modernize playback APIs

THe APIs were already recently updated so modernize it by using bool
where the return value is semantically boolean value. Using TRUE/FALSE is
inherently ambiguous because it is not obvious from the prototype if
success is 0 or TRUE (1).

src/aja_win32_stub.cpp

@@ -66,15 +66,15 @@ static const struct video_capture_info vidcap_aja_info = {
 REGISTER_MODULE(aja, &vidcap_aja_info, LIBRARY_CLASS_VIDEO_CAPTURE, VIDEO_CAPTURE_ABI_VERSION);
 
 extern "C" {
-__declspec(dllimport) int display_aja_get_property(void *state, int property, void *val, size_t *len);
+__declspec(dllimport) bool display_aja_get_property(void *state, int property, void *val, size_t *len);
 __declspec(dllimport) void display_aja_probe(struct device_info **available_cards, int *count, void (**deleter)(void *));
-__declspec(dllimport) int display_aja_reconfigure(void *state, struct video_desc desc);
+__declspec(dllimport) bool display_aja_reconfigure(void *state, struct video_desc desc);
 __declspec(dllimport) void *display_aja_init(struct module * /* parent */, const char *fmt, unsigned int flags);
 __declspec(dllimport) void display_aja_done(void *state);
 __declspec(dllimport) struct video_frame *display_aja_getf(void *state);
 __declspec(dllimport) bool display_aja_putf(void *state, struct video_frame *frame, long long nonblock);
 __declspec(dllimport) void display_aja_put_audio_frame(void *state, const struct audio_frame *frame);
-__declspec(dllimport) int display_aja_reconfigure_audio(void *state, int quant_samples, int channels,
+__declspec(dllimport) bool display_aja_reconfigure_audio(void *state, int quant_samples, int channels,
                 int sample_rate);
 }
 

src/audio/audio.cpp

@@ -1128,7 +1128,13 @@ void audio_sdi_send(struct state_audio *s, struct audio_frame *frame) {
         sdi_capture_new_incoming_frame(sdi_capture, frame);
 }
 
-void audio_register_display_callbacks(struct state_audio *s, void *udata, void (*putf)(void *, const struct audio_frame *), int (*reconfigure)(void *, int, int, int), int (*get_property)(void *, int, void *, size_t *))
+void
+audio_register_display_callbacks(struct state_audio *s, void *udata,
+                                 void (*putf)(void *,
+                                              const struct audio_frame *),
+                                 bool (*reconfigure)(void *, int, int, int),
+                                 bool (*get_property)(void *, int, void *,
+                                                     size_t *))
 {
         struct state_sdi_playback *sdi_playback;
         if(!audio_playback_get_display_flags(s->audio_playback_device))

src/audio/audio.h

@@ -97,7 +97,12 @@ void audio_join(struct state_audio *s);
 void audio_sdi_send(struct state_audio *s, struct audio_frame *frame);
 struct audio_frame * sdi_get_frame(void *state);
 void sdi_put_frame(void *state, struct audio_frame *frame);
-void audio_register_display_callbacks(struct state_audio *s, void *udata, void (*putf)(void *, const struct audio_frame *), int (*reconfigure)(void *, int, int, int), int (*get_property)(void *, int, void *, size_t *));
+void audio_register_display_callbacks(struct state_audio *s, void *udata,
+                                      void (*putf)(void *,
+                                                   const struct audio_frame *),
+                                      bool (*reconfigure)(void *, int, int, int),
+                                      bool (*get_property)(void *, int, void *,
+                                                          size_t *));
 
 struct audio_frame * audio_get_frame(struct state_audio *s);
 

src/audio/audio_playback.c

@@ -160,7 +160,7 @@ bool audio_playback_ctl(struct state_audio_playback *s, int request, void *data,
         return s->funcs->ctl(s->state, request, data, len);
 }
 
-int audio_playback_reconfigure(struct state_audio_playback *s, int quant_samples, int channels,
+bool audio_playback_reconfigure(struct state_audio_playback *s, int quant_samples, int channels,
                 int sample_rate)
 {
         return s->funcs->reconfigure(s->state, (struct audio_desc){quant_samples / 8, sample_rate, channels, AC_PCM});

src/audio/audio_playback.h

@@ -42,9 +42,11 @@ struct audio_frame;
 
 #ifdef __cplusplus
 extern "C" {
+#else
+#include <stdbool.h>
 #endif
 
-#define AUDIO_PLAYBACK_ABI_VERSION 10
+#define AUDIO_PLAYBACK_ABI_VERSION 11
 
 /** @anchor audio_playback_ctl_reqs
  * @name Audio playback control requests
@@ -81,7 +83,7 @@ struct audio_playback_info {
         void (*write)(void *state, const struct audio_frame *frame);
         /** Returns device supported format that matches best with propsed audio desc */
         bool (*ctl)(void *state, int request, void *data, size_t *len);
-        int (*reconfigure)(void *state, struct audio_desc);
+        bool (*reconfigure)(void *state, struct audio_desc);
         void (*done)(void *state);
 };
 
@@ -114,12 +116,10 @@ bool audio_playback_ctl(struct state_audio_playback *s, int request, void *data,
  * @param[in] quant_samples number of quantization bits
  * @param[in] channels      number of channels to be played back
  * @param[in] sample_rate   sample rate
- * @retval TRUE             if reconfiguration succeeded
- * @retval FALSE            if reconfiguration failed
  */
-int                             audio_playback_reconfigure(struct state_audio_playback *state,
-                int quant_samples, int channels,
-                int sample_rate);
+bool audio_playback_reconfigure(struct state_audio_playback *state,
+                                int quant_samples, int channels,
+                                int sample_rate);
 void                            audio_playback_put_frame(struct state_audio_playback *state, const struct audio_frame *frame);
 void                            audio_playback_finish(struct state_audio_playback *state);
 void                            audio_playback_done(struct state_audio_playback *state);

src/audio/playback/alsa.c

@@ -435,7 +435,7 @@ ADD_TO_PARAM("alsa-play-period-size", "* alsa-play-period-size=<frames>\n"
  * with PulseAudio. However, may underrun with different drivers/devices (Juli@?) where
  * the buffer size is significantly lower.
  */
-static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
+static bool audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
 {
         struct state_alsa_playback *s = (struct state_alsa_playback *) state;
         snd_pcm_hw_params_t *params;
@@ -479,7 +479,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
         if (rc < 0) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot obtain default hw parameters: %s\n",
                         snd_strerror(rc));
-                return FALSE;
+                return false;
         }
 
         /* Set the desired hardware parameters. */
@@ -495,7 +495,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
                 if (rc < 0) {
                         log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set non-interleaved hw access: %s\n",
                                         snd_strerror(rc));
-                        return FALSE;
+                        return false;
                 }
                 s->non_interleaved = true;
         } else {
@@ -505,7 +505,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
         if (desc.bps > 4 || desc.bps < 1) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "Unsupported BPS for audio (%d).\n",
                                 desc.bps * 8);
-                return FALSE;
+                return false;
 
         }
         format = bps_to_snd_fmts[desc.bps];
@@ -516,15 +516,15 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
         if (rc < 0) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set format: %s\n",
                         snd_strerror(rc));
-                return FALSE;
+                return false;
         }
 
         /* Two channels (stereo) */
         rc = snd_pcm_hw_params_set_channels(s->handle, params, desc.ch_count);
         if (rc < 0) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set requested channel count: %s\n",
                                 snd_strerror(rc));
-                return FALSE;
+                return false;
         }
 
         /* we want to resample if device doesn't support default sample rate */
@@ -545,7 +545,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
         if (rc < 0) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "cannot set requested sample rate: %s\n",
                         snd_strerror(rc));
-                return FALSE;
+                return false;
         }
 
         /* Set period to its minimal size.
@@ -599,7 +599,7 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
         if (rc < 0) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "unable to set hw parameters: %s\n",
                         snd_strerror(rc));
-                return FALSE;
+                return false;
         }
 
         snd_pcm_hw_params_current(s->handle, params);
@@ -653,9 +653,9 @@ static int audio_play_alsa_reconfigure(void *state, struct audio_desc desc)
                 }
         }
 
-        return TRUE;
+        return true;
 error:
-        return FALSE;
+        return false;
 }
 
 static void audio_play_alsa_probe(struct device_info **available_devices, int *count, void (**deleter)(void *))

src/audio/playback/coreaudio.cpp

@@ -150,7 +150,7 @@ static bool audio_play_ca_ctl(void *state [[gnu::unused]], int request, void *da
 
 ADD_TO_PARAM("ca-disable-adaptive-buf", "* ca-disable-adaptive-buf\n"
                 "  Core Audio - use fixed audio playback buffer instead of an adaptive one\n");
-static int audio_play_ca_reconfigure(void *state, struct audio_desc desc)
+static bool audio_play_ca_reconfigure(void *state, struct audio_desc desc)
 {
         struct state_ca_playback *s = (struct state_ca_playback *)state;
         AudioStreamBasicDescription stream_desc;
@@ -242,10 +242,10 @@ static int audio_play_ca_reconfigure(void *state, struct audio_desc desc)
 
         s->initialized = true;
 
-        return TRUE;
+        return true;
 
 error:
-        return FALSE;
+        return false;
 }
 
 

src/audio/playback/decklink.cpp

@@ -394,7 +394,7 @@ static bool audio_play_decklink_ctl(void *state [[gnu::unused]], int request, vo
         }
 }
 
-static int audio_play_decklink_reconfigure(void *state, struct audio_desc desc) {
+static bool audio_play_decklink_reconfigure(void *state, struct audio_desc desc) {
         struct state_decklink *s = (struct state_decklink *)state;
         BMDAudioSampleType sample_type;
 
@@ -407,7 +407,7 @@ static int audio_play_decklink_reconfigure(void *state, struct audio_desc desc)
                         s->audio_desc.ch_count != 16) {
                 fprintf(stderr, "[decklink] requested channel count isn't supported: "
                         "%d\n", s->audio_desc.ch_count);
-                return FALSE;
+                return false;
         }
 
         /* toggle one channel to supported two */
@@ -419,7 +419,7 @@ static int audio_play_decklink_reconfigure(void *state, struct audio_desc desc)
                         desc.sample_rate != 48000) {
                 LOG(LOG_LEVEL_ERROR) << "[decklink] audio format isn't supported: " <<
                         desc;
-                return FALSE;
+                return false;
         }
         switch(desc.bps) {
                 case 2:
@@ -429,7 +429,7 @@ static int audio_play_decklink_reconfigure(void *state, struct audio_desc desc)
                         sample_type = bmdAudioSampleType32bitInteger;
                         break;
                 default:
-                        return FALSE;
+                        return false;
         }
 
         s->deckLinkOutput->EnableAudioOutput(bmdAudioSampleRate48kHz,
@@ -438,7 +438,7 @@ static int audio_play_decklink_reconfigure(void *state, struct audio_desc desc)
                         bmdAudioOutputStreamContinuous);
         s->deckLinkOutput->StartScheduledPlayback(0, s->frameRateScale, s->frameRateDuration);
 
-        return TRUE;
+        return true;
 }
 
 static void audio_play_decklink_done(void *state)

src/audio/playback/dummy.c

@@ -91,10 +91,10 @@ static bool audio_play_dummy_ctl(void *state, int request, void *data, size_t *l
         }
 }
 
-static int audio_play_dummy_reconfigure(void *state, struct audio_desc desc)
+static bool audio_play_dummy_reconfigure(void *state, struct audio_desc desc)
 {
         UNUSED(state), UNUSED(desc);
-        return TRUE;
+        return true;
 }
 
 static const struct audio_playback_info aplay_dummy_info = {

src/audio/playback/dump.cpp

@@ -132,7 +132,7 @@ static void audio_play_dump_put_frame(void *state, const struct audio_frame *f)
         audio_export(s->exporter.get(), f);
 }
 
-static int audio_play_dump_reconfigure(void *state, struct audio_desc new_desc)
+static bool audio_play_dump_reconfigure(void *state, struct audio_desc new_desc)
 {
         auto *s = static_cast<audio_dump_state *>(state);
         s->desc = new_desc;

src/audio/playback/jack.c

@@ -294,7 +294,7 @@ static bool audio_play_jack_ctl(void *state, int request, void *data, size_t *le
         }
 }
 
-static int audio_play_jack_reconfigure(void *state, struct audio_desc desc)
+static bool audio_play_jack_reconfigure(void *state, struct audio_desc desc)
 {
         struct state_jack_playback *s = (struct state_jack_playback *) state;
         const char **ports;
@@ -307,7 +307,7 @@ static int audio_play_jack_reconfigure(void *state, struct audio_desc desc)
         ports = s->libjack->get_ports(s->client, s->jack_ports_pattern, NULL, JackPortIsInput);
         if(ports == NULL) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "Unable to input ports matching %s.\n", s->jack_ports_pattern);
-                return FALSE;
+                return false;
         }
 
         if(desc.ch_count > s->jack_ports_count) {
@@ -356,18 +356,18 @@ static int audio_play_jack_reconfigure(void *state, struct audio_desc desc)
 
         if (s->libjack->activate(s->client)) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "Cannot activate client.\n");
-                return FALSE;
+                return false;
         }
 
         for(i = 0; i < desc.ch_count; ++i) {
                 if (s->libjack->connect (s->client, s->libjack->port_name (s->output_port[i]), ports[i])) {
                         log_msg(LOG_LEVEL_ERROR, MOD_NAME "Cannot connect output port: %d.\n", i);
-                        return FALSE;
+                        return false;
                 }
         }
         free(ports);
 
-        return TRUE;
+        return true;
 }
 
 static void audio_play_jack_put_frame(void *state, const struct audio_frame *frame)

src/audio/playback/mixer.cpp

@@ -473,11 +473,11 @@ static bool audio_play_mixer_ctl(void *state, int request, void *data, size_t *l
         }
 }
 
-static int audio_play_mixer_reconfigure(void *state [[gnu::unused]], struct audio_desc desc)
+static bool audio_play_mixer_reconfigure(void *state [[gnu::unused]], struct audio_desc desc)
 {
         audio_desc requested{BPS, SAMPLE_RATE, CHANNELS, AC_PCM};
         assert(desc == requested);
-        return TRUE;
+        return true;
 }
 
 static const struct audio_playback_info aplay_mixer_info = {

src/audio/playback/none.c

@@ -100,14 +100,14 @@ static bool audio_play_none_ctl(void *state, int request, void *data, size_t *le
         return false;
 }
 
-static int audio_play_none_reconfigure(void *state, struct audio_desc desc)
+static bool audio_play_none_reconfigure(void *state, struct audio_desc desc)
 {
         UNUSED(desc);
         struct state_audio_playback_none *s = 
                 (struct state_audio_playback_none *) state;
         assert(s->magic == AUDIO_PLAYBACK_NONE_MAGIC);
 
-        return TRUE;
+        return true;
 }
 
 static const struct audio_playback_info aplay_none_info = {

src/audio/playback/portaudio.c

@@ -103,7 +103,7 @@ static int callback( const void *inputBuffer, void *outputBuffer,
                 PaStreamCallbackFlags statusFlags,
                 void *userData );
 static void     cleanup(struct state_portaudio_playback * s);
-static int audio_play_portaudio_reconfigure(void *state, struct audio_desc);
+static bool audio_play_portaudio_reconfigure(void *state, struct audio_desc);
 
  /*
   * Shared functions
@@ -317,7 +317,7 @@ static bool audio_play_portaudio_ctl(void *state, int request, void *data, size_
         }
 }
 
-static int audio_play_portaudio_reconfigure(void *state, struct audio_desc desc)
+static bool audio_play_portaudio_reconfigure(void *state, struct audio_desc desc)
 {
         struct state_portaudio_playback * s = 
                 (struct state_portaudio_playback *) state;
@@ -344,7 +344,7 @@ static int audio_play_portaudio_reconfigure(void *state, struct audio_desc desc)
         if (error != paNoError) {
                 log_msg(LOG_LEVEL_ERROR, MOD_NAME "error initializing portaudio\n");
                 log_msg(LOG_LEVEL_ERROR, "\tPortAudio error: %s\n", Pa_GetErrorText( error ) );
-                return FALSE;
+                return false;
         }
 
         log_msg(LOG_LEVEL_INFO, "Using PortAudio version: %s\n", Pa_GetVersionText());
@@ -393,14 +393,14 @@ static int audio_play_portaudio_reconfigure(void *state, struct audio_desc desc)
         if (error != paNoError) {
                 log_msg(LOG_LEVEL_ERROR, "[Portaudio] Unable to open stream: %s\n",
                                 Pa_GetErrorText(error));
-                return FALSE;
+                return false;
         }
 
         if (!portaudio_start_stream(s->stream)) {
-                return FALSE;
+                return false;
         }
 
-        return TRUE;
+        return true;
 }                        
 
 /* This routine will be called by the PortAudio engine when audio is needed.