avformat/argo_asf: fix enforcement of chunk count

[ffmpeg] / libavcodec / libopusdec.c

diff --git a/libavcodec/libopusdec.c b/libavcodec/libopusdec.c
index 4f7f4755c242ae2b93908114a6737aeea7e2a4f5..1724a49906f64a5ddbc0b8049d4f0b53dc640ca2 100644 (file)
--- a/libavcodec/libopusdec.c
+++ b/libavcodec/libopusdec.c
@@ -25,6 +25,7 @@
 #include "libavutil/internal.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/ffmath.h"
+#include "libavutil/opt.h"
 
 #include "avcodec.h"
 #include "internal.h"
@@ -33,11 +34,15 @@
 #include "libopus.h"
 
 struct libopus_context {
+    AVClass *class;
     OpusMSDecoder *dec;
     int pre_skip;
 #ifndef OPUS_SET_GAIN
     union { int i; double d; } gain;
 #endif
+#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
+    int apply_phase_inv;
+#endif
 };
 
 #define OPUS_HEAD_SIZE 19
@@ -58,6 +63,8 @@ static av_cold int libopus_decode_init(AVCodecContext *avc)
     avc->sample_rate    = 48000;
     avc->sample_fmt     = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
                           AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
+    avc->channel_layout = avc->channels > 8 ? 0 :
+                          ff_vorbis_channel_layouts[avc->channels - 1];
 
     if (avc->extradata_size >= OPUS_HEAD_SIZE) {
         opus->pre_skip = AV_RL16(avc->extradata + 10);
@@ -81,35 +88,14 @@ static av_cold int libopus_decode_init(AVCodecContext *avc)
         mapping    = mapping_arr;
     }
 
-    if (channel_map == 1) {
-        avc->channel_layout = avc->channels > 8 ? 0 :
-                              ff_vorbis_channel_layouts[avc->channels - 1];
-        if (avc->channels > 2 && avc->channels <= 8) {
-            const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
-            int ch;
-
-            /* Remap channels from Vorbis order to ffmpeg order */
-            for (ch = 0; ch < avc->channels; ch++)
-                mapping_arr[ch] = mapping[vorbis_offset[ch]];
-            mapping = mapping_arr;
-        }
-    } else if (channel_map == 2) {
-        int ambisonic_order = ff_sqrt(avc->channels) - 1;
-        if (avc->channels != (ambisonic_order + 1) * (ambisonic_order + 1) &&
-            avc->channels != (ambisonic_order + 1) * (ambisonic_order + 1) + 2) {
-            av_log(avc, AV_LOG_ERROR,
-                   "Channel mapping 2 is only specified for channel counts"
-                   " which can be written as (n + 1)^2 or (n + 2)^2 + 2"
-                   " for nonnegative integer n\n");
-            return AVERROR_INVALIDDATA;
-        }
-        if (avc->channels > 227) {
-            av_log(avc, AV_LOG_ERROR, "Too many channels\n");
-            return AVERROR_INVALIDDATA;
-        }
-        avc->channel_layout = 0;
-    } else {
-        avc->channel_layout = 0;
+    if (avc->channels > 2 && avc->channels <= 8) {
+        const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
+        int ch;
+
+        /* Remap channels from Vorbis order to ffmpeg order */
+        for (ch = 0; ch < avc->channels; ch++)
+            mapping_arr[ch] = mapping[vorbis_offset[ch]];
+        mapping = mapping_arr;
     }
 
     opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels,
@@ -136,6 +122,15 @@ static av_cold int libopus_decode_init(AVCodecContext *avc)
     }
 #endif
 
+#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
+    ret = opus_multistream_decoder_ctl(opus->dec,
+                                       OPUS_SET_PHASE_INVERSION_DISABLED(!opus->apply_phase_inv));
+    if (ret != OPUS_OK)
+        av_log(avc, AV_LOG_WARNING,
+               "Unable to set phase inversion: %s\n",
+               opus_strerror(ret));
+#endif
+
     /* Decoder delay (in samples) at 48kHz */
     avc->delay = avc->internal->skip_samples = opus->pre_skip;
 
@@ -146,7 +141,10 @@ static av_cold int libopus_decode_close(AVCodecContext *avc)
 {
     struct libopus_context *opus = avc->priv_data;
 
-    opus_multistream_decoder_destroy(opus->dec);
+    if (opus->dec) {
+        opus_multistream_decoder_destroy(opus->dec);
+        opus->dec = NULL;
+    }
     return 0;
 }
 
@@ -209,6 +207,24 @@ static void libopus_flush(AVCodecContext *avc)
     avc->internal->skip_samples = opus->pre_skip;
 }
 
+
+#define OFFSET(x) offsetof(struct libopus_context, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
+static const AVOption libopusdec_options[] = {
+#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
+    { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, FLAGS },
+#endif
+    { NULL },
+};
+
+static const AVClass libopusdec_class = {
+    .class_name = "libopusdec",
+    .item_name  = av_default_item_name,
+    .option     = libopusdec_options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+
 AVCodec ff_libopus_decoder = {
     .name           = "libopus",
     .long_name      = NULL_IF_CONFIG_SMALL("libopus Opus"),
@@ -220,8 +236,10 @@ AVCodec ff_libopus_decoder = {
     .decode         = libopus_decode,
     .flush          = libopus_flush,
     .capabilities   = AV_CODEC_CAP_DR1,
+    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
                                                      AV_SAMPLE_FMT_S16,
                                                      AV_SAMPLE_FMT_NONE },
+    .priv_class     = &libopusdec_class,
     .wrapper_name   = "libopus",
 };