ref: 2182b8e13d9ed233138fc465050344ab0f37d079
parent: 10db50bf910563bfa08aec0a5090db6da87b20ac
author: qwx <qwx@sciops.net>
date: Thu Aug 20 19:41:49 EDT 2020
add seeking for opusdec
--- /dev/null
+++ b/opus-tools-seek
@@ -1,0 +1,118 @@
+--- /mnt/git/branch/heads/test3-appveyor/tree/src/opusdec.c Fri Aug 21 01:36:31 2020
++++ src/opusdec.c Fri Aug 21 01:32:19 2020
+@@ -469,7 +469,7 @@
+
+ opus_int64 audio_write(float *pcm, int channels, int frame_size, FILE *fout,
+ SpeexResamplerState *resampler, float *clipmem, shapestate *shapemem,
+- int file, int rate, opus_int64 link_read, opus_int64 link_out, int fp)
++ int file, int rate, opus_int64 link_read, opus_int64 link_out, int fp, opus_int64 *seek)
+ {+ opus_int64 sampout=0;
+ opus_int64 maxout;
+@@ -503,6 +503,14 @@
+ frame_size=0;
+ }
+
++ if (*seek > 0)
++ {++ ret = *seek>out_len?out_len:*seek;
++ ret *= (fp?sizeof(float):sizeof(short))*channels;
++ *seek -= ret;
++ goto next;
++ }
++
+ if (!file||!fp)
+ {+ /*Convert to short and save to output file*/
+@@ -546,6 +554,7 @@
+ #endif
+ ret=fwrite(fp?(char *)output:(char *)out,
+ (fp?sizeof(float):sizeof(short))*channels, out_len, fout);
++next:
+ sampout+=ret;
+ maxout-=ret;
+ }
+@@ -632,7 +641,7 @@
+ static void drain_resampler(FILE *fout, int file_output,
+ SpeexResamplerState *resampler, int channels, int rate,
+ opus_int64 link_read, opus_int64 link_out, float *clipmem,
+- shapestate *shapemem, opus_int64 *audio_size, int fp)
++ shapestate *shapemem, opus_int64 *audio_size, int fp, opus_int64 *seek)
+ {+ float *zeros;
+ int drain;
+@@ -643,7 +652,7 @@
+ opus_int64 outsamp;
+ int tmp=MINI(drain, 100);
+ outsamp=audio_write(zeros, channels, tmp, fout, resampler, clipmem,
+- shapemem, file_output, rate, link_read, link_out, fp);
++ shapemem, file_output, rate, link_read, link_out, fp, seek);
+ link_out+=outsamp;
+ (*audio_size)+=(fp?sizeof(float):sizeof(short))*outsamp*channels;
+ drain-=tmp;
+@@ -690,6 +699,7 @@
+ {0, 0, 0, 0}+ };
+ opus_int64 audio_size=0;
++ opus_int64 seek=-1;
+ double last_coded_seconds=0;
+ float loss_percent=-1;
+ float manual_gain=0;
+@@ -725,7 +735,7 @@
+ /*Process options*/
+ while (1)
+ {+- c = getopt_long(argc_utf8, argv_utf8, "hV",
++ c = getopt_long(argc_utf8, argv_utf8, "hVs:",
+ long_options, &option_index);
+ if (c==-1)
+ break;
+@@ -777,6 +787,9 @@
+ case 'h':
+ usage();
+ goto done;
++ case 's':
++ seek = atoi(optarg);
++ break;
+ case 'V':
+ version();
+ goto done;
+@@ -990,6 +1003,11 @@
+ op_set_decode_callback(st, (op_decode_cb_func)decode_cb, &cb_ctx);
+ }
+
++ if (seek != -1)
++ {++ seek *= rate;
++ }
++
+ /*Main decoding loop*/
+ while (1)
+ {+@@ -1033,7 +1051,7 @@
+ {+ drain_resampler(fout, file_output, resampler, channels, rate,
+ link_read, link_out, clipmem, dither?&shapemem:NULL, &audio_size,
+- fp);
++ fp, &seek);
+ /*Neither speex_resampler_reset_mem() nor
+ speex_resampler_skip_zeros() clear the number of fractional
+ samples properly, so we just destroy it. It will get re-created
+@@ -1141,7 +1159,7 @@
+ }
+ outsamp=audio_write(permuted_output?permuted_output:output, channels,
+ nb_read, fout, resampler, clipmem, dither?&shapemem:0, file_output,
+- rate, link_read, link_out, fp);
++ rate, link_read, link_out, fp, &seek);
+ link_out+=outsamp;
+ audio_size+=(fp?sizeof(float):sizeof(short))*outsamp*channels;
+ }
+@@ -1149,7 +1167,7 @@
+ if (resampler!=NULL)
+ {+ drain_resampler(fout, file_output, resampler, channels, rate,
+- link_read, link_out, clipmem, dither?&shapemem:NULL, &audio_size, fp);
++ link_read, link_out, clipmem, dither?&shapemem:NULL, &audio_size, fp, &seek);
+ speex_resampler_destroy(resampler);
+ }
+