audio/out: channel map selection

Make all AOs use what has been introduced in the previous commit.

Note that even AOs which can handle all possible layouts (like ao_null)
use the new functions. This might be important if in the future
ao_select_champ() possibly honors global user options about downmixing
and so on.

audio/chmap.h

@@ -129,6 +129,4 @@ void mp_chmap_print_help(int msgt, int msgl);
 #define mp_chmap_is_waveext mp_chmap_is_lavc
 #define mp_chmap_reorder_to_waveext mp_chmap_reorder_to_lavc
 
-#define mp_chmap_reorder_to_alsa(x) mp_chmap_from_channels_alsa((x), (x)->num)
-
 #endif

audio/out/ao_alsa.c

@@ -330,22 +330,35 @@ static const char *device_channel_layouts[][2] = {
     {"surround41",      "fl-fr-bl-br-lfe"},
     {"surround51",      "fl-fr-bl-br-fc-lfe"},
     {"surround71",      "fl-fr-bl-br-fc-lfe-sl-sr"},
-    {0}
 };
 
-// Find a device that contains exactly all the requested speakers.
-// Set *request to the required channel order.
-static const char *find_device(struct mp_chmap *request)
+#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))
+
+#define NUM_ALSA_CHMAPS ARRAY_LEN(device_channel_layouts)
+
+static const char *select_chmap(struct ao *ao)
 {
-    for (int n = 0; device_channel_layouts[n][0]; n++) {
-        struct mp_chmap map = {0};
-        mp_chmap_from_str(&map, bstr0(device_channel_layouts[n][1]));
-        if (mp_chmap_equals_reordered(&map, request)) {
-            *request = map;
+    struct mp_chmap_sel sel = {0};
+    struct mp_chmap maps[NUM_ALSA_CHMAPS];
+    for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
+        mp_chmap_from_str(&maps[n], bstr0(device_channel_layouts[n][1]));
+        mp_chmap_sel_add_map(&sel, &maps[n]);
+    };
+
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return NULL;
+
+    for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
+        if (mp_chmap_equals(&ao->channels, &maps[n]))
             return device_channel_layouts[n][0];
-        }
     }
-    return NULL;
+
+    char *name = mp_chmap_to_str(&ao->channels);
+    mp_tmsg(MSGT_AO, MSGL_ERR,
+            "[AO_ALSA] channel layout %s (%d ch) not supported.\n",
+            name, ao->channels.num);
+    talloc_free(name);
+    return "default";
 }
 
 static int try_open_device(struct ao *ao, const char *device, int open_mode,
@@ -449,16 +462,7 @@ static int init(struct ao *ao, char *params)
                "alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n",
                ao->channels.num);
     } else {
-        device.str = find_device(&ao->channels);
-        if (!device.str) {
-            char *name = mp_chmap_to_str(&ao->channels);
-            device.str = "default";
-            mp_chmap_from_channels(&ao->channels, ao->channels.num);
-            mp_tmsg(MSGT_AO, MSGL_ERR,
-                    "[AO_ALSA] channel layout %s (%d ch) not supported.\n",
-                    name, ao->channels.num);
-            talloc_free(name);
-        }
+        device.str = select_chmap(ao);
         if (strcmp(device.str, "default") != 0 && ao->format == AF_FORMAT_FLOAT_NE)
         {
             // hack - use the converter plugin (why the heck?)
@@ -535,8 +539,11 @@ static int init(struct ao *ao, char *params)
             (p->alsa, alsa_hwparams, &num_channels);
     CHECK_ALSA_ERROR("Unable to set channels");
 
-    if (num_channels != ao->channels.num)
-        mp_chmap_from_channels(&ao->channels, num_channels);
+    if (num_channels != ao->channels.num) {
+        mp_tmsg(MSGT_AO, MSGL_ERR,
+                "[AO_ALSA] Couldn't get requested number of channels.\n");
+        mp_chmap_from_channels_alsa(&ao->channels, num_channels);
+    }
 
     /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
         prefer our own resampler, since that allows users to choose the resampler,

audio/out/ao_coreaudio.c

@@ -499,6 +499,11 @@ int device_id, display_help = 0;
     // Save selected device id
     ao->i_selected_dev = devid_def;
 
+    struct mp_chmap_sel chmap_sel = {0};
+    mp_chmap_sel_add_waveext(&chmap_sel);
+    if (!ao_chmap_sel_adjust(&ao_data, &ao_data.channels, &chmap_sel))
+        goto err_out;
+
 	// Build Description for the input format
 	inDesc.mSampleRate=rate;
 	inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
@@ -605,8 +610,8 @@ int device_id, display_help = 0;
 	ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
 
 	ao_data.samplerate = inDesc.mSampleRate;
-        mp_chmap_from_channels(&ao_data.channels, inDesc.mChannelsPerFrame);
-        mp_chmap_reorder_to_waveext(&ao_data.channels);
+        if (!ao_chmap_sel_get_def(&ao_data, &chmap_sel, &ao_data.channels, inDesc.mChannelsPerFrame))
+            goto err_out2;
     ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
     ao_data.outburst = ao->chunk_size;
 	ao_data.buffersize = ao_data.bps;
@@ -838,7 +843,7 @@ static int OpenSPDIF(void)
     ao->chunk_size = ao->stream_format.mBytesPerPacket;
 
     ao_data.samplerate = ao->stream_format.mSampleRate;
-    // Applies default ordering; ok becazse AC3 data is always in mpv internal channel order
+    // Applies default ordering; ok because AC3 data is always in mpv internal channel order
     mp_chmap_from_channels(&ao_data.channels, ao->stream_format.mChannelsPerFrame);
     ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
     ao_data.outburst = ao->chunk_size;

audio/out/ao_dsound.c

@@ -390,18 +390,14 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
 	DSBUFFERDESC dsbpridesc;
 	DSBUFFERDESC dsbdesc;
 
-	//check if the channel count and format is supported in general
-	if (ao_data.channels.num > 8) {
-                // More than 8 channels might just work, but needs testing
-		UninitDirectSound();
-		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: > 8 channel audio not yet supported\n");
-		return 0;
-	}
-
-	if (AF_FORMAT_IS_AC3(format))
+	if (AF_FORMAT_IS_AC3(format)) {
 		format = AF_FORMAT_AC3_NE;
-        else
-                mp_chmap_reorder_to_waveext(&ao_data.channels);
+        } else {
+                struct mp_chmap_sel sel = {0};
+                mp_chmap_sel_add_waveext(&sel);
+                if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
+                        return 0;
+        }
 	switch(format){
 		case AF_FORMAT_AC3_NE:
 		case AF_FORMAT_S24_LE:

audio/out/ao_jack.c

@@ -50,7 +50,7 @@ static const ao_info_t info =
 LIBAO_EXTERN(jack)
 
 //! maximum number of channels supported, avoids lots of mallocs
-#define MAX_CHANS 8
+#define MAX_CHANS MP_NUM_CHANNELS
 static jack_port_t *ports[MAX_CHANS];
 static int num_ports; ///< Number of used ports == number of channels
 static jack_client_t *client;
@@ -223,11 +223,12 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
     print_help();
     return 0;
   }
-  if (ao_data.channels.num > MAX_CHANS) {
-    mp_msg(MSGT_AO, MSGL_FATAL, "[JACK] Invalid number of channels: %i\n",
-           ao_data.channels.num);
+
+  struct mp_chmap_sel sel = {0};
+  mp_chmap_sel_add_waveext(&sel);
+  if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
     goto err_out;
-  }
+
   if (!client_name) {
     client_name = malloc(40);
     sprintf(client_name, "mpv [%d]", getpid());
@@ -283,8 +284,9 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
                  / (float)rate;
   callback_interval = 0;
 
-  mp_chmap_from_channels(&ao_data.channels, num_ports);
-  mp_chmap_reorder_to_alsa(&ao_data.channels); // what order does hack use?
+  if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, num_ports))
+    goto err_out;
+
   ao_data.samplerate = rate;
   ao_data.format = AF_FORMAT_FLOAT_NE;
   ao_data.bps = ao_data.channels.num * rate * sizeof(float);

audio/out/ao_lavc.c

@@ -102,6 +102,10 @@ static int init(struct ao *ao, char *params)
 
     ac->stream->codec->sample_rate = ao->samplerate;
 
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_any(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return -1;
     mp_chmap_reorder_to_lavc(&ao->channels);
     ac->stream->codec->channels = ao->channels.num;
     ac->stream->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);

audio/out/ao_null.c

@@ -48,6 +48,12 @@ static int init(struct ao *ao, char *params)
 {
     struct priv *priv = talloc_zero(ao, struct priv);
     ao->priv = priv;
+
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_any(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return -1;
+
     int samplesize = af_fmt2bits(ao->format) / 8;
     ao->outburst = 256 * ao->channels.num * samplesize;
     // A "buffer" for about 0.2 seconds of audio

audio/out/ao_openal.c

@@ -53,7 +53,7 @@ static const ao_info_t info =
 
 LIBAO_EXTERN(openal)
 
-#define MAX_CHANS 8
+#define MAX_CHANS MP_NUM_CHANNELS
 #define NUM_BUF 128
 #define CHUNK_SIZE 512
 static ALuint buffers[MAX_CHANS][NUM_BUF];
@@ -151,11 +151,11 @@ static int init(int rate, const struct mp_chmap *channels, int format,
     list_devices();
     goto err_out;
   }
-  if (ao_data.channels.num > MAX_CHANS) {
-    mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Invalid number of channels: %i\n",
-           ao_data.channels.num);
+  struct mp_chmap_sel sel = {0};
+  for (i = 0; speaker_pos[i].id != -1; i++)
+    mp_chmap_sel_add_speaker(&sel, speaker_pos[i].id);
+  if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
     goto err_out;
-  }
   struct speaker speakers[MAX_CHANS];
   for (i = 0; i < ao_data.channels.num; i++) {
     speakers[i].id = -1;

audio/out/ao_oss.c

@@ -340,7 +340,10 @@ ac3_retry:
     af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
 
   if(!AF_FORMAT_IS_AC3(format)) {
-    mp_chmap_reorder_to_alsa(&ao_data.channels);
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_alsa_def(&sel);
+    if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
+      return 0;
     int reqchannels = ao_data.channels.num;
     // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
     if (reqchannels > 2) {
@@ -357,7 +360,8 @@ ac3_retry:
 	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", reqchannels);
 	return 0;
       }
-      mp_chmap_from_channels(&ao_data.channels, c + 1);
+      if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, c + 1))
+        return 0;
     }
     mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels.num, reqchannels);
     // set rate

audio/out/ao_pcm.c

@@ -146,7 +146,10 @@ static int init(struct ao *ao, char *params)
         }
     }
 
-    mp_chmap_reorder_to_waveext(&ao->channels);
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_waveext(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        return -1;
 
     ao->outburst = 65536;
     ao->bps = ao->channels.num * ao->samplerate * (af_fmt2bits(ao->format) / 8);

audio/out/ao_portaudio.c

@@ -273,7 +273,11 @@ static int init(struct ao *ao, char *params)
     if (pa_device == paNoDevice)
         goto error_exit;
 
-    mp_chmap_reorder_to_alsa(&ao->channels);
+    // The actual channel order probably depends on the platform.
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_waveext_def(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        goto error_exit;
 
     PaStreamParameters sp = {
         .device = pa_device,

audio/out/ao_pulse.c

@@ -187,6 +187,15 @@ static bool chmap_pa_from_mp(pa_channel_map *dst, struct mp_chmap *src)
     return true;
 }
 
+static bool select_chmap(struct ao *ao, pa_channel_map *dst)
+{
+    struct mp_chmap_sel sel = {0};
+    for (int n = 0; speaker_map[n][1] != -1; n++)
+        mp_chmap_sel_add_speaker(&sel, speaker_map[n][1]);
+    return ao_chmap_sel_adjust(ao, &sel, &ao->channels) &&
+           chmap_pa_from_mp(dst, &ao->channels);
+}
+
 static void uninit(struct ao *ao, bool cut_audio)
 {
     struct priv *priv = ao->priv;
@@ -301,16 +310,8 @@ static int init(struct ao *ao, char *params)
         goto fail;
     }
 
-    if (!chmap_pa_from_mp(&map, &ao->channels)) {
-        char *name = mp_chmap_to_str(&ao->channels);
-        mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Can't map %s channel layout\n",
-               name);
-        talloc_free(name);
-        // Not a really good fallback, since this doesn't trigger if the
-        // channel map is valid, but unsupported by the output device.
-        ao->channels = (struct mp_chmap) MP_CHMAP_INIT_STEREO;
-        pa_channel_map_init_stereo(&map);
-    }
+    if (!select_chmap(ao, &map))
+        goto fail;
 
     ao->bps = pa_bytes_per_second(&ss);
 

audio/out/ao_rsound.c

@@ -121,7 +121,13 @@ static int init(struct ao *ao, char *params)
         free(port);
     }
 
-    mp_chmap_reorder_to_alsa(&ao->channels);
+    // Actual channel layout unknown.
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_waveext_def(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) {
+        rsd_free(priv->rd);
+        return -1;
+    }
 
     rsd_set_param(priv->rd, RSD_SAMPLERATE, &ao->samplerate);
     rsd_set_param(priv->rd, RSD_CHANNELS, &ao->channels.num);

audio/out/ao_sdl.c

@@ -160,6 +160,13 @@ static int init(struct ao *ao, char *params)
         return -1;
     }
 
+    struct mp_chmap_sel sel = {0};
+    mp_chmap_sel_add_waveext_def(&sel);
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) {
+        uninit(ao, true);
+        return -1;
+    }
+
     SDL_AudioSpec desired, obtained;
 
     int bytes = 0;
@@ -236,9 +243,12 @@ static int init(struct ao *ao, char *params)
             return -1;
     }
 
+    if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, obtained.channels)) {
+        uninit(ao, true);
+        return -1;
+    }
+
     ao->samplerate = obtained.freq;
-    mp_chmap_from_channels(&ao->channels, obtained.channels);
-    mp_chmap_reorder_to_alsa(&ao->channels);
     ao->bps = ao->channels.num * ao->samplerate * bytes;
     ao->buffersize = obtained.size * bufcnt;
     ao->outburst = obtained.size;