diff options
| -rw-r--r-- | Makefile | 2 | ||||
| -rw-r--r-- | audio_read.c | 977 | ||||
| -rw-r--r-- | audio_read.h | 9 | ||||
| -rw-r--r-- | common.h | 12 | ||||
| -rw-r--r-- | options.h | 1 | ||||
| -rw-r--r-- | toolame.c | 66 | ||||
| -rw-r--r-- | toolame.h | 2 |
7 files changed, 656 insertions, 413 deletions
@@ -96,7 +96,7 @@ CC_SWITCHES = $(OPTIM) $(REQUIRED) $(ARCH) $(PG) $(TWEAKS) $(WARNINGS) $(NEW_02L PGM = toolame -LIBS = -lm -lzmq +LIBS = -lm -lzmq -ljack -lpthread #nick burch's OS/2 fix gagravarr@SoftHome.net UNAME = $(shell uname) diff --git a/audio_read.c b/audio_read.c index afc4ad6..f5a4816 100644 --- a/audio_read.c +++ b/audio_read.c @@ -5,154 +5,349 @@ #include "encoder.h" #include "options.h" #include "portableio.h" +#include <pthread.h> +#include <jack/jack.h> +#include <jack/ringbuffer.h> #include "audio_read.h" +jack_port_t *input_port_left; +jack_port_t *input_port_right; +jack_client_t *client; +pthread_mutex_t encode_thread_lock = PTHREAD_MUTEX_INITIALIZER; +pthread_cond_t data_ready = PTHREAD_COND_INITIALIZER; + +/* shutdown can tell get_audio to stop */ +typedef struct _thread_info { + volatile int connected; +} jack_thread_info_t; + +jack_thread_info_t thread_info; +const size_t sample_size = sizeof(jack_default_audio_sample_t); + +#define DEFAULT_RB_SIZE 16384 /* ringbuffer size in frames */ +jack_ringbuffer_t *rb; + +/* setup_jack() + * + * PURPOSE: connect to jack, setup the ports, the ringbuffer + * + * frame_header is needed (fill information about sampling rate) + */ + +void setup_jack(frame_header *header, const char* jackname) { + const char *client_name = jackname; + const char *server_name = NULL; + jack_options_t options = JackNullOption; + jack_status_t status; + + /* open a client connection to the JACK server */ + + client = jack_client_open(client_name, options, &status, server_name); + if (client == NULL) { + fprintf(stderr, "jack_client_open() failed, " + "status = 0x%2.0x\n", status); + if (status & JackServerFailed) { + fprintf(stderr, "Unable to connect to JACK server\n"); + } + exit(1); + } + if (status & JackServerStarted) { + fprintf(stderr, "JACK server started\n"); + } + if (status & JackNameNotUnique) { + client_name = jack_get_client_name(client); + fprintf(stderr, "unique name `%s' assigned\n", client_name); + } + + thread_info.connected = 1; + + /* tell the JACK server to call `process()' whenever + there is work to be done. + */ + + jack_set_process_callback(client, process, &thread_info); + + /* tell the JACK server to call `jack_shutdown()' if + it ever shuts down, either entirely, or if it + just decides to stop calling us. + */ + + jack_on_shutdown(client, jack_shutdown, &thread_info); + + /* display the current sample rate. + */ + + printf ("engine sample rate: %" PRIu32 "\n", + jack_get_sample_rate(client)); + + if ((header->sampling_frequency = SmpFrqIndex((long) jack_get_sample_rate(client), &header->version)) < 0) { + fprintf (stderr, "invalid sample rate\n"); + exit(1); + } + + /* create two ports */ + + input_port_left = jack_port_register(client, "in_left", + JACK_DEFAULT_AUDIO_TYPE, + JackPortIsInput, 0); + input_port_right = jack_port_register(client, "in_right", + JACK_DEFAULT_AUDIO_TYPE, + JackPortIsInput, 0); + + if ((input_port_left == NULL) || (input_port_right == NULL)) { + fprintf(stderr, "no more JACK ports available\n"); + exit(1); + } + + + /* setup the ringbuffer */ + rb = jack_ringbuffer_create(2 * sample_size * DEFAULT_RB_SIZE); + fprintf(stderr, "jack sample_size: %zu\n", sample_size); + + + /* take the mutex */ + pthread_setcanceltype(PTHREAD_CANCEL_ASYNCHRONOUS, NULL); + pthread_mutex_lock(&encode_thread_lock); + + /* Tell the JACK server that we are ready to roll. Our + * process() callback will start running now. */ + + if (jack_activate(client)) { + fprintf (stderr, "cannot activate client"); + exit(1); + } +} + +/** + * The process callback for this JACK application is called in a + * special realtime thread once for each audio cycle. + * + * It fills the ringbuffer + */ +int process(jack_nframes_t nframes, void *arg) { + int i; + int samp; + //jack_thread_info_t *info = (jack_thread_info_t *) arg; + + jack_default_audio_sample_t *in_left, *in_right; + in_left = jack_port_get_buffer(input_port_left, nframes); + in_right = jack_port_get_buffer(input_port_right, nframes); + + /* Sndfile requires interleaved data. It is simpler here to + * just queue interleaved samples to a single ringbuffer. */ + //fprintf(stderr, "process()\n"); + for (i = 0; i < nframes; i++) { + /* + jack_ringbuffer_write(rb, (void *)(in_left + i), sample_size); + jack_ringbuffer_write(rb, (void *)(in_right + i), sample_size); + */ + /* convert to shorts, then insert into ringbuffer */ + samp = lrintf(in_left[i] * 1.0 * 0x7FFF); + jack_ringbuffer_write(rb, (char*)&samp, 2); + samp = lrintf(in_right[i] * 1.0 * 0x7FFF); + jack_ringbuffer_write(rb, (char*)&samp, 2); + + } + //fprintf(stderr, "PROCESS()\n"); + + /* tell read_samples that we've got new data */ + pthread_cond_signal(&data_ready); + + return 0; +} + +/** + * JACK calls this shutdown_callback if the server ever shuts down or + * decides to disconnect the client. + */ +void jack_shutdown(void *arg) +{ + jack_thread_info_t *info = (jack_thread_info_t *) arg; + + info->connected = 0; + /* tell read_samples to move on */ + + pthread_cond_signal(&data_ready); +} /************************************************************************ -* -* read_samples() -* -* PURPOSE: reads the PCM samples from a file to the buffer -* -* SEMANTICS: -* Reads #samples_read# number of shorts from #musicin# filepointer -* into #sample_buffer[]#. Returns the number of samples read. -* -************************************************************************/ - -unsigned long -read_samples (FILE * musicin, short sample_buffer[2304], - unsigned long num_samples, unsigned long frame_size) + * + * read_samples() + * + * PURPOSE: reads the PCM samples from a file to the buffer + * + * SEMANTICS: + * Reads #samples_read# number of shorts from #musicin# filepointer + * into #sample_buffer[]#. Returns the number of samples read. + * + ************************************************************************/ + +unsigned long read_samples (music_in_t* musicin, short sample_buffer[2304], + unsigned long num_samples, unsigned long frame_size) { - unsigned long samples_read; - static unsigned long samples_to_read; - static char init = TRUE; - - if (init) { - samples_to_read = num_samples; - init = FALSE; - } - if (samples_to_read >= frame_size) - samples_read = frame_size; - else - samples_read = samples_to_read; - if ((samples_read = - fread (sample_buffer, sizeof (short), (int) samples_read, - musicin)) == 0) - fprintf (stderr, "Hit end of audio data\n"); - /* - Samples are big-endian. If this is a little-endian machine - we must swap - */ - if (NativeByteOrder == order_unknown) { - NativeByteOrder = DetermineByteOrder (); + unsigned long samples_read; + static unsigned long samples_to_read; + static char init = TRUE; + + void* jack_sample_buffer; + + if (init) { + samples_to_read = num_samples; + init = FALSE; + } + if (samples_to_read >= frame_size) + samples_read = frame_size; + else + samples_read = samples_to_read; + + if (glopts.enable_jack) { + int f = 2; + while (jack_ringbuffer_read_space(rb) < f * samples_read) { + /* wait until process() signals more data */ + pthread_cond_wait(&data_ready, &encode_thread_lock); + + if (thread_info.connected == 0) { + pthread_mutex_unlock(&encode_thread_lock); + jack_client_close(client); + jack_ringbuffer_free(rb); + return 0; + } + } + + jack_sample_buffer = malloc(f * (int)samples_read); + int bytes_read = jack_ringbuffer_read(rb, jack_sample_buffer, f * (int)samples_read); + //fprintf(stderr, " read_bytes / f = %d, should be %d\n", (int)bytes_read/f, samples_read); + samples_read = bytes_read / f; + if (bytes_read % f != 0) { + fprintf(stderr, "cannot divide bytes_read by f: %d mod f = %d", bytes_read, bytes_read % f); + } + //fprintf(stderr, " #%d(%d) \n", (int)bytes_read, samples_read); + //f2les_array(jack_sample_buffer, sample_buffer, samples_read, 1); + + memcpy(sample_buffer, jack_sample_buffer, bytes_read); + + free(jack_sample_buffer); + + } + else { + if ((samples_read = + fread (sample_buffer, sizeof (short), (int) samples_read, + musicin->wav_input)) == 0) + fprintf (stderr, "Hit end of audio data\n"); + } + /* + Samples are big-endian. If this is a little-endian machine + we must swap + */ if (NativeByteOrder == order_unknown) { - fprintf (stderr, "byte order not determined\n"); - exit (1); + NativeByteOrder = DetermineByteOrder (); + if (NativeByteOrder == order_unknown) { + fprintf (stderr, "byte order not determined\n"); + exit (1); + } } - } - if (NativeByteOrder != order_littleEndian || (glopts.byteswap == TRUE)) - SwapBytesInWords (sample_buffer, samples_read); - - if (num_samples != MAX_U_32_NUM) - samples_to_read -= samples_read; - - if (samples_read < frame_size && samples_read > 0) { - /* fill out frame with zeros */ - for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); - samples_to_read = 0; - samples_read = frame_size; - } - return (samples_read); + if (NativeByteOrder != order_littleEndian || (glopts.byteswap == TRUE)) + SwapBytesInWords (sample_buffer, samples_read); + + if (num_samples != MAX_U_32_NUM) + samples_to_read -= samples_read; + + if (samples_read < frame_size && samples_read > 0) { + /* fill out frame with zeros */ + for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); + samples_to_read = 0; + samples_read = frame_size; + } + return (samples_read); } /************************************************************************ -* -* get_audio() -* -* PURPOSE: reads a frame of audio data from a file to the buffer, -* aligns the data for future processing, and separates the -* left and right channels -* -* -************************************************************************/ -unsigned long -get_audio (FILE * musicin, short buffer[2][1152], unsigned long num_samples, - int nch, frame_header *header) + * + * get_audio() + * + * PURPOSE: reads a frame of audio data from a file to the buffer, + * aligns the data for future processing, and separates the + * left and right channels + * + * + ************************************************************************/ + unsigned long +get_audio (music_in_t* musicin, short buffer[2][1152], unsigned long num_samples, + int nch, frame_header *header) { - int j; - short insamp[2304]; - unsigned long samples_read; - - if (nch == 2) { /* stereo */ - samples_read = - read_samples (musicin, insamp, num_samples, (unsigned long) 2304); - if (glopts.channelswap == TRUE) { - for (j = 0; j < 1152; j++) { - buffer[1][j] = insamp[2 * j]; - buffer[0][j] = insamp[2 * j + 1]; - } - } else { - for (j = 0; j < 1152; j++) { - buffer[0][j] = insamp[2 * j]; - buffer[1][j] = insamp[2 * j + 1]; - } - } - } else if (glopts.downmix == TRUE) { - samples_read = - read_samples (musicin, insamp, num_samples, (unsigned long) 2304); - for (j = 0; j < 1152; j++) { - buffer[0][j] = 0.5 * (insamp[2 * j] + insamp[2 * j + 1]); - } - } else { /* mono */ - samples_read = - read_samples (musicin, insamp, num_samples, (unsigned long) 1152); - for (j = 0; j < 1152; j++) { - buffer[0][j] = insamp[j]; - /* buffer[1][j] = 0; don't bother zeroing this buffer. MFC Nov 99 */ + int j; + short insamp[2304]; + unsigned long samples_read; + + if (nch == 2) { /* stereo */ + samples_read = + read_samples (musicin, insamp, num_samples, (unsigned long) 2304); + if (glopts.channelswap == TRUE) { + for (j = 0; j < 1152; j++) { + buffer[1][j] = insamp[2 * j]; + buffer[0][j] = insamp[2 * j + 1]; + } + } else { + for (j = 0; j < 1152; j++) { + buffer[0][j] = insamp[2 * j]; + buffer[1][j] = insamp[2 * j + 1]; + } + } + } else if (glopts.downmix == TRUE) { + samples_read = + read_samples (musicin, insamp, num_samples, (unsigned long) 2304); + for (j = 0; j < 1152; j++) { + buffer[0][j] = 0.5 * (insamp[2 * j] + insamp[2 * j + 1]); + } + } else { /* mono */ + samples_read = + read_samples (musicin, insamp, num_samples, (unsigned long) 1152); + for (j = 0; j < 1152; j++) { + buffer[0][j] = insamp[j]; + /* buffer[1][j] = 0; don't bother zeroing this buffer. MFC Nov 99 */ + } } - } - return (samples_read); + return (samples_read); } /***************************************************************************** -* -* Routines to determine byte order and swap bytes -* -*****************************************************************************/ + * + * Routines to determine byte order and swap bytes + * + *****************************************************************************/ enum byte_order DetermineByteOrder (void) { - char s[sizeof (long) + 1]; - union { - long longval; - char charval[sizeof (long)]; - } probe; - probe.longval = 0x41424344L; /* ABCD in ASCII */ - strncpy (s, probe.charval, sizeof (long)); - s[sizeof (long)] = '\0'; - /* fprintf( stderr, "byte order is %s\n", s ); */ - if (strcmp (s, "ABCD") == 0) - return order_bigEndian; - else if (strcmp (s, "DCBA") == 0) - return order_littleEndian; - else - return order_unknown; + char s[sizeof (long) + 1]; + union { + long longval; + char charval[sizeof (long)]; + } probe; + probe.longval = 0x41424344L; /* ABCD in ASCII */ + strncpy (s, probe.charval, sizeof (long)); + s[sizeof (long)] = '\0'; + /* fprintf( stderr, "byte order is %s\n", s ); */ + if (strcmp (s, "ABCD") == 0) + return order_bigEndian; + else if (strcmp (s, "DCBA") == 0) + return order_littleEndian; + else + return order_unknown; } void SwapBytesInWords (short *loc, int words) { - int i; - short thisval; - char *dst, *src; - src = (char *) &thisval; - for (i = 0; i < words; i++) { - thisval = *loc; - dst = (char *) loc++; - dst[0] = src[1]; - dst[1] = src[0]; - } + int i; + short thisval; + char *dst, *src; + src = (char *) &thisval; + for (i = 0; i < words; i++) { + thisval = *loc; + dst = (char *) loc++; + dst[0] = src[1]; + dst[1] = src[0]; + } } /***************************************************************************** @@ -163,62 +358,62 @@ void SwapBytesInWords (short *loc, int words) int aiff_read_headers (FILE * file_ptr, IFF_AIFF * aiff_ptr) { - int chunkSize, subSize, sound_position; - - if (fseek (file_ptr, 0, SEEK_SET) != 0) - return -1; - - if (Read32BitsHighLow (file_ptr) != IFF_ID_FORM) - return -1; - - chunkSize = Read32BitsHighLow (file_ptr); - - if (Read32BitsHighLow (file_ptr) != IFF_ID_AIFF) - return -1; - - sound_position = 0; - while (chunkSize > 0) { - chunkSize -= 4; - switch (Read32BitsHighLow (file_ptr)) { - - case IFF_ID_COMM: - chunkSize -= subSize = Read32BitsHighLow (file_ptr); - aiff_ptr->numChannels = Read16BitsHighLow (file_ptr); - subSize -= 2; - aiff_ptr->numSampleFrames = Read32BitsHighLow (file_ptr); - subSize -= 4; - aiff_ptr->sampleSize = Read16BitsHighLow (file_ptr); - subSize -= 2; - aiff_ptr->sampleRate = ReadIeeeExtendedHighLow (file_ptr); - subSize -= 10; - while (subSize > 0) { - getc (file_ptr); - subSize -= 1; - } - break; - - case IFF_ID_SSND: - chunkSize -= subSize = Read32BitsHighLow (file_ptr); - aiff_ptr->blkAlgn.offset = Read32BitsHighLow (file_ptr); - subSize -= 4; - aiff_ptr->blkAlgn.blockSize = Read32BitsHighLow (file_ptr); - subSize -= 4; - sound_position = ftell (file_ptr) + aiff_ptr->blkAlgn.offset; - if (fseek (file_ptr, (long) subSize, SEEK_CUR) != 0) + int chunkSize, subSize, sound_position; + + if (fseek (file_ptr, 0, SEEK_SET) != 0) + return -1; + + if (Read32BitsHighLow (file_ptr) != IFF_ID_FORM) return -1; - aiff_ptr->sampleType = IFF_ID_SSND; - break; - - default: - chunkSize -= subSize = Read32BitsHighLow (file_ptr); - while (subSize > 0) { - getc (file_ptr); - subSize -= 1; - } - break; + + chunkSize = Read32BitsHighLow (file_ptr); + + if (Read32BitsHighLow (file_ptr) != IFF_ID_AIFF) + return -1; + + sound_position = 0; + while (chunkSize > 0) { + chunkSize -= 4; + switch (Read32BitsHighLow (file_ptr)) { + + case IFF_ID_COMM: + chunkSize -= subSize = Read32BitsHighLow (file_ptr); + aiff_ptr->numChannels = Read16BitsHighLow (file_ptr); + subSize -= 2; + aiff_ptr->numSampleFrames = Read32BitsHighLow (file_ptr); + subSize -= 4; + aiff_ptr->sampleSize = Read16BitsHighLow (file_ptr); + subSize -= 2; + aiff_ptr->sampleRate = ReadIeeeExtendedHighLow (file_ptr); + subSize -= 10; + while (subSize > 0) { + getc (file_ptr); + subSize -= 1; + } + break; + + case IFF_ID_SSND: + chunkSize -= subSize = Read32BitsHighLow (file_ptr); + aiff_ptr->blkAlgn.offset = Read32BitsHighLow (file_ptr); + subSize -= 4; + aiff_ptr->blkAlgn.blockSize = Read32BitsHighLow (file_ptr); + subSize -= 4; + sound_position = ftell (file_ptr) + aiff_ptr->blkAlgn.offset; + if (fseek (file_ptr, (long) subSize, SEEK_CUR) != 0) + return -1; + aiff_ptr->sampleType = IFF_ID_SSND; + break; + + default: + chunkSize -= subSize = Read32BitsHighLow (file_ptr); + while (subSize > 0) { + getc (file_ptr); + subSize -= 1; + } + break; + } } - } - return sound_position; + return sound_position; } /***************************************************************************** @@ -229,236 +424,236 @@ int aiff_read_headers (FILE * file_ptr, IFF_AIFF * aiff_ptr) int aiff_seek_to_sound_data (FILE * file_ptr) { - if (fseek - (file_ptr, AIFF_FORM_HEADER_SIZE + AIFF_SSND_HEADER_SIZE, - SEEK_SET) != 0) - return (-1); - return (0); + if (fseek + (file_ptr, AIFF_FORM_HEADER_SIZE + AIFF_SSND_HEADER_SIZE, + SEEK_SET) != 0) + return (-1); + return (0); } /************************************************************ -* parse_input_file() -* Determine the type of sound file. (stdin, wav, aiff, raw pcm) -* Determine Sampling Frequency -* number of samples -* whether the new sample is stereo or mono. -* -* If file is coming from /dev/stdin assume it is raw PCM. (it's what I use. YMMV) -* -* This is just a hacked together function. The aiff parsing comes from the ISO code. -* The WAV code comes from Nick Burch -* The ugly /dev/stdin hack comes from me. -* MFC Dec 99 -**************************************************************/ -void + * parse_input_file() + * Determine the type of sound file. (stdin, wav, aiff, raw pcm) + * Determine Sampling Frequency + * number of samples + * whether the new sample is stereo or mono. + * + * If file is coming from /dev/stdin assume it is raw PCM. (it's what I use. YMMV) + * + * This is just a hacked together function. The aiff parsing comes from the ISO code. + * The WAV code comes from Nick Burch + * The ugly /dev/stdin hack comes from me. + * MFC Dec 99 + **************************************************************/ + void parse_input_file (FILE * musicin, char inPath[MAX_NAME_SIZE], frame_header *header, - unsigned long *num_samples) + unsigned long *num_samples) { - IFF_AIFF pcm_aiff_data; - long soundPosition; - - unsigned char wave_header_buffer[40]; //HH fixed - int wave_header_read = 0; - int wave_header_stereo = -1; - int wave_header_16bit = -1; - unsigned long samplerate; - - /*************************** STDIN ********************************/ - /* check if we're reading from stdin. Assume it's a raw PCM file. */ - /* Of course, you could be piping a WAV file into stdin. Not done in this code */ - /* this code is probably very dodgy and was written to suit my needs. MFC Dec 99 */ - if ((strcmp (inPath, "/dev/stdin") == 0)) { - fprintf (stderr, "Reading from stdin\n"); - fprintf (stderr, "Remember to set samplerate with '-s'.\n"); - *num_samples = MAX_U_32_NUM; /* huge sound file */ - return; - } - - /**************************** AIFF ********************************/ - if ((soundPosition = aiff_read_headers (musicin, &pcm_aiff_data)) != -1) { - fprintf (stderr, ">>> Using Audio IFF sound file headers\n"); - aiff_check (inPath, &pcm_aiff_data, &header->version); - if (fseek (musicin, soundPosition, SEEK_SET) != 0) { - fprintf (stderr, "Could not seek to PCM sound data in \"%s\".\n", - inPath); - exit (1); - } - fprintf (stderr, "Parsing AIFF audio file \n"); - header->sampling_frequency = - SmpFrqIndex ((long) pcm_aiff_data.sampleRate, &header->version); - fprintf (stderr, ">>> %f Hz sampling frequency selected\n", - pcm_aiff_data.sampleRate); - - /* Determine number of samples in sound file */ - *num_samples = pcm_aiff_data.numChannels * pcm_aiff_data.numSampleFrames; - - if (pcm_aiff_data.numChannels == 1) { - header->mode = MPG_MD_MONO; - header->mode_ext = 0; - } - return; - } - - /**************************** WAVE *********************************/ - /* Nick Burch <The_Leveller@newmail.net> */ - /*********************************/ - /* Wave File Headers: (Dec) */ - /* 8-11 = "WAVE" */ - /* 22 = Stereo / Mono */ - /* 01 = mono, 02 = stereo */ - /* 24 = Sampling Frequency */ - /* 32 = Data Rate */ - /* 01 = x1 (8bit Mono) */ - /* 02 = x2 (8bit Stereo or */ - /* 16bit Mono) */ - /* 04 = x4 (16bit Stereo) */ - /*********************************/ - - fseek (musicin, 0, SEEK_SET); - fread (wave_header_buffer, 1, 40, musicin); - - if (wave_header_buffer[8] == 'W' && wave_header_buffer[9] == 'A' - && wave_header_buffer[10] == 'V' && wave_header_buffer[11] == 'E') { - fprintf (stderr, "Parsing Wave File Header\n"); - if (NativeByteOrder == order_unknown) { - NativeByteOrder = DetermineByteOrder (); - if (NativeByteOrder == order_unknown) { - fprintf (stderr, "byte order not determined\n"); - exit (1); - } - } - if (NativeByteOrder == order_littleEndian) { - samplerate = wave_header_buffer[24] + - (wave_header_buffer[25] << 8) + - (wave_header_buffer[26] << 16) + - (wave_header_buffer[27] << 24); - } else { - samplerate = wave_header_buffer[27] + - (wave_header_buffer[26] << 8) + - (wave_header_buffer[25] << 16) + - (wave_header_buffer[24] << 24); - } - /* Wave File */ - wave_header_read = 1; - switch (samplerate) { - case 44100: - case 48000: - case 32000: - case 24000: - case 22050: - case 16000: - fprintf (stderr, ">>> %ld Hz sampling freq selected\n", samplerate); - break; - default: - /* Unknown Unsupported Frequency */ - fprintf (stderr, ">>> Unknown samp freq %ld Hz in Wave Header\n", - samplerate); - fprintf (stderr, ">>> Default 44.1 kHz samp freq selected\n"); - samplerate = 44100; + IFF_AIFF pcm_aiff_data; + long soundPosition; + + unsigned char wave_header_buffer[40]; //HH fixed + int wave_header_read = 0; + int wave_header_stereo = -1; + int wave_header_16bit = -1; + unsigned long samplerate; + + /*************************** STDIN ********************************/ + /* check if we're reading from stdin. Assume it's a raw PCM file. */ + /* Of course, you could be piping a WAV file into stdin. Not done in this code */ + /* this code is probably very dodgy and was written to suit my needs. MFC Dec 99 */ + if ((strcmp (inPath, "/dev/stdin") == 0)) { + fprintf (stderr, "Reading from stdin\n"); + fprintf (stderr, "Remember to set samplerate with '-s'.\n"); + *num_samples = MAX_U_32_NUM; /* huge sound file */ + return; } - if ((header->sampling_frequency = - SmpFrqIndex ((long) samplerate, &header->version)) < 0) { - fprintf (stderr, "invalid sample rate\n"); - exit (0); + /**************************** AIFF ********************************/ + if ((soundPosition = aiff_read_headers (musicin, &pcm_aiff_data)) != -1) { + fprintf (stderr, ">>> Using Audio IFF sound file headers\n"); + aiff_check (inPath, &pcm_aiff_data, &header->version); + if (fseek (musicin, soundPosition, SEEK_SET) != 0) { + fprintf (stderr, "Could not seek to PCM sound data in \"%s\".\n", + inPath); + exit (1); + } + fprintf (stderr, "Parsing AIFF audio file \n"); + header->sampling_frequency = + SmpFrqIndex ((long) pcm_aiff_data.sampleRate, &header->version); + fprintf (stderr, ">>> %f Hz sampling frequency selected\n", + pcm_aiff_data.sampleRate); + + /* Determine number of samples in sound file */ + *num_samples = pcm_aiff_data.numChannels * pcm_aiff_data.numSampleFrames; + + if (pcm_aiff_data.numChannels == 1) { + header->mode = MPG_MD_MONO; + header->mode_ext = 0; + } + return; } - if ((long) wave_header_buffer[22] == 1) { - fprintf (stderr, ">>> Input Wave File is Mono\n"); - wave_header_stereo = 0; - header->mode = MPG_MD_MONO; - header->mode_ext = 0; - } - if ((long) wave_header_buffer[22] == 2) { - fprintf (stderr, ">>> Input Wave File is Stereo\n"); - wave_header_stereo = 1; + /**************************** WAVE *********************************/ + /* Nick Burch <The_Leveller@newmail.net> */ + /*********************************/ + /* Wave File Headers: (Dec) */ + /* 8-11 = "WAVE" */ + /* 22 = Stereo / Mono */ + /* 01 = mono, 02 = stereo */ + /* 24 = Sampling Frequency */ + /* 32 = Data Rate */ + /* 01 = x1 (8bit Mono) */ + /* 02 = x2 (8bit Stereo or */ + /* 16bit Mono) */ + /* 04 = x4 (16bit Stereo) */ + /*********************************/ + + fseek (musicin, 0, SEEK_SET); + fread (wave_header_buffer, 1, 40, musicin); + + if (wave_header_buffer[8] == 'W' && wave_header_buffer[9] == 'A' + && wave_header_buffer[10] == 'V' && wave_header_buffer[11] == 'E') { + fprintf (stderr, "Parsing Wave File Header\n"); + if (NativeByteOrder == order_unknown) { + NativeByteOrder = DetermineByteOrder (); + if (NativeByteOrder == order_unknown) { + fprintf (stderr, "byte order not determined\n"); + exit (1); + } + } + if (NativeByteOrder == order_littleEndian) { + samplerate = wave_header_buffer[24] + + (wave_header_buffer[25] << 8) + + (wave_header_buffer[26] << 16) + + (wave_header_buffer[27] << 24); + } else { + samplerate = wave_header_buffer[27] + + (wave_header_buffer[26] << 8) + + (wave_header_buffer[25] << 16) + + (wave_header_buffer[24] << 24); + } + /* Wave File */ + wave_header_read = 1; + switch (samplerate) { + case 44100: + case 48000: + case 32000: + case 24000: + case 22050: + case 16000: + fprintf (stderr, ">>> %ld Hz sampling freq selected\n", samplerate); + break; + default: + /* Unknown Unsupported Frequency */ + fprintf (stderr, ">>> Unknown samp freq %ld Hz in Wave Header\n", + samplerate); + fprintf (stderr, ">>> Default 44.1 kHz samp freq selected\n"); + samplerate = 44100; + } + + if ((header->sampling_frequency = + SmpFrqIndex ((long) samplerate, &header->version)) < 0) { + fprintf (stderr, "invalid sample rate\n"); + exit (0); + } + + if ((long) wave_header_buffer[22] == 1) { + fprintf (stderr, ">>> Input Wave File is Mono\n"); + wave_header_stereo = 0; + header->mode = MPG_MD_MONO; + header->mode_ext = 0; + } + if ((long) wave_header_buffer[22] == 2) { + fprintf (stderr, ">>> Input Wave File is Stereo\n"); + wave_header_stereo = 1; + } + if ((long) wave_header_buffer[32] == 1) { + fprintf (stderr, ">>> Input Wave File is 8 Bit\n"); + wave_header_16bit = 0; + fprintf (stderr, "Input File must be 16 Bit! Please Re-sample"); + exit (1); + } + if ((long) wave_header_buffer[32] == 2) { + if (wave_header_stereo == 1) { + fprintf (stderr, ">>> Input Wave File is 8 Bit\n"); + wave_header_16bit = 0; + fprintf (stderr, "Input File must be 16 Bit! Please Re-sample"); + exit (1); + } else { + /* fprintf(stderr, ">>> Input Wave File is 16 Bit\n" ); */ + wave_header_16bit = 1; + } + } + if ((long) wave_header_buffer[32] == 4) { + /* fprintf(stderr, ">>> Input Wave File is 16 Bit\n" ); */ + wave_header_16bit = 1; + } + /* should probably use the wave header to determine size here FIXME MFC Feb 2003 */ + *num_samples = MAX_U_32_NUM; + if (fseek (musicin, 44, SEEK_SET) != 0) { /* there's a way of calculating the size of the + wave header. i'll just jump 44 to start with */ + fprintf (stderr, "Could not seek to PCM sound data in \"%s\".\n", + inPath); + exit (1); + } + return; } - if ((long) wave_header_buffer[32] == 1) { - fprintf (stderr, ">>> Input Wave File is 8 Bit\n"); - wave_header_16bit = 0; - fprintf (stderr, "Input File must be 16 Bit! Please Re-sample"); - exit (1); - } - if ((long) wave_header_buffer[32] == 2) { - if (wave_header_stereo == 1) { - fprintf (stderr, ">>> Input Wave File is 8 Bit\n"); - wave_header_16bit = 0; - fprintf (stderr, "Input File must be 16 Bit! Please Re-sample"); - exit (1); - } else { - /* fprintf(stderr, ">>> Input Wave File is 16 Bit\n" ); */ - wave_header_16bit = 1; - } - } - if ((long) wave_header_buffer[32] == 4) { - /* fprintf(stderr, ">>> Input Wave File is 16 Bit\n" ); */ - wave_header_16bit = 1; - } - /* should probably use the wave header to determine size here FIXME MFC Feb 2003 */ + + /*************************** PCM **************************/ + fprintf (stderr, "No header found. Assuming Raw PCM sound file\n"); + /* Raw PCM. No header. Reset the input file to read from the start */ + fseek (musicin, 0, SEEK_SET); + /* Assume it is a huge sound file since there's no real info available */ + /* FIXME: Could always fstat the file? Probably not worth it. MFC Feb 2003 */ *num_samples = MAX_U_32_NUM; - if (fseek (musicin, 44, SEEK_SET) != 0) { /* there's a way of calculating the size of the - wave header. i'll just jump 44 to start with */ - fprintf (stderr, "Could not seek to PCM sound data in \"%s\".\n", - inPath); - exit (1); - } - return; - } - - /*************************** PCM **************************/ - fprintf (stderr, "No header found. Assuming Raw PCM sound file\n"); - /* Raw PCM. No header. Reset the input file to read from the start */ - fseek (musicin, 0, SEEK_SET); - /* Assume it is a huge sound file since there's no real info available */ - /* FIXME: Could always fstat the file? Probably not worth it. MFC Feb 2003 */ - *num_samples = MAX_U_32_NUM; } /************************************************************************ -* -* aiff_check -* -* PURPOSE: Checks AIFF header information to make sure it is valid. -* Exits if not. -* -************************************************************************/ + * + * aiff_check + * + * PURPOSE: Checks AIFF header information to make sure it is valid. + * Exits if not. + * + ************************************************************************/ void aiff_check (char *file_name, IFF_AIFF * pcm_aiff_data, int *version) { - if (pcm_aiff_data->sampleType != IFF_ID_SSND) { - fprintf (stderr, "Sound data is not PCM in \"%s\".\n", file_name); - exit (1); - } - - if (SmpFrqIndex ((long) pcm_aiff_data->sampleRate, version) < 0) { - fprintf (stderr, "in \"%s\".\n", file_name); - exit (1); - } - - if (pcm_aiff_data->sampleSize != sizeof (short) * BITS_IN_A_BYTE) { - fprintf (stderr, "Sound data is not %d bits in \"%s\".\n", - sizeof (short) * BITS_IN_A_BYTE, file_name); - exit (1); - } - - if (pcm_aiff_data->numChannels != MONO - && pcm_aiff_data->numChannels != STEREO) { - fprintf (stderr, "Sound data is not mono or stereo in \"%s\".\n", - file_name); - exit (1); - } - - if (pcm_aiff_data->blkAlgn.blockSize != 0) { - fprintf (stderr, "Block size is not %d bytes in \"%s\".\n", 0, file_name); - exit (1); - } - - if (pcm_aiff_data->blkAlgn.offset != 0) { - fprintf (stderr, "Block offset is not %d bytes in \"%s\".\n", 0, - file_name); - exit (1); - } + if (pcm_aiff_data->sampleType != IFF_ID_SSND) { + fprintf (stderr, "Sound data is not PCM in \"%s\".\n", file_name); + exit (1); + } + + if (SmpFrqIndex ((long) pcm_aiff_data->sampleRate, version) < 0) { + fprintf (stderr, "in \"%s\".\n", file_name); + exit (1); + } + + if (pcm_aiff_data->sampleSize != sizeof (short) * BITS_IN_A_BYTE) { + fprintf (stderr, "Sound data is not %zu bits in \"%s\".\n", + sizeof (short) * BITS_IN_A_BYTE, file_name); + exit (1); + } + + if (pcm_aiff_data->numChannels != MONO + && pcm_aiff_data->numChannels != STEREO) { + fprintf (stderr, "Sound data is not mono or stereo in \"%s\".\n", + file_name); + exit (1); + } + + if (pcm_aiff_data->blkAlgn.blockSize != 0) { + fprintf (stderr, "Block size is not %d bytes in \"%s\".\n", 0, file_name); + exit (1); + } + + if (pcm_aiff_data->blkAlgn.offset != 0) { + fprintf (stderr, "Block offset is not %d bytes in \"%s\".\n", 0, + file_name); + exit (1); + } } diff --git a/audio_read.h b/audio_read.h index 7ab3f01..ab17a8f 100644 --- a/audio_read.h +++ b/audio_read.h @@ -28,6 +28,10 @@ typedef struct IFF_AIFF_struct } IFF_AIFF; +void setup_jack(frame_header *header, const char* jackname); +int process(jack_nframes_t nframes, void *arg); +void jack_shutdown(void *arg); + void parse_input_file (FILE *musicin, char *, frame_header *header, unsigned long *num_samples); void aiff_check (char *file_name, IFF_AIFF * pcm_aiff_data, int *version); @@ -35,7 +39,8 @@ int aiff_read_headers (FILE *, IFF_AIFF *); int aiff_seek_to_sound_data (FILE *); enum byte_order DetermineByteOrder (void); void SwapBytesInWords (short *loc, int words); - unsigned long read_samples (FILE *, short[2304], unsigned long, + unsigned long read_samples (music_in_t*, short[2304], unsigned long, unsigned long); - unsigned long get_audio (FILE *, short[2][1152], unsigned long, + unsigned long get_audio (music_in_t*, short[2][1152], unsigned long, int, frame_header *header); + @@ -89,6 +89,7 @@ #include <stdio.h> #include <stdlib.h> +#include <jack/jack.h> /* Structure for Reading Layer II Allocation Tables from File */ @@ -162,6 +163,17 @@ enum byte_order { order_unknown, order_bigEndian, order_littleEndian }; extern enum byte_order NativeByteOrder; + +typedef struct music_in_s +{ + /* Data for the wav input */ + FILE* wav_input; + + /* Data for the jack input */ + jack_client_t* jack_client; + const char* jack_name; +} music_in_t; + /* "bit_stream.h" Type Definitions */ @@ -16,6 +16,7 @@ typedef struct float athlevel; /* 0 extra argument to the ATH equation - used for VBR in LAME */ int verbosity; /* 2 by default. 0 is no output at all */ + int enable_jack; /* 1=use JACK input, 2=use wav input */ } options; @@ -1,6 +1,9 @@ #include <stdio.h> #include <stdlib.h> #include <string.h> +#include <pthread.h> +#include <jack/jack.h> +#include <jack/ringbuffer.h> #include "common.h" #include "encoder.h" #include "musicin.h" @@ -24,7 +27,7 @@ #include <assert.h> -FILE *musicin; +music_in_t musicin; Bit_stream_struc bs; char *programName; char toolameversion[] = "0.2l-opendigitalradio"; @@ -44,6 +47,7 @@ void global_init (void) glopts.vbrlevel = 0; glopts.athlevel = 0; glopts.verbosity = 2; + glopts.enable_jack = 0; } /************************************************************************ @@ -194,7 +198,7 @@ int main (int argc, char **argv) nch = frame.nch; error_protection = header.error_protection; - while (get_audio (musicin, buffer, num_samples, nch, &header) > 0) { + while (get_audio (&musicin, buffer, num_samples, nch, &header) > 0) { if (glopts.verbosity > 1) if (++frameNum % 10 == 0) { if (mot_file) { @@ -535,9 +539,11 @@ int main (int argc, char **argv) (FLOAT) sentBits / (frameNum * 1152) * s_freq[header.version][header.sampling_frequency]); - if (fclose (musicin) != 0) { - fprintf (stderr, "Could not close \"%s\".\n", original_file_name); - exit (2); + if (!glopts.enable_jack) { + if ( fclose (musicin.wav_input) != 0) { + fprintf (stderr, "Could not close \"%s\".\n", original_file_name); + exit (2); + } } fprintf (stderr, "\nDone\n"); @@ -561,9 +567,16 @@ void print_config (frame_info * frame, int *psy, char *inPath, return; fprintf (stderr, "--------------------------------------------\n"); - fprintf (stderr, "Input File : '%s' %.1f kHz\n", - (strcmp (inPath, "-") ? inPath : "stdin"), - s_freq[header->version][header->sampling_frequency]); + if (glopts.enable_jack) { + fprintf (stderr, "Input JACK\n"); + fprintf (stderr, " name %s\n", musicin.jack_name); + } + else { + fprintf (stderr, "Input File : '%s' %.1f kHz\n", + (strcmp (inPath, "-") ? inPath : "stdin"), + s_freq[header->version][header->sampling_frequency]); + } + fprintf (stderr, "Output File: '%s'\n", (strcmp (outPath, "-") ? outPath : "stdout")); fprintf (stderr, "%d kbps ", bitrate[header->version][header->bitrate_index]); @@ -620,6 +633,7 @@ void usage (void) fprintf (stdout, "\t-a downmix from stereo to mono\n"); fprintf (stdout, "\t-x force byte-swapping of input\n"); fprintf (stdout, "\t-g swap channels of input file\n"); + fprintf (stdout, "\t-j enable jack input\n"); fprintf (stdout, "Output\n"); fprintf (stdout, "\t-m mode channel mode : s/d/j/m (dflt %4c)\n", DFLT_MOD); @@ -670,7 +684,7 @@ void short_usage (void) fprintf (stderr, "tooLAME version %s\n (http://opendigitalradio.org)\n", toolameversion); fprintf (stderr, "MPEG Audio Layer II encoder for DAB\n\n"); - fprintf (stderr, "USAGE: %s [options] <infile> [outfile]\n\n", programName); + fprintf (stderr, "USAGE: %s [options] (<infile>|<jackname>) [outfile]\n\n", programName); fprintf (stderr, "Try \"%s -h\" for more information.\n", programName); exit (0); } @@ -686,6 +700,7 @@ void short_usage (void) * SEMANTICS: The command line is parsed according to the following * syntax: * + * -j turns on JACK input * -m is followed by the mode * -y is followed by the psychoacoustic model number * -s is followed by the sampling rate @@ -832,6 +847,10 @@ void parse_args (int argc, char **argv, frame_info * frame, int *psy, err = 1; break; + case 'j': + glopts.enable_jack = 1; + break; + case 'b': argUsed = 1; brate = atoi (arg); @@ -958,23 +977,34 @@ void parse_args (int argc, char **argv, frame_info * frame, int *psy, } - if (err || inPath[0] == '\0') - usage (); /* If no infile defined, or err has occured, then call usage() */ + if (err) + usage (); /* If err has occured, then call usage() */ + + if (!glopts.enable_jack && inPath[0] == '\0') + usage (); /* If not in jack-mode and no file specified, then call usage() */ if (outPath[0] == '\0') { /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */ new_ext (inPath, DFLT_EXT, outPath); } - if (!strcmp (inPath, "-")) { - musicin = stdin; /* read from stdin */ + if (glopts.enable_jack) { + musicin.jack_name = inPath; *num_samples = MAX_U_32_NUM; - } else { - if ((musicin = fopen (inPath, "rb")) == NULL) { - fprintf (stderr, "Could not find \"%s\".\n", inPath); - exit (1); + + setup_jack(header, musicin.jack_name); + } + else { + if (!strcmp (inPath, "-")) { + musicin.wav_input = stdin; /* read from stdin */ + *num_samples = MAX_U_32_NUM; + } else { + if ((musicin.wav_input = fopen (inPath, "rb")) == NULL) { + fprintf (stderr, "Could not find \"%s\".\n", inPath); + exit (1); + } + parse_input_file (musicin.wav_input, inPath, header, num_samples); } - parse_input_file (musicin, inPath, header, num_samples); } /* check for a valid bitrate */ @@ -12,5 +12,5 @@ void print_config (frame_info *, int *, void usage (void); - void smr_dump(double smr[2][SBLIMIT], int nch); + |