Use av_packet_rescale_ts() to simplify code.

[ffmpeg] / libavcodec / libopusdec.c

diff --git a/libavcodec/libopusdec.c b/libavcodec/libopusdec.c
index 9d24e6e33b48e3b8cfc9642201f7c104a14958bc..398450f4d45ea5963b5be7c6b79d7ba2ff788be4 100644 (file)
--- a/libavcodec/libopusdec.c
+++ b/libavcodec/libopusdec.c
@@ -22,62 +22,25 @@
 #include <opus.h>
 #include <opus_multistream.h>
 
-#include "libavutil/common.h"
 #include "libavutil/avassert.h"
 #include "libavutil/intreadwrite.h"
 #include "avcodec.h"
+#include "internal.h"
 #include "vorbis.h"
 #include "mathops.h"
+#include "libopus.h"
 
 struct libopus_context {
     OpusMSDecoder *dec;
-    AVFrame frame;
 };
 
-static int opus_error_to_averror(int err)
-{
-    switch (err) {
-    case OPUS_BAD_ARG:
-        return AVERROR(EINVAL);
-    case OPUS_BUFFER_TOO_SMALL:
-        return AVERROR_UNKNOWN;
-    case OPUS_INTERNAL_ERROR:
-        return AVERROR(EFAULT);
-    case OPUS_INVALID_PACKET:
-        return AVERROR_INVALIDDATA;
-    case OPUS_UNIMPLEMENTED:
-        return AVERROR(ENOSYS);
-    case OPUS_INVALID_STATE:
-        return AVERROR_UNKNOWN;
-    case OPUS_ALLOC_FAIL:
-        return AVERROR(ENOMEM);
-    default:
-        return AVERROR(EINVAL);
-    }
-}
-
-static inline void reorder(uint8_t *data, unsigned channels, unsigned bps,
-                           unsigned samples, const uint8_t *map)
-{
-    uint8_t tmp[8 * 4];
-    unsigned i;
-
-    av_assert1(channels * bps <= sizeof(tmp));
-    for (; samples > 0; samples--) {
-        for (i = 0; i < channels; i++)
-            memcpy(tmp + bps * i, data + bps * map[i], bps);
-        memcpy(data, tmp, bps * channels);
-        data += bps * channels;
-    }
-}
-
 #define OPUS_HEAD_SIZE 19
 
 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;
-    uint8_t mapping_stereo[] = { 0, 1 }, *mapping;
+    uint8_t mapping_arr[8] = { 0, 1 }, *mapping;
 
     avc->sample_rate    = 48000;
     avc->sample_fmt     = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
@@ -103,7 +66,17 @@ static av_cold int libopus_decode_init(AVCodecContext *avc)
         }
         nb_streams = 1;
         nb_coupled = avc->channels > 1;
-        mapping    = mapping_stereo;
+        mapping    = mapping_arr;
+    }
+
+    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 libav 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,
@@ -112,7 +85,7 @@ static av_cold int libopus_decode_init(AVCodecContext *avc)
     if (!opus->dec) {
         av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n",
                opus_strerror(ret));
-        return opus_error_to_averror(ret);
+        return ff_opus_error_to_averror(ret);
     }
 
     ret = opus_multistream_decoder_ctl(opus->dec, OPUS_SET_GAIN(gain_db));
@@ -121,8 +94,7 @@ static av_cold int libopus_decode_init(AVCodecContext *avc)
                opus_strerror(ret));
 
     avc->delay = 3840;  /* Decoder delay (in samples) at 48kHz */
-    avcodec_get_frame_defaults(&opus->frame);
-    avc->coded_frame = &opus->frame;
+
     return 0;
 }
 
@@ -136,14 +108,15 @@ static av_cold int libopus_decode_close(AVCodecContext *avc)
 
 #define MAX_FRAME_SIZE (960 * 6)
 
-static int libopus_decode(AVCodecContext *avc, void *frame,
+static int libopus_decode(AVCodecContext *avc, void *data,
                           int *got_frame_ptr, AVPacket *pkt)
 {
     struct libopus_context *opus = avc->priv_data;
+    AVFrame *frame               = data;
     int ret, nb_samples;
 
-    opus->frame.nb_samples = MAX_FRAME_SIZE;
-    ret = avc->get_buffer(avc, &opus->frame);
+    frame->nb_samples = MAX_FRAME_SIZE;
+    ret = ff_get_buffer(avc, frame, 0);
     if (ret < 0) {
         av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
@@ -151,30 +124,22 @@ static int libopus_decode(AVCodecContext *avc, void *frame,
 
     if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
         nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
-                                             (opus_int16 *)opus->frame.data[0],
-                                             opus->frame.nb_samples, 0);
+                                             (opus_int16 *)frame->data[0],
+                                             frame->nb_samples, 0);
     else
         nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
-                                                   (float *)opus->frame.data[0],
-                                                   opus->frame.nb_samples, 0);
+                                                   (float *)frame->data[0],
+                                                   frame->nb_samples, 0);
 
     if (nb_samples < 0) {
         av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
                opus_strerror(nb_samples));
-        return opus_error_to_averror(nb_samples);
+        return ff_opus_error_to_averror(nb_samples);
     }
 
-    if (avc->channels > 3 && avc->channels <= 8) {
-        const uint8_t *m = ff_vorbis_channel_layout_offsets[avc->channels - 1];
-        if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
-            reorder(opus->frame.data[0], avc->channels, 2, nb_samples, m);
-        else
-            reorder(opus->frame.data[0], avc->channels, 4, nb_samples, m);
-    }
+    frame->nb_samples = nb_samples;
+    *got_frame_ptr    = 1;
 
-    opus->frame.nb_samples = nb_samples;
-    *(AVFrame *)frame = opus->frame;
-    *got_frame_ptr = 1;
     return pkt->size;
 }
 
@@ -187,6 +152,7 @@ static void libopus_flush(AVCodecContext *avc)
 
 AVCodec ff_libopus_decoder = {
     .name           = "libopus",
+    .long_name      = NULL_IF_CONFIG_SMALL("libopus Opus"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_OPUS,
     .priv_data_size = sizeof(struct libopus_context),
@@ -195,7 +161,6 @@ AVCodec ff_libopus_decoder = {
     .decode         = libopus_decode,
     .flush          = libopus_flush,
     .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("libopus Opus"),
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
                                                      AV_SAMPLE_FMT_S16,
                                                      AV_SAMPLE_FMT_NONE },