/* * Licensed to the Apache Software Foundation (ASF) under one * or more contributor license agreements. See the NOTICE file * distributed with this work for additional information * regarding copyright ownership. The ASF licenses this file * to you under the Apache License, Version 2.0 (the * "License"); you may not use this file except in compliance * with the License. You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, * software distributed under the License is distributed on an * "AS IS" BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY * KIND, either express or implied. See the License for the * specific language governing permissions and limitations * under the License. */ #include "config.h" #include "audio_input.h" #include "dvc.h" #include "ptr_string.h" #include "rdp.h" #include #include #include #include #include #include #include #include #include #include /** * Parses the given raw audio mimetype, producing the corresponding rate, * number of channels, and bytes per sample. * * @param mimetype * The raw auduio mimetype to parse. * * @param rate * A pointer to an int where the sample rate for the PCM format described * by the given mimetype should be stored. * * @param channels * A pointer to an int where the number of channels used by the PCM format * described by the given mimetype should be stored. * * @param bps * A pointer to an int where the number of bytes used the PCM format for * each sample (independent of number of channels) described by the given * mimetype should be stored. * * @return * Zero if the given mimetype is a raw audio mimetype and has been parsed * successfully, non-zero otherwise. */ static int guac_rdp_audio_parse_mimetype(const char* mimetype, int* rate, int* channels, int* bps) { int parsed_rate = -1; int parsed_channels = 1; int parsed_bps; /* PCM audio with one byte per sample */ if (strncmp(mimetype, "audio/L8;", 9) == 0) { mimetype += 8; /* Advance to semicolon ONLY */ parsed_bps = 1; } /* PCM audio with two bytes per sample */ else if (strncmp(mimetype, "audio/L16;", 10) == 0) { mimetype += 9; /* Advance to semicolon ONLY */ parsed_bps = 2; } /* Unsupported mimetype */ else return 1; /* Parse each parameter name/value pair within the mimetype */ do { /* Advance to first character of parameter (current is either a * semicolon or a comma) */ mimetype++; /* Parse number of channels */ if (strncmp(mimetype, "channels=", 9) == 0) { mimetype += 9; parsed_channels = strtol(mimetype, (char**) &mimetype, 10); /* Fail if value invalid / out of range */ if (errno == EINVAL || errno == ERANGE) return 1; } /* Parse number of rate */ else if (strncmp(mimetype, "rate=", 5) == 0) { mimetype += 5; parsed_rate = strtol(mimetype, (char**) &mimetype, 10); /* Fail if value invalid / out of range */ if (errno == EINVAL || errno == ERANGE) return 1; } /* Advance to next parameter */ mimetype = strchr(mimetype, ','); } while (mimetype != NULL); /* Mimetype is invalid if rate was not specified */ if (parsed_rate == -1) return 1; /* Parse success */ *rate = parsed_rate; *channels = parsed_channels; *bps = parsed_bps; return 0; } int guac_rdp_audio_handler(guac_user* user, guac_stream* stream, char* mimetype) { guac_client* client = user->client; guac_rdp_client* rdp_client = (guac_rdp_client*) client->data; int rate; int channels; int bps; /* Parse mimetype, abort on parse error */ if (guac_rdp_audio_parse_mimetype(mimetype, &rate, &channels, &bps)) { guac_user_log(user, GUAC_LOG_WARNING, "Denying user audio stream with " "unsupported mimetype: \"%s\"", mimetype); guac_protocol_send_ack(user->socket, stream, "Unsupported audio " "mimetype", GUAC_PROTOCOL_STATUS_CLIENT_BAD_TYPE); return 0; } /* Init stream data */ stream->blob_handler = guac_rdp_audio_blob_handler; stream->end_handler = guac_rdp_audio_end_handler; /* Associate stream with audio buffer */ guac_rdp_audio_buffer_set_stream(rdp_client->audio_input, user, stream, rate, channels, bps); return 0; } int guac_rdp_audio_blob_handler(guac_user* user, guac_stream* stream, void* data, int length) { guac_client* client = user->client; guac_rdp_client* rdp_client = (guac_rdp_client*) client->data; /* Write blob to audio stream, buffering if necessary */ guac_rdp_audio_buffer_write(rdp_client->audio_input, data, length); return 0; } int guac_rdp_audio_end_handler(guac_user* user, guac_stream* stream) { /* Ignore - the AUDIO_INPUT channel will simply not receive anything */ return 0; } void guac_rdp_audio_load_plugin(rdpContext* context, guac_rdp_dvc_list* list) { guac_client* client = ((rdp_freerdp_context*) context)->client; /* Add "AUDIO_INPUT" channel */ guac_rdp_dvc_list_add(list, "guacai", guac_rdp_ptr_to_string(client), NULL); } guac_rdp_audio_buffer* guac_rdp_audio_buffer_alloc() { guac_rdp_audio_buffer* buffer = calloc(1, sizeof(guac_rdp_audio_buffer)); pthread_mutex_init(&(buffer->lock), NULL); return buffer; } /** * Sends an "ack" instruction over the socket associated with the Guacamole * stream over which audio data is being received. The "ack" instruction will * only be sent if the Guacamole audio stream has been established (through * receipt of an "audio" instruction), is still open (has not received an "end" * instruction nor been associated with an "ack" having an error code), and is * associated with an active RDP AUDIO_INPUT channel. * * @param audio_buffer * The audio buffer associated with the guac_stream for which the "ack" * instruction should be sent, if any. If there is no associated * guac_stream, this function has no effect. * * @param message * An arbitrary human-readable message to send along with the "ack". * * @param status * The Guacamole protocol status code to send with the "ack". This should * be GUAC_PROTOCOL_STATUS_SUCCESS if the audio stream has been set up * successfully or GUAC_PROTOCOL_STATUS_RESOURCE_CLOSED if the audio stream * has been closed (but may usable again if reopened). */ static void guac_rdp_audio_buffer_ack(guac_rdp_audio_buffer* audio_buffer, const char* message, guac_protocol_status status) { guac_user* user = audio_buffer->user; guac_stream* stream = audio_buffer->stream; /* Do not send ack unless both sides of the audio stream are ready */ if (user == NULL || stream == NULL || audio_buffer->packet == NULL) return; /* Send ack instruction */ guac_protocol_send_ack(user->socket, stream, message, status); guac_socket_flush(user->socket); } void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer, guac_user* user, guac_stream* stream, int rate, int channels, int bps) { pthread_mutex_lock(&(audio_buffer->lock)); /* Associate received stream */ audio_buffer->user = user; audio_buffer->stream = stream; audio_buffer->in_format.rate = rate; audio_buffer->in_format.channels = channels; audio_buffer->in_format.bps = bps; /* Reset input counter */ audio_buffer->total_bytes_received = 0; /* Acknowledge stream creation (if buffer is ready to receive) */ guac_rdp_audio_buffer_ack(audio_buffer, "OK", GUAC_PROTOCOL_STATUS_SUCCESS); guac_user_log(user, GUAC_LOG_DEBUG, "User is requesting to provide audio " "input as %i-channel, %i Hz PCM audio at %i bytes/sample.", audio_buffer->in_format.channels, audio_buffer->in_format.rate, audio_buffer->in_format.bps); pthread_mutex_unlock(&(audio_buffer->lock)); } void guac_rdp_audio_buffer_set_output(guac_rdp_audio_buffer* audio_buffer, int rate, int channels, int bps) { pthread_mutex_lock(&(audio_buffer->lock)); /* Set output format */ audio_buffer->out_format.rate = rate; audio_buffer->out_format.channels = channels; audio_buffer->out_format.bps = bps; /* Reset output counter */ audio_buffer->total_bytes_sent = 0; pthread_mutex_unlock(&(audio_buffer->lock)); } void guac_rdp_audio_buffer_begin(guac_rdp_audio_buffer* audio_buffer, int packet_frames, guac_rdp_audio_buffer_flush_handler* flush_handler, void* data) { pthread_mutex_lock(&(audio_buffer->lock)); /* Reset buffer state to provided values */ audio_buffer->bytes_written = 0; audio_buffer->flush_handler = flush_handler; audio_buffer->data = data; /* Calculate size of each packet in bytes */ audio_buffer->packet_size = packet_frames * audio_buffer->out_format.channels * audio_buffer->out_format.bps; /* Allocate new buffer */ free(audio_buffer->packet); audio_buffer->packet = malloc(audio_buffer->packet_size); /* Acknowledge stream creation (if stream is ready to receive) */ guac_rdp_audio_buffer_ack(audio_buffer, "OK", GUAC_PROTOCOL_STATUS_SUCCESS); pthread_mutex_unlock(&(audio_buffer->lock)); } /** * Reads a single sample from the given buffer of data, using the input * format defined within the given audio buffer. Each read sample is * translated to a signed 16-bit value, even if the input format is 8-bit. * The offset into the given buffer will be determined according to the * input and output formats, the number of bytes sent thus far, and the * number of bytes received (excluding the contents of the buffer). * * @param audio_buffer * The audio buffer dictating the format of the given data buffer, as * well as the offset from which the sample should be read. * * @param buffer * The buffer of raw PCM audio data from which the sample should be read. * This buffer MUST NOT contain data already taken into account by the * audio buffer's total_bytes_received counter. * * @param length * The number of bytes within the given buffer of PCM data. * * @param sample * A pointer to the int16_t in which the read sample should be stored. If * the input format is 8-bit, the sample will be shifted left by 8 bits * to produce a 16-bit sample. * * @return * Non-zero if a sample was successfully read, zero if no data remains * within the given buffer that has not already been mapped to an * output sample. */ static int guac_rdp_audio_buffer_read_sample( guac_rdp_audio_buffer* audio_buffer, const char* buffer, int length, int16_t* sample) { int in_bps = audio_buffer->in_format.bps; int in_rate = audio_buffer->in_format.rate; int in_channels = audio_buffer->in_format.channels; int out_bps = audio_buffer->out_format.bps; int out_rate = audio_buffer->out_format.rate; int out_channels = audio_buffer->out_format.channels; /* Calculate position within audio output */ int current_sample = audio_buffer->total_bytes_sent / out_bps; int current_frame = current_sample / out_channels; int current_channel = current_sample % out_channels; /* Map output channel to input channel */ if (current_channel >= in_channels) current_channel = in_channels - 1; /* Transform output position to input position */ current_frame = (int) current_frame * ((double) in_rate / out_rate); current_sample = current_frame * in_channels + current_channel; /* Calculate offset within given buffer from absolute input position */ int offset = current_sample * in_bps - audio_buffer->total_bytes_received; /* It should be impossible for the offset to ever go negative */ assert(offset >= 0); /* Apply offset to buffer */ buffer += offset; length -= offset; /* Read only if sufficient data is present in the given buffer */ if (length < in_bps) return 0; /* Simply read sample directly if input is 16-bit */ if (in_bps == 2) { *sample = *((int16_t*) buffer); return 1; } /* Translate to 16-bit if input is 8-bit */ if (in_bps == 1) { *sample = *buffer << 8; return 1; } /* Accepted audio formats are required to be 8- or 16-bit */ return 0; } void guac_rdp_audio_buffer_write(guac_rdp_audio_buffer* audio_buffer, char* buffer, int length) { int16_t sample; pthread_mutex_lock(&(audio_buffer->lock)); /* Ignore packet if there is no buffer */ if (audio_buffer->packet_size == 0 || audio_buffer->packet == NULL) { pthread_mutex_unlock(&(audio_buffer->lock)); return; } int out_bps = audio_buffer->out_format.bps; /* Continuously write packets until no data remains */ while (guac_rdp_audio_buffer_read_sample(audio_buffer, buffer, length, &sample) > 0) { char* current = audio_buffer->packet + audio_buffer->bytes_written; /* Store as 16-bit or 8-bit, depending on output format */ if (out_bps == 2) *((int16_t*) current) = sample; else if (out_bps == 1) *current = sample >> 8; /* Accepted audio formats are required to be 8- or 16-bit */ else assert(0); /* Update byte counters */ audio_buffer->bytes_written += out_bps; audio_buffer->total_bytes_sent += out_bps; /* Invoke flush handler if full */ if (audio_buffer->bytes_written == audio_buffer->packet_size) { /* Only actually invoke if defined */ if (audio_buffer->flush_handler) audio_buffer->flush_handler(audio_buffer->packet, audio_buffer->bytes_written, audio_buffer->data); /* Reset buffer in all cases */ audio_buffer->bytes_written = 0; } } /* end packet write loop */ /* Track current position in audio stream */ audio_buffer->total_bytes_received += length; pthread_mutex_unlock(&(audio_buffer->lock)); } void guac_rdp_audio_buffer_end(guac_rdp_audio_buffer* audio_buffer) { pthread_mutex_lock(&(audio_buffer->lock)); /* The stream is now closed */ guac_rdp_audio_buffer_ack(audio_buffer, "CLOSED", GUAC_PROTOCOL_STATUS_RESOURCE_CLOSED); /* Unset user and stream */ audio_buffer->user = NULL; audio_buffer->stream = NULL; /* Reset buffer state */ audio_buffer->bytes_written = 0; audio_buffer->packet_size = 0; audio_buffer->flush_handler = NULL; /* Free packet (if any) */ free(audio_buffer->packet); audio_buffer->packet = NULL; pthread_mutex_unlock(&(audio_buffer->lock)); } void guac_rdp_audio_buffer_free(guac_rdp_audio_buffer* audio_buffer) { pthread_mutex_destroy(&(audio_buffer->lock)); free(audio_buffer->packet); free(audio_buffer); }