libopus: convert to new channel layout API

Signed-off-by: James Almer <jamrial@gmail.com>

1 files changed, 22 insertions, 17 deletions

diff --git a/libavcodec/libopusdec.c b/libavcodec/libopusdec.c
index 86ef715205..e4d127ad74 100644
--- a/libavcodec/libopusdec.c
+++ b/libavcodec/libopusdec.c
@@ -50,55 +50,60 @@ struct libopus_context {
 static av_cold int libopus_decode_init(AVCodecContext *avc)
 {
     struct libopus_context *opus = avc->priv_data;
-    int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled;
+    int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled, channels;
     uint8_t mapping_arr[8] = { 0, 1 }, *mapping;
 
-    avc->channels = avc->extradata_size >= 10 ? avc->extradata[9] : (avc->channels == 1) ? 1 : 2;
-    if (avc->channels <= 0) {
+    channels = avc->extradata_size >= 10 ? avc->extradata[9] : (avc->ch_layout.nb_channels == 1) ? 1 : 2;
+    if (channels <= 0) {
         av_log(avc, AV_LOG_WARNING,
-               "Invalid number of channels %d, defaulting to stereo\n", avc->channels);
-        avc->channels = 2;
+               "Invalid number of channels %d, defaulting to stereo\n", channels);
+        channels = 2;
     }
 
     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];
+    av_channel_layout_uninit(&avc->ch_layout);
+    if (channels > 8) {
+        avc->ch_layout.order       = AV_CHANNEL_ORDER_UNSPEC;
+        avc->ch_layout.nb_channels = channels;
+    } else {
+        av_channel_layout_copy(&avc->ch_layout, &ff_vorbis_ch_layouts[channels - 1]);
+    }
 
     if (avc->extradata_size >= OPUS_HEAD_SIZE) {
         opus->pre_skip = AV_RL16(avc->extradata + 10);
         gain_db     = sign_extend(AV_RL16(avc->extradata + 16), 16);
         channel_map = AV_RL8 (avc->extradata + 18);
     }
-    if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + avc->channels) {
+    if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + channels) {
         nb_streams = avc->extradata[OPUS_HEAD_SIZE + 0];
         nb_coupled = avc->extradata[OPUS_HEAD_SIZE + 1];
-        if (nb_streams + nb_coupled != avc->channels)
+        if (nb_streams + nb_coupled != channels)
             av_log(avc, AV_LOG_WARNING, "Inconsistent channel mapping.\n");
         mapping = avc->extradata + OPUS_HEAD_SIZE + 2;
     } else {
-        if (avc->channels > 2 || channel_map) {
+        if (channels > 2 || channel_map) {
             av_log(avc, AV_LOG_ERROR,
-                   "No channel mapping for %d channels.\n", avc->channels);
+                   "No channel mapping for %d channels.\n", channels);
             return AVERROR(EINVAL);
         }
         nb_streams = 1;
-        nb_coupled = avc->channels > 1;
+        nb_coupled = channels > 1;
         mapping    = mapping_arr;
     }
 
-    if (avc->channels > 2 && avc->channels <= 8) {
-        const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
+    if (channels > 2 && channels <= 8) {
+        const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[channels - 1];
         int ch;
 
         /* Remap channels from Vorbis order to ffmpeg order */
-        for (ch = 0; ch < avc->channels; ch++)
+        for (ch = 0; ch < channels; ch++)
             mapping_arr[ch] = mapping[vorbis_offset[ch]];
         mapping = mapping_arr;
     }
 
-    opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels,
+    opus->dec = opus_multistream_decoder_create(avc->sample_rate, channels,
                                                 nb_streams, nb_coupled,
                                                 mapping, &ret);
     if (!opus->dec) {
@@ -178,7 +183,7 @@ static int libopus_decode(AVCodecContext *avc, void *data,
 
 #ifndef OPUS_SET_GAIN
     {
-        int i = avc->channels * nb_samples;
+        int i = avc->ch_layout.nb_channels * nb_samples;
         if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) {
             float *pcm = (float *)frame->data[0];
             for (; i > 0; i--, pcm++)