avcodec/aacenc: Remove always-false check

The sample rates have already been checked generically via AVCodec.supported_samplerates.

Andreas Rheinhardt committed 1 year ago

Commit ab1bc2f1 parent 516bcfc4 files · +10 −12Browse files

Unified · Split

4 files

     }
     /* Samplerate */
-    for (i = 0; i < 16; i++)
+    for (int i = 0;; i++) {
-        if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
+        av_assert1(i < 13);
+        if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i]) {
+            s->samplerate_index = i;
             break;
-    s->samplerate_index = i;
+        }
-    ERROR_IF(s->samplerate_index == 16 ||
+    }
-             s->samplerate_index >= ff_aac_swb_size_1024_len ||
-             s->samplerate_index >= ff_aac_swb_size_128_len,
-             "Unsupported sample rate %d\n", avctx->sample_rate);
     /* Bitrate limiting */
     WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,

@@ -103,6 +103,3 @@ const uint8_t *const ff_aac_swb_size_1024[] = { · 102 unmodified lines
103	swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,	103	swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
104	swb_size_1024_8	104	swb_size_1024_8
105	};	105	};
106
107	const int ff_aac_swb_size_128_len = FF_ARRAY_ELEMS(ff_aac_swb_size_128);
108	const int ff_aac_swb_size_1024_len = FF_ARRAY_ELEMS(ff_aac_swb_size_1024);

@@ -41,9 +41,7 @@ · 40 unmodified lines
41	#define AAC_MAX_CHANNELS 16	41	#define AAC_MAX_CHANNELS 16
42		42
43	extern const uint8_t *const ff_aac_swb_size_1024[];	43	extern const uint8_t *const ff_aac_swb_size_1024[];
44	extern const int ff_aac_swb_size_1024_len;
45	extern const uint8_t *const ff_aac_swb_size_128[];	44	extern const uint8_t *const ff_aac_swb_size_128[];
46	extern const int ff_aac_swb_size_128_len;
47		45
48	/* Supported layouts without using a PCE */	46	/* Supported layouts without using a PCE */
49	static const AVChannelLayout aac_normal_chan_layouts[7] = {	47	static const AVChannelLayout aac_normal_chan_layouts[7] = {

 #ifndef AVCODEC_MPEG4AUDIO_SAMPLE_RATES_H
 #define AVCODEC_MPEG4AUDIO_SAMPLE_RATES_H
+// This table contains only 13 real elements and is padded with zeroes.
+// It is used by the AAC encoder as sample rate table, so the encoder
+// needs to actually support all of these rates and it needs to have
+// a trailing zero.
 const int ff_mpeg4audio_sample_rates[16] = {
     96000, 88200, 64000, 48000, 44100, 32000,
     24000, 22050, 16000, 12000, 11025, 8000, 7350

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