[streaming/xcode] configurable MP3 streaming bitrate

forked-daapd.conf.in

@@ -367,3 +367,16 @@ sqlite {
 	# but may reduce database size). Default is yes.
 #	vacuum = yes
 }
+
+
+# Streaming audio settings for remote connections (ie stream.mp3)
+streaming {
+	# Sample rate, typically 44100 or 48000
+#	sample_rate = 44100
+ 
+	# Channels
+#	channels = 2
+ 
+	# Set the MP3 streaming bit rate (in kbps), valid options: 64 / 96 / 128 / 192 / 320
+#	bit_rate = 192
+}

src/conffile.c

@@ -60,6 +60,7 @@ static cfg_opt_t sec_general[] =
 #else
     CFG_BOOL("high_resolution_clock", cfg_true, CFGF_NONE),
 #endif
+    CFG_INT("streaming_bitrate", 192, CFGF_NONE),
     // Hidden options
     CFG_INT("db_pragma_cache_size", -1, CFGF_NONE),
     CFG_STR("db_pragma_journal_mode", NULL, CFGF_NONE),
@@ -186,6 +187,15 @@ static cfg_opt_t sec_mpd[] =
     CFG_END()
   };
 
+/* streaming section structure */
+static cfg_opt_t sec_streaming[] =
+  {
+    CFG_INT("sample_rate", 44100, CFGF_NONE),
+    CFG_INT("channels", 2, CFGF_NONE),
+    CFG_INT("bit_rate", 192, CFGF_NONE),
+    CFG_END()
+  };
+
 /* Config file structure */
 static cfg_opt_t toplvl_cfg[] =
   {
@@ -198,6 +208,7 @@ static cfg_opt_t toplvl_cfg[] =
     CFG_SEC("spotify", sec_spotify, CFGF_NONE),
     CFG_SEC("sqlite", sec_sqlite, CFGF_NONE),
     CFG_SEC("mpd", sec_mpd, CFGF_NONE),
+    CFG_SEC("streaming", sec_streaming, CFGF_NONE),
     CFG_END()
   };
 

src/httpd.c

@@ -1034,7 +1034,7 @@ httpd_request_parse(struct evhttp_request *req, struct httpd_uri_parsed *uri_par
 void
 httpd_stream_file(struct evhttp_request *req, int id)
 {
-  struct media_quality quality = { HTTPD_STREAM_SAMPLE_RATE, HTTPD_STREAM_BPS, HTTPD_STREAM_CHANNELS };
+  struct media_quality quality = { HTTPD_STREAM_SAMPLE_RATE, HTTPD_STREAM_BPS, HTTPD_STREAM_CHANNELS, 0 };
   struct media_file_info *mfi;
   struct stream_ctx *st;
   void (*stream_cb)(int fd, short event, void *arg);

src/httpd_streaming.c

@@ -52,6 +52,8 @@ extern struct event_base *evbase_httpd;
 #define STREAMING_MP3_SAMPLE_RATE 44100
 #define STREAMING_MP3_BPS         16
 #define STREAMING_MP3_CHANNELS    2
+#define STREAMING_MP3_BIT_RATE    192000
+static int streaming_mp3_bit_rate = STREAMING_MP3_BIT_RATE;
 
 
 // Linked list of mp3 streaming requests
@@ -182,7 +184,7 @@ streaming_end(void)
 static void
 streaming_meta_cb(evutil_socket_t fd, short event, void *arg)
 {
-  struct media_quality mp3_quality = { STREAMING_MP3_SAMPLE_RATE, STREAMING_MP3_BPS, STREAMING_MP3_CHANNELS };
+  struct media_quality mp3_quality = { STREAMING_MP3_SAMPLE_RATE, STREAMING_MP3_BPS, STREAMING_MP3_CHANNELS, streaming_mp3_bit_rate };
   struct media_quality quality;
   struct decode_ctx *decode_ctx;
   int ret;
@@ -204,6 +206,9 @@ streaming_meta_cb(evutil_socket_t fd, short event, void *arg)
   if (!decode_ctx)
     goto error;
 
+  if (quality.bit_rate)
+    mp3_quality.bit_rate = quality.bit_rate;
+
   streaming_encode_ctx = transcode_encode_setup(XCODE_MP3, &mp3_quality, decode_ctx, NULL, 0, 0);
   transcode_decode_cleanup(&decode_ctx);
   if (!streaming_encode_ctx)
@@ -618,6 +623,23 @@ streaming_init(void)
 {
   int ret;
 
+  streaming_mp3_bit_rate = cfg_getint(cfg_getsec(cfg, "streaming"), "bit_rate");
+  switch (streaming_mp3_bit_rate)
+  {
+    case  64:
+    case  96:
+    case 128:
+    case 192:
+    case 320:
+      streaming_mp3_bit_rate *= 1000;
+      break;
+
+    default:
+      DPRINTF(E_WARN, L_STREAMING, "streaming bit_rate=%d not 128/192/320, defaulting\n", streaming_mp3_bit_rate);
+      streaming_mp3_bit_rate = STREAMING_MP3_BIT_RATE;
+  }
+  DPRINTF(E_INFO, L_STREAMING, "streaming bit_rate=%d\n", streaming_mp3_bit_rate);
+
   pthread_mutex_init(&streaming_sessions_lck, NULL);
 
   // Non-blocking because otherwise httpd and player thread may deadlock

src/misc.h

@@ -31,6 +31,7 @@ struct media_quality {
   int sample_rate;
   int bits_per_sample;
   int channels;
+  int bit_rate;
 };
 
 struct onekeyval {

src/outputs/alsa.c

@@ -140,7 +140,7 @@ static int alsa_latency_history_size;
 
 // We will try to play the music with the source quality, but if the card
 // doesn't support that we resample to the fallback quality
-static struct media_quality alsa_fallback_quality = { 44100, 16, 2 };
+static struct media_quality alsa_fallback_quality = { 44100, 16, 2, 0 };
 static struct media_quality alsa_last_quality;
 
 

src/outputs/cast.c

@@ -446,7 +446,7 @@ static struct cast_session *cast_sessions;
 static struct cast_master_session *cast_master_session;
 //static struct timeval heartbeat_timeout = { HEARTBEAT_TIMEOUT, 0 };
 static struct timeval reply_timeout = { REPLY_TIMEOUT, 0 };
-static struct media_quality cast_quality_default = { CAST_QUALITY_SAMPLE_RATE_DEFAULT, CAST_QUALITY_BITS_PER_SAMPLE_DEFAULT, CAST_QUALITY_CHANNELS_DEFAULT };
+static struct media_quality cast_quality_default = { CAST_QUALITY_SAMPLE_RATE_DEFAULT, CAST_QUALITY_BITS_PER_SAMPLE_DEFAULT, CAST_QUALITY_CHANNELS_DEFAULT, 0 };
 
 
 /* ------------------------------- MISC HELPERS ----------------------------- */

src/outputs/fifo.c

@@ -61,7 +61,7 @@ struct fifo_buffer
 
 static struct fifo_buffer buffer;
 
-static struct media_quality fifo_quality = { 44100, 16, 2 };
+static struct media_quality fifo_quality = { 44100, 16, 2, 0 };
 
 
 static void

src/outputs/pulse.c

@@ -86,7 +86,7 @@ static struct pulse_session *sessions;
 static uint32_t pulse_known_devices[PULSE_MAX_DEVICES];
 
 static struct media_quality pulse_last_quality;
-static struct media_quality pulse_fallback_quality = { 44100, 16, 2 };
+static struct media_quality pulse_fallback_quality = { 44100, 16, 2, 0 };
 
 // Converts from 0 - 100 to Pulseaudio's scale
 static inline pa_volume_t

src/transcode.c

@@ -79,6 +79,7 @@ struct settings_ctx
   int sample_rate;
   uint64_t channel_layout;
   int channels;
+  int bit_rate;
   enum AVSampleFormat sample_format;
   bool wavheader;
   bool icy;
@@ -262,6 +263,11 @@ init_settings(struct settings_ctx *settings, enum transcode_profile profile, str
       settings->channel_layout = av_get_default_channel_layout(quality->channels);
     }
 
+  if (quality && quality->bit_rate)
+    {
+      settings->bit_rate    = quality->bit_rate;
+    }
+
   if (quality && quality->bits_per_sample && (quality->bits_per_sample != 8 * av_get_bytes_per_sample(settings->sample_format)))
     {
       DPRINTF(E_LOG, L_XCODE, "Bug! Mismatch between profile and media quality\n");
@@ -281,6 +287,7 @@ stream_settings_set(struct stream_ctx *s, struct settings_ctx *settings, enum AV
       s->codec->channels       = settings->channels;
       s->codec->sample_fmt     = settings->sample_format;
       s->codec->time_base      = (AVRational){1, settings->sample_rate};
+      s->codec->bit_rate       = settings->bit_rate;
     }
   else if (type == AVMEDIA_TYPE_VIDEO)
     {
@@ -731,7 +738,7 @@ open_decoder(unsigned int *stream_index, struct decode_ctx *ctx, enum AVMediaTyp
   if ((*stream_index < 0) || (!decoder))
     {
       if (!ctx->settings.silent)
-	DPRINTF(E_LOG, L_XCODE, "No stream data or decoder for stream #%d\n", *stream_index);
+       DPRINTF(E_LOG, L_XCODE, "No stream data or decoder for stream #%d\n", *stream_index);
       return NULL;
     }