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Diffstat (limited to 'audio/out/ao_lavc.c')
-rw-r--r--audio/out/ao_lavc.c74
1 files changed, 30 insertions, 44 deletions
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index 0631d54aea..bf42a75ca3 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -71,13 +71,7 @@ static int init(struct ao *ao)
AVCodec *codec;
if (!encode_lavc_available(ao->encode_lavc_ctx)) {
- mp_msg(MSGT_ENCODE, MSGL_ERR,
- "ao-lavc: the option --o (output file) must be specified\n");
- return -1;
- }
-
- if (ac->stream) {
- mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: rejecting reinitialization\n");
+ MP_ERR(ao, "the option --o (output file) must be specified\n");
return -1;
}
@@ -85,7 +79,7 @@ static int init(struct ao *ao)
AVMEDIA_TYPE_AUDIO);
if (!ac->stream) {
- mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: could not get a new audio stream\n");
+ MP_ERR(ao, "could not get a new audio stream\n");
return -1;
}
@@ -243,8 +237,7 @@ out_takefirst:
}
if (!found_format && !found_planar_format) {
// shouldn't happen
- mp_msg(MSGT_ENCODE, MSGL_ERR,
- "ao-lavc: sample format not found\n");
+ MP_ERR(ao, "sample format not found\n");
}
}
@@ -285,8 +278,7 @@ out_takefirst:
ao->priv = ac;
if (ac->planarize)
- mp_msg(MSGT_ENCODE, MSGL_WARN,
- "ao-lavc: need to planarize audio data\n");
+ MP_WARN(ao, "need to planarize audio data\n");
return 0;
}
@@ -311,8 +303,7 @@ static void uninit(struct ao *ao, bool cut_audio)
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
if (!encode_lavc_start(ectx)) {
- mp_msg(MSGT_ENCODE, MSGL_WARN,
- "ao-lavc: not even ready to encode audio at end -> dropped");
+ MP_WARN(ao, "not even ready to encode audio at end -> dropped");
return;
}
@@ -329,8 +320,7 @@ static void uninit(struct ao *ao, bool cut_audio)
ac->sample_size, ac->sample_padding);
int written = play(ao, paddingbuf, ao->buffer.len + extralen, 0);
if (written < ao->buffer.len) {
- mp_msg(MSGT_ENCODE, MSGL_ERR,
- "ao-lavc: did not write enough data at the end\n");
+ MP_ERR(ao, "did not write enough data at the end\n");
}
talloc_free(paddingbuf);
ao->buffer.len = 0;
@@ -392,7 +382,7 @@ static int encode(struct ao *ao, double apts, void *data)
ac->stream->codec->sample_fmt, data,
audiolen, 1))
{
- mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: error filling\n");
+ MP_ERR(ao, "error filling\n");
return -1;
}
@@ -408,9 +398,8 @@ static int encode(struct ao *ao, double apts, void *data)
if (ac->lastpts != MP_NOPTS_VALUE && frame_pts <= ac->lastpts) {
// this indicates broken video
// (video pts failing to increase fast enough to match audio)
- mp_msg(MSGT_ENCODE, MSGL_WARN, "ao-lavc: audio frame pts went backwards "
- "(%d <- %d), autofixed\n", (int)frame->pts,
- (int)ac->lastpts);
+ MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n",
+ (int)frame->pts, (int)ac->lastpts);
frame_pts = ac->lastpts + 1;
frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, ac->stream->codec->time_base);
}
@@ -436,17 +425,15 @@ static int encode(struct ao *ao, double apts, void *data)
status = avcodec_encode_audio2(ac->stream->codec, &packet, NULL, &gotpacket);
}
- if(status)
- {
- mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: error encoding\n");
+ if(status) {
+ MP_ERR(ao, "error encoding\n");
return -1;
}
if(!gotpacket)
return 0;
- mp_msg(MSGT_ENCODE, MSGL_DBG2,
- "ao-lavc: got pts %f (playback time: %f); out size: %d\n",
+ 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->stream);
@@ -455,7 +442,7 @@ static int encode(struct ao *ao, double apts, void *data)
// Do we need this at all? Better be safe than sorry...
if (packet.pts == AV_NOPTS_VALUE) {
- mp_msg(MSGT_ENCODE, MSGL_WARN, "ao-lavc: encoder lost pts, why?\n");
+ MP_WARN(ao, "encoder lost pts, why?\n");
if (ac->savepts != MP_NOPTS_VALUE)
packet.pts = ac->savepts;
}
@@ -475,7 +462,7 @@ static int encode(struct ao *ao, double apts, void *data)
ac->savepts = MP_NOPTS_VALUE;
if (encode_lavc_write_frame(ao->encode_lavc_ctx, &packet) < 0) {
- mp_msg(MSGT_ENCODE, MSGL_ERR, "ao-lavc: error writing at %f %f/%f\n",
+ 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;
@@ -501,13 +488,11 @@ static int play(struct ao *ao, void *data, int len, int flags)
len /= ac->sample_size * ao->channels.num;
if (!encode_lavc_start(ectx)) {
- mp_msg(MSGT_ENCODE, MSGL_WARN,
- "ao-lavc: not ready yet for encoding audio\n");
+ MP_WARN(ao, "not ready yet for encoding audio\n");
return 0;
}
if (pts == MP_NOPTS_VALUE) {
- mp_msg(MSGT_ENCODE, MSGL_WARN,
- "ao-lavc: frame without pts, please report; synthesizing pts instead\n");
+ MP_WARN(ao, "frame without pts, please report; synthesizing pts instead\n");
// synthesize pts from previous expected next pts
pts = ac->expected_next_pts;
}
@@ -516,19 +501,21 @@ static int play(struct ao *ao, void *data, int len, int flags)
//if (ac->stream->codec->time_base.num / ac->stream->codec->time_base.den >= ac->stream->time_base.num / ac->stream->time_base.den)
if (ac->stream->codec->time_base.num * (double) ac->stream->time_base.den >=
ac->stream->time_base.num * (double) ac->stream->codec->time_base.den) {
- mp_msg(MSGT_ENCODE, MSGL_V, "ao-lavc: NOTE: using codec time base "
- "(%d/%d) for pts adjustment; the stream base (%d/%d) is "
- "not worse.\n", (int)ac->stream->codec->time_base.num,
- (int)ac->stream->codec->time_base.den, (int)ac->stream->time_base.num,
- (int)ac->stream->time_base.den);
+ MP_VERBOSE(ao, "NOTE: using codec time base (%d/%d) for pts "
+ "adjustment; the stream base (%d/%d) is not worse.\n",
+ (int)ac->stream->codec->time_base.num,
+ (int)ac->stream->codec->time_base.den,
+ (int)ac->stream->time_base.num,
+ (int)ac->stream->time_base.den);
ac->worst_time_base = ac->stream->codec->time_base;
ac->worst_time_base_is_stream = 0;
} else {
- mp_msg(MSGT_ENCODE, MSGL_WARN, "ao-lavc: NOTE: not using codec time "
- "base (%d/%d) for pts adjustment; the stream base (%d/%d) "
- "is worse.\n", (int)ac->stream->codec->time_base.num,
- (int)ac->stream->codec->time_base.den, (int)ac->stream->time_base.num,
- (int)ac->stream->time_base.den);
+ MP_WARN(ao, "NOTE: not using codec time base (%d/%d) for pts "
+ "adjustment; the stream base (%d/%d) is worse.\n",
+ (int)ac->stream->codec->time_base.num,
+ (int)ac->stream->codec->time_base.den,
+ (int)ac->stream->time_base.num,
+ (int)ac->stream->time_base.den);
ac->worst_time_base = ac->stream->time_base;
ac->worst_time_base_is_stream = 1;
}
@@ -596,7 +583,7 @@ static int play(struct ao *ao, void *data, int len, int flags)
*/
int finalbufpos = len - (len - bufpos) % ac->aframesize;
if (finalbufpos < 0) {
- mp_msg(MSGT_ENCODE, MSGL_WARN, "ao-lavc: cannot attain the "
+ MP_WARN(ao, "cannot attain the "
"exact requested audio sync; shifting by %d frames\n",
-finalbufpos);
bufpos -= finalbufpos;
@@ -611,8 +598,7 @@ static int play(struct ao *ao, void *data, int len, int flags)
ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
}
else if (fabs(nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts) > 30) {
- mp_msg(MSGT_ENCODE, MSGL_WARN,
- "ao-lavc: detected an unexpected discontinuity (pts jumped by "
+ MP_WARN(ao, "detected an unexpected discontinuity (pts jumped by "
"%f seconds)\n",
nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts);
ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-16 19:50:26 +0000