diff options
Diffstat (limited to 'audio/wavaudio.c')
-rw-r--r-- | audio/wavaudio.c | 482 |
1 files changed, 482 insertions, 0 deletions
diff --git a/audio/wavaudio.c b/audio/wavaudio.c new file mode 100644 index 0000000..8a500b9 --- /dev/null +++ b/audio/wavaudio.c @@ -0,0 +1,482 @@ +/* + * QEMU WAV audio driver + * + * Copyright (c) 2007 The Android Open Source Project + * Copyright (c) 2004-2005 Vassili Karpov (malc) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#define AUDIO_CAP "wav" +#include "qemu-timer.h" +#include "audio_int.h" +#include "qemu_file.h" + +#define WAV_AUDIO_IN 1 + +/** VOICE OUT (Saving to a .WAV file) + **/ +typedef struct WAVVoiceOut { + HWVoiceOut hw; + QEMUFile *f; + int64_t old_ticks; + void *pcm_buf; + int total_samples; +} WAVVoiceOut; + +static struct { + audsettings_t settings; + const char *wav_path; +} conf_out = { + { + 44100, + 2, + AUD_FMT_S16, + 0 + }, + "qemu.wav" +}; + +static int wav_out_run (HWVoiceOut *hw) +{ + WAVVoiceOut *wav = (WAVVoiceOut *) hw; + int rpos, live, decr, samples; + uint8_t *dst; + st_sample_t *src; + int64_t now = qemu_get_clock (vm_clock); + int64_t ticks = now - wav->old_ticks; + int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec; + + if (bytes > INT_MAX) { + samples = INT_MAX >> hw->info.shift; + } + else { + samples = bytes >> hw->info.shift; + } + + live = audio_pcm_hw_get_live_out (hw); + if (!live) { + return 0; + } + + wav->old_ticks = now; + decr = audio_MIN (live, samples); + samples = decr; + rpos = hw->rpos; + while (samples) { + int left_till_end_samples = hw->samples - rpos; + int convert_samples = audio_MIN (samples, left_till_end_samples); + + src = hw->mix_buf + rpos; + dst = advance (wav->pcm_buf, rpos << hw->info.shift); + + hw->clip (dst, src, convert_samples); + qemu_put_buffer (wav->f, dst, convert_samples << hw->info.shift); + + rpos = (rpos + convert_samples) % hw->samples; + samples -= convert_samples; + wav->total_samples += convert_samples; + } + + hw->rpos = rpos; + return decr; +} + +static int wav_out_write (SWVoiceOut *sw, void *buf, int len) +{ + return audio_pcm_sw_write (sw, buf, len); +} + +/* VICE code: Store number as little endian. */ +static void le_store (uint8_t *buf, uint32_t val, int len) +{ + int i; + for (i = 0; i < len; i++) { + buf[i] = (uint8_t) (val & 0xff); + val >>= 8; + } +} + +static int wav_out_init (HWVoiceOut *hw, audsettings_t *as) +{ + WAVVoiceOut *wav = (WAVVoiceOut *) hw; + int bits16 = 0, stereo = 0; + uint8_t hdr[] = { + 0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, + 0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, + 0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04, + 0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00 + }; + audsettings_t wav_as = conf_out.settings; + + (void) as; + + stereo = wav_as.nchannels == 2; + switch (wav_as.fmt) { + case AUD_FMT_S8: + case AUD_FMT_U8: + bits16 = 0; + break; + + case AUD_FMT_S16: + case AUD_FMT_U16: + bits16 = 1; + break; + + case AUD_FMT_S32: + case AUD_FMT_U32: + dolog ("WAVE files can not handle 32bit formats\n"); + return -1; + } + + hdr[34] = bits16 ? 0x10 : 0x08; + + wav_as.endianness = 0; + audio_pcm_init_info (&hw->info, &wav_as); + + hw->samples = 1024; + wav->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); + if (!wav->pcm_buf) { + dolog ("Could not allocate buffer (%d bytes)\n", + hw->samples << hw->info.shift); + return -1; + } + + le_store (hdr + 22, hw->info.nchannels, 2); + le_store (hdr + 24, hw->info.freq, 4); + le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4); + le_store (hdr + 32, 1 << (bits16 + stereo), 2); + + wav->f = qemu_fopen (conf_out.wav_path, "wb"); + if (!wav->f) { + dolog ("Failed to open wave file `%s'\nReason: %s\n", + conf_out.wav_path, strerror (errno)); + qemu_free (wav->pcm_buf); + wav->pcm_buf = NULL; + return -1; + } + + qemu_put_buffer (wav->f, hdr, sizeof (hdr)); + return 0; +} + +static void wav_out_fini (HWVoiceOut *hw) +{ + WAVVoiceOut *wav = (WAVVoiceOut *) hw; + uint8_t rlen[4]; + uint8_t dlen[4]; + uint32_t datalen = wav->total_samples << hw->info.shift; + uint32_t rifflen = datalen + 36; + + if (!wav->f) { + return; + } + + le_store (rlen, rifflen, 4); + le_store (dlen, datalen, 4); + + qemu_fseek (wav->f, 4, SEEK_SET); + qemu_put_buffer (wav->f, rlen, 4); + + qemu_fseek (wav->f, 32, SEEK_CUR); + qemu_put_buffer (wav->f, dlen, 4); + + qemu_fclose (wav->f); + wav->f = NULL; + + qemu_free (wav->pcm_buf); + wav->pcm_buf = NULL; +} + +static int wav_out_ctl (HWVoiceOut *hw, int cmd, ...) +{ + (void) hw; + (void) cmd; + return 0; +} + + +#if WAV_AUDIO_IN + +/** WAV IN (Reading from a .WAV file) + **/ + + static struct { + const char *wav_path; +} conf_in = { + "qemu.wav" +}; + +typedef struct WAVVoiceIn { + HWVoiceIn hw; + QEMUFile* f; + int64_t old_ticks; + void* pcm_buf; + int total_samples; + int total_size; +} WAVVoiceIn; + + +static int +le_read( const uint8_t* p, int size ) { + int shift = 0; + int result = 0; + for ( ; size > 0; size-- ) { + result = result | (p[0] << shift); + p += 1; + shift += 8; + } + return result; +} + +static int +wav_in_init (HWVoiceIn *hw, audsettings_t *as) +{ + WAVVoiceIn* wav = (WAVVoiceIn *) hw; + const char* path = conf_in.wav_path; + uint8_t hdr[44]; + audsettings_t wav_as = *as; + int nchannels, freq, format, bits; + + wav->f = qemu_fopen (path, "rb"); + if (wav->f == NULL) { + dolog("Failed to open wave file '%s'\nReason: %s\n", path, + strerror(errno)); + return -1; + } + + if (qemu_get_buffer (wav->f, hdr, sizeof(hdr)) != (int)sizeof(hdr)) { + dolog("File '%s' to be a .wav file\n", path); + goto Fail; + } + + /* check that this is a wave file */ + if ( hdr[0] != 'R' || hdr[1] != 'I' || hdr[2] != 'F' || hdr[3] != 'F' || + hdr[8] != 'W' || hdr[9] != 'A' || hdr[10]!= 'V' || hdr[11]!= 'E' || + hdr[12]!= 'f' || hdr[13]!= 'm' || hdr[14]!= 't' || hdr[15]!= ' ' || + hdr[40]!= 'd' || hdr[41]!= 'a' || hdr[42]!= 't' || hdr[43]!= 'a') { + dolog("File '%s' is not a valid .wav file\n", path); + goto Fail; + } + + nchannels = le_read( hdr+22, 2 ); + freq = le_read( hdr+24, 4 ); + format = le_read( hdr+32, 2 ); + bits = le_read( hdr+34, 2 ); + + wav->total_size = le_read( hdr+40, 4 ); + + /* perform some sainty checks */ + switch (nchannels) { + case 1: + case 2: break; + default: + dolog("unsupported number of channels (%d) in '%s'\n", + nchannels, path); + goto Fail; + } + + switch (format) { + case 1: + case 2: + case 4: break; + default: + dolog("unsupported bytes per sample (%d) in '%s'\n", + format, path); + goto Fail; + } + + if (format*8/nchannels != bits) { + dolog("invalid bits per sample (%d, expected %d) in '%s'\n", + bits, format*8/nchannels, path); + goto Fail; + } + + wav_as.nchannels = nchannels; + wav_as.fmt = (bits == 8) ? AUD_FMT_U8 : AUD_FMT_S16; + wav_as.freq = freq; + wav_as.endianness = 0; /* always little endian */ + + audio_pcm_init_info (&hw->info, &wav_as); + + hw->samples = 1024; + wav->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); + if (!wav->pcm_buf) { + goto Fail; + } + return 0; + +Fail: + qemu_fclose (wav->f); + wav->f = NULL; + return -1; +} + + +static void wav_in_fini (HWVoiceIn *hw) +{ + WAVVoiceIn *wav = (WAVVoiceIn *) hw; + + if (!wav->f) { + return; + } + + qemu_fclose (wav->f); + wav->f = NULL; + + qemu_free (wav->pcm_buf); + wav->pcm_buf = NULL; +} + +static int wav_in_run (HWVoiceIn *hw) +{ + WAVVoiceIn* wav = (WAVVoiceIn *) hw; + int wpos, live, decr, samples; + uint8_t* src; + st_sample_t* dst; + + int64_t now = qemu_get_clock (vm_clock); + int64_t ticks = now - wav->old_ticks; + int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec; + + if (bytes > INT_MAX) { + samples = INT_MAX >> hw->info.shift; + } + else { + samples = bytes >> hw->info.shift; + } + + live = audio_pcm_hw_get_live_in (hw); + if (!live) { + return 0; + } + + wav->old_ticks = now; + + decr = audio_MIN (live, samples); + samples = decr; + wpos = hw->wpos; + while (samples) { + int left_till_end_samples = hw->samples - wpos; + int convert_samples = audio_MIN (samples, left_till_end_samples); + + dst = hw->conv_buf + wpos; + src = advance (wav->pcm_buf, wpos << hw->info.shift); + + qemu_get_buffer (wav->f, src, convert_samples << hw->info.shift); + memcpy (dst, src, convert_samples << hw->info.shift); + + wpos = (wpos + convert_samples) % hw->samples; + samples -= convert_samples; + wav->total_samples += convert_samples; + } + + hw->wpos = wpos; + return decr; +} + +static int wav_in_read (SWVoiceIn *sw, void *buf, int len) +{ + return audio_pcm_sw_read (sw, buf, len); +} + +static int wav_in_ctl (HWVoiceIn *hw, int cmd, ...) +{ + (void) hw; + (void) cmd; + return 0; +} + +#endif /* WAV_AUDIO_IN */ + +/** COMMON CODE + **/ +static void *wav_audio_init (void) +{ + return &conf_out; +} + +static void wav_audio_fini (void *opaque) +{ + (void) opaque; + ldebug ("wav_fini"); +} + +struct audio_option wav_options[] = { + {"FREQUENCY", AUD_OPT_INT, &conf_out.settings.freq, + "Frequency", NULL, 0}, + + {"FORMAT", AUD_OPT_FMT, &conf_out.settings.fmt, + "Format", NULL, 0}, + + {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &conf_out.settings.nchannels, + "Number of channels (1 - mono, 2 - stereo)", NULL, 0}, + + {"PATH", AUD_OPT_STR, &conf_out.wav_path, + "Path to output .wav file", NULL, 0}, + +#if WAV_AUDIO_IN + {"IN_PATH", AUD_OPT_STR, &conf_in.wav_path, + "Path to input .wav file", NULL, 0}, +#endif + {NULL, 0, NULL, NULL, NULL, 0} +}; + +struct audio_pcm_ops wav_pcm_ops = { + wav_out_init, + wav_out_fini, + wav_out_run, + wav_out_write, + wav_out_ctl, + +#if WAV_AUDIO_IN + wav_in_init, + wav_in_fini, + wav_in_run, + wav_in_read, + wav_in_ctl +#else + NULL, + NULL, + NULL, + NULL, + NULL +#endif +}; + +struct audio_driver wav_audio_driver = { + INIT_FIELD (name = ) "wav", + INIT_FIELD (descr = ) + "WAV file read/write (www.wikipedia.org/wiki/WAV)", + INIT_FIELD (options = ) wav_options, + INIT_FIELD (init = ) wav_audio_init, + INIT_FIELD (fini = ) wav_audio_fini, + INIT_FIELD (pcm_ops = ) &wav_pcm_ops, + INIT_FIELD (can_be_default = ) 0, +#if WAV_AUDIO_IN + INIT_FIELD (max_voices_in = ) 1, + INIT_FIELD (max_voices_out = ) 1, + INIT_FIELD (voice_size_out = ) sizeof (WAVVoiceOut), + INIT_FIELD (voice_size_in = ) sizeof (WAVVoiceIn) +#else + INIT_FIELD (max_voices_out = ) 1, + INIT_FIELD (max_voices_in = ) 0, + INIT_FIELD (voice_size_out = ) sizeof (WAVVoiceOut), + INIT_FIELD (voice_size_in = ) 0 +#endif +}; |