diff options
| -rw-r--r-- | audio_alsa.c | 23 | ||||
| -rw-r--r-- | common.h | 5 | ||||
| -rw-r--r-- | dacp.c | 14 | ||||
| -rw-r--r-- | dacp.h | 6 | ||||
| -rw-r--r-- | metadata_hub.h | 14 | ||||
| -rw-r--r-- | player.c | 28 | ||||
| -rw-r--r-- | player.h | 12 | ||||
| -rw-r--r-- | rtp.c | 27 | ||||
| -rw-r--r-- | rtsp.c | 12 | ||||
| -rw-r--r-- | shairport.c | 26 |
10 files changed, 91 insertions, 76 deletions
diff --git a/audio_alsa.c b/audio_alsa.c index 36d5bfa..3315ed9 100644 --- a/audio_alsa.c +++ b/audio_alsa.c @@ -92,9 +92,12 @@ static int has_softvol = 0; static int volume_set_request = 0; // set when an external request is made to set the volume. int mute_request_pending = 0; // set when an external request is made to mute or unmute. int overriding_mute_state_requested = 0; // 1 = mute; 0 = unmute requested -int mixer_volume_setting_gives_mute = 0; // set when it is discovered that particular mixer volume setting causes a mute. -long alsa_mix_mute; // setting the volume to this value mutes output, if mixer_volume_setting_gives_mute is true -int volume_based_mute_is_active = 0; // set when muting is being done by a setting the volume to a magic value +int mixer_volume_setting_gives_mute = + 0; // set when it is discovered that particular mixer volume setting causes a mute. +long alsa_mix_mute; // setting the volume to this value mutes output, if + // mixer_volume_setting_gives_mute is true +int volume_based_mute_is_active = + 0; // set when muting is being done by a setting the volume to a magic value static snd_pcm_sframes_t (*alsa_pcm_write)(snd_pcm_t *, const void *, snd_pcm_uframes_t) = snd_pcm_writei; @@ -152,7 +155,7 @@ void close_mixer() { } } -void do_snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t * mix_elem, double vol) { +void do_snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t *mix_elem, double vol) { if (snd_mixer_selem_set_playback_dB_all(mix_elem, vol, 0) != 0) { debug(1, "Can't set playback volume accurately to %f dB.", vol); if (snd_mixer_selem_set_playback_dB_all(mix_elem, vol, -1) != 0) @@ -377,7 +380,8 @@ static int init(int argc, char **argv) { // For instance, the Raspberry Pi does this debug(1, "Lowest dB value is a mute"); mixer_volume_setting_gives_mute = 1; - alsa_mix_mute = SND_CTL_TLV_DB_GAIN_MUTE; // this may not be necessary -- it's always going to be SND_CTL_TLV_DB_GAIN_MUTE, right? + alsa_mix_mute = SND_CTL_TLV_DB_GAIN_MUTE; // this may not be necessary -- it's always + // going to be SND_CTL_TLV_DB_GAIN_MUTE, right? // debug(1, "Try minimum volume + 1 as lowest true attenuation value"); if (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv + 1, &alsa_mix_mindb) != 0) @@ -429,7 +433,8 @@ static int init(int argc, char **argv) { } } if ((config.alsa_use_playback_switch_for_mute == 1) && - (snd_mixer_selem_has_playback_switch(alsa_mix_elem)) || mixer_volume_setting_gives_mute) { + (snd_mixer_selem_has_playback_switch(alsa_mix_elem)) || + mixer_volume_setting_gives_mute) { audio_alsa.mute = &mute; // insert the mute function now we know it can do muting stuff // debug(1, "Has mixer and mute ability we will use."); } else { @@ -961,11 +966,11 @@ void do_volume(double vol) { // caller is assumed to have the alsa_mutex when us "control."); } } else { - if (volume_based_mute_is_active==0) { + if (volume_based_mute_is_active == 0) { // debug(1,"Set alsa volume."); do_snd_mixer_selem_set_playback_dB_all(alsa_mix_elem, vol); } else { - debug(1,"Not setting volume because volume-based mute is active"); + debug(1, "Not setting volume because volume-based mute is active"); } } volume_set_request = 0; // any external request that has been made is now satisfied @@ -1022,7 +1027,7 @@ void do_mute(int mute_state_requested) { // If the hardware isn't there, or we are not allowed to use it, nothing will be done // The caller must have the alsa mutex - if ((config.alsa_use_playback_switch_for_mute == 1) || mixer_volume_setting_gives_mute){ + if ((config.alsa_use_playback_switch_for_mute == 1) || mixer_volume_setting_gives_mute) { if (mute_request_pending == 0) local_mute_state_requested = mute_state_requested; if (open_mixer()) { @@ -122,8 +122,9 @@ typedef struct { mdns_backend *mdns; int buffer_start_fill; int64_t userSuppliedLatency; // overrides all other latencies -- use with caution - int64_t fixedLatencyOffset; // add this to all automatic latencies supplied to get the actual total latency - // the total latency will be limited to the min and max-latency values, if supplied + int64_t fixedLatencyOffset; // add this to all automatic latencies supplied to get the actual + // total latency + // the total latency will be limited to the min and max-latency values, if supplied int daemonise; int daemonise_store_pid; // don't try to save a PID file char *piddir; @@ -615,7 +615,7 @@ int32_t dacp_get_client_volume(void) { int32_t item_size; if (reply_size >= 8) { if (dacp_tlv_crawl(&sp, &item_size) == 'cmgt') { - debug(1,"Volume:",item_size); + debug(1, "Volume:", item_size); sp -= item_size; // drop down into the array -- don't skip over it reply_size -= 8; while (reply_size >= 8) { @@ -627,12 +627,12 @@ int32_t dacp_get_client_volume(void) { } } } else { - debug(1,"Unexpected payload response from getproperty?properties=dmcp.volume"); + debug(1, "Unexpected payload response from getproperty?properties=dmcp.volume"); } } else { - debug(1,"Too short a response from getproperty?properties=dmcp.volume"); + debug(1, "Too short a response from getproperty?properties=dmcp.volume"); } - debug(1,"Overall Volume is %d.",overall_volume); + debug(1, "Overall Volume is %d.", overall_volume); free(server_reply); } else { debug(1, "Unexpected response %d to dacp volume control request", response); @@ -658,8 +658,7 @@ int dacp_set_speaker_volume(int64_t machine_number, int32_t vo) { // should return 204 } -int dacp_get_speaker_list(dacp_spkr_stuff *speaker_info, - int max_size_of_array) { +int dacp_get_speaker_list(dacp_spkr_stuff *speaker_info, int max_size_of_array) { char *server_reply = NULL; int speaker_index = -1; // will be incremented before use int reply = -1; // will bve fixed if there is no problem @@ -775,7 +774,7 @@ void dacp_get_volume(void) { // calculate the real volume int32_t overall_volume = dacp_get_client_volume(); - debug(1,"DACP Volume: %d.",overall_volume); + debug(1, "DACP Volume: %d.", overall_volume); int speaker_count = dacp_get_speaker_list((dacp_spkr_stuff *)&speaker_info, 50); // debug(1,"DACP Speaker Count: %d.",speaker_count); @@ -804,4 +803,3 @@ void dacp_get_volume(void) { run_metadata_watchers(); } } - @@ -24,11 +24,11 @@ uint32_t dacp_tlv_crawl( int dacp_set_speaker_volume(int64_t machine_number, int32_t vo); -int dacp_get_speaker_list(dacp_spkr_stuff *speaker_array, - int max_size_of_array); +int dacp_get_speaker_list(dacp_spkr_stuff *speaker_array, int max_size_of_array); void set_dacp_server_information(rtsp_conn_info *conn); // tell the DACP conversation thread that // the dacp server information has been set // or changed int send_simple_dacp_command(const char *command); -void dacp_get_volume(void); // get the speaker volume information from the DACP source and store it in the metadata_hub +void dacp_get_volume(void); // get the speaker volume information from the DACP source and store it + // in the metadata_hub diff --git a/metadata_hub.h b/metadata_hub.h index d8a238d..c84cd89 100644 --- a/metadata_hub.h +++ b/metadata_hub.h @@ -58,14 +58,16 @@ typedef struct metadata_bundle { unsigned char item_composite_id[16]; // seems to be nowplaying 4 ids: dbid, plid, playlistItem, itemid - + // - - enum play_status_type player_state; // this is the state of the actual player itself, which can be a bit noisy. - - int speaker_volume; // this is the actual speaker volume, allowing for the main volume and the speaker volume control + + enum play_status_type + player_state; // this is the state of the actual player itself, which can be a bit noisy. + + int speaker_volume; // this is the actual speaker volume, allowing for the main volume and the + // speaker volume control int previous_speaker_volume; // this is needed to prevent a loop - + metadata_watcher watchers[number_of_watchers]; // functions to call if the metadata is changed. void *watchers_data[number_of_watchers]; // their individual data @@ -816,10 +816,10 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { // say we have started playing here #ifdef HAVE_METADATA_HUB - if (metadata_store.player_state != PS_PLAYING) { - metadata_store.player_state = PS_PLAYING; - run_metadata_watchers(); - } + if (metadata_store.player_state != PS_PLAYING) { + metadata_store.player_state = PS_PLAYING; + run_metadata_watchers(); + } #endif if (reference_timestamp) { // if we have a reference time // debug(1,"First frame seen with timestamp..."); @@ -1366,9 +1366,10 @@ static void *player_thread_func(void *arg) { conn->first_packet_timestamp = 0; conn->flush_requested = 0; // conn->fix_volume = 0x10000; - - if (conn->latency==0) { - debug(1,"No latency has (yet) been specified. Setting 99225 (2.25 seconds) frames as a default."); + + if (conn->latency == 0) { + debug(1, + "No latency has (yet) been specified. Setting 99225 (2.25 seconds) frames as a default."); conn->latency = 99225; } @@ -2500,9 +2501,8 @@ void player_flush(int64_t timestamp, rtsp_conn_info *conn) { if (metadata_store.player_state != PS_PAUSED) { metadata_store.player_state = PS_PAUSED; run_metadata_watchers(); - } + } #endif - } int player_play(rtsp_conn_info *conn) { @@ -2532,7 +2532,7 @@ int player_play(rtsp_conn_info *conn) { if (metadata_store.player_state != PS_PLAYING) { metadata_store.player_state = PS_PLAYING; run_metadata_watchers(); - } + } #endif return 0; } @@ -2550,10 +2550,10 @@ void player_stop(rtsp_conn_info *conn) { free(conn->player_thread); conn->player_thread = NULL; #ifdef HAVE_METADATA_HUB - if (metadata_store.player_state != PS_STOPPED) { - metadata_store.player_state = PS_STOPPED; - run_metadata_watchers(); - } + if (metadata_store.player_state != PS_STOPPED) { + metadata_store.player_state = PS_STOPPED; + run_metadata_watchers(); + } #endif } else { debug(3, "player thread of RTSP conversation %d is already deleted.", conn->connection_number); @@ -54,11 +54,13 @@ typedef struct { } stream_cfg; typedef struct { - int connection_number; // for debug ID purposes, nothing else... - int64_t latency; // the actual latency used for this play session - int64_t minimum_latency; // set if an a=min-latency: line appears in the ANNOUNCE message; zero otherwise - int64_t maximum_latency; // set if an a=max-latency: line appears in the ANNOUNCE message; zero otherwise - + int connection_number; // for debug ID purposes, nothing else... + int64_t latency; // the actual latency used for this play session + int64_t minimum_latency; // set if an a=min-latency: line appears in the ANNOUNCE message; zero + // otherwise + int64_t maximum_latency; // set if an a=max-latency: line appears in the ANNOUNCE message; zero + // otherwise + int fd; int authorized; // set if a password is required and has been supplied stream_cfg stream; @@ -224,25 +224,28 @@ void *rtp_control_receiver(void *arg) { // debug(1,"Remote Sync Time: %0llx.",remote_time_of_sync); sync_rtp_timestamp = monotonic_timestamp(ntohl(*((uint32_t *)&packet[16])), conn); - + if (config.userSuppliedLatency) { if (config.userSuppliedLatency != conn->latency) { - debug(1,"Using the user-supplied latency: %lld.",config.userSuppliedLatency); + debug(1, "Using the user-supplied latency: %lld.", config.userSuppliedLatency); } conn->latency = config.userSuppliedLatency; - } else if (packet[0] & 0x10) { // only set latency if it's a packet just after a flush or resume - int64_t rtp_timestamp_less_latency = monotonic_timestamp(ntohl(*((uint32_t *)&packet[4])), conn); - int64_t la = - sync_rtp_timestamp - rtp_timestamp_less_latency + config.fixedLatencyOffset; - if ((conn->maximum_latency) && (conn->maximum_latency<la)) + } else if (packet[0] & + 0x10) { // only set latency if it's a packet just after a flush or resume + int64_t rtp_timestamp_less_latency = + monotonic_timestamp(ntohl(*((uint32_t *)&packet[4])), conn); + int64_t la = sync_rtp_timestamp - rtp_timestamp_less_latency + config.fixedLatencyOffset; + if ((conn->maximum_latency) && (conn->maximum_latency < la)) la = conn->maximum_latency; - if ((conn->minimum_latency) && (conn->minimum_latency>la)) - la = conn->minimum_latency; - + if ((conn->minimum_latency) && (conn->minimum_latency > la)) + la = conn->minimum_latency; + if (la != conn->latency) { conn->latency = la; - debug(1,"New latency: %lld, sync latency: %lld, minimum latency: %lld, maximum latency: %lld, fixed offset: %lld.", - la,sync_rtp_timestamp - rtp_timestamp_less_latency,conn->minimum_latency,conn->maximum_latency,config.fixedLatencyOffset); + debug(1, "New latency: %lld, sync latency: %lld, minimum latency: %lld, maximum " + "latency: %lld, fixed offset: %lld.", + la, sync_rtp_timestamp - rtp_timestamp_less_latency, conn->minimum_latency, + conn->maximum_latency, config.fixedLatencyOffset); } } @@ -771,7 +771,7 @@ static void handle_setup(rtsp_conn_info *conn, rtsp_message *req, rtsp_message * send_metadata('ssnc', 'daid', ar, strlen(ar), req, 1); #endif } - + char *hdr = msg_get_header(req, "Transport"); if (!hdr) goto error; @@ -1197,7 +1197,7 @@ void *metadata_thread_function(void *ignore) { pc_queue_get_item(&metadata_queue, &pack); if (config.metadata_enabled) metadata_process(pack.type, pack.code, pack.data, pack.length); - + if (pack.carrier) msg_free(pack.carrier); // release the message else if (pack.data) @@ -1473,18 +1473,18 @@ static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_messag if (!strncmp(cp, "a=max-latency:", 14)) pmaxlatency = cp + 14; - + cp = next; } - + if (pminlatency) { conn->minimum_latency = atoi(pminlatency); - debug(1,"Minimum latency %d specified",conn->minimum_latency); + debug(1, "Minimum latency %d specified", conn->minimum_latency); } if (pmaxlatency) { conn->maximum_latency = atoi(pmaxlatency); - debug(1,"Maximum latency %d specified",conn->maximum_latency); + debug(1, "Maximum latency %d specified", conn->maximum_latency); } if ((paesiv == NULL) && (prsaaeskey == NULL)) { diff --git a/shairport.c b/shairport.c index 77a45e4..fbeda85 100644 --- a/shairport.c +++ b/shairport.c @@ -62,7 +62,6 @@ #include "dacp.h" #endif - #if defined(HAVE_METADATA_HUB) #include "metadata_hub.h" #endif @@ -396,16 +395,19 @@ int parse_options(int argc, char **argv) { inform("Warning: the option -R or --reconnectToOutput is deprecated."); break; case 'A': - inform("Warning: the option -A or --AirPlayLatency is deprecated and ignored. This setting is now " + inform("Warning: the option -A or --AirPlayLatency is deprecated and ignored. This setting " + "is now " "automatically received from the AirPlay device."); break; case 'i': - inform("Warning: the option -i or --iTunesLatency is deprecated and ignored. This setting is now " + inform("Warning: the option -i or --iTunesLatency is deprecated and ignored. This setting is " + "now " "automatically received from iTunes"); break; case 'f': - inform("Warning: the option --forkedDaapdLatency is deprecated and ignored. This setting is now " - "automatically received from forkedDaapd"); + inform( + "Warning: the option --forkedDaapdLatency is deprecated and ignored. This setting is now " + "automatically received from forkedDaapd"); break; case 'r': inform("Warning: the option -r or --resync is deprecated. Please use the " @@ -1108,9 +1110,9 @@ int main(int argc, char **argv) { strcat(configuration_file_path, ".conf"); config.configfile = configuration_file_path; - //config.statistics_requested = 0; // don't print stats in the log - //config.userSuppliedLatency = 0; // zero means none supplied - config.resyncthreshold = 0.05; // 50 ms + // config.statistics_requested = 0; // don't print stats in the log + // config.userSuppliedLatency = 0; // zero means none supplied + config.resyncthreshold = 0.05; // 50 ms config.timeout = 120; // this number of seconds to wait for [more] audio before switching to idle. config.tolerance = 0.002; // this number of seconds of timing error before attempting to correct it. @@ -1341,9 +1343,11 @@ int main(int argc, char **argv) { } /* Mess around with the latency options */ - // Basically, we expect the source to set the latency and add a fixed offset of 11025 frames to it, which sounds right - // If this latency is outside the max and min latensies that may be set by the source, clamp it to fit. - + // Basically, we expect the source to set the latency and add a fixed offset of 11025 frames to + // it, which sounds right + // If this latency is outside the max and min latensies that may be set by the source, clamp it to + // fit. + // If they specify a non-standard latency, we suggest the user to use the // audio_backend_latency_offset instead. |