkt->dts=pkt->pts= s->streams[0]->cur_dts;
+
+ if(ret>5 && (pkt->data[0]&0x3F) == 9 && (pkt->data[1]&0x3F)==1 && !(pkt->data[2+2+1] & 0x0F)){
+ pkt->flags = AV_PKT_FLAG_KEY;
+ }
return ret;
}
*/
#define LIBAVUTIL_VERSION_MAJOR 51
- #define LIBAVUTIL_VERSION_MINOR 26
-#define LIBAVUTIL_VERSION_MINOR 18
++#define LIBAVUTIL_VERSION_MINOR 27
#define LIBAVUTIL_VERSION_MICRO 0
#define LIBAVUTIL_VERSION_INT AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \
#include <string.h>
typedef struct SampleFmtInfo {
- const char *name;
+ char name[4];
int bits;
+ int planar;
} SampleFmtInfo;
/** this table gives more information about formats */
0, 0, 460800, 0x54aedafe
1, 0, 4096, 0x00000000
1, 2090, 4096, 0x4dfae7a6
- 0, 3003, 460800, 0x54aedafe
+ 0, 3003, 460800, 0xb7aa8b56
1, 4180, 4096, 0x3fd9f5c6
- 0, 6006, 460800, 0x54aedafe
+ 0, 6006, 460800, 0x283ea3b5
1, 6269, 4096, 0x7b86e310
1, 8359, 4096, 0x611cece5
- 0, 9009, 460800, 0x54aedafe
+ 0, 9009, 460800, 0x283ea3b5
1, 10449, 4096, 0xb7d8e872
- 0, 12012, 460800, 0xb7aa8b56
+ 0, 12012, 460800, 0x10e577de
1, 12539, 4096, 0x072ef72b
1, 14629, 4096, 0xb3560144
- 0, 15015, 460800, 0x283ea3b5
+ 0, 15015, 460800, 0x4e091ee2
1, 16718, 4096, 0x0a3d119e
- 0, 18018, 460800, 0x283ea3b5
+ 0, 18018, 460800, 0x2ea88828
1, 18808, 4096, 0xbe391aa4
1, 20898, 4096, 0x28f7c6e5
- 0, 21021, 460800, 0x10e577de
+ 0, 21021, 460800, 0x4b7f4df0
1, 22988, 4096, 0xca9d9df2
- 0, 24024, 460800, 0x4e091ee2
-0, 24024, 460800, 0xb30eb322
++0, 24024, 460800, 0xa57f20d0
1, 25078, 4096, 0x5c6b95a9
- 0, 27027, 460800, 0x2ea88828
1, 27167, 4096, 0x0bdfc0bf
1, 29257, 4096, 0xd95a9277
- 0, 30030, 460800, 0x4b7f4df0
1, 31347, 4096, 0xae2bef2c
- 0, 33033, 460800, 0xa57f20d0
1, 33437, 4096, 0xbf031e83
1, 35527, 4096, 0x4c83e2d1
projects / ffmpeg / commitdiff