diff options
| -rw-r--r-- | Makefile.am | 1 | ||||
| -rw-r--r-- | codec.c | 42 | ||||
| -rw-r--r-- | codec.h | 65 | ||||
| -rw-r--r-- | main.c | 3 | ||||
| -rw-r--r-- | playlist.c | 1 | ||||
| -rw-r--r-- | plugins/flac/flac.c | 2 | ||||
| -rw-r--r-- | streamer.c | 291 |
7 files changed, 183 insertions, 222 deletions
diff --git a/Makefile.am b/Makefile.am index 7509601a..1276047a 100644 --- a/Makefile.am +++ b/Makefile.am @@ -34,7 +34,6 @@ deadbeef_SOURCES =\ plugins.c plugins.h moduleconf.h\ playlist.c playlist.h \ streamer.c streamer.h\ - codec.c codec.h\ messagepump.c messagepump.h\ conf.c conf.h\ cgme.c cdumb.c csid.cpp\ diff --git a/codec.c b/codec.c deleted file mode 100644 index 686c5bcf..00000000 --- a/codec.c +++ /dev/null @@ -1,42 +0,0 @@ -/* - DeaDBeeF - ultimate music player for GNU/Linux systems with X11 - Copyright (C) 2009-2010 Alexey Yakovenko <waker@users.sourceforge.net> - - This program is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 2 of the License, or - (at your option) any later version. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with this program. If not, see <http://www.gnu.org/licenses/>. -*/ -#include "codec.h" -#include "threading.h" - -static uintptr_t mutex; - -void -codec_init_locking (void) { - mutex = mutex_create (); -} - -void -codec_free_locking (void) { - mutex_free (mutex); -} - -void -codec_lock (void) { - mutex_lock (mutex); -} - -void -codec_unlock (void) { - mutex_unlock (mutex); -} - diff --git a/codec.h b/codec.h deleted file mode 100644 index 1b6ad2a3..00000000 --- a/codec.h +++ /dev/null @@ -1,65 +0,0 @@ -/* - DeaDBeeF - ultimate music player for GNU/Linux systems with X11 - Copyright (C) 2009-2010 Alexey Yakovenko <waker@users.sourceforge.net> - - This program is free software: you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation, either version 2 of the License, or - (at your option) any later version. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with this program. If not, see <http://www.gnu.org/licenses/>. -*/ -#ifndef __CODEC_H -#define __CODEC_H - -#include <stdint.h> -//#include "playlist.h" - -#if 0 -typedef struct { - int bitsPerSample; - int channels; - int samplesPerSecond; -// float position; - float readposition; -} fileinfo_t; - -struct playItem_s; - -typedef struct codec_s { - int (*init) (struct playItem_s *it); - void (*free) (void); - // player is responsible for starting next song if -1 is returned - int (*read) (char *bytes, int size); - int (*seek) (float time); - struct playItem_s * (*insert) (struct playItem_s *after, const char *fname); // after==NULL means "prepend to beginning" - const char ** (*getexts) (void); - int (*numvoices) (void); - void (*mutevoice) (int voice, int mute); - const char *id; // codec id used for playlist serialization - const char *filetypes[20]; // NULL terminated array of const strings, representing supported file types (can be NULL) - fileinfo_t info; -} codec_t; -#endif - -//codec_t *get_codec_for_file (const char *fname); - -void -codec_init_locking (void); - -void -codec_free_locking (void); - -void -codec_lock (void); - -void -codec_unlock (void); - -#endif // __CODEC_H @@ -45,7 +45,6 @@ #include "unistd.h" #include "threading.h" #include "messagepump.h" -#include "codec.h" #include "streamer.h" #include "conf.h" #include "volume.h" @@ -605,7 +604,6 @@ main (int argc, char *argv[]) { } plug_trigger_event_playlistchanged (); session_load (sessfile); - codec_init_locking (); streamer_init (); // this runs in main thread (blocks right here) @@ -629,7 +627,6 @@ main (int argc, char *argv[]) { plug_unload_all (); // at this point we can simply do exit(0), but let's clean up for debugging - codec_free_locking (); session_save (sessfile); pl_free (); conf_free (); @@ -31,7 +31,6 @@ #endif #include <limits.h> #include "playlist.h" -#include "codec.h" #include "streamer.h" #include "messagepump.h" #include "playback.h" diff --git a/plugins/flac/flac.c b/plugins/flac/flac.c index b45b0205..82656795 100644 --- a/plugins/flac/flac.c +++ b/plugins/flac/flac.c @@ -366,7 +366,7 @@ cflac_read_float32 (char *bytes, int size) { do { if (remaining) { int n_input_frames = remaining / sizeof (float) / n_output_channels; - int n_output_frames = size / n_output_channels / sizeof (int16_t); + int n_output_frames = size / n_output_channels / sizeof (float); int n = min (n_input_frames, n_output_frames); float *in = (float *)buffer; @@ -26,7 +26,6 @@ #endif #include <sys/time.h> #include "threading.h" -#include "codec.h" #include "playlist.h" #include "common.h" #include "streamer.h" @@ -38,26 +37,24 @@ #include "volume.h" #include "vfs.h" -#define trace(...) { fprintf(stderr, __VA_ARGS__); } -//#define trace(fmt,...) +//#define trace(...) { fprintf(stderr, __VA_ARGS__); } +#define trace(fmt,...) static intptr_t streamer_tid; static int src_quality; static SRC_STATE *src; static SRC_DATA srcdata; -static int codecleft; +static int src_remaining; // number of input samples in SRC buffer static int conf_replaygain_mode = 0; static int conf_replaygain_scale = 1; -// that's buffer for resampling. -// our worst case is 192KHz downsampling to 22050Hz with 2048 sample output buffer -#define INPUT_BUFFER_SIZE (2048*192000/22050*8) -static char g_readbuffer[INPUT_BUFFER_SIZE]; -// fbuffer contains readbuffer converted to floating point, to pass to SRC -static float g_fbuffer[INPUT_BUFFER_SIZE]; -// output SRC buffer - can't really exceed INPUT_BUFFER_SIZE -#define SRC_BUFFER_SIZE (INPUT_BUFFER_SIZE*2) -static float g_srcbuffer[SRC_BUFFER_SIZE]; + +#define SRC_BUFFER 16000 +static int src_in_remaining = 0; +//static int16_t g_src_in_buffer[SRC_BUFFER*2]; +static float g_src_in_fbuffer[SRC_BUFFER*2]; +static float g_src_out_fbuffer[SRC_BUFFER*2]; + static int streaming_terminate; // buffer up to 3 seconds at 44100Hz stereo @@ -256,7 +253,7 @@ streamer_thread (void *ctx) { #ifdef __linux__ prctl (PR_SET_NAME, "deadbeef-stream", 0, 0, 0, 0); #endif - codecleft = 0; + src_remaining = 0; while (!streaming_terminate) { struct timeval tm1; @@ -266,9 +263,7 @@ streamer_thread (void *ctx) { int sng = nextsong; int pstate = nextsong_pstate; nextsong = -1; - codec_lock (); - codecleft = 0; - codec_unlock (); + src_remaining = 0; if (badsong == sng) { trace ("looped to bad file. stopping...\n"); streamer_set_nextsong (-2, 1); @@ -449,9 +444,7 @@ streamer_thread (void *ctx) { streamer_lock (); streambuffer_fill = 0; streambuffer_pos = 0; - codec_lock (); - codecleft = 0; - codec_unlock (); + src_remaining = 0; if (str_playing_song.decoder->seek (pos) >= 0) { playpos = str_playing_song.decoder->info.readpos; } @@ -543,7 +536,7 @@ streamer_free (void) { void streamer_reset (int full) { // must be called when current song changes by external reasons - codecleft = 0; + src_remaining = 0; if (full) { streambuffer_pos = 0; streambuffer_fill = 0; @@ -694,17 +687,177 @@ float32_to_int16 (float *in, int16_t *out, int nsamples) { fpu_restore (ctl); } +/* + + src algorithm + + initsize = size; + while (size > 0) { + need_frames = SRC_BUFFER - src_remaining + read_samples (need_frames) + src_process (output_frames) + convert_to_int16 (bytes, size) + // handle errors + } + return initsize-size; + +*/ + +static int +streamer_read_data_for_src (int16_t *buffer, int frames) { + DB_decoder_t *decoder = str_streaming_song.decoder; + int channels = decoder->info.channels; + if (channels > 2) { + channels = 2; + } + int bytesread = decoder->read_int16 ((uint8_t*)buffer, frames * sizeof (int16_t) * channels); + if (channels == 1) { + // convert to stereo + int n = frames-1; + while (n >= 0) { + buffer[n*2+0] = buffer[n]; + buffer[n*2+1] = buffer[n]; + } + } + return bytesread / (sizeof (int16_t) * channels); +} + +static int +streamer_read_data_for_src_float (float *buffer, int frames) { + DB_decoder_t *decoder = str_streaming_song.decoder; + int channels = decoder->info.channels; + if (channels > 2) { + channels = 2; + } + if (decoder->read_float32) { +// trace ("call read_float32 (%d frames -> %d bytes)\n", frames, frames * sizeof (float) * channels); + int bytesread = decoder->read_float32 ((uint8_t*)buffer, frames * sizeof (float) * channels); +// trace ("got %d bytes -> %d frames\n", bytesread, bytesread / (sizeof (float) * channels)); + if (channels == 1) { + // convert to stereo + int n = frames-1; + while (n >= 0) { + buffer[n*2+0] = buffer[n]; + buffer[n*2+1] = buffer[n]; + } + bytesread *= 2; + } + return bytesread / (sizeof (float) * channels); + } +// trace ("read_float32 not impl\n"); + int16_t intbuffer[frames*2]; + int res = streamer_read_data_for_src (intbuffer, frames); + for (int i = 0; i < res; i++) { + buffer[i*2+0] = intbuffer[i*2+0]/(float)0x7fff; + buffer[i*2+1] = intbuffer[i*2+1]/(float)0x7fff; + } + return res; +} + +static int +streamer_decode_src_libsamplerate (uint8_t *bytes, int size) { + int initsize = size; + int16_t *out = (int16_t *)bytes; + DB_decoder_t *decoder = str_streaming_song.decoder; + int samplerate = decoder->info.samplerate; + float ratio = (float)p_get_rate ()/samplerate; + while (size > 0) { + int n_output_frames = size / sizeof (int16_t) / 2; + int n_input_frames = n_output_frames * samplerate / p_get_rate () + 100; + // read data from decoder + if (n_input_frames > SRC_BUFFER - src_remaining ) { + n_input_frames = SRC_BUFFER - src_remaining; + } + int nread = streamer_read_data_for_src_float (&g_src_in_fbuffer[src_remaining*2], n_input_frames); + src_remaining += nread; + // resample + + srcdata.data_in = g_src_in_fbuffer; + srcdata.data_out = g_src_out_fbuffer; + srcdata.input_frames = src_remaining; + srcdata.output_frames = size/4; + srcdata.src_ratio = ratio; +// trace ("SRC from %d to %d\n", samplerate, p_get_rate ()); + srcdata.end_of_input = (nread == n_input_frames) ? 0 : 1; + src_process (src, &srcdata); + // convert back to s16 format +// trace ("out frames: %d\n", srcdata.output_frames_gen); + int genbytes = srcdata.output_frames_gen * 4; + int bytesread = min(size, genbytes); +// trace ("bytesread: %d\n", bytesread); + float32_to_int16 ((float*)g_src_out_fbuffer, (int16_t*)bytes, bytesread>>1); + size -= bytesread; +// trace ("size: %d\n", size); + bytes += bytesread; + // calculate how many unused input samples left + src_remaining -= srcdata.input_frames_used; + + // copy spare samples for next update + if (src_remaining > 0) { + memmove (g_src_in_fbuffer, &g_src_in_fbuffer[srcdata.input_frames_used*2], src_remaining * sizeof (float) * 2); + } + + if (nread != n_input_frames) { + trace ("nread=%d, n_input_frames=%d, mismatch!\n", nread, n_input_frames); + break; + } + } + return initsize-size; +} + +#if 0 +static int +streamer_decode_src (uint8_t *bytes, int size) { + int initsize = size; + int16_t *out = (int16_t *)bytes; + DB_decoder_t *decoder = str_streaming_song.decoder; + int samplerate = decoder->info.samplerate; + float ratio = (float)samplerate / p_get_rate (); + while (size > 0) { + int n_output_frames = size / sizeof (int16_t) / 2; + int n_input_frames = n_output_frames * samplerate / p_get_rate () + 100; + // read data from decoder + if (n_input_frames > SRC_BUFFER - src_remaining ) { + n_input_frames = SRC_BUFFER - src_remaining; + } + int nread = streamer_read_data_for_src (&g_src_in_buffer[src_remaining*2], n_input_frames); + src_remaining += nread; + // resample + float idx = 0; + int i = 0; + while (i < n_output_frames && idx < src_remaining) { + int k = (int)idx; + out[0] = g_src_in_buffer[k*2+0]; + out[1] = g_src_in_buffer[k*2+1]; + i++; + idx += ratio; + out += 2; + } + size -= i*4; + src_remaining -= idx; + if (src_remaining < 0) { + src_remaining = 0; + } + else if (src_remaining > 0) { + memmove (g_src_in_buffer, &g_src_in_buffer[(SRC_BUFFER-src_remaining)*2], src_remaining*2*sizeof (int16_t)); + } + if (nread != n_input_frames) { + break; + } + } + return initsize-size; +} +#endif + // returns number of bytes been read static int streamer_read_async (char *bytes, int size) { int initsize = size; for (;;) { int bytesread = 0; - codec_lock (); DB_decoder_t *decoder = str_streaming_song.decoder; if (!decoder) { // means there's nothing left to stream, so just do nothing - codec_unlock (); break; } if (decoder->info.samplerate != -1) { @@ -713,108 +866,28 @@ streamer_read_async (char *bytes, int size) { nchannels = 2; } int samplerate = decoder->info.samplerate; - if (decoder->info.samplerate == p_get_rate ()) { + if (decoder->info.samplerate == p_get_rate (), 0) { // samplerate match if (nchannels == 2) { bytesread = decoder->read_int16 (bytes, size); apply_replay_gain_int16 (&str_streaming_song, bytes, size); - codec_unlock (); } else { - bytesread = decoder->read_int16 (g_readbuffer, size>>1); - apply_replay_gain_int16 (&str_streaming_song, g_readbuffer, size>>1); - mono_int16_to_stereo_int16 ((int16_t*)g_readbuffer, (int16_t*)bytes, size>>2); + uint8_t buffer[size>>1]; + bytesread = decoder->read_int16 (buffer, size>>1); + apply_replay_gain_int16 (&str_streaming_song, buffer, size>>1); + mono_int16_to_stereo_int16 ((int16_t*)buffer, (int16_t*)bytes, size>>2); bytesread *= 2; - codec_unlock (); } } else if (src_is_valid_ratio (p_get_rate ()/(double)samplerate)) { - // read and do SRC - int nsamples = size/4; - nsamples = nsamples * samplerate / p_get_rate () * 2; - // read data at source samplerate (with some room for SRC) - int nbytes = (nsamples - codecleft) * 2 * nchannels; - int samplesize = 2; - if (nbytes <= 0) { - nbytes = 0; - } - else { - if (!decoder->read_float32) { - if (nbytes > INPUT_BUFFER_SIZE) { - trace ("input buffer overflow\n"); - nbytes = INPUT_BUFFER_SIZE; - } - bytesread = decoder->read_int16 (g_readbuffer, nbytes); - apply_replay_gain_int16 (&str_streaming_song, g_readbuffer, nbytes); - } - else { - samplesize = 4; - } - } - codec_unlock (); - // recalculate nsamples according to how many bytes we've got - if (nbytes != 0) { - if (!decoder->read_float32) { - nsamples = bytesread / (samplesize * nchannels) + codecleft; - // convert to float - float *fbuffer = g_fbuffer + codecleft*2; - int n = nsamples - codecleft; - if (nchannels == 2) { - n <<= 1; - int16_to_float32 ((int16_t*)g_readbuffer, fbuffer, n); - } - else if (nchannels == 1) { // convert mono to stereo - mono_int16_to_stereo_float32 ((int16_t*)g_readbuffer, fbuffer, n); - } - } - else { - float *fbuffer = g_fbuffer + codecleft*2; - if (nchannels == 1) { - codec_lock (); - bytesread = decoder->read_float32 (g_readbuffer, nbytes*2); - codec_unlock (); - apply_replay_gain_float32 (&str_streaming_song, g_readbuffer, nbytes*2); - nsamples = bytesread / (samplesize * nchannels) + codecleft; - mono_float32_to_stereo_float32 ((float *)g_readbuffer, fbuffer, nsamples-codecleft); - } - else { - codec_lock (); - bytesread = decoder->read_float32 ((char *)fbuffer, nbytes*2); - codec_unlock (); - apply_replay_gain_float32 (&str_streaming_song, (char *)fbuffer, nbytes*2); - nsamples = bytesread / (samplesize * nchannels) + codecleft; - } - } - } - //codec_lock (); - // convert samplerate - mutex_lock (decodemutex); - srcdata.data_in = g_fbuffer; - srcdata.data_out = g_srcbuffer; - srcdata.input_frames = nsamples; - srcdata.output_frames = size/4; - srcdata.src_ratio = p_get_rate ()/(double)samplerate; - srcdata.end_of_input = 0; - // src_set_ratio (src, srcdata.src_ratio); - src_process (src, &srcdata); - //codec_unlock (); - // convert back to s16 format - nbytes = size; - int genbytes = srcdata.output_frames_gen * 4; - bytesread = min(size, genbytes); - float32_to_int16 ((float*)g_srcbuffer, (int16_t*)bytes, bytesread>>1); - // calculate how many unused input samples left - codecleft = nsamples - srcdata.input_frames_used; - mutex_unlock (decodemutex); - // copy spare samples for next update - memmove (g_fbuffer, &g_fbuffer[srcdata.input_frames_used*2], codecleft * 8); + bytesread = streamer_decode_src_libsamplerate (bytes, size); } else { fprintf (stderr, "invalid ratio! %d / %d = %f", p_get_rate (), samplerate, p_get_rate ()/(float)samplerate); } } else { - codec_unlock (); } bytes += bytesread; size -= bytesread; |