FFmpeg ADPCM 디코더를 사용하여 오디오 디코딩

mp4serv-use-ffmpeg-adpcm-decoder.patch

From c69c805fc1194e225c3f2ddd2c7429f827d3684e Mon Sep 17 00:00:00 2001
From: InCheol Moon <moonpfe@emstone.com>
Date: Thu, 23 Aug 2018 14:03:55 +0900
Subject: [PATCH] mp4serv: use ffmpeg adpcm decoder

---
 src/mp4serv.c | 84 ++++++++++++++++++++-------------------------------
 1 file changed, 32 insertions(+), 52 deletions(-)

diff --git a/src/mp4serv.c b/src/mp4serv.c
index 4b84abc30..fcff66c59 100644
--- a/src/mp4serv.c
+++ b/src/mp4serv.c
@@ -123,17 +123,13 @@ struct _Mp4ServClient
 
   struct
   {
-    EdcAdpcm        *decoder;
+    AVCodecContext  *decoder;
     AVFilterGraph   *graph;
     AVFilterContext *src;
     AVFilterContext *sink;
     AVCodecContext  *encoder;
     AVStream        *stream;
 
-    char            *buf;
-    int              buf_len;
-    int              buf_alloc;
-
     int        input_sample_fmt;
     int        input_channel_layout;
     AVRational input_time_base;
@@ -558,12 +554,18 @@ mp4serv_init_audio (Mp4ServClient     *client,
       return;
     }
 
-  client->audio.decoder = edc_adpcm_new ();
-  client->audio.buf_alloc = 1920 * 2;
-  client->audio.buf = g_malloc (client->audio.buf_alloc + AV_INPUT_BUFFER_PADDING_SIZE);
-  client->audio.buf_len = 0;
-  client->audio.input_sample_fmt = in_par->format;
-  client->audio.input_channel_layout = in_par->channel_layout;
+  client->audio.decoder = avcodec_alloc_context3 (decoder);
+  avcodec_parameters_to_context (client->audio.decoder, in_par);
+
+  ret = avcodec_open2 (client->audio.decoder, decoder, NULL);
+  if (ret < 0)
+    {
+      mp4serv_debug (client, "failed to open audio decoder: '%s'(%d)",
+                     av_err2str (ret), ret);
+      return;
+    }
+
+  avcodec_parameters_from_context (in_par, client->audio.decoder);
 
   ret = mp4serv_init_audio_resampling_filter (client, in_par);
   if (ret < 0)
@@ -579,16 +581,11 @@ mp4serv_deinit_audio (Mp4ServClient *client)
 {
   if (client->audio.decoder)
     {
-      edc_adpcm_unref (client->audio.decoder);
+      avcodec_free_context (&client->audio.decoder);
       client->audio.decoder = NULL;
     }
   if (client->audio.graph)
     avfilter_graph_free (&client->audio.graph);
-
-  g_free (client->audio.buf);
-  client->audio.buf = NULL;
-  client->audio.buf_alloc = 0;
-  client->audio.buf_len = 0;
 }
 
 static void
@@ -1345,45 +1342,29 @@ mp4serv_transcode (Mp4ServClient    *client,
                       av_ts2str (packet->duration),
                       av_ts2str (packet->pos),
                       packet->size);
-      do
+
+      ret = avcodec_send_packet (client->audio.decoder, packet);
+      if (ret < 0 && ret != AVERROR (EAGAIN) && ret != AVERROR_EOF)
+        mp4serv_debug (client, "failed to send audio packet to decoder: '%s'(%d)",
+                      av_err2str (ret), ret);
+
+      while (TRUE)
         {
-          AVFrame *frame;
-          int pcm_len;
-          int dec_size;
           AVRational time_base = { 1, client->audio.sample_rate };
+          AVFrame *frame;
 
-          dec_size = (packet->size - 4) * 4;
-          if (dec_size > client->audio.buf_alloc)
-            {
-              client->audio.buf_alloc = dec_size + AV_INPUT_BUFFER_PADDING_SIZE;
-              client->audio.buf = g_realloc (client->audio.buf,
-                                            client->audio.buf_alloc);
-            }
+          frame = av_frame_alloc ();
 
-          pcm_len = edc_adpcm_decode (client->audio.decoder,
-                                      (const uint8_t *) packet->data,
-                                      (int16_t *) client->audio.buf,
-                                      packet->size);
-          if (pcm_len <= 0)
-            break;
+          ret = avcodec_receive_frame (client->audio.decoder, frame);
+          if (ret < 0)
+            {
+              if (0 && ret != AVERROR (EAGAIN) && ret != AVERROR_EOF)
+                mp4serv_debug (client, "failed to get decoded audio frame: '%s'(%d)",
+                              av_err2str (ret), ret);
 
-          frame = av_frame_alloc ();
-          frame->pts = packet->pts;
-          frame->pkt_dts = frame->pts;
-          frame->channel_layout = client->audio.input_channel_layout;
-          frame->channels = av_get_channel_layout_nb_channels (frame->channel_layout);
-          frame->sample_rate = client->audio.sample_rate;
-          frame->format = client->audio.input_sample_fmt;
-          frame->nb_samples = pcm_len
-                            / av_get_bytes_per_sample (frame->format)
-                            / frame->channels;
-
-          avcodec_fill_audio_frame (frame,
-                                    frame->channels,
-                                    frame->format,
-                                    (const uint8_t *) client->audio.buf,
-                                    client->audio.buf_alloc,
-                                    0);
+              av_frame_free (&frame);
+              break;
+            }
 
           frame->pts = av_rescale_delta (client->audio.input_time_base,
                                          frame->pts,
@@ -1413,7 +1394,6 @@ mp4serv_transcode (Mp4ServClient    *client,
 
           av_frame_free (&frame);
         }
-      while (0);
 
       while (TRUE)
         {
-- 
2.18.0