mov: Fix little endian audio detection

[ffmpeg] / libavformat / riffenc.c

diff --git a/libavformat/riffenc.c b/libavformat/riffenc.c
index 8479d9da3890e17903daff7d5ee1ac898cecc601..81b3b2da12e7fffb62e89fc37fb3f08517d6effc 100644 (file)
--- a/libavformat/riffenc.c
+++ b/libavformat/riffenc.c
@@ -62,9 +62,7 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
     /* We use the known constant frame size for the codec if known, otherwise
      * fall back on using AVCodecContext.frame_size, which is not as reliable
      * for indicating packet duration. */
-    frame_size = av_get_audio_frame_duration(enc, 0);
-    if (!frame_size)
-        frame_size = enc->frame_size;
+    frame_size = av_get_audio_frame_duration(enc, enc->block_align);
 
     waveformatextensible = (enc->channels > 2 && enc->channel_layout) ||
                            enc->sample_rate > 48000 ||
@@ -96,12 +94,10 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
                enc->bits_per_coded_sample, bps);
     }
 
-    if (enc->codec_id == AV_CODEC_ID_MP2 ||
-        enc->codec_id == AV_CODEC_ID_MP3) {
-        /* This is wrong, but it seems many demuxers do not work if this
-         * is set correctly. */
+    if (enc->codec_id == AV_CODEC_ID_MP2) {
         blkalign = frame_size;
-        // blkalign = 144 * enc->bit_rate/enc->sample_rate;
+    } else if (enc->codec_id == AV_CODEC_ID_MP3) {
+        blkalign = 576 * (enc->sample_rate <= 24000 ? 1 : 2);
     } else if (enc->codec_id == AV_CODEC_ID_AC3) {
         blkalign = 3840;                /* maximum bytes per frame */
     } else if (enc->block_align != 0) { /* specified by the codec */
@@ -209,32 +205,33 @@ void ff_put_bmp_header(AVIOContext *pb, AVCodecContext *enc,
         avio_w8(pb, 0);
 }
 
-void ff_parse_specific_params(AVCodecContext *stream, int *au_rate,
+void ff_parse_specific_params(AVStream *st, int *au_rate,
                               int *au_ssize, int *au_scale)
 {
+    AVCodecContext *codec = st->codec;
     int gcd;
     int audio_frame_size;
 
     /* We use the known constant frame size for the codec if known, otherwise
      * fall back on using AVCodecContext.frame_size, which is not as reliable
      * for indicating packet duration. */
-    audio_frame_size = av_get_audio_frame_duration(stream, 0);
+    audio_frame_size = av_get_audio_frame_duration(codec, 0);
     if (!audio_frame_size)
-        audio_frame_size = stream->frame_size;
+        audio_frame_size = codec->frame_size;
 
-    *au_ssize = stream->block_align;
-    if (audio_frame_size && stream->sample_rate) {
+    *au_ssize = codec->block_align;
+    if (audio_frame_size && codec->sample_rate) {
         *au_scale = audio_frame_size;
-        *au_rate  = stream->sample_rate;
-    } else if (stream->codec_type == AVMEDIA_TYPE_VIDEO ||
-               stream->codec_type == AVMEDIA_TYPE_DATA ||
-               stream->codec_type == AVMEDIA_TYPE_SUBTITLE) {
-        *au_scale = stream->time_base.num;
-        *au_rate  = stream->time_base.den;
+        *au_rate  = codec->sample_rate;
+    } else if (codec->codec_type == AVMEDIA_TYPE_VIDEO ||
+               codec->codec_type == AVMEDIA_TYPE_DATA ||
+               codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
+        *au_scale = st->time_base.num;
+        *au_rate  = st->time_base.den;
     } else {
-        *au_scale = stream->block_align ? stream->block_align * 8 : 8;
-        *au_rate  = stream->bit_rate ? stream->bit_rate :
-                    8 * stream->sample_rate;
+        *au_scale = codec->block_align ? codec->block_align * 8 : 8;
+        *au_rate  = codec->bit_rate ? codec->bit_rate :
+                    8 * codec->sample_rate;
     }
     gcd        = av_gcd(*au_scale, *au_rate);
     *au_scale /= gcd;