| /* |
| * 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 "channels/audio-input/audio-buffer.h" |
| #include "rdp.h" |
| |
| #include <guacamole/client.h> |
| #include <guacamole/protocol.h> |
| #include <guacamole/socket.h> |
| #include <guacamole/stream.h> |
| #include <guacamole/user.h> |
| |
| #include <assert.h> |
| #include <pthread.h> |
| #include <stdint.h> |
| #include <stdlib.h> |
| |
| 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); |
| } |
| |