audio: introduce a new type to hold audio frames

This is pretty pointless, but I believe it allows us to claim that the
new code is not affected by the copyright of the old code. This is
needed, because the original mp_audio struct was written by someone who
has disagreed with LGPL relicensing (it was called af_data at the time,
and was defined in af.h).

The "GPL'ed" struct contents that surive are pretty trivial: just the
data pointer, and some metadata like the format, samplerate, etc. - but
at least in this case, any new code would be extremely similar anyway,
and I'm not really sure whether it's OK to claim different copyright. So
what we do is we just use AVFrame (which of course is LGPL with 100%
certainty), and add some accessors around it to adapt it to mpv
conventions.

Also, this gets rid of some annoying conventions of mp_audio, like the
struct fields that require using an accessor to write to them anyway.

For the most part, this change is only dumb replacements of mp_audio
related functions and fields. One minor actual change is that you can't
allocate the new type on the stack anymore.

Some code still uses mp_audio. All audio filter code will be deleted, so
it makes no sense to convert this code. (Audio filters which are LGPL
and which we keep will have to be ported to a new filter infrastructure
anyway.) player/audio.c uses it because it interacts with the old filter
code. push.c has some complex use of mp_audio and mp_audio_buffer, but
this and pull.c will most likely be rewritten to do something else.

5 files changed, 59 insertions, 59 deletions

diff --git a/audio/decode/ad.h b/audio/decode/ad.h
index 0af05e1827..a8384c277f 100644
--- a/audio/decode/ad.h
+++ b/audio/decode/ad.h
@@ -23,7 +23,7 @@
 #include "demux/demux.h"
 
 #include "audio/format.h"
-#include "audio/audio.h"
+#include "audio/aframe.h"
 #include "dec_audio.h"
 
 struct mp_decoder_list;
@@ -39,7 +39,7 @@ struct ad_functions {
     bool (*send_packet)(struct dec_audio *da, struct demux_packet *pkt);
     // Return whether decoding is still going on (false if EOF was reached).
     // Never returns false & *out set, but can return true with !*out.
-    bool (*receive_frame)(struct dec_audio *da, struct mp_audio **out);
+    bool (*receive_frame)(struct dec_audio *da, struct mp_aframe **out);
 };
 
 enum ad_ctrl {
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index d701630fc6..fb429d567b 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -40,7 +40,6 @@
 struct priv {
     AVCodecContext *avctx;
     AVFrame *avframe;
-    struct mp_audio frame;
     bool force_channel_map;
     uint32_t skip_samples, trim_samples;
     bool preroll_done;
@@ -191,7 +190,7 @@ static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
     return true;
 }
 
-static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
+static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
 {
     struct priv *priv = da->priv;
     AVCodecContext *avctx = priv->avctx;
@@ -217,25 +216,18 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
 
     double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase);
 
-    struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
+    struct mp_aframe *mpframe = mp_aframe_from_avframe(priv->avframe);
     if (!mpframe)
         return true;
 
-    struct mp_chmap lavc_chmap = mpframe->channels;
-    if (lavc_chmap.num != avctx->channels)
-        mp_chmap_from_channels(&lavc_chmap, avctx->channels);
-    if (priv->force_channel_map) {
-        if (lavc_chmap.num == da->codec->channels.num)
-            lavc_chmap = da->codec->channels;
-    }
-    mp_audio_set_channels(mpframe, &lavc_chmap);
+    if (priv->force_channel_map)
+        mp_aframe_set_chmap(mpframe, &da->codec->channels);
 
-    mpframe->pts = out_pts;
+    if (out_pts == MP_NOPTS_VALUE)
+        out_pts = priv->next_pts;
+    mp_aframe_set_pts(mpframe, out_pts);
 
-    if (mpframe->pts == MP_NOPTS_VALUE)
-        mpframe->pts = priv->next_pts;
-    if (mpframe->pts != MP_NOPTS_VALUE)
-        priv->next_pts = mpframe->pts + mpframe->samples / (double)mpframe->rate;
+    priv->next_pts = mp_aframe_end_pts(mpframe);
 
 #if LIBAVCODEC_VERSION_MICRO >= 100
     AVFrameSideData *sd =
@@ -254,14 +246,14 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
         priv->preroll_done = true;
     }
 
-    uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
+    uint32_t skip = MPMIN(priv->skip_samples, mp_aframe_get_size(mpframe));
     if (skip) {
-        mp_audio_skip_samples(mpframe, skip);
+        mp_aframe_skip_samples(mpframe, skip);
         priv->skip_samples -= skip;
     }
-    uint32_t trim = MPMIN(priv->trim_samples, mpframe->samples);
+    uint32_t trim = MPMIN(priv->trim_samples, mp_aframe_get_size(mpframe));
     if (trim) {
-        mpframe->samples -= trim;
+        mp_aframe_set_size(mpframe, mp_aframe_get_size(mpframe) - trim);
         priv->trim_samples -= trim;
     }
 
@@ -269,7 +261,6 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
 
     av_frame_unref(priv->avframe);
 
-    MP_DBG(da, "Decoded %d samples\n", mpframe->samples);
     return true;
 }
 
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 4b3e8149ec..0ca20e5485 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -40,8 +40,9 @@ struct spdifContext {
     uint8_t          out_buffer[OUTBUF_SIZE];
     bool             need_close;
     bool             use_dts_hd;
-    struct mp_audio  fmt;
-    struct mp_audio_pool *pool;
+    struct mp_aframe *fmt;
+    int              sstride;
+    struct mp_aframe_pool *pool;
     bool             got_eof;
     struct demux_packet *queued_packet;
 };
@@ -84,7 +85,7 @@ static int init(struct dec_audio *da, const char *decoder)
     da->priv = spdif_ctx;
     spdif_ctx->log = da->log;
     spdif_ctx->use_dts_hd = da->opts->dtshd;
-    spdif_ctx->pool = mp_audio_pool_create(spdif_ctx);
+    spdif_ctx->pool = mp_aframe_pool_create(spdif_ctx);
 
     if (strcmp(decoder, "spdif_dts_hd") == 0)
         spdif_ctx->use_dts_hd = true;
@@ -198,6 +199,9 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
 
     AVDictionary *format_opts = NULL;
 
+    spdif_ctx->fmt = mp_aframe_create();
+    talloc_steal(spdif_ctx, spdif_ctx->fmt);
+
     int num_channels = 0;
     int sample_format = 0;
     int samplerate = 0;
@@ -246,9 +250,14 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
     default:
         abort();
     }
-    mp_audio_set_num_channels(&spdif_ctx->fmt, num_channels);
-    mp_audio_set_format(&spdif_ctx->fmt, sample_format);
-    spdif_ctx->fmt.rate = samplerate;
+
+    struct mp_chmap chmap;
+    mp_chmap_from_channels(&chmap, num_channels);
+    mp_aframe_set_chmap(spdif_ctx->fmt, &chmap);
+    mp_aframe_set_format(spdif_ctx->fmt, sample_format);
+    mp_aframe_set_rate(spdif_ctx->fmt, samplerate);
+
+    spdif_ctx->sstride = mp_aframe_get_sstride(spdif_ctx->fmt);
 
     if (avformat_write_header(lavf_ctx, &format_opts) < 0) {
         MP_FATAL(da, "libavformat spdif initialization failed.\n");
@@ -279,7 +288,7 @@ static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
     return true;
 }
 
-static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
+static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
 {
     struct spdifContext *spdif_ctx = da->priv;
 
@@ -308,13 +317,21 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
         goto done;
     }
 
-    int samples = spdif_ctx->out_buffer_len / spdif_ctx->fmt.sstride;
-    *out = mp_audio_pool_get(spdif_ctx->pool, &spdif_ctx->fmt, samples);
-    if (!*out)
+    *out = mp_aframe_new_ref(spdif_ctx->fmt);
+    int samples = spdif_ctx->out_buffer_len / spdif_ctx->sstride;
+    if (mp_aframe_pool_allocate(spdif_ctx->pool, *out, samples) < 0) {
+        TA_FREEP(out);
         goto done;
+    }
+
+    uint8_t **data = mp_aframe_get_data_rw(*out);
+    if (!data) {
+        TA_FREEP(out);
+        goto done;
+    }
 
-    memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
-    (*out)->pts = pts;
+    memcpy(data[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
+    mp_aframe_set_pts(*out, pts);
 
 done:
     talloc_free(spdif_ctx->queued_packet);
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index 1351cb8ecd..401e26fb7b 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -38,8 +38,6 @@
 #include "dec_audio.h"
 #include "ad.h"
 #include "audio/format.h"
-#include "audio/audio.h"
-#include "audio/audio_buffer.h"
 
 #include "audio/filter/af.h"
 
@@ -179,25 +177,24 @@ static void fix_audio_pts(struct dec_audio *da)
     if (!da->current_frame)
         return;
 
-    if (da->current_frame->pts != MP_NOPTS_VALUE) {
-        double newpts = da->current_frame->pts;
-
+    double frame_pts = mp_aframe_get_pts(da->current_frame);
+    if (frame_pts != MP_NOPTS_VALUE) {
         if (da->pts != MP_NOPTS_VALUE)
-            MP_STATS(da, "value %f audio-pts-err", da->pts - newpts);
+            MP_STATS(da, "value %f audio-pts-err", da->pts - frame_pts);
 
         // Keep the interpolated timestamp if it doesn't deviate more
         // than 1 ms from the real one. (MKV rounded timestamps.)
-        if (da->pts == MP_NOPTS_VALUE || fabs(da->pts - newpts) > 0.001)
-            da->pts = newpts;
+        if (da->pts == MP_NOPTS_VALUE || fabs(da->pts - frame_pts) > 0.001)
+            da->pts = frame_pts;
     }
 
     if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
         da->pts = 0;
 
-    da->current_frame->pts = da->pts;
+    mp_aframe_set_pts(da->current_frame, da->pts);
 
     if (da->pts != MP_NOPTS_VALUE)
-        da->pts += da->current_frame->samples / (double)da->current_frame->rate;
+        da->pts += mp_aframe_duration(da->current_frame);
 }
 
 void audio_work(struct dec_audio *da)
@@ -228,11 +225,6 @@ void audio_work(struct dec_audio *da)
 
     bool progress = da->ad_driver->receive_frame(da, &da->current_frame);
 
-    if (da->current_frame && !mp_audio_config_valid(da->current_frame)) {
-        talloc_free(da->current_frame);
-        da->current_frame = NULL;
-    }
-
     da->current_state = da->current_frame ? DATA_OK : DATA_AGAIN;
     if (!progress)
         da->current_state = DATA_EOF;
@@ -242,10 +234,11 @@ void audio_work(struct dec_audio *da)
     bool segment_end = da->current_state == DATA_EOF;
 
     if (da->current_frame) {
-        mp_audio_clip_timestamps(da->current_frame, da->start, da->end);
-        if (da->current_frame->pts != MP_NOPTS_VALUE && da->start != MP_NOPTS_VALUE)
-            segment_end = da->current_frame->pts >= da->end;
-        if (da->current_frame->samples == 0) {
+        mp_aframe_clip_timestamps(da->current_frame, da->start, da->end);
+        double frame_pts = mp_aframe_get_pts(da->current_frame);
+        if (frame_pts != MP_NOPTS_VALUE && da->start != MP_NOPTS_VALUE)
+            segment_end = frame_pts >= da->end;
+        if (mp_aframe_get_size(da->current_frame) == 0) {
             talloc_free(da->current_frame);
             da->current_frame = NULL;
         }
@@ -280,7 +273,7 @@ void audio_work(struct dec_audio *da)
 //  DATA_WAIT:  waiting for demuxer; will receive a wakeup signal
 //  DATA_EOF:   end of file, no more frames to be expected
 //  DATA_AGAIN: dropped frame or something similar
-int audio_get_frame(struct dec_audio *da, struct mp_audio **out_frame)
+int audio_get_frame(struct dec_audio *da, struct mp_aframe **out_frame)
 {
     *out_frame = NULL;
     if (da->current_frame) {
diff --git a/audio/decode/dec_audio.h b/audio/decode/dec_audio.h
index 886b617b58..ea504328df 100644
--- a/audio/decode/dec_audio.h
+++ b/audio/decode/dec_audio.h
@@ -19,11 +19,10 @@
 #define MPLAYER_DEC_AUDIO_H
 
 #include "audio/chmap.h"
-#include "audio/audio.h"
+#include "audio/aframe.h"
 #include "demux/demux.h"
 #include "demux/stheader.h"
 
-struct mp_audio_buffer;
 struct mp_decoder_list;
 
 struct dec_audio {
@@ -48,7 +47,7 @@ struct dec_audio {
     double start, end;
     struct demux_packet *packet;
     struct demux_packet *new_segment;
-    struct mp_audio *current_frame;
+    struct mp_aframe *current_frame;
     int current_state;
 };
 
@@ -57,7 +56,7 @@ int audio_init_best_codec(struct dec_audio *d_audio);
 void audio_uninit(struct dec_audio *d_audio);
 
 void audio_work(struct dec_audio *d_audio);
-int audio_get_frame(struct dec_audio *d_audio, struct mp_audio **out_frame);
+int audio_get_frame(struct dec_audio *d_audio, struct mp_aframe **out_frame);
 
 void audio_reset_decoding(struct dec_audio *d_audio);