diff --git a/src/protocols/rdp/audio_input.c b/src/protocols/rdp/audio_input.c index 0eb757d8..eefb231b 100644 --- a/src/protocols/rdp/audio_input.c +++ b/src/protocols/rdp/audio_input.c @@ -30,6 +30,7 @@ #include #include +#include #include #include #include @@ -246,6 +247,9 @@ void guac_rdp_audio_buffer_set_stream(guac_rdp_audio_buffer* audio_buffer, 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); @@ -270,6 +274,9 @@ void guac_rdp_audio_buffer_set_output(guac_rdp_audio_buffer* audio_buffer, 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)); } @@ -302,12 +309,99 @@ void guac_rdp_audio_buffer_begin(guac_rdp_audio_buffer* audio_buffer, } +/** + * 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) { - pthread_mutex_lock(&(audio_buffer->lock)); + int16_t sample; - /* FIXME: Assuming mimetype of "audio/L16;rate=44100,channels=2" */ + pthread_mutex_lock(&(audio_buffer->lock)); /* Ignore packet if there is no buffer */ if (audio_buffer->packet_size == 0 || audio_buffer->packet == NULL) { @@ -315,27 +409,27 @@ void guac_rdp_audio_buffer_write(guac_rdp_audio_buffer* audio_buffer, return; } + int out_bps = audio_buffer->out_format.bps; + /* Continuously write packets until no data remains */ - while (length > 0) { + while (guac_rdp_audio_buffer_read_sample(audio_buffer, + buffer, length, &sample) > 0) { - /* Calculate ideal size of chunk based on available space */ - int chunk_size = audio_buffer->packet_size - - audio_buffer->bytes_written; + char* current = audio_buffer->packet + audio_buffer->bytes_written; - /* Shrink chunk size if insufficient bytes are provided */ - if (length < chunk_size) - chunk_size = length; + /* 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; - /* Append buffer */ - memcpy(audio_buffer->packet + audio_buffer->bytes_written, - buffer, chunk_size); + /* Accepted audio formats are required to be 8- or 16-bit */ + else + assert(0); /* Update byte counters */ - length -= chunk_size; - audio_buffer->bytes_written += chunk_size; - - /* Advance to next chunk */ - buffer += chunk_size; + 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) { @@ -352,6 +446,9 @@ void guac_rdp_audio_buffer_write(guac_rdp_audio_buffer* audio_buffer, } /* end packet write loop */ + /* Track current position in audio stream */ + audio_buffer->total_bytes_received += length; + pthread_mutex_unlock(&(audio_buffer->lock)); } diff --git a/src/protocols/rdp/audio_input.h b/src/protocols/rdp/audio_input.h index 87eee9a9..bd84fe10 100644 --- a/src/protocols/rdp/audio_input.h +++ b/src/protocols/rdp/audio_input.h @@ -121,6 +121,18 @@ typedef struct guac_rdp_audio_buffer { */ int bytes_written; + /** + * The total number of bytes having ever been received by the Guacamole + * server for the current audio stream. + */ + int total_bytes_received; + + /** + * The total number of bytes having ever been sent to the RDP server for + * the current audio stream. + */ + int total_bytes_sent; + /** * All audio data being prepared for sending to the AUDIO_INPUT channel. */