From aa79d13f51aa820c7e5f07784a2512434e68bc46 Mon Sep 17 00:00:00 2001
From: James Almer <jamrial@gmail.com>
Date: Wed, 21 Sep 2022 00:01:40 -0300
Subject: [PATCH] avformat/cafenc: derive Opus frame size from the relevant
 stream parameters

Use the stream duration as last resort, as an off-by-one result of the
"st->duration / (caf->packets - 1)" calculation can break playback on some
devices.
Also, don't write the sample_rate value propagated by encoders like libopus.
The sample rate of the audio fed to it is irrelevant after being encoded.

Fixes ticket #9930.

Signed-off-by: James Almer <jamrial@gmail.com>
---
 libavformat/cafenc.c | 19 ++++++++++++++-----
 1 file changed, 14 insertions(+), 5 deletions(-)

diff --git a/libavformat/cafenc.c b/libavformat/cafenc.c
index fedb430b17..b90811d46f 100644
--- a/libavformat/cafenc.c
+++ b/libavformat/cafenc.c
@@ -53,7 +53,11 @@ static uint32_t codec_flags(enum AVCodecID codec_id) {
     }
 }
 
-static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int block_align) {
+static uint32_t samples_per_packet(const AVCodecParameters *par) {
+    enum AVCodecID codec_id = par->codec_id;
+    int channels = par->ch_layout.nb_channels, block_align = par->block_align;
+    int frame_size = par->frame_size, sample_rate = par->sample_rate;
+
     switch (codec_id) {
     case AV_CODEC_ID_PCM_S8:
     case AV_CODEC_ID_PCM_S16LE:
@@ -83,6 +87,8 @@ static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int bl
         return 320;
     case AV_CODEC_ID_MP1:
         return 384;
+    case AV_CODEC_ID_OPUS:
+        return frame_size * 48000 / sample_rate;
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MP3:
         return 1152;
@@ -110,7 +116,7 @@ static int caf_write_header(AVFormatContext *s)
     AVDictionaryEntry *t = NULL;
     unsigned int codec_tag = ff_codec_get_tag(ff_codec_caf_tags, par->codec_id);
     int64_t chunk_size = 0;
-    int frame_size = par->frame_size;
+    int frame_size = par->frame_size, sample_rate = par->sample_rate;
 
     if (s->nb_streams != 1) {
         av_log(s, AV_LOG_ERROR, "CAF files have exactly one stream\n");
@@ -139,7 +145,10 @@ static int caf_write_header(AVFormatContext *s)
     }
 
     if (par->codec_id != AV_CODEC_ID_MP3 || frame_size != 576)
-        frame_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
+        frame_size = samples_per_packet(par);
+
+    if (par->codec_id == AV_CODEC_ID_OPUS)
+        sample_rate = 48000;
 
     ffio_wfourcc(pb, "caff"); //< mFileType
     avio_wb16(pb, 1);         //< mFileVersion
@@ -147,7 +156,7 @@ static int caf_write_header(AVFormatContext *s)
 
     ffio_wfourcc(pb, "desc");                         //< Audio Description chunk
     avio_wb64(pb, 32);                                //< mChunkSize
-    avio_wb64(pb, av_double2int(par->sample_rate));   //< mSampleRate
+    avio_wb64(pb, av_double2int(sample_rate));        //< mSampleRate
     avio_wl32(pb, codec_tag);                         //< mFormatID
     avio_wb32(pb, codec_flags(par->codec_id));        //< mFormatFlags
     avio_wb32(pb, par->block_align);                  //< mBytesPerPacket
@@ -248,7 +257,7 @@ static int caf_write_trailer(AVFormatContext *s)
         avio_seek(pb, caf->data, SEEK_SET);
         avio_wb64(pb, file_size - caf->data - 8);
         if (!par->block_align) {
-            int packet_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
+            int packet_size = samples_per_packet(par);
             if (!packet_size) {
                 packet_size = st->duration / (caf->packets - 1);
                 avio_seek(pb, FRAME_SIZE_OFFSET, SEEK_SET);