ao_lavc, vo_lavc: Migrate to new encoding API.

Also marked some places for possible later refactoring, as they became
quite similar in this commit.

1 files changed, 121 insertions, 76 deletions

diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index 572874d27c..6b4279ca87 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -39,8 +39,6 @@
 #include "common/encode_lavc.h"
 
 struct priv {
-    uint8_t *buffer;
-    size_t buffer_size;
     AVStream *stream;
     AVCodecContext *codec;
     int pcmhack;
@@ -146,18 +144,10 @@ static int init(struct ao *ao)
     if (ac->codec->frame_size <= 1)
         ac->pcmhack = av_get_bits_per_sample(ac->codec->codec_id) / 8;
 
-    if (ac->pcmhack) {
+    if (ac->pcmhack)
         ac->aframesize = 16384; // "enough"
-        ac->buffer_size =
-            ac->aframesize * ac->pcmhack * ao->channels.num * 2 + 200;
-    } else {
+    else
         ac->aframesize = ac->codec->frame_size;
-        ac->buffer_size =
-            ac->aframesize * ac->sample_size * ao->channels.num * 2 + 200;
-    }
-    if (ac->buffer_size < FF_MIN_BUFFER_SIZE)
-        ac->buffer_size = FF_MIN_BUFFER_SIZE;
-    ac->buffer = talloc_size(ac, ac->buffer_size);
 
     // enough frames for at least 0.25 seconds
     ac->framecount = ceil(ao->samplerate * 0.25 / ac->aframesize);
@@ -182,7 +172,7 @@ fail:
 }
 
 // close audio device
-static int encode(struct ao *ao, double apts, void **data);
+static void encode(struct ao *ao, double apts, void **data);
 static void uninit(struct ao *ao)
 {
     struct priv *ac = ao->priv;
@@ -199,12 +189,12 @@ static void uninit(struct ao *ao)
         return;
     }
 
-    if (ac->buffer) {
+    if (ac->stream) {
         double outpts = ac->expected_next_pts;
         if (!ectx->options->rawts && ectx->options->copyts)
             outpts += ectx->discontinuity_pts_offset;
         outpts += encode_lavc_getoffset(ectx, ac->codec);
-        while (encode(ao, outpts, NULL) > 0) ;
+        encode(ao, outpts, NULL);
     }
 
     pthread_mutex_unlock(&ectx->lock);
@@ -220,24 +210,130 @@ static int get_space(struct ao *ao)
     return ac->aframesize * ac->framecount;
 }
 
+static void write_packet(struct ao *ao, AVPacket *packet)
+{
+    // TODO: Can we unify this with the equivalent video code path?
+    struct priv *ac = ao->priv;
+
+    packet->stream_index = ac->stream->index;
+    if (packet->pts != AV_NOPTS_VALUE) {
+        packet->pts = av_rescale_q(packet->pts,
+                                   ac->codec->time_base,
+                                   ac->stream->time_base);
+    } else {
+        // Do we need this at all? Better be safe than sorry...
+        MP_WARN(ao, "encoder lost pts, why?\n");
+        if (ac->savepts != MP_NOPTS_VALUE) {
+            packet->pts = av_rescale_q(ac->savepts,
+                                       ac->codec->time_base,
+                                       ac->stream->time_base);
+        }
+    }
+    if (packet->dts != AV_NOPTS_VALUE) {
+        packet->dts = av_rescale_q(packet->dts,
+                                   ac->codec->time_base,
+                                   ac->stream->time_base);
+    }
+    if (packet->duration > 0) {
+        packet->duration = av_rescale_q(packet->duration,
+                                        ac->codec->time_base,
+                                        ac->stream->time_base);
+    }
+
+    ac->savepts = AV_NOPTS_VALUE;
+
+    if (encode_lavc_write_frame(ao->encode_lavc_ctx,
+                                ac->stream, packet) < 0) {
+        MP_ERR(ao, "error writing at %d %d/%d\n",
+               (int) packet->pts,
+               ac->stream->time_base.num,
+               ac->stream->time_base.den);
+        return;
+    }
+}
+
+static void encode_audio_and_write(struct ao *ao, AVFrame *frame)
+{
+    // TODO: Can we unify this with the equivalent video code path?
+    struct priv *ac = ao->priv;
+    AVPacket packet = {0};
+
+#if HAVE_AVCODEC_NEW_CODEC_API
+    int status = avcodec_send_frame(ac->codec, frame);
+    if (status < 0) {
+        MP_ERR(ao, "error encoding at %d %d/%d\n",
+               frame ? (int) frame->pts : -1,
+               ac->codec->time_base.num,
+               ac->codec->time_base.den);
+        return;
+    }
+    for (;;) {
+        av_init_packet(&packet);
+        status = avcodec_receive_packet(ac->codec, &packet);
+        if (status == AVERROR(EAGAIN)) { // No more packets for now.
+            if (frame == NULL) {
+                MP_ERR(ao, "sent flush frame, got EAGAIN");
+            }
+            break;
+        }
+        if (status == AVERROR_EOF) { // No more packets, ever.
+            if (frame != NULL) {
+                MP_ERR(ao, "sent audio frame, got EOF");
+            }
+            break;
+        }
+        if (status < 0) {
+            MP_ERR(ao, "error encoding at %d %d/%d\n",
+                   frame ? (int) frame->pts : -1,
+                   ac->codec->time_base.num,
+                   ac->codec->time_base.den);
+            break;
+        }
+        if (frame) {
+            if (ac->savepts == AV_NOPTS_VALUE)
+                ac->savepts = frame->pts;
+        }
+        encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec);
+        write_packet(ao, &packet);
+        av_packet_unref(&packet);
+    }
+#else
+    av_init_packet(&packet);
+    int got_packet = 0;
+    int status = avcodec_encode_audio2(ac->codec, &packet, frame, &got_packet);
+    if (status < 0) {
+        MP_ERR(ao, "error encoding at %d %d/%d\n",
+               frame ? (int) frame->pts : -1,
+               ac->codec->time_base.num,
+               ac->codec->time_base.den);
+        return;
+    }
+    if (!got_packet) {
+        return;
+    }
+    if (frame) {
+        if (ac->savepts == AV_NOPTS_VALUE)
+            ac->savepts = frame->pts;
+    }
+    encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec);
+    write_packet(ao, &packet);
+    av_packet_unref(&packet);
+#endif
+}
+
 // must get exactly ac->aframesize amount of data
-static int encode(struct ao *ao, double apts, void **data)
+static void encode(struct ao *ao, double apts, void **data)
 {
-    AVPacket packet;
     struct priv *ac = ao->priv;
     struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
     double realapts = ac->aframecount * (double) ac->aframesize /
                       ao->samplerate;
-    int status, gotpacket;
 
     ac->aframecount++;
 
     if (data)
         ectx->audio_pts_offset = realapts - apts;
 
-    av_init_packet(&packet);
-    packet.data = ac->buffer;
-    packet.size = ac->buffer_size;
     if(data) {
         AVFrame *frame = av_frame_alloc();
         frame->format = af_to_avformat(ao->format);
@@ -270,64 +366,11 @@ static int encode(struct ao *ao, double apts, void **data)
         ac->lastpts = frame_pts;
 
         frame->quality = ac->codec->global_quality;
-        status = avcodec_encode_audio2(ac->codec, &packet, frame, &gotpacket);
-
-        if (!status) {
-            if (ac->savepts == AV_NOPTS_VALUE)
-                ac->savepts = frame->pts;
-        }
-
+        encode_audio_and_write(ao, frame);
         av_frame_free(&frame);
     }
     else
-    {
-        status = avcodec_encode_audio2(ac->codec, &packet, NULL, &gotpacket);
-    }
-
-    if(status) {
-        MP_ERR(ao, "error encoding\n");
-        return -1;
-    }
-
-    if(!gotpacket)
-        return 0;
-
-    MP_DBG(ao, "got pts %f (playback time: %f); out size: %d\n",
-           apts, realapts, packet.size);
-
-    encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec);
-
-    packet.stream_index = ac->stream->index;
-
-    // Do we need this at all? Better be safe than sorry...
-    if (packet.pts == AV_NOPTS_VALUE) {
-        MP_WARN(ao, "encoder lost pts, why?\n");
-        if (ac->savepts != MP_NOPTS_VALUE)
-            packet.pts = ac->savepts;
-    }
-
-    if (packet.pts != AV_NOPTS_VALUE)
-        packet.pts = av_rescale_q(packet.pts, ac->codec->time_base,
-                ac->stream->time_base);
-
-    if (packet.dts != AV_NOPTS_VALUE)
-        packet.dts = av_rescale_q(packet.dts, ac->codec->time_base,
-                ac->stream->time_base);
-
-    if(packet.duration > 0)
-        packet.duration = av_rescale_q(packet.duration, ac->codec->time_base,
-                ac->stream->time_base);
-
-    ac->savepts = AV_NOPTS_VALUE;
-
-    if (encode_lavc_write_frame(ao->encode_lavc_ctx, ac->stream, &packet) < 0) {
-        MP_ERR(ao, "error writing at %f %f/%f\n",
-               realapts, (double) ac->stream->time_base.num,
-               (double) ac->stream->time_base.den);
-        return -1;
-    }
-
-    return packet.size;
+        encode_audio_and_write(ao, NULL);
 }
 
 // this should round samples down to frame sizes
@@ -492,3 +535,5 @@ const struct ao_driver audio_out_lavc = {
     .play      = play,
     .drain     = drain,
 };
+
+// vim: sw=4 ts=4 et tw=80