From: Jörn Nettingsmeier Date: Fri, 15 Feb 2019 17:58:50 +0000 (+0000) Subject: NOP: remove commented-out code segments that were re-implemented X-Git-Tag: 3.3RC0~52^2~35 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=444c114231c20758efd289544fc6d58be20bf0ad;p=thirdparty%2Fshairport-sync.git NOP: remove commented-out code segments that were re-implemented --- diff --git a/audio_jack.c b/audio_jack.c index 544255a8..3d3fadda 100644 --- a/audio_jack.c +++ b/audio_jack.c @@ -106,49 +106,9 @@ int play(void *buf, int samples) { if (bytes_transferred < bytes_to_transfer) { debug(1, "JACK ringbuffer overrun. Only wrote %d of %d bytes.", bytes_transferred, bytes_to_transfer); } -/* - size_t space_to_end_of_buffer = audio_umb - audio_eoq; - if (space_to_end_of_buffer >= bytes_to_transfer) { - memcpy(audio_eoq, buf, bytes_to_transfer); - pthread_mutex_lock(&buffer_mutex); - audio_occupancy += samples; - audio_eoq += bytes_to_transfer; - pthread_mutex_unlock(&buffer_mutex); - } else { - memcpy(audio_eoq, buf, space_to_end_of_buffer); - buf += space_to_end_of_buffer; - memcpy(audio_lmb, buf, bytes_to_transfer - space_to_end_of_buffer); - pthread_mutex_lock(&buffer_mutex); - audio_occupancy += samples; - audio_eoq = audio_lmb + bytes_to_transfer - space_to_end_of_buffer; - pthread_mutex_unlock(&buffer_mutex); - } -*/ return 0; } -void deinterleave_and_convert_stream(const char *interleaved_frames, - const jack_default_audio_sample_t *jack_frame_buffer, - jack_nframes_t number_of_frames, enum ift_type side) { - jack_nframes_t i; - short *ifp = (short *)interleaved_frames; - jack_default_audio_sample_t *fp = (jack_default_audio_sample_t *)jack_frame_buffer; - if (side == IFT_frame_right_sample) - ifp++; - for (i = 0; i < number_of_frames; i++) { - short sample = *ifp; - jack_default_audio_sample_t converted_value; - if (sample >= 0) - converted_value = (1.0 * sample) / SHRT_MAX; - else - converted_value = -(1.0 * sample) / SHRT_MIN; - *fp = converted_value; - ifp++; - ifp++; - fp++; - } -} - static inline jack_default_audio_sample_t sample_conv(short sample) { return ((sample < 0) ? (-1.0 * sample / SHRT_MIN) : (1.0 * sample / SHRT_MAX)); } @@ -212,50 +172,6 @@ int jack_stream_write_cb(jack_nframes_t nframes, __attribute__((unused)) void *a frames_written++; nframes--; } - -/* - size_t frames_we_can_transfer = nframes; - // lock - pthread_mutex_lock(&buffer_mutex); - if (audio_occupancy < frames_we_can_transfer) { - frames_we_can_transfer = audio_occupancy; - // This means we effectively have underflow from the Shairport Sync source. - // In fact, it may be that there is nothing at all coming from the source, - // but the Shairport Sync client is open and active, so it must continue to output something. - } - - if (frames_we_can_transfer * 2 * 2 <= (size_t)(audio_umb - audio_toq)) { - // the bytes are all in a row in the audio buffer - deinterleave_and_convert_stream(audio_toq, &left_buffer[0], frames_we_can_transfer, - IFT_frame_left_sample); - deinterleave_and_convert_stream(audio_toq, &right_buffer[0], frames_we_can_transfer, - IFT_frame_right_sample); - audio_toq += frames_we_can_transfer * 2 * 2; - } else { - // the bytes are in two places in the audio buffer - size_t first_portion_to_write = (audio_umb - audio_toq) / (2 * 2); - if (first_portion_to_write != 0) { - deinterleave_and_convert_stream(audio_toq, &left_buffer[0], first_portion_to_write, - IFT_frame_left_sample); - deinterleave_and_convert_stream(audio_toq, &right_buffer[0], first_portion_to_write, - IFT_frame_right_sample); - } - deinterleave_and_convert_stream(audio_lmb, &left_buffer[first_portion_to_write], - frames_we_can_transfer - first_portion_to_write, - IFT_frame_left_sample); - deinterleave_and_convert_stream(audio_lmb, &right_buffer[first_portion_to_write], - frames_we_can_transfer - first_portion_to_write, - IFT_frame_right_sample); - audio_toq = audio_lmb + (frames_we_can_transfer - first_portion_to_write) * 2 * 2; - } - audio_occupancy -= frames_we_can_transfer; - jack_port_get_latency_range(left_port, JackPlaybackLatency, &latest_left_latency_range); - jack_port_get_latency_range(right_port, JackPlaybackLatency, &latest_right_latency_range); - time_of_latest_transfer = get_absolute_time_in_fp(); - pthread_mutex_unlock(&buffer_mutex); - // unlock - -*/ return 0; } @@ -265,26 +181,6 @@ void default_jack_error_callback(const char *desc) { debug(2, "jackd error: \"%s void default_jack_info_callback(const char *desc) { inform("jackd information: \"%s\"", desc); } -/* -void default_jack_set_latency_callback(jack_latency_callback_mode_t mode, - __attribute__((unused)) void *arg) { - if (mode == JackPlaybackLatency) { - jack_latency_range_t left_latency_range, right_latency_range; - jack_port_get_latency_range(left_port, JackPlaybackLatency, &left_latency_range); - jack_port_get_latency_range(right_port, JackPlaybackLatency, &right_latency_range); - - jack_nframes_t b_latency = (left_latency_range.min + left_latency_range.max) / 2; - if (b_latency == 0) - b_latency = (right_latency_range.min + right_latency_range.max) / 2; - // jack_latency = b_latency; // actually, we are not interested in the latency of the jack - // devices connected... - // FIXME: yes, we are. - jack_latency = 0; - debug(1, "playback latency callback: %" PRIu32 ".", jack_latency); - } -} -*/ - int jack_is_running() { int reply = -1; // meaning jack is not running if (client_is_open) { @@ -329,19 +225,12 @@ int jack_client_open_if_needed(void) { sample_rate = jack_get_sample_rate(client); // debug(1, "jackaudio sample rate = %" PRId32 ".", sample_rate); if (sample_rate == 44100) { - // FIXME: shairport-sync has no need for a latency callback, it's a leaf node with only outputs. - // If in the future some jack video player wants to resync to shairplay-sync audio, then yes, but - // the current usage seems to suggest a misunderstanding. -// if (jack_set_latency_callback(client, default_jack_set_latency_callback, NULL) == 0) { if (jack_activate(client)) { debug(1, "jackaudio cannot activate client"); } else { debug(2, "jackaudio client opened."); client_is_open = 1; } -// } else { -// debug(1, "jackaudio cannot set latency callback"); -// } } else { inform( "jackaudio is running at the wrong speed (%d) for Shairport Sync, which must be 44100", @@ -453,16 +342,6 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a jack_set_error_function(default_jack_error_callback); jack_set_info_function(default_jack_info_callback); -/* - // allocate space for the audio buffer - audio_lmb = malloc(buffer_size); - if (audio_lmb == NULL) - die("Can't allocate %d bytes for jackaudio buffer.", buffer_size); - audio_toq = audio_eoq = audio_lmb; - audio_umb = audio_lmb + buffer_size; - audio_occupancy = 0; // frames -*/ - client_is_open = 0; // now, if selected, start a thread to automatically open a client when there is a server. @@ -534,12 +413,6 @@ int jack_delay(long *the_delay) { void jack_flush() { debug(1, "Only the consumer can safely flush a lock-free ringbuffer. Asking the process callback to do it..."); flush_please = 1; -/* - // debug(1,"jack flush"); - audio_toq = audio_eoq = audio_lmb; - audio_umb = audio_lmb + buffer_size; - audio_occupancy = 0; // frames -*/ } void jack_stop(void) {