Merge branch 'master' into release/current

7 files changed, 65 insertions, 16 deletions

diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index c30aff7fd5..8a6bb4061b 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -45,6 +45,7 @@ struct priv {
     uint32_t skip_samples, trim_samples;
     bool preroll_done;
     double next_pts;
+    AVRational codec_timebase;
 };
 
 static void uninit(struct dec_audio *da);
@@ -84,6 +85,8 @@ static int init(struct dec_audio *da, const char *decoder)
     struct priv *ctx = talloc_zero(NULL, struct priv);
     da->priv = ctx;
 
+    ctx->codec_timebase = (AVRational){0};
+
     ctx->force_channel_map = c->force_channels;
 
     lavc_codec = avcodec_find_decoder_by_name(decoder);
@@ -128,8 +131,7 @@ static int init(struct dec_audio *da, const char *decoder)
     // demux_mkv
     mp_lavc_set_extradata(lavc_context, c->extradata, c->extradata_size);
 
-    if (c->lav_headers)
-        mp_copy_lav_codec_headers(lavc_context, c->lav_headers);
+    mp_set_lav_codec_headers(lavc_context, c);
 
     mp_set_avcodec_threads(da->log, lavc_context, opts->threads);
 
@@ -185,10 +187,23 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
     int in_len = mpkt ? mpkt->len : 0;
 
     AVPacket pkt;
-    mp_set_av_packet(&pkt, mpkt, NULL);
+    mp_set_av_packet(&pkt, mpkt, &priv->codec_timebase);
 
     int got_frame = 0;
     av_frame_unref(priv->avframe);
+
+#if HAVE_AVCODEC_NEW_CODEC_API
+    int ret = avcodec_send_packet(avctx, &pkt);
+    if (ret >= 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+        if (ret >= 0 && mpkt)
+            mpkt->len = 0;
+        ret = avcodec_receive_frame(avctx, priv->avframe);
+        if (ret >= 0)
+            got_frame = 1;
+        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+            ret = 0;
+    }
+#else
     int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
     if (mpkt) {
         // At least "shorten" decodes sub-frames, instead of the whole packet.
@@ -205,6 +220,7 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
             return 0;
         }
     }
+#endif
     if (ret < 0) {
         MP_ERR(da, "Error decoding audio.\n");
         return -1;
@@ -212,7 +228,7 @@ static int decode_packet(struct dec_audio *da, struct demux_packet *mpkt,
     if (!got_frame)
         return 0;
 
-    double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
+    double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, &priv->codec_timebase);
 
     struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
     if (!mpframe)
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 7298d9e7d7..eb2e2bb0a9 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -171,7 +171,11 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
     if (!stream)
         goto fail;
 
+#if HAVE_AVCODEC_HAS_CODECPAR
+    stream->codecpar->codec_id = spdif_ctx->codec_id;
+#else
     stream->codec->codec_id = spdif_ctx->codec_id;
+#endif
 
     AVDictionary *format_opts = NULL;
 
diff --git a/audio/format.c b/audio/format.c
index 06360b884b..bbe0cec9bb 100644
--- a/audio/format.c
+++ b/audio/format.c
@@ -245,6 +245,37 @@ void af_get_best_sample_formats(int src_format, int out_formats[AF_FORMAT_COUNT]
     out_formats[num] = 0;
 }
 
+// Return the best match to src_samplerate from the list provided in the array
+// *available, which must be terminated by 0, or itself NULL. If *available is
+// empty or NULL, return a negative value. Exact match to src_samplerate is
+// most preferred, followed by the lowest integer multiple, followed by the
+// maximum of *available.
+int af_select_best_samplerate(int src_samplerate, const int *available)
+{
+    if (!available)
+        return -1;
+
+    int min_mult_rate = INT_MAX;
+    int max_rate      = INT_MIN;
+    for (int i = 0; available[i]; i++) {
+        if (available[i] == src_samplerate)
+            return available[i];
+
+        if (!(available[i] % src_samplerate))
+            min_mult_rate = MPMIN(min_mult_rate, available[i]);
+
+        max_rate = MPMAX(max_rate, available[i]);
+    }
+
+    if (min_mult_rate < INT_MAX)
+        return min_mult_rate;
+
+    if (max_rate > INT_MIN)
+        return max_rate;
+
+    return -1;
+}
+
 // Return the number of samples that make up one frame in this format.
 // You get the byte size by multiplying them with sample size and channel count.
 int af_format_sample_alignment(int format)
diff --git a/audio/format.h b/audio/format.h
index de1ff1a30d..3d5cc1024b 100644
--- a/audio/format.h
+++ b/audio/format.h
@@ -76,6 +76,7 @@ int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int sampler
 void af_fill_silence(void *dst, size_t bytes, int format);
 
 void af_get_best_sample_formats(int src_format, int out_formats[AF_FORMAT_COUNT]);
+int af_select_best_samplerate(int src_sampelrate, const int *available);
 
 int af_format_sample_alignment(int format);
 
diff --git a/audio/out/ao_coreaudio_exclusive.c b/audio/out/ao_coreaudio_exclusive.c
index af07fe0f3e..510fa3a529 100644
--- a/audio/out/ao_coreaudio_exclusive.c
+++ b/audio/out/ao_coreaudio_exclusive.c
@@ -367,7 +367,7 @@ static void uninit(struct ao *ao)
     CHECK_CA_WARN("failed to remove device render callback");
 
     if (!ca_change_physical_format_sync(ao, p->stream, p->original_asbd))
-        MP_WARN(ao, "can't revert to original device format");
+        MP_WARN(ao, "can't revert to original device format\n");
 
     err = ca_enable_mixing(ao, p->device, p->changed_mixing);
     CHECK_CA_WARN("can't re-enable mixing");
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index b671301922..bf1728682a 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -108,6 +108,11 @@ static int init(struct ao *ao)
 
     codec = encode_lavc_get_codec(ao->encode_lavc_ctx, ac->stream);
 
+    int samplerate = af_select_best_samplerate(ao->samplerate,
+                                               codec->supported_samplerates);
+    if (samplerate > 0)
+        ao->samplerate = samplerate;
+
     // TODO: Remove this redundancy with encode_lavc_alloc_stream also
     // setting the time base.
     // Using codec->time_bvase is deprecated, but needed for older lavf.
diff --git a/audio/out/ao_wasapi_utils.c b/audio/out/ao_wasapi_utils.c
index 8d0ea30bd7..0e7dec885a 100644
--- a/audio/out/ao_wasapi_utils.c
+++ b/audio/out/ao_wasapi_utils.c
@@ -364,17 +364,9 @@ static bool search_samplerates(struct ao *ao, WAVEFORMATEXTENSIBLE *wformat,
         }
     }
 
-    for (int i = 0; supported[i]; i++) {
-        // first choose the lowest integer multiple of the sample rate
-        if (!(supported[i] % ao->samplerate)) {
-            change_waveformat_samplerate(wformat, supported[i]);
-            return true;
-        }
-    }
-
-    // then choose the highest supported (if any)
-    if (n) {
-        change_waveformat_samplerate(wformat, supported[n-1]);
+    int samplerate = af_select_best_samplerate(ao->samplerate, supported);
+    if (samplerate > 0) {
+        change_waveformat_samplerate(wformat, samplerate);
         return true;
     }