/****************************************************************************** Copyright (C) 2013 by Hugh Bailey 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 . ******************************************************************************/ #include #include #include "../util/threading.h" #include "../util/darray.h" #include "../util/circlebuf.h" #include "../util/platform.h" #include "audio-io.h" #include "audio-resampler.h" /* #define DEBUG_AUDIO */ #define nop() do {int invalid = 0;} while(0) struct audio_input { struct audio_convert_info conversion; audio_resampler_t resampler; void (*callback)(void *param, struct audio_data *data); void *param; }; static inline void audio_input_free(struct audio_input *input) { audio_resampler_destroy(input->resampler); } struct audio_line { char *name; struct audio_output *audio; struct circlebuf buffers[MAX_AV_PLANES]; pthread_mutex_t mutex; DARRAY(uint8_t) volume_buffers[MAX_AV_PLANES]; uint64_t base_timestamp; uint64_t last_timestamp; /* states whether this line is still being used. if not, then when the * buffer is depleted, it's destroyed */ bool alive; struct audio_line **prev_next; struct audio_line *next; }; static inline void audio_line_destroy_data(struct audio_line *line) { for (size_t i = 0; i < MAX_AV_PLANES; i++) { circlebuf_free(&line->buffers[i]); da_free(line->volume_buffers[i]); } pthread_mutex_destroy(&line->mutex); bfree(line->name); bfree(line); } struct audio_output { struct audio_output_info info; size_t block_size; size_t channels; size_t planes; pthread_t thread; os_event_t stop_event; DARRAY(uint8_t) mix_buffers[MAX_AV_PLANES]; bool initialized; pthread_mutex_t line_mutex; struct audio_line *first_line; pthread_mutex_t input_mutex; DARRAY(struct audio_input) inputs; }; static inline void audio_output_removeline(struct audio_output *audio, struct audio_line *line) { pthread_mutex_lock(&audio->line_mutex); *line->prev_next = line->next; if (line->next) line->next->prev_next = line->prev_next; pthread_mutex_unlock(&audio->line_mutex); audio_line_destroy_data(line); } /* ------------------------------------------------------------------------- */ /* the following functions are used to calculate frame offsets based upon * timestamps. this will actually work accurately as long as you handle the * values correctly */ static inline double ts_to_frames(audio_t audio, uint64_t ts) { double audio_offset_d = (double)ts; audio_offset_d /= 1000000000.0; audio_offset_d *= (double)audio->info.samples_per_sec; return audio_offset_d; } static inline double positive_round(double val) { return floor(val+0.5); } static size_t ts_diff_frames(audio_t audio, uint64_t ts1, uint64_t ts2) { double diff = ts_to_frames(audio, ts1) - ts_to_frames(audio, ts2); return (size_t)positive_round(diff); } static size_t ts_diff_bytes(audio_t audio, uint64_t ts1, uint64_t ts2) { return ts_diff_frames(audio, ts1, ts2) * audio->block_size; } /* unless the value is 3+ hours worth of frames, this won't overflow */ static inline uint64_t conv_frames_to_time(audio_t audio, uint32_t frames) { return (uint64_t)frames * 1000000000ULL / (uint64_t)audio->info.samples_per_sec; } /* ------------------------------------------------------------------------- */ /* this only really happens with the very initial data insertion. can be * ignored safely. */ static inline void clear_excess_audio_data(struct audio_line *line, uint64_t prev_time) { size_t size = ts_diff_bytes(line->audio, prev_time, line->base_timestamp); /*blog(LOG_DEBUG, "Excess audio data for audio line '%s', somehow " "audio data went back in time by %"PRIu32" bytes. " "prev_time: %"PRIu64", line->base_timestamp: %"PRIu64, line->name, (uint32_t)size, prev_time, line->base_timestamp);*/ for (size_t i = 0; i < line->audio->planes; i++) { size_t clear_size = (size < line->buffers[i].size) ? size : line->buffers[i].size; circlebuf_pop_front(&line->buffers[i], NULL, clear_size); } } static inline uint64_t min_uint64(uint64_t a, uint64_t b) { return a < b ? a : b; } static inline size_t min_size(size_t a, size_t b) { return a < b ? a : b; } #ifndef CLAMP #define CLAMP(val, minval, maxval) \ ((val > maxval) ? maxval : ((val < minval) ? minval : val)) #endif #define MIX_BUFFER_SIZE 256 /* TODO: optimize mixing */ static void mix_float(uint8_t *mix_in, struct circlebuf *buf, size_t size) { float *mix = (float*)mix_in; float vals[MIX_BUFFER_SIZE]; register float mix_val; while (size) { size_t pop_count = min_size(size, sizeof(vals)); size -= pop_count; circlebuf_pop_front(buf, vals, pop_count); pop_count /= sizeof(float); /* This sequence provides hints for MSVC to use packed SSE * instructions addps, minps, maxps, etc. */ for (size_t i = 0; i < pop_count; i++) { mix_val = *mix + vals[i]; /* clamp confuses the optimisation */ mix_val = (mix_val > 1.0f) ? 1.0f : mix_val; mix_val = (mix_val < -1.0f) ? -1.0f : mix_val; *(mix++) = mix_val; } } } static inline bool mix_audio_line(struct audio_output *audio, struct audio_line *line, size_t size, uint64_t timestamp) { size_t time_offset = ts_diff_bytes(audio, line->base_timestamp, timestamp); if (time_offset > size) return false; size -= time_offset; #ifdef DEBUG_AUDIO blog(LOG_DEBUG, "shaved off %lu bytes", size); #endif for (size_t i = 0; i < audio->planes; i++) { size_t pop_size = min_size(size, line->buffers[i].size); mix_float(audio->mix_buffers[i].array + time_offset, &line->buffers[i], pop_size); } return true; } static bool resample_audio_output(struct audio_input *input, struct audio_data *data) { bool success = true; if (input->resampler) { uint8_t *output[MAX_AV_PLANES]; uint32_t frames; uint64_t offset; memset(output, 0, sizeof(output)); success = audio_resampler_resample(input->resampler, output, &frames, &offset, (const uint8_t *const *)data->data, data->frames); for (size_t i = 0; i < MAX_AV_PLANES; i++) data->data[i] = output[i]; data->frames = frames; data->timestamp -= offset; } return success; } static inline void do_audio_output(struct audio_output *audio, uint64_t timestamp, uint32_t frames) { struct audio_data data; for (size_t i = 0; i < MAX_AV_PLANES; i++) data.data[i] = audio->mix_buffers[i].array; data.frames = frames; data.timestamp = timestamp; data.volume = 1.0f; pthread_mutex_lock(&audio->input_mutex); for (size_t i = 0; i < audio->inputs.num; i++) { struct audio_input *input = audio->inputs.array+i; if (resample_audio_output(input, &data)) input->callback(input->param, &data); } pthread_mutex_unlock(&audio->input_mutex); } static uint64_t mix_and_output(struct audio_output *audio, uint64_t audio_time, uint64_t prev_time) { struct audio_line *line = audio->first_line; uint32_t frames = (uint32_t)ts_diff_frames(audio, audio_time, prev_time); size_t bytes = frames * audio->block_size; #ifdef DEBUG_AUDIO blog(LOG_DEBUG, "audio_time: %llu, prev_time: %llu, bytes: %lu", audio_time, prev_time, bytes); #endif /* return an adjusted audio_time according to the amount * of data that was sampled to ensure seamless transmission */ audio_time = prev_time + conv_frames_to_time(audio, frames); /* resize and clear mix buffers */ for (size_t i = 0; i < audio->planes; i++) { da_resize(audio->mix_buffers[i], bytes); memset(audio->mix_buffers[i].array, 0, bytes); } /* mix audio lines */ while (line) { struct audio_line *next = line->next; /* if line marked for removal, destroy and move to the next */ if (!line->buffers[0].size) { if (!line->alive) { audio_output_removeline(audio, line); line = next; continue; } } pthread_mutex_lock(&line->mutex); if (line->buffers[0].size && line->base_timestamp < prev_time) { clear_excess_audio_data(line, prev_time); line->base_timestamp = prev_time; } if (mix_audio_line(audio, line, bytes, prev_time)) line->base_timestamp = audio_time; pthread_mutex_unlock(&line->mutex); line = next; } /* output */ do_audio_output(audio, prev_time, frames); return audio_time; } /* sample audio 40 times a second */ #define AUDIO_WAIT_TIME (1000/40) static void *audio_thread(void *param) { struct audio_output *audio = param; uint64_t buffer_time = audio->info.buffer_ms * 1000000; uint64_t prev_time = os_gettime_ns() - buffer_time; uint64_t audio_time; while (os_event_try(audio->stop_event) == EAGAIN) { os_sleep_ms(AUDIO_WAIT_TIME); pthread_mutex_lock(&audio->line_mutex); audio_time = os_gettime_ns() - buffer_time; audio_time = mix_and_output(audio, audio_time, prev_time); prev_time = audio_time; pthread_mutex_unlock(&audio->line_mutex); } return NULL; } /* ------------------------------------------------------------------------- */ static size_t audio_get_input_idx(audio_t video, void (*callback)(void *param, struct audio_data *data), void *param) { for (size_t i = 0; i < video->inputs.num; i++) { struct audio_input *input = video->inputs.array+i; if (input->callback == callback && input->param == param) return i; } return DARRAY_INVALID; } static inline bool audio_input_init(struct audio_input *input, struct audio_output *audio) { if (input->conversion.format != audio->info.format || input->conversion.samples_per_sec != audio->info.samples_per_sec || input->conversion.speakers != audio->info.speakers) { struct resample_info from = { .format = audio->info.format, .samples_per_sec = audio->info.samples_per_sec, .speakers = audio->info.speakers }; struct resample_info to = { .format = input->conversion.format, .samples_per_sec = input->conversion.samples_per_sec, .speakers = input->conversion.speakers }; input->resampler = audio_resampler_create(&to, &from); if (!input->resampler) { blog(LOG_ERROR, "audio_input_init: Failed to " "create resampler"); return false; } } else { input->resampler = NULL; } return true; } bool audio_output_connect(audio_t audio, const struct audio_convert_info *conversion, void (*callback)(void *param, struct audio_data *data), void *param) { bool success = false; if (!audio) return false; pthread_mutex_lock(&audio->input_mutex); if (audio_get_input_idx(audio, callback, param) == DARRAY_INVALID) { struct audio_input input; input.callback = callback; input.param = param; if (conversion) { input.conversion = *conversion; } else { input.conversion.format = audio->info.format; input.conversion.speakers = audio->info.speakers; input.conversion.samples_per_sec = audio->info.samples_per_sec; } if (input.conversion.format == AUDIO_FORMAT_UNKNOWN) input.conversion.format = audio->info.format; if (input.conversion.speakers == SPEAKERS_UNKNOWN) input.conversion.speakers = audio->info.speakers; if (input.conversion.samples_per_sec == 0) input.conversion.samples_per_sec = audio->info.samples_per_sec; success = audio_input_init(&input, audio); if (success) da_push_back(audio->inputs, &input); } pthread_mutex_unlock(&audio->input_mutex); return success; } void audio_output_disconnect(audio_t audio, void (*callback)(void *param, struct audio_data *data), void *param) { if (!audio) return; pthread_mutex_lock(&audio->input_mutex); size_t idx = audio_get_input_idx(audio, callback, param); if (idx != DARRAY_INVALID) { audio_input_free(audio->inputs.array+idx); da_erase(audio->inputs, idx); } pthread_mutex_unlock(&audio->input_mutex); } static inline bool valid_audio_params(struct audio_output_info *info) { return info->format && info->name && info->samples_per_sec > 0 && info->speakers > 0; } int audio_output_open(audio_t *audio, struct audio_output_info *info) { struct audio_output *out; pthread_mutexattr_t attr; bool planar = is_audio_planar(info->format); if (!valid_audio_params(info)) return AUDIO_OUTPUT_INVALIDPARAM; out = bzalloc(sizeof(struct audio_output)); if (!out) goto fail; memcpy(&out->info, info, sizeof(struct audio_output_info)); pthread_mutex_init_value(&out->line_mutex); out->channels = get_audio_channels(info->speakers); out->planes = planar ? out->channels : 1; out->block_size = (planar ? 1 : out->channels) * get_audio_bytes_per_channel(info->format); if (pthread_mutexattr_init(&attr) != 0) goto fail; if (pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE) != 0) goto fail; if (pthread_mutex_init(&out->line_mutex, &attr) != 0) goto fail; if (pthread_mutex_init(&out->input_mutex, NULL) != 0) goto fail; if (os_event_init(&out->stop_event, OS_EVENT_TYPE_MANUAL) != 0) goto fail; if (pthread_create(&out->thread, NULL, audio_thread, out) != 0) goto fail; out->initialized = true; *audio = out; return AUDIO_OUTPUT_SUCCESS; fail: audio_output_close(out); return AUDIO_OUTPUT_FAIL; } void audio_output_close(audio_t audio) { void *thread_ret; struct audio_line *line; if (!audio) return; if (audio->initialized) { os_event_signal(audio->stop_event); pthread_join(audio->thread, &thread_ret); } line = audio->first_line; while (line) { struct audio_line *next = line->next; audio_line_destroy_data(line); line = next; } for (size_t i = 0; i < audio->inputs.num; i++) audio_input_free(audio->inputs.array+i); for (size_t i = 0; i < MAX_AV_PLANES; i++) da_free(audio->mix_buffers[i]); da_free(audio->inputs); os_event_destroy(audio->stop_event); pthread_mutex_destroy(&audio->line_mutex); bfree(audio); } audio_line_t audio_output_create_line(audio_t audio, const char *name) { if (!audio) return NULL; struct audio_line *line = bzalloc(sizeof(struct audio_line)); line->alive = true; line->audio = audio; if (pthread_mutex_init(&line->mutex, NULL) != 0) { blog(LOG_ERROR, "audio_output_createline: Failed to create " "mutex"); bfree(line); return NULL; } pthread_mutex_lock(&audio->line_mutex); if (audio->first_line) { audio->first_line->prev_next = &line->next; line->next = audio->first_line; } line->prev_next = &audio->first_line; audio->first_line = line; pthread_mutex_unlock(&audio->line_mutex); line->name = bstrdup(name ? name : "(unnamed audio line)"); return line; } const struct audio_output_info *audio_output_get_info(audio_t audio) { return audio ? &audio->info : NULL; } void audio_line_destroy(struct audio_line *line) { if (line) { if (!line->buffers[0].size) audio_output_removeline(line->audio, line); else line->alive = false; } } bool audio_output_active(audio_t audio) { if (!audio) return false; return audio->inputs.num != 0; } size_t audio_output_get_block_size(audio_t audio) { return audio ? audio->block_size : 0; } size_t audio_output_get_planes(audio_t audio) { return audio ? audio->planes : 0; } size_t audio_output_get_channels(audio_t audio) { return audio ? audio->channels : 0; } uint32_t audio_output_get_sample_rate(audio_t audio) { return audio ? audio->info.samples_per_sec : 0; } /* TODO: optimize these two functions */ static inline void mul_vol_float(float *array, float volume, size_t count) { for (size_t i = 0; i < count; i++) array[i] *= volume; } static void audio_line_place_data_pos(struct audio_line *line, const struct audio_data *data, size_t position) { bool planar = line->audio->planes > 1; size_t total_num = data->frames * (planar ? 1 : line->audio->channels); size_t total_size = data->frames * line->audio->block_size; for (size_t i = 0; i < line->audio->planes; i++) { da_copy_array(line->volume_buffers[i], data->data[i], total_size); uint8_t *array = line->volume_buffers[i].array; switch (line->audio->info.format) { case AUDIO_FORMAT_FLOAT: case AUDIO_FORMAT_FLOAT_PLANAR: mul_vol_float((float*)array, data->volume, total_num); break; default: blog(LOG_ERROR, "audio_line_place_data_pos: " "Unsupported or unknown format"); break; } circlebuf_place(&line->buffers[i], position, line->volume_buffers[i].array, total_size); } } static void audio_line_place_data(struct audio_line *line, const struct audio_data *data) { size_t pos = ts_diff_bytes(line->audio, data->timestamp, line->base_timestamp); #ifdef DEBUG_AUDIO blog(LOG_DEBUG, "data->timestamp: %llu, line->base_timestamp: %llu, " "pos: %lu, bytes: %lu, buf size: %lu", data->timestamp, line->base_timestamp, pos, data->frames * line->audio->block_size, line->buffers[0].size); #endif audio_line_place_data_pos(line, data, pos); } void audio_line_output(audio_line_t line, const struct audio_data *data) { /* TODO: prevent insertation of data too far away from expected * audio timing */ if (!line || !data) return; pthread_mutex_lock(&line->mutex); if (!line->buffers[0].size) { line->base_timestamp = data->timestamp - line->audio->info.buffer_ms * 1000000; audio_line_place_data(line, data); } else if (line->base_timestamp <= data->timestamp) { audio_line_place_data(line, data); } else { blog(LOG_DEBUG, "Bad timestamp for audio line '%s', " "data->timestamp: %"PRIu64", " "line->base_timestamp: %"PRIu64". This can " "sometimes happen when there's a pause in " "the threads.", line->name, data->timestamp, line->base_timestamp); } pthread_mutex_unlock(&line->mutex); }