| /* |
| * 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 <freerdp/freerdp.h> |
| #include <freerdp/channels/channels.h> |
| #include <guacamole/protocol.h> |
| #include <guacamole/socket.h> |
| #include <guacamole/stream.h> |
| #include <guacamole/user.h> |
| |
| #include <assert.h> |
| #include <errno.h> |
| #include <stdlib.h> |
| #include <pthread.h> |
| |
| /** |
| * 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; |
| char client_ref[GUAC_RDP_PTR_STRING_LENGTH]; |
| |
| /* Add "AUDIO_INPUT" channel */ |
| guac_rdp_ptr_to_string(client, client_ref); |
| guac_rdp_dvc_list_add(list, "guacai", client_ref, 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; |
| |
| /* 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; |
| |
| 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; |
| |
| /* Reset I/O counters */ |
| audio_buffer->total_bytes_sent = 0; |
| audio_buffer->total_bytes_received = 0; |
| |
| /* 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); |
| } |
| |