/** codec.c * * Copyright (C) 2013-2015 Tox project All Rights Reserved. * * This file is part of Tox. * * Tox 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 3 of the License, or * (at your option) any later version. * * Tox 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 Tox. If not, see . * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif /* HAVE_CONFIG_H */ #include "../toxcore/logger.h" #include "../toxcore/util.h" #include #include #include #include #include #include #include "msi.h" #include "rtp.h" #include "codec.h" #define DEFAULT_JBUF 6 /* Good quality encode. */ #define MAX_DECODE_TIME_US 0 #define MAX_VIDEOFRAME_SIZE 0x40000 /* 256KiB */ #define VIDEOFRAME_HEADER_SIZE 0x2 /* FIXME: Might not be enough */ #define VIDEO_DECODE_BUFFER_SIZE 20 #define ARRAY(TYPE__) struct { uint16_t size; TYPE__ data[]; } typedef ARRAY(uint8_t) Payload; typedef struct { uint16_t size; /* Max size */ uint16_t start; uint16_t end; Payload **packets; } PayloadBuffer; static bool buffer_full(const PayloadBuffer *b) { return (b->end + 1) % b->size == b->start; } static bool buffer_empty(const PayloadBuffer *b) { return b->end == b->start; } static void buffer_write(PayloadBuffer *b, Payload *p) { b->packets[b->end] = p; b->end = (b->end + 1) % b->size; if (b->end == b->start) b->start = (b->start + 1) % b->size; /* full, overwrite */ } static void buffer_read(PayloadBuffer *b, Payload **p) { *p = b->packets[b->start]; b->start = (b->start + 1) % b->size; } static void buffer_clear(PayloadBuffer *b) { while (!buffer_empty(b)) { Payload *p; buffer_read(b, &p); free(p); } } static PayloadBuffer *buffer_new(int size) { PayloadBuffer *buf = calloc(sizeof(PayloadBuffer), 1); if (!buf) return NULL; buf->size = size + 1; /* include empty elem */ if (!(buf->packets = calloc(buf->size, sizeof(Payload *)))) { free(buf); return NULL; } return buf; } static void buffer_free(PayloadBuffer *b) { if (b) { buffer_clear(b); free(b->packets); free(b); } } /* JITTER BUFFER WORK */ typedef struct JitterBuffer_s { RTPMessage **queue; uint32_t size; uint32_t capacity; uint16_t bottom; uint16_t top; } JitterBuffer; static JitterBuffer *jbuf_new(uint32_t capacity) { unsigned int size = 1; while (size <= (capacity * 4)) { size *= 2; } JitterBuffer *q; if ( !(q = calloc(sizeof(JitterBuffer), 1)) ) return NULL; if (!(q->queue = calloc(sizeof(RTPMessage *), size))) { free(q); return NULL; } q->size = size; q->capacity = capacity; return q; } static void jbuf_clear(JitterBuffer *q) { for (; q->bottom != q->top; ++q->bottom) { if (q->queue[q->bottom % q->size]) { rtp_free_msg(NULL, q->queue[q->bottom % q->size]); q->queue[q->bottom % q->size] = NULL; } } } static void jbuf_free(JitterBuffer *q) { if (!q) return; jbuf_clear(q); free(q->queue); free(q); } static int jbuf_write(JitterBuffer *q, RTPMessage *m) { uint16_t sequnum = m->header->sequnum; unsigned int num = sequnum % q->size; if ((uint32_t)(sequnum - q->bottom) > q->size) { jbuf_clear(q); q->bottom = sequnum - q->capacity; q->queue[num] = m; q->top = sequnum + 1; return 0; } if (q->queue[num]) return -1; q->queue[num] = m; if ((sequnum - q->bottom) >= (q->top - q->bottom)) q->top = sequnum + 1; return 0; } /* Success is 0 when there is nothing to dequeue, * 1 when there's a good packet, * 2 when there's a lost packet */ static RTPMessage *jbuf_read(JitterBuffer *q, int32_t *success) { if (q->top == q->bottom) { *success = 0; return NULL; } unsigned int num = q->bottom % q->size; if (q->queue[num]) { RTPMessage *ret = q->queue[num]; q->queue[num] = NULL; ++q->bottom; *success = 1; return ret; } if ((uint32_t)(q->top - q->bottom) > q->capacity) { ++q->bottom; *success = 2; return NULL; } *success = 0; return NULL; } static int convert_bw_to_sampling_rate(int bw) { switch(bw) { case OPUS_BANDWIDTH_NARROWBAND: return 8000; case OPUS_BANDWIDTH_MEDIUMBAND: return 12000; case OPUS_BANDWIDTH_WIDEBAND: return 16000; case OPUS_BANDWIDTH_SUPERWIDEBAND: return 24000; case OPUS_BANDWIDTH_FULLBAND: return 48000; default: return -1; } } int cs_set_receiving_audio_bitrate(CSSession *cs, int32_t rate) { if (cs->audio_decoder == NULL) return -1; int rc = opus_decoder_ctl(cs->audio_decoder, OPUS_SET_BITRATE(rate)); if ( rc != OPUS_OK ) { LOGGER_ERROR("Error while setting decoder ctl: %s", opus_strerror(rc)); return -1; } LOGGER_DEBUG("Set new decoder bitrate to: %d", rate); return 0; } int cs_set_receiving_audio_sampling_rate(CSSession* cs, int32_t rate) { /* TODO Find a better way? */ if (cs->audio_decoder == NULL) return -1; if (cs->decoder_sample_rate == rate) return 0; int channels = cs->decoder_channels; cs_disable_audio_receiving(cs); cs->decoder_sample_rate = rate; cs->decoder_channels = channels; LOGGER_DEBUG("Set new encoder sampling rate: %d", rate); return cs_enable_audio_receiving(cs); } int cs_set_receiving_audio_channels(CSSession* cs, int32_t count) { /* TODO Find a better way? */ if (cs->audio_decoder == NULL) return -1; if (cs->decoder_channels == count) return 0; int srate = cs->decoder_sample_rate; cs_disable_audio_receiving(cs); cs->decoder_channels = count; cs->decoder_sample_rate = srate; LOGGER_DEBUG("Set new encoder channel count: %d", count); return cs_enable_audio_receiving(cs); } /* PUBLIC */ void cs_do(CSSession *cs) { /* Codec session should always be protected by call mutex so no need to check for cs validity */ if (!cs) return; Payload *p; int rc; int success = 0; pthread_mutex_lock(cs->queue_mutex); if (cs->audio_decoder) { /* If receiving enabled */ RTPMessage *msg; uint16_t fsize = 10000; /* Should be enough for all normal frequences */ int16_t tmp[fsize * 2]; while ((msg = jbuf_read(cs->j_buf, &success)) || success == 2) { pthread_mutex_unlock(cs->queue_mutex); if (success == 2) { rc = opus_decode(cs->audio_decoder, 0, 0, tmp, fsize, 1); } else { /* Get values from packet and decode. * It also checks for validity of an opus packet */ rc = convert_bw_to_sampling_rate(opus_packet_get_bandwidth(msg->data)); if (rc != -1) { cs->last_packet_sampling_rate = rc; cs->last_pack_channels = opus_packet_get_nb_channels(msg->data); cs->last_packet_frame_duration = ( opus_packet_get_samples_per_frame(msg->data, cs->last_packet_sampling_rate) * 1000 ) / cs->last_packet_sampling_rate; } else { LOGGER_WARNING("Failed to load packet values!"); rtp_free_msg(NULL, msg); continue; } cs_set_receiving_audio_sampling_rate(cs, cs->last_packet_sampling_rate); cs_set_receiving_audio_channels(cs, cs->last_pack_channels); LOGGER_DEBUG("Decoding packet of length: %d", msg->length); rc = opus_decode(cs->audio_decoder, msg->data, msg->length, tmp, fsize, 0); rtp_free_msg(NULL, msg); } if (rc < 0) { LOGGER_WARNING("Decoding error: %s", opus_strerror(rc)); } else if (cs->acb.first) { /* Play */ LOGGER_DEBUG("Playing audio frame size: %d chans: %d srate: %d", rc, cs->last_pack_channels, cs->last_packet_sampling_rate); cs->acb.first(cs->agent, cs->friend_id, tmp, rc, cs->last_pack_channels, cs->last_packet_sampling_rate, cs->acb.second); } pthread_mutex_lock(cs->queue_mutex); } } if (cs->vbuf_raw && !buffer_empty(cs->vbuf_raw)) { /* Decode video */ buffer_read(cs->vbuf_raw, &p); /* Leave space for (possibly) other thread to queue more data after we read it here */ pthread_mutex_unlock(cs->queue_mutex); rc = vpx_codec_decode(cs->v_decoder, p->data, p->size, NULL, MAX_DECODE_TIME_US); free(p); if (rc != VPX_CODEC_OK) { LOGGER_ERROR("Error decoding video: %s", vpx_codec_err_to_string(rc)); } else { vpx_codec_iter_t iter = NULL; vpx_image_t *dest = vpx_codec_get_frame(cs->v_decoder, &iter); /* Play decoded images */ for (; dest; dest = vpx_codec_get_frame(cs->v_decoder, &iter)) { if (cs->vcb.first) cs->vcb.first(cs->agent, cs->friend_id, dest->d_w, dest->d_h, (const uint8_t**)dest->planes, dest->stride, cs->vcb.second); vpx_img_free(dest); } } return; } pthread_mutex_unlock(cs->queue_mutex); } CSSession *cs_new(uint32_t peer_video_frame_piece_size) { CSSession *cs = calloc(sizeof(CSSession), 1); if (!cs) { LOGGER_WARNING("Allocation failed! Application might misbehave!"); return NULL; } if (create_recursive_mutex(cs->queue_mutex) != 0) { LOGGER_WARNING("Failed to create recursive mutex!"); free(cs); return NULL; } cs->peer_video_frame_piece_size = peer_video_frame_piece_size; cs->decoder_sample_rate = 48000; cs->decoder_channels = 2; return cs; } void cs_kill(CSSession *cs) { if (!cs) return; /* NOTE: queue_message() will not be called since * the callback is unregistered before cs_kill is called. */ cs_disable_audio_sending(cs); cs_disable_audio_receiving(cs); cs_disable_video_sending(cs); cs_disable_video_receiving(cs); pthread_mutex_destroy(cs->queue_mutex); LOGGER_DEBUG("Terminated codec state: %p", cs); free(cs); } void cs_init_video_splitter_cycle(CSSession* cs) { cs->split_video_frame[0] = cs->frameid_out++; cs->split_video_frame[1] = 0; } int cs_update_video_splitter_cycle(CSSession *cs, const uint8_t *payload, uint16_t length) { cs->processing_video_frame = payload; cs->processing_video_frame_size = length; return ((length - 1) / VIDEOFRAME_PIECE_SIZE) + 1; } const uint8_t *cs_iterate_split_video_frame(CSSession *cs, uint16_t *size) { if (!cs || !size) return NULL; if (cs->processing_video_frame_size > VIDEOFRAME_PIECE_SIZE) { memcpy(cs->split_video_frame + VIDEOFRAME_HEADER_SIZE, cs->processing_video_frame, VIDEOFRAME_PIECE_SIZE); cs->processing_video_frame += VIDEOFRAME_PIECE_SIZE; cs->processing_video_frame_size -= VIDEOFRAME_PIECE_SIZE; *size = VIDEOFRAME_PIECE_SIZE + VIDEOFRAME_HEADER_SIZE; } else { memcpy(cs->split_video_frame + VIDEOFRAME_HEADER_SIZE, cs->processing_video_frame, cs->processing_video_frame_size); *size = cs->processing_video_frame_size + VIDEOFRAME_HEADER_SIZE; } cs->split_video_frame[1]++; return cs->split_video_frame; } int cs_set_sending_video_resolution(CSSession *cs, uint16_t width, uint16_t height) { if (!cs->v_encoding) return -1; /* TODO FIXME reference is safe? */ vpx_codec_enc_cfg_t cfg = *cs->v_encoder[0].config.enc; if (cfg.g_w == width && cfg.g_h == height) return 0; /* if (width * height > cs->max_width * cs->max_height) { vpx_codec_ctx_t v_encoder = cs->v_encoder; if (init_video_encoder(cs, width, height, cs->video_bitrate) == -1) { cs->v_encoder = v_encoder; return cs_ErrorSettingVideoResolution; } vpx_codec_destroy(&v_encoder); return 0; }*/ LOGGER_DEBUG("New video resolution: %u %u", width, height); cfg.g_w = width; cfg.g_h = height; int rc = vpx_codec_enc_config_set(cs->v_encoder, &cfg); if ( rc != VPX_CODEC_OK) { LOGGER_ERROR("Failed to set encoder control setting: %s", vpx_codec_err_to_string(rc)); return cs_ErrorSettingVideoResolution; } return 0; } int cs_set_sending_video_bitrate(CSSession *cs, uint32_t bitrate) { if (!cs->v_encoding) return -1; /* TODO FIXME reference is safe? */ vpx_codec_enc_cfg_t cfg = *cs->v_encoder[0].config.enc; if (cfg.rc_target_bitrate == bitrate) return 0; LOGGER_DEBUG("New video bitrate: %u", bitrate); cfg.rc_target_bitrate = bitrate; int rc = vpx_codec_enc_config_set(cs->v_encoder, &cfg); if ( rc != VPX_CODEC_OK) { LOGGER_ERROR("Failed to set encoder control setting: %s", vpx_codec_err_to_string(rc)); return cs_ErrorSettingVideoBitrate; } return 0; } int cs_enable_video_sending(CSSession* cs, uint32_t bitrate) { if (cs->v_encoding) return 0; vpx_codec_enc_cfg_t cfg; int rc = vpx_codec_enc_config_default(VIDEO_CODEC_ENCODER_INTERFACE, &cfg, 0); if (rc != VPX_CODEC_OK) { LOGGER_ERROR("Failed to get config: %s", vpx_codec_err_to_string(rc)); return -1; } rc = vpx_codec_enc_init_ver(cs->v_encoder, VIDEO_CODEC_ENCODER_INTERFACE, &cfg, 0, VPX_ENCODER_ABI_VERSION); if ( rc != VPX_CODEC_OK) { LOGGER_ERROR("Failed to initialize encoder: %s", vpx_codec_err_to_string(rc)); return -1; } /* So that we can use cs_disable_video_sending to clean up */ cs->v_encoding = true; if ( !(cs->split_video_frame = calloc(VIDEOFRAME_PIECE_SIZE + VIDEOFRAME_HEADER_SIZE, 1)) ) goto FAILURE; cfg.rc_target_bitrate = bitrate; cfg.g_w = 800; cfg.g_h = 600; cfg.g_pass = VPX_RC_ONE_PASS; cfg.g_error_resilient = VPX_ERROR_RESILIENT_DEFAULT | VPX_ERROR_RESILIENT_PARTITIONS; cfg.g_lag_in_frames = 0; cfg.kf_min_dist = 0; cfg.kf_max_dist = 48; cfg.kf_mode = VPX_KF_AUTO; rc = vpx_codec_control(cs->v_encoder, VP8E_SET_CPUUSED, 8); if ( rc != VPX_CODEC_OK) { LOGGER_ERROR("Failed to set encoder control setting: %s", vpx_codec_err_to_string(rc)); goto FAILURE; } return 0; FAILURE: cs_disable_video_sending(cs); return -1; } int cs_enable_video_receiving(CSSession* cs) { if (cs->v_decoding) return 0; int rc = vpx_codec_dec_init_ver(cs->v_decoder, VIDEO_CODEC_DECODER_INTERFACE, NULL, 0, VPX_DECODER_ABI_VERSION); if ( rc != VPX_CODEC_OK) { LOGGER_ERROR("Init video_decoder failed: %s", vpx_codec_err_to_string(rc)); return -1; } /* So that we can use cs_disable_video_sending to clean up */ cs->v_decoding = true; if ( !(cs->frame_buf = calloc(MAX_VIDEOFRAME_SIZE, 1)) ) goto FAILURE; if ( !(cs->vbuf_raw = buffer_new(VIDEO_DECODE_BUFFER_SIZE)) ) goto FAILURE; return 0; FAILURE: cs_disable_video_receiving(cs); return -1; } void cs_disable_video_sending(CSSession* cs) { if (cs->v_encoding) { cs->v_encoding = false; free(cs->split_video_frame); cs->split_video_frame = NULL; vpx_codec_destroy(cs->v_encoder); } } void cs_disable_video_receiving(CSSession* cs) { if (cs->v_decoding) { cs->v_decoding = false; buffer_free(cs->vbuf_raw); cs->vbuf_raw = NULL; free(cs->frame_buf); cs->frame_buf = NULL; vpx_codec_destroy(cs->v_decoder); } } int cs_set_sending_audio_bitrate(CSSession *cs, int32_t rate) { if (cs->audio_encoder == NULL) return -1; int rc = opus_encoder_ctl(cs->audio_encoder, OPUS_SET_BITRATE(rate)); if ( rc != OPUS_OK ) { LOGGER_ERROR("Error while setting encoder ctl: %s", opus_strerror(rc)); return -1; } LOGGER_DEBUG("Set new encoder bitrate to: %d", rate); return 0; } int cs_set_sending_audio_sampling_rate(CSSession* cs, int32_t rate) { /* TODO Find a better way? */ if (cs->audio_encoder == NULL) return -1; if (cs->encoder_sample_rate == rate) return 0; int rc = OPUS_OK; int bitrate = 0; int channels = cs->encoder_channels; rc = opus_encoder_ctl(cs->audio_encoder, OPUS_GET_BITRATE(&bitrate)); if ( rc != OPUS_OK ) { LOGGER_ERROR("Error while getting encoder ctl: %s", opus_strerror(rc)); return -1; } cs_disable_audio_sending(cs); cs->encoder_sample_rate = rate; LOGGER_DEBUG("Set new encoder sampling rate: %d", rate); return cs_enable_audio_sending(cs, bitrate, channels); } int cs_set_sending_audio_channels(CSSession* cs, int32_t count) { /* TODO Find a better way? */ if (cs->audio_encoder == NULL) return -1; if (cs->encoder_channels == count) return 0; int rc = OPUS_OK; int bitrate = 0; rc = opus_encoder_ctl(cs->audio_encoder, OPUS_GET_BITRATE(&bitrate)); if ( rc != OPUS_OK ) { LOGGER_ERROR("Error while getting encoder ctl: %s", opus_strerror(rc)); return -1; } cs_disable_audio_sending(cs); LOGGER_DEBUG("Set new encoder channel count: %d", count); return cs_enable_audio_sending(cs, bitrate, count); } void cs_disable_audio_sending(CSSession* cs) { if ( cs->audio_encoder ) { opus_encoder_destroy(cs->audio_encoder); cs->audio_encoder = NULL; cs->encoder_channels = 0; } } void cs_disable_audio_receiving(CSSession* cs) { if ( cs->audio_decoder ) { opus_decoder_destroy(cs->audio_decoder); cs->audio_decoder = NULL; jbuf_free(cs->j_buf); cs->j_buf = NULL; /* It's used for measuring iteration interval so this has to be some value. * To avoid unecessary checking we set this to 500 */ cs->last_packet_frame_duration = 500; cs->decoder_sample_rate = 48000; cs->decoder_channels = 2; } } int cs_enable_audio_sending(CSSession* cs, uint32_t bitrate, int channels) { if (cs->audio_encoder) return 0; if (!cs->encoder_sample_rate) cs->encoder_sample_rate = 48000; cs->encoder_channels = channels; int rc = OPUS_OK; cs->audio_encoder = opus_encoder_create(cs->encoder_sample_rate, channels, OPUS_APPLICATION_AUDIO, &rc); if ( rc != OPUS_OK ) { LOGGER_ERROR("Error while starting audio encoder: %s", opus_strerror(rc)); return -1; } rc = opus_encoder_ctl(cs->audio_encoder, OPUS_SET_BITRATE(bitrate)); if ( rc != OPUS_OK ) { LOGGER_ERROR("Error while setting encoder ctl: %s", opus_strerror(rc)); goto FAILURE; } rc = opus_encoder_ctl(cs->audio_encoder, OPUS_SET_COMPLEXITY(10)); if ( rc != OPUS_OK ) { LOGGER_ERROR("Error while setting encoder ctl: %s", opus_strerror(rc)); goto FAILURE; } return 0; FAILURE: cs_disable_audio_sending(cs); return -1; } int cs_enable_audio_receiving(CSSession* cs) { if (cs->audio_decoder) return 0; int rc; cs->audio_decoder = opus_decoder_create(cs->decoder_sample_rate, cs->decoder_channels, &rc ); if ( rc != OPUS_OK ) { LOGGER_ERROR("Error while starting audio decoder: %s", opus_strerror(rc)); return -1; } if ( !(cs->j_buf = jbuf_new(DEFAULT_JBUF)) ) { LOGGER_WARNING("Jitter buffer creaton failed!"); opus_decoder_destroy(cs->audio_decoder); cs->audio_decoder = NULL; return -1; } /* It's used for measuring iteration interval so this has to be some value. * To avoid unecessary checking we set this to 500 */ cs->last_packet_frame_duration = 500; return 0; } /* Called from RTP */ void queue_message(RTPSession *session, RTPMessage *msg) { /* This function is unregistered during call termination befor destroing * Codec session so no need to check for validity of cs */ CSSession *cs = session->cs; if (!cs) return; /* Audio */ if (session->payload_type == rtp_TypeAudio % 128) { pthread_mutex_lock(cs->queue_mutex); int ret = jbuf_write(cs->j_buf, msg); pthread_mutex_unlock(cs->queue_mutex); if (ret == -1) { rtp_free_msg(NULL, msg); } } /* Video */ else { uint8_t *packet = msg->data; uint32_t packet_size = msg->length; if (packet_size < VIDEOFRAME_HEADER_SIZE) goto end; uint8_t diff = packet[0] - cs->frameid_in; if (diff != 0) { if (diff < 225) { /* New frame */ /* Flush last frames' data and get ready for this frame */ Payload *p = malloc(sizeof(Payload) + cs->frame_size); if (p) { pthread_mutex_lock(cs->queue_mutex); if (buffer_full(cs->vbuf_raw)) { LOGGER_DEBUG("Dropped video frame"); Payload *tp; buffer_read(cs->vbuf_raw, &tp); free(tp); } else { p->size = cs->frame_size; memcpy(p->data, cs->frame_buf, cs->frame_size); } buffer_write(cs->vbuf_raw, p); pthread_mutex_unlock(cs->queue_mutex); } else { LOGGER_WARNING("Allocation failed! Program might misbehave!"); goto end; } cs->last_timestamp = msg->header->timestamp; cs->frameid_in = packet[0]; memset(cs->frame_buf, 0, cs->frame_size); cs->frame_size = 0; } else { /* Old frame; drop */ LOGGER_DEBUG("Old packet: %u", packet[0]); goto end; } } uint8_t piece_number = packet[1]; uint32_t length_before_piece = ((piece_number - 1) * cs->peer_video_frame_piece_size); uint32_t framebuf_new_length = length_before_piece + (packet_size - VIDEOFRAME_HEADER_SIZE); if (framebuf_new_length > MAX_VIDEOFRAME_SIZE) { goto end; } /* Otherwise it's part of the frame so just process */ /* LOGGER_DEBUG("Video Packet: %u %u", packet[0], packet[1]); */ memcpy(cs->frame_buf + length_before_piece, packet + VIDEOFRAME_HEADER_SIZE, packet_size - VIDEOFRAME_HEADER_SIZE); if (framebuf_new_length > cs->frame_size) cs->frame_size = framebuf_new_length; end: rtp_free_msg(NULL, msg); } }