From c537d5958e8b421ac4f9bef6c8b9e7425a6cf167 Mon Sep 17 00:00:00 2001
From: Michael Sevakis <jethead71@rockbox.org>
Date: Wed, 27 Apr 2011 03:08:23 +0000
Subject: Commit FS#12069 - Playback rework - first stages. Gives as thorough
 as possible a treatment of codec management, track change and metadata logic
 as possible while maintaining fairly narrow focus and not rewriting
 everything all at once. Please see the rockbox-dev mail archive on 2011-04-25
 (Playback engine rework) for a more thorough manifest of what was addressed.
 Plugins and codecs become incompatible.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29785 a1c6a512-1295-4272-9138-f99709370657
---
 apps/codecs/aac.c | 62 ++++++++++++++++++++++++-------------------------------
 1 file changed, 27 insertions(+), 35 deletions(-)

(limited to 'apps/codecs/aac.c')

diff --git a/apps/codecs/aac.c b/apps/codecs/aac.c
index 5638dc49a9..90cf0438ce 100644
--- a/apps/codecs/aac.c
+++ b/apps/codecs/aac.c
@@ -33,7 +33,19 @@ CODEC_HEADER
 #define FAAD_BYTE_BUFFER_SIZE (2048-12)
 
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
+    }
+
+    return CODEC_OK;
+}
+
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
     /* Note that when dealing with QuickTime/MPEG4 files, terminology is
      * a bit confusing. Files with sound are split up in chunks, where
@@ -59,25 +71,15 @@ enum codec_status codec_main(void)
     uint32_t sbr_fac = 1;
     unsigned char c = 0;
     void *ret;
-
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
-
-next_track:
-    err = CODEC_OK;
+    intptr_t param;
 
     /* Clean and initialize decoder structures */
     memset(&demux_res , 0, sizeof(demux_res));
     if (codec_init()) {
         LOGF("FAAD: Codec init error\n");
-        err = CODEC_ERROR;
-        goto exit;
+        return CODEC_ERROR;
     }
 
-    if (codec_wait_taginfo() != 0)
-        goto done;
-  
     file_offset = ci->id3->offset;
 
     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -85,12 +87,13 @@ next_track:
 
     stream_create(&input_stream,ci);
 
+    ci->seek_buffer(ci->id3->first_frame_offset);
+
     /* if qtmovie_read returns successfully, the stream is up to
      * the movie data, which can be used directly by the decoder */
     if (!qtmovie_read(&input_stream, &demux_res)) {
         LOGF("FAAD: File init error\n");
-        err = CODEC_ERROR;
-        goto done;
+        return CODEC_ERROR;
     }
 
     /* initialise the sound converter */
@@ -98,8 +101,7 @@ next_track:
 
     if (!decoder) {
         LOGF("FAAD: Decode open error\n");
-        err = CODEC_ERROR;
-        goto done;
+        return CODEC_ERROR;
     }
 
     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
@@ -109,8 +111,7 @@ next_track:
     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
     if (err) {
         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
-        err = CODEC_ERROR;
-        goto done;
+        return CODEC_ERROR;
     }
 
 #ifdef SBR_DEC
@@ -150,20 +151,19 @@ next_track:
 
     /* The main decoding loop */
     while (i < demux_res.num_sample_byte_sizes) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
 
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
             break;
-        }
 
         /* Deal with any pending seek requests */
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
             /* Seek to the desired time position. Important: When seeking in SBR
              * upsampling files the seek_time must be divided by 2 when calling 
              * m4a_seek and the resulting sound_samples_done must be expanded 
              * by a factor 2. This is done via using sbr_fac. */
             if (m4a_seek(&demux_res, &input_stream,
-                          ((ci->seek_time-1)/10/sbr_fac)*(ci->id3->frequency/100),
+                          (param/10/sbr_fac)*(ci->id3->frequency/100),
                           &sound_samples_done, (int*) &i)) {
                 sound_samples_done *= sbr_fac;
                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
@@ -194,8 +194,7 @@ next_track:
         else if (file_offset == 0)
         {
             LOGF("AAC: get_sample_offset error\n");
-            err = CODEC_ERROR;
-            goto done;
+            return CODEC_ERROR;
         }
         
         /* Request the required number of bytes from the input buffer */
@@ -207,8 +206,7 @@ next_track:
         /* NeAACDecDecode may sometimes return NULL without setting error. */
         if (ret == NULL || frame_info.error > 0) {
             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
-            err = CODEC_ERROR;
-            goto done;
+            return CODEC_ERROR;
         }
 
         /* Advance codec buffer (no need to call set_offset because of this) */
@@ -251,12 +249,6 @@ next_track:
         i++;
     }
 
-done:
     LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
-
-    if (ci->request_next_track())
-        goto next_track;
-
-exit:
-    return err;
+    return CODEC_OK;
 }
-- 
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