audio: move decoder context from sh_audio into new struct

Move all state that basically changes during decoding or is needed in
order to manage decoding itself into a new struct (dec_audio).

sh_audio (defined in stheader.h) is supposed to be the audio stream
header. This should reflect the file headers for the stream. Putting the
decoder context there is strange design, to say the least.

1 files changed, 47 insertions, 38 deletions

diff --git a/mpvcore/player/audio.c b/mpvcore/player/audio.c
index 3bf338c21d..ce161a58d5 100644
--- a/mpvcore/player/audio.c
+++ b/mpvcore/player/audio.c
@@ -41,18 +41,18 @@
 
 static int build_afilter_chain(struct MPContext *mpctx)
 {
-    struct sh_audio *sh_audio = mpctx->sh_audio;
+    struct dec_audio *d_audio = mpctx->d_audio;
     struct ao *ao = mpctx->ao;
     struct MPOpts *opts = mpctx->opts;
 
-    if (!sh_audio->initialized)
+    if (!d_audio)
         return 0;
 
     struct mp_audio in_format;
-    mp_audio_buffer_get_format(mpctx->sh_audio->decode_buffer, &in_format);
+    mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
 
     int new_srate;
-    if (af_control_any_rev(sh_audio->afilter, AF_CONTROL_SET_PLAYBACK_SPEED,
+    if (af_control_any_rev(d_audio->afilter, AF_CONTROL_SET_PLAYBACK_SPEED,
                            &opts->playback_speed))
         new_srate = in_format.rate;
     else {
@@ -66,13 +66,13 @@ static int build_afilter_chain(struct MPContext *mpctx)
             opts->playback_speed = new_srate / (double)in_format.rate;
         }
     }
-    return init_audio_filters(sh_audio, new_srate,
+    return audio_init_filters(d_audio, new_srate,
                               &ao->samplerate, &ao->channels, &ao->format);
 }
 
 static int recreate_audio_filters(struct MPContext *mpctx)
 {
-    assert(mpctx->sh_audio);
+    assert(mpctx->d_audio);
 
     // init audio filters:
     if (!build_afilter_chain(mpctx)) {
@@ -80,19 +80,19 @@ static int recreate_audio_filters(struct MPContext *mpctx)
         return -1;
     }
 
-    mixer_reinit_audio(mpctx->mixer, mpctx->ao, mpctx->sh_audio->afilter);
+    mixer_reinit_audio(mpctx->mixer, mpctx->ao, mpctx->d_audio->afilter);
 
     return 0;
 }
 
 int reinit_audio_filters(struct MPContext *mpctx)
 {
-    struct sh_audio *sh_audio = mpctx->sh_audio;
-    if (!sh_audio)
+    struct dec_audio *d_audio = mpctx->d_audio;
+    if (!d_audio)
         return -2;
 
-    af_uninit(mpctx->sh_audio->afilter);
-    if (af_init(mpctx->sh_audio->afilter) < 0)
+    af_uninit(mpctx->d_audio->afilter);
+    if (af_init(mpctx->d_audio->afilter) < 0)
         return -1;
     if (recreate_audio_filters(mpctx) < 0)
         return -1;
@@ -103,20 +103,25 @@ int reinit_audio_filters(struct MPContext *mpctx)
 void reinit_audio_chain(struct MPContext *mpctx)
 {
     struct MPOpts *opts = mpctx->opts;
-    init_demux_stream(mpctx, STREAM_AUDIO);
-    if (!mpctx->sh_audio) {
+    struct sh_stream *sh = init_demux_stream(mpctx, STREAM_AUDIO);
+    if (!sh) {
         uninit_player(mpctx, INITIALIZED_AO);
         goto no_audio;
     }
 
     if (!(mpctx->initialized_flags & INITIALIZED_ACODEC)) {
-        if (!init_best_audio_codec(mpctx->sh_audio, opts->audio_decoders))
-            goto init_error;
         mpctx->initialized_flags |= INITIALIZED_ACODEC;
+        assert(!mpctx->d_audio);
+        mpctx->d_audio = talloc_zero(NULL, struct dec_audio);
+        mpctx->d_audio->opts = opts;
+        mpctx->d_audio->header = sh;
+        if (!audio_init_best_codec(mpctx->d_audio, opts->audio_decoders))
+            goto init_error;
     }
+    assert(mpctx->d_audio);
 
     struct mp_audio in_format;
-    mp_audio_buffer_get_format(mpctx->sh_audio->decode_buffer, &in_format);
+    mp_audio_buffer_get_format(mpctx->d_audio->decode_buffer, &in_format);
 
     int ao_srate = opts->force_srate;
     int ao_format = opts->audio_output_format;
@@ -137,7 +142,7 @@ void reinit_audio_chain(struct MPContext *mpctx)
     // Determine what the filter chain outputs. build_afilter_chain() also
     // needs this for testing whether playback speed is changed by resampling
     // or using a special filter.
-    if (!init_audio_filters(mpctx->sh_audio,  // preliminary init
+    if (!audio_init_filters(mpctx->d_audio,  // preliminary init
                             // input:
                             in_format.rate,
                             // output:
@@ -188,31 +193,31 @@ no_audio:
 // ao so far.
 double written_audio_pts(struct MPContext *mpctx)
 {
-    sh_audio_t *sh_audio = mpctx->sh_audio;
-    if (!sh_audio || !sh_audio->initialized)
+    struct dec_audio *d_audio = mpctx->d_audio;
+    if (!d_audio)
         return MP_NOPTS_VALUE;
 
     struct mp_audio in_format;
-    mp_audio_buffer_get_format(mpctx->sh_audio->decode_buffer, &in_format);
+    mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
 
     // first calculate the end pts of audio that has been output by decoder
-    double a_pts = sh_audio->pts;
+    double a_pts = d_audio->pts;
     if (a_pts == MP_NOPTS_VALUE)
         return MP_NOPTS_VALUE;
 
-    // sh_audio->pts is the timestamp of the latest input packet with
-    // known pts that the decoder has decoded. sh_audio->pts_bytes is
+    // d_audio->pts is the timestamp of the latest input packet with
+    // known pts that the decoder has decoded. d_audio->pts_bytes is
     // the amount of bytes the decoder has written after that timestamp.
-    a_pts += sh_audio->pts_offset / (double)in_format.rate;
+    a_pts += d_audio->pts_offset / (double)in_format.rate;
 
     // Now a_pts hopefully holds the pts for end of audio from decoder.
     // Subtract data in buffers between decoder and audio out.
 
     // Decoded but not filtered
-    a_pts -= mp_audio_buffer_seconds(sh_audio->decode_buffer);
+    a_pts -= mp_audio_buffer_seconds(d_audio->decode_buffer);
 
     // Data buffered in audio filters, measured in seconds of "missing" output
-    double buffered_output = af_calc_delay(sh_audio->afilter);
+    double buffered_output = af_calc_delay(d_audio->afilter);
 
     // Data that was ready for ao but was buffered because ao didn't fully
     // accept everything to internal buffers yet
@@ -275,11 +280,13 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize)
 {
     struct ao *ao = mpctx->ao;
     struct MPOpts *opts = mpctx->opts;
-    sh_audio_t * const sh_audio = mpctx->sh_audio;
+    struct dec_audio *d_audio = mpctx->d_audio;
     int res;
 
+    assert(d_audio);
+
     // Timing info may not be set without
-    res = decode_audio(sh_audio, ao->buffer, 1);
+    res = audio_decode(d_audio, ao->buffer, 1);
     if (res < 0)
         return res;
 
@@ -300,10 +307,10 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize)
         samples = ptsdiff * real_samplerate;
 
         // ogg demuxers give packets without timing
-        if (written_pts <= 1 && sh_audio->pts == MP_NOPTS_VALUE) {
+        if (written_pts <= 1 && d_audio->pts == MP_NOPTS_VALUE) {
             if (!did_retry) {
                 // Try to read more data to see packets that have pts
-                res = decode_audio(sh_audio, ao->buffer, ao->samplerate);
+                res = audio_decode(d_audio, ao->buffer, ao->samplerate);
                 if (res < 0)
                     return res;
                 did_retry = true;
@@ -321,12 +328,12 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize)
         mpctx->syncing_audio = false;
         int skip_samples = -samples;
         int a = MPMIN(skip_samples, MPMAX(playsize, 2500));
-        res = decode_audio(sh_audio, ao->buffer, a);
+        res = audio_decode(d_audio, ao->buffer, a);
         if (skip_samples <= mp_audio_buffer_samples(ao->buffer)) {
             mp_audio_buffer_skip(ao->buffer, skip_samples);
             if (res < 0)
                 return res;
-            return decode_audio(sh_audio, ao->buffer, playsize);
+            return audio_decode(d_audio, ao->buffer, playsize);
         }
         mp_audio_buffer_clear(ao->buffer);
         if (res < 0)
@@ -346,7 +353,7 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize)
     }
     mpctx->syncing_audio = false;
     mp_audio_buffer_prepend_silence(ao->buffer, samples);
-    return decode_audio(sh_audio, ao->buffer, playsize);
+    return audio_decode(d_audio, ao->buffer, playsize);
 }
 
 int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
@@ -358,9 +365,11 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
     bool audio_eof = false;
     bool signal_eof = false;
     bool partial_fill = false;
-    sh_audio_t * const sh_audio = mpctx->sh_audio;
+    struct dec_audio *d_audio = mpctx->d_audio;
     bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK);
 
+    assert(d_audio);
+
     if (mpctx->paused)
         playsize = 1;   // just initialize things (audio pts at least)
     else
@@ -378,7 +387,7 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
     if (mpctx->syncing_audio || mpctx->hrseek_active)
         res = audio_start_sync(mpctx, playsize);
     else
-        res = decode_audio(sh_audio, ao->buffer, playsize);
+        res = audio_decode(d_audio, ao->buffer, playsize);
 
     if (res < 0) {  // EOF, error or format change
         if (res == -2) {
@@ -392,7 +401,7 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
             return -1;
         } else if (res == ASYNC_PLAY_DONE)
             return 0;
-        else if (demux_stream_eof(mpctx->sh_audio->gsh))
+        else if (demux_stream_eof(d_audio->header))
             audio_eof = true;
     }
 
@@ -455,6 +464,6 @@ void clear_audio_output_buffers(struct MPContext *mpctx)
 // Drop decoded data queued for filtering.
 void clear_audio_decode_buffers(struct MPContext *mpctx)
 {
-    if (mpctx->sh_audio && mpctx->sh_audio->decode_buffer)
-        mp_audio_buffer_clear(mpctx->sh_audio->decode_buffer);
+    if (mpctx->d_audio)
+        mp_audio_buffer_clear(mpctx->d_audio->decode_buffer);
 }