Mercurial > libavformat.hg
view beosaudio.cpp @ 2136:b53a19eed95d libavformat
remove duplicate PAT scaning code and actual PAT scan
also disabling the nonsense 3yr old rawts change (it would after the
above chage totally break the demuxer if its left enabled)
author | michael |
---|---|
date | Mon, 04 Jun 2007 14:44:52 +0000 |
parents | 62792a60f740 |
children | bbfeec27c98c |
line wrap: on
line source
/* * BeOS audio play interface * Copyright (c) 2000, 2001 Fabrice Bellard. * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include <signal.h> #include <stdlib.h> #include <stdio.h> #include <string.h> #include <unistd.h> #include <sys/time.h> #include <Application.h> #include <SoundPlayer.h> extern "C" { #include "avformat.h" } #ifdef HAVE_BSOUNDRECORDER #include <SoundRecorder.h> using namespace BPrivate::Media::Experimental; #endif /* enable performance checks */ //#define PERF_CHECK /* enable Media Kit latency checks */ //#define LATENCY_CHECK #define AUDIO_BLOCK_SIZE 4096 #define AUDIO_BLOCK_COUNT 8 #define AUDIO_BUFFER_SIZE (AUDIO_BLOCK_SIZE*AUDIO_BLOCK_COUNT) typedef struct { int fd; // UNUSED int sample_rate; int channels; int frame_size; /* in bytes ! */ CodecID codec_id; uint8_t buffer[AUDIO_BUFFER_SIZE]; int buffer_ptr; /* ring buffer */ sem_id input_sem; int input_index; sem_id output_sem; int output_index; BSoundPlayer *player; #ifdef HAVE_BSOUNDRECORDER BSoundRecorder *recorder; #endif int has_quit; /* signal callbacks not to wait */ volatile bigtime_t starve_time; } AudioData; static thread_id main_thid; static thread_id bapp_thid; static int own_BApp_created = 0; static int refcount = 0; /* create the BApplication and Run() it */ static int32 bapp_thread(void *arg) { new BApplication("application/x-vnd.ffmpeg"); own_BApp_created = 1; be_app->Run(); /* kill the process group */ // kill(0, SIGINT); // kill(main_thid, SIGHUP); return B_OK; } /* create the BApplication only if needed */ static void create_bapp_if_needed(void) { if (refcount++ == 0) { /* needed by libmedia */ if (be_app == NULL) { bapp_thid = spawn_thread(bapp_thread, "ffmpeg BApplication", B_NORMAL_PRIORITY, NULL); resume_thread(bapp_thid); while (!own_BApp_created) snooze(50000); } } } static void destroy_bapp_if_needed(void) { if (--refcount == 0 && own_BApp_created) { be_app->Lock(); be_app->Quit(); be_app = NULL; } } /* called back by BSoundPlayer */ static void audioplay_callback(void *cookie, void *buffer, size_t bufferSize, const media_raw_audio_format &format) { AudioData *s; size_t len, amount; unsigned char *buf = (unsigned char *)buffer; s = (AudioData *)cookie; if (s->has_quit) return; while (bufferSize > 0) { #ifdef PERF_CHECK bigtime_t t; t = system_time(); #endif len = MIN(AUDIO_BLOCK_SIZE, bufferSize); if (acquire_sem_etc(s->output_sem, len, B_CAN_INTERRUPT, 0LL) < B_OK) { s->has_quit = 1; s->player->SetHasData(false); return; } amount = MIN(len, (AUDIO_BUFFER_SIZE - s->output_index)); memcpy(buf, &s->buffer[s->output_index], amount); s->output_index += amount; if (s->output_index >= AUDIO_BUFFER_SIZE) { s->output_index %= AUDIO_BUFFER_SIZE; memcpy(buf + amount, &s->buffer[s->output_index], len - amount); s->output_index += len-amount; s->output_index %= AUDIO_BUFFER_SIZE; } release_sem_etc(s->input_sem, len, 0); #ifdef PERF_CHECK t = system_time() - t; s->starve_time = MAX(s->starve_time, t); #endif buf += len; bufferSize -= len; } } #ifdef HAVE_BSOUNDRECORDER /* called back by BSoundRecorder */ static void audiorecord_callback(void *cookie, bigtime_t timestamp, void *buffer, size_t bufferSize, const media_multi_audio_format &format) { AudioData *s; size_t len, amount; unsigned char *buf = (unsigned char *)buffer; s = (AudioData *)cookie; if (s->has_quit) return; while (bufferSize > 0) { len = MIN(bufferSize, AUDIO_BLOCK_SIZE); //printf("acquire_sem(input, %d)\n", len); if (acquire_sem_etc(s->input_sem, len, B_CAN_INTERRUPT, 0LL) < B_OK) { s->has_quit = 1; return; } amount = MIN(len, (AUDIO_BUFFER_SIZE - s->input_index)); memcpy(&s->buffer[s->input_index], buf, amount); s->input_index += amount; if (s->input_index >= AUDIO_BUFFER_SIZE) { s->input_index %= AUDIO_BUFFER_SIZE; memcpy(&s->buffer[s->input_index], buf + amount, len - amount); s->input_index += len - amount; } release_sem_etc(s->output_sem, len, 0); //printf("release_sem(output, %d)\n", len); buf += len; bufferSize -= len; } } #endif static int audio_open(AudioData *s, int is_output, const char *audio_device) { int p[2]; int ret; media_raw_audio_format format; media_multi_audio_format iformat; #ifndef HAVE_BSOUNDRECORDER if (!is_output) return AVERROR(EIO); /* not for now */ #endif s->input_sem = create_sem(AUDIO_BUFFER_SIZE, "ffmpeg_ringbuffer_input"); if (s->input_sem < B_OK) return AVERROR(EIO); s->output_sem = create_sem(0, "ffmpeg_ringbuffer_output"); if (s->output_sem < B_OK) { delete_sem(s->input_sem); return AVERROR(EIO); } s->input_index = 0; s->output_index = 0; create_bapp_if_needed(); s->frame_size = AUDIO_BLOCK_SIZE; /* bump up the priority (avoid realtime though) */ set_thread_priority(find_thread(NULL), B_DISPLAY_PRIORITY+1); #ifdef HAVE_BSOUNDRECORDER if (!is_output) { bool wait_for_input = false; if (audio_device && !strcmp(audio_device, "wait:")) wait_for_input = true; s->recorder = new BSoundRecorder(&iformat, wait_for_input, "ffmpeg input", audiorecord_callback); if (wait_for_input && (s->recorder->InitCheck() == B_OK)) { s->recorder->WaitForIncomingConnection(&iformat); } if (s->recorder->InitCheck() != B_OK || iformat.format != media_raw_audio_format::B_AUDIO_SHORT) { delete s->recorder; s->recorder = NULL; if (s->input_sem) delete_sem(s->input_sem); if (s->output_sem) delete_sem(s->output_sem); return AVERROR(EIO); } s->codec_id = (iformat.byte_order == B_MEDIA_LITTLE_ENDIAN)?CODEC_ID_PCM_S16LE:CODEC_ID_PCM_S16BE; s->channels = iformat.channel_count; s->sample_rate = (int)iformat.frame_rate; s->frame_size = iformat.buffer_size; s->recorder->SetCookie(s); s->recorder->SetVolume(1.0); s->recorder->Start(); return 0; } #endif format = media_raw_audio_format::wildcard; format.format = media_raw_audio_format::B_AUDIO_SHORT; format.byte_order = B_HOST_IS_LENDIAN ? B_MEDIA_LITTLE_ENDIAN : B_MEDIA_BIG_ENDIAN; format.channel_count = s->channels; format.buffer_size = s->frame_size; format.frame_rate = s->sample_rate; s->player = new BSoundPlayer(&format, "ffmpeg output", audioplay_callback); if (s->player->InitCheck() != B_OK) { delete s->player; s->player = NULL; if (s->input_sem) delete_sem(s->input_sem); if (s->output_sem) delete_sem(s->output_sem); return AVERROR(EIO); } s->player->SetCookie(s); s->player->SetVolume(1.0); s->player->Start(); s->player->SetHasData(true); return 0; } static int audio_close(AudioData *s) { if (s->input_sem) delete_sem(s->input_sem); if (s->output_sem) delete_sem(s->output_sem); s->has_quit = 1; if (s->player) { s->player->Stop(); } if (s->player) delete s->player; #ifdef HAVE_BSOUNDRECORDER if (s->recorder) delete s->recorder; #endif destroy_bapp_if_needed(); return 0; } /* sound output support */ static int audio_write_header(AVFormatContext *s1) { AudioData *s = (AudioData *)s1->priv_data; AVStream *st; int ret; st = s1->streams[0]; s->sample_rate = st->codec->sample_rate; s->channels = st->codec->channels; ret = audio_open(s, 1, NULL); if (ret < 0) return AVERROR(EIO); return 0; } static int audio_write_packet(AVFormatContext *s1, int stream_index, const uint8_t *buf, int size, int64_t force_pts) { AudioData *s = (AudioData *)s1->priv_data; int len, ret; #ifdef LATENCY_CHECK bigtime_t lat1, lat2; lat1 = s->player->Latency(); #endif #ifdef PERF_CHECK bigtime_t t = s->starve_time; s->starve_time = 0; printf("starve_time: %lld \n", t); #endif while (size > 0) { int amount; len = MIN(size, AUDIO_BLOCK_SIZE); if (acquire_sem_etc(s->input_sem, len, B_CAN_INTERRUPT, 0LL) < B_OK) return AVERROR(EIO); amount = MIN(len, (AUDIO_BUFFER_SIZE - s->input_index)); memcpy(&s->buffer[s->input_index], buf, amount); s->input_index += amount; if (s->input_index >= AUDIO_BUFFER_SIZE) { s->input_index %= AUDIO_BUFFER_SIZE; memcpy(&s->buffer[s->input_index], buf + amount, len - amount); s->input_index += len - amount; } release_sem_etc(s->output_sem, len, 0); buf += len; size -= len; } #ifdef LATENCY_CHECK lat2 = s->player->Latency(); printf("#### BSoundPlayer::Latency(): before= %lld, after= %lld\n", lat1, lat2); #endif return 0; } static int audio_write_trailer(AVFormatContext *s1) { AudioData *s = (AudioData *)s1->priv_data; audio_close(s); return 0; } /* grab support */ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) { AudioData *s = (AudioData *)s1->priv_data; AVStream *st; int ret; if (!ap || ap->sample_rate <= 0 || ap->channels <= 0) return -1; st = av_new_stream(s1, 0); if (!st) { return AVERROR(ENOMEM); } s->sample_rate = ap->sample_rate; s->channels = ap->channels; ret = audio_open(s, 0, s1->filename); if (ret < 0) { av_free(st); return AVERROR(EIO); } /* take real parameters */ st->codec->codec_type = CODEC_TYPE_AUDIO; st->codec->codec_id = s->codec_id; st->codec->sample_rate = s->sample_rate; st->codec->channels = s->channels; return 0; av_set_pts_info(s1, 48, 1, 1000000); /* 48 bits pts in us */ } static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) { AudioData *s = (AudioData *)s1->priv_data; int size; size_t len, amount; unsigned char *buf; status_t err; if (av_new_packet(pkt, s->frame_size) < 0) return AVERROR(EIO); buf = (unsigned char *)pkt->data; size = pkt->size; while (size > 0) { len = MIN(AUDIO_BLOCK_SIZE, size); //printf("acquire_sem(output, %d)\n", len); while ((err=acquire_sem_etc(s->output_sem, len, B_CAN_INTERRUPT, 0LL)) == B_INTERRUPTED); if (err < B_OK) { av_free_packet(pkt); return AVERROR(EIO); } amount = MIN(len, (AUDIO_BUFFER_SIZE - s->output_index)); memcpy(buf, &s->buffer[s->output_index], amount); s->output_index += amount; if (s->output_index >= AUDIO_BUFFER_SIZE) { s->output_index %= AUDIO_BUFFER_SIZE; memcpy(buf + amount, &s->buffer[s->output_index], len - amount); s->output_index += len-amount; s->output_index %= AUDIO_BUFFER_SIZE; } release_sem_etc(s->input_sem, len, 0); //printf("release_sem(input, %d)\n", len); buf += len; size -= len; } //XXX: add pts info return 0; } static int audio_read_close(AVFormatContext *s1) { AudioData *s = (AudioData *)s1->priv_data; audio_close(s); return 0; } static AVInputFormat audio_demuxer = { "audio_device", "audio grab and output", sizeof(AudioData), NULL, audio_read_header, audio_read_packet, audio_read_close, NULL, AVFMT_NOFILE, }; AVOutputFormat audio_muxer = { "audio_device", "audio grab and output", "", "", sizeof(AudioData), #ifdef WORDS_BIGENDIAN CODEC_ID_PCM_S16BE, #else CODEC_ID_PCM_S16LE, #endif CODEC_ID_NONE, audio_write_header, audio_write_packet, audio_write_trailer, AVFMT_NOFILE, }; extern "C" { int audio_init(void) { main_thid = find_thread(NULL); av_register_input_format(&audio_demuxer); av_register_output_format(&audio_muxer); return 0; } } // "C"