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Merge pull request #90 from glyptodon/new-media-streams

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GUAC-1354: Implement raw encoder.
This commit is contained in:
James Muehlner 2015-10-02 23:42:39 -07:00
commit 3dddc8c5a4
18 changed files with 399 additions and 781 deletions
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12
src/libguac/Makefile.am
View File

@ -67,7 +67,7 @@ noinst_HEADERS = \
encode-jpeg.h \
encode-png.h \
palette.h \
wav_encoder.h
raw_encoder.h
libguac_la_SOURCES = \
audio.c \
@ -82,18 +82,12 @@ libguac_la_SOURCES = \
plugin.c \
pool.c \
protocol.c \
raw_encoder.c \
socket.c \
socket-fd.c \
socket-nest.c \
timestamp.c \
unicode.c \
wav_encoder.c
# Compile OGG support if available
if ENABLE_OGG
libguac_la_SOURCES += ogg_encoder.c
noinst_HEADERS += ogg_encoder.h
endif
unicode.c
# Compile WebP support if available
if ENABLE_WEBP

163
src/libguac/audio.c
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@ -22,11 +22,7 @@
#include "config.h"
#ifdef ENABLE_OGG
#include "ogg_encoder.h"
#endif
#include "wav_encoder.h"
#include "raw_encoder.h"
#include <guacamole/audio.h>
#include <guacamole/client.h>
@ -36,7 +32,8 @@
#include <stdlib.h>
#include <string.h>
guac_audio_stream* guac_audio_stream_alloc(guac_client* client, guac_audio_encoder* encoder) {
guac_audio_stream* guac_audio_stream_alloc(guac_client* client,
guac_audio_encoder* encoder, int rate, int channels, int bps) {
guac_audio_stream* audio;
@ -50,17 +47,15 @@ guac_audio_stream* guac_audio_stream_alloc(guac_client* client, guac_audio_encod
const char* mimetype = client->info.audio_mimetypes[i];
#ifdef ENABLE_OGG
/* If Ogg is supported, done. */
if (strcmp(mimetype, ogg_encoder->mimetype) == 0) {
encoder = ogg_encoder;
/* If 16-bit raw audio is supported, done. */
if (bps == 16 && strcmp(mimetype, raw16_encoder->mimetype) == 0) {
encoder = raw16_encoder;
break;
}
#endif
/* If wav is supported, done. */
if (strcmp(mimetype, wav_encoder->mimetype) == 0) {
encoder = wav_encoder;
/* If 8-bit raw audio is supported, done. */
if (bps == 8 && strcmp(mimetype, raw8_encoder->mimetype) == 0) {
encoder = raw8_encoder;
break;
}
@ -73,130 +68,84 @@ guac_audio_stream* guac_audio_stream_alloc(guac_client* client, guac_audio_encod
}
/* Allocate stream */
audio = (guac_audio_stream*) malloc(sizeof(guac_audio_stream));
audio = (guac_audio_stream*) calloc(1, sizeof(guac_audio_stream));
audio->client = client;
/* Reset buffer stats */
audio->used = 0;
audio->length = 0x40000;
audio->encoded_data_used = 0;
audio->encoded_data_length = 0x40000;
/* Allocate buffers */
audio->pcm_data = malloc(audio->length);
audio->encoded_data = malloc(audio->encoded_data_length);
/* Assign encoder */
audio->encoder = encoder;
audio->stream = guac_client_alloc_stream(client);
return audio;
}
void guac_audio_stream_begin(guac_audio_stream* audio, int rate, int channels, int bps) {
/* Load PCM properties */
audio->rate = rate;
audio->channels = channels;
audio->bps = bps;
/* Reset write counter */
audio->pcm_bytes_written = 0;
/* Call handler, if defined */
if (audio->encoder->begin_handler)
audio->encoder->begin_handler(audio);
/* Call handler */
return audio;
}
void guac_audio_stream_reset(guac_audio_stream* audio,
guac_audio_encoder* encoder, int rate, int channels, int bps) {
/* Do nothing if nothing is changing */
if ((encoder == NULL || encoder == audio->encoder)
&& rate == audio->rate
&& channels == audio->channels
&& bps == audio->bps) {
return;
}
/* Free old encoder data */
if (audio->encoder->end_handler)
audio->encoder->end_handler(audio);
/* Assign new encoder, if changed */
if (encoder != NULL)
audio->encoder = encoder;
/* Set PCM properties */
audio->rate = rate;
audio->channels = channels;
audio->bps = bps;
/* Init encoder with new data */
if (audio->encoder->begin_handler)
audio->encoder->begin_handler(audio);
}
void guac_audio_stream_end(guac_audio_stream* audio) {
void guac_audio_stream_free(guac_audio_stream* audio) {
double duration;
/* Flush stream and finish encoding */
/* Flush stream encoding */
guac_audio_stream_flush(audio);
/* Clean up encoder */
if (audio->encoder->end_handler)
audio->encoder->end_handler(audio);
/* Calculate duration of PCM data */
duration = ((double) (audio->pcm_bytes_written * 1000 * 8))
/ audio->rate / audio->channels / audio->bps;
/* Send audio */
guac_protocol_send_audio(audio->client->socket, audio->stream,
audio->stream->index, audio->encoder->mimetype, duration);
guac_protocol_send_blob(audio->client->socket, audio->stream,
audio->encoded_data, audio->encoded_data_used);
guac_protocol_send_end(audio->client->socket, audio->stream);
/* Clear data */
audio->encoded_data_used = 0;
}
void guac_audio_stream_free(guac_audio_stream* audio) {
free(audio->pcm_data);
/* Free associated data */
free(audio);
}
void guac_audio_stream_write_pcm(guac_audio_stream* audio,
const unsigned char* data, int length) {
/* Update counter */
audio->pcm_bytes_written += length;
/* Resize audio buffer if necessary */
if (length > audio->length) {
/* Resize to double provided length */
audio->length = length*2;
audio->pcm_data = realloc(audio->pcm_data, audio->length);
}
/* Flush if necessary */
if (audio->used + length > audio->length)
guac_audio_stream_flush(audio);
/* Append to buffer */
memcpy(&(audio->pcm_data[audio->used]), data, length);
audio->used += length;
/* Write data */
if (audio->encoder->write_handler)
audio->encoder->write_handler(audio, data, length);
}
void guac_audio_stream_flush(guac_audio_stream* audio) {
/* If data in buffer */
if (audio->used != 0) {
/* Write data */
audio->encoder->write_handler(audio,
audio->pcm_data, audio->used);
/* Reset buffer */
audio->used = 0;
}
}
void guac_audio_stream_write_encoded(guac_audio_stream* audio,
const unsigned char* data, int length) {
/* Resize audio buffer if necessary */
if (audio->encoded_data_used + length > audio->encoded_data_length) {
/* Increase to double concatenated size to accomodate */
audio->encoded_data_length = (audio->encoded_data_length + length)*2;
audio->encoded_data = realloc(audio->encoded_data,
audio->encoded_data_length);
}
/* Append to buffer */
memcpy(&(audio->encoded_data[audio->encoded_data_used]), data, length);
audio->encoded_data_used += length;
/* Flush any buffered data */
if (audio->encoder->flush_handler)
audio->encoder->flush_handler(audio);
}

9
src/libguac/guacamole/audio-fntypes.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (C) 2014 Glyptodon LLC
* Copyright (C) 2015 Glyptodon LLC
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
@ -36,13 +36,18 @@
*/
typedef void guac_audio_encoder_begin_handler(guac_audio_stream* audio);
/**
* Handler which is called when the audio stream needs to be flushed.
*/
typedef void guac_audio_encoder_flush_handler(guac_audio_stream* audio);
/**
* Handler which is called when the audio stream is closed.
*/
typedef void guac_audio_encoder_end_handler(guac_audio_stream* audio);
/**
* Handler which is called when the audio stream is flushed.
* Handler which is called when PCM data is written to the audio stream.
*/
typedef void guac_audio_encoder_write_handler(guac_audio_stream* audio,
const unsigned char* pcm_data, int length);

2
src/libguac/guacamole/audio-types.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (C) 2014 Glyptodon LLC
* Copyright (C) 2015 Glyptodon LLC
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal

160
src/libguac/guacamole/audio.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (C) 2013 Glyptodon LLC
* Copyright (C) 2015 Glyptodon LLC
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
@ -44,15 +44,21 @@ struct guac_audio_encoder {
const char* mimetype;
/**
* Handler which will be called when the audio stream is opened.
* Handler which will be called when the audio stream is first created.
*/
guac_audio_encoder_begin_handler* begin_handler;
/**
* Handler which will be called when the audio stream is flushed.
* Handler which will be called when PCM data is written to the audio
* stream for encoding.
*/
guac_audio_encoder_write_handler* write_handler;
/**
* Handler which will be called when the audio stream is flushed.
*/
guac_audio_encoder_flush_handler* flush_handler;
/**
* Handler which will be called when the audio stream is closed.
*/
@ -63,38 +69,7 @@ struct guac_audio_encoder {
struct guac_audio_stream {
/**
* PCM data buffer, 16-bit samples, 2-channel, 44100 Hz.
*/
unsigned char* pcm_data;
/**
* Number of bytes in buffer.
*/
int used;
/**
* Maximum number of bytes in buffer.
*/
int length;
/**
* Encoded audio data buffer, as written by the encoder.
*/
unsigned char* encoded_data;
/**
* Number of bytes in the encoded data buffer.
*/
int encoded_data_used;
/**
* Maximum number of bytes in the encoded data buffer.
*/
int encoded_data_length;
/**
* Arbitrary codec encoder. When the PCM buffer is flushed, PCM data will
* be sent to this encoder.
* Arbitrary codec encoder which will receive raw PCM data.
*/
guac_audio_encoder* encoder;
@ -125,11 +100,6 @@ struct guac_audio_stream {
*/
int bps;
/**
* The number of PCM bytes written since the audio chunk began.
*/
int pcm_bytes_written;
/**
* Encoder-specific state data.
*/
@ -141,79 +111,83 @@ struct guac_audio_stream {
* Allocates a new audio stream which encodes audio data using the given
* encoder. If NULL is specified for the encoder, an appropriate encoder
* will be selected based on the encoders built into libguac and the level
* of client support.
* of client support. The PCM format specified here (via rate, channels, and
* bps) must be the format used for all PCM data provided to the audio stream.
* The format may only be changed using guac_audio_stream_reset().
*
* @param client The guac_client for which this audio stream is being
* allocated.
* @param encoder The guac_audio_encoder to use when encoding audio, or
* NULL if libguac should select an appropriate built-in
* encoder on its own.
* @return The newly allocated guac_audio_stream, or NULL if no audio
* stream could be allocated due to lack of client support.
* @param client
* The guac_client for which this audio stream is being allocated.
*
* @param encoder
* The guac_audio_encoder to use when encoding audio, or NULL if libguac
* should select an appropriate built-in encoder on its own.
*
* @param rate
* The number of samples per second of PCM data sent to this stream.
*
* @param channels
* The number of audio channels per sample of PCM data. Legal values are
* 1 or 2.
*
* @param bps
* The number of bits per sample per channel for PCM data. Legal values are
* 8 or 16.
*
* @return
* The newly allocated guac_audio_stream, or NULL if no audio stream could
* be allocated due to lack of client support.
*/
guac_audio_stream* guac_audio_stream_alloc(guac_client* client,
guac_audio_encoder* encoder);
guac_audio_encoder* encoder, int rate, int channels, int bps);
/**
* Frees the given audio stream.
* Resets the given audio stream, switching to the given encoder, rate,
* channels, and bits per sample. If NULL is specified for the encoder, the
* encoder is left unchanged. If the encoder, rate, channels, and bits per
* sample are all identical to the current settings, this function has no
* effect.
*
* @param stream The guac_audio_stream to free.
* @param encoder
* The guac_audio_encoder to use when encoding audio, or NULL to leave this
* unchanged.
*/
void guac_audio_stream_reset(guac_audio_stream* audio,
guac_audio_encoder* encoder, int rate, int channels, int bps);
/**
* Closes and frees the given audio stream.
*
* @param stream
* The guac_audio_stream to free.
*/
void guac_audio_stream_free(guac_audio_stream* stream);
/**
* Begins a new audio packet within the given audio stream. This packet will be
* built up with repeated writes of PCM data, finally being sent when complete
* via guac_audio_stream_end().
*
* @param stream The guac_audio_stream which should start a new audio packet.
* @param rate The audio rate of the packet, in Hz.
* @param channels The number of audio channels.
* @param bps The number of bits per audio sample.
*/
void guac_audio_stream_begin(guac_audio_stream* stream, int rate, int channels, int bps);
/**
* Ends the current audio packet, writing the finished packet as an audio
* instruction.
*
* @param stream The guac_audio_stream whose current audio packet should be
* completed and sent.
*/
void guac_audio_stream_end(guac_audio_stream* stream);
/**
* Writes PCM data to the given audio stream. This PCM data will be
* automatically encoded by the audio encoder associated with this stream. This
* function must only be called after an audio packet has been started with
* guac_audio_stream_begin().
* automatically encoded by the audio encoder associated with this stream. The
* PCM data must be 2-channel, 44100 Hz, with signed 16-bit samples.
*
* @param stream The guac_audio_stream to write PCM data through.
* @param data The PCM data to write.
* @param length The number of bytes of PCM data provided.
* @param stream
* The guac_audio_stream to write PCM data through.
*
* @param data
* The PCM data to write.
*
* @param length
* The number of bytes of PCM data provided.
*/
void guac_audio_stream_write_pcm(guac_audio_stream* stream,
const unsigned char* data, int length);
/**
* Flushes the given audio stream.
* Flushes the underlying audio buffer, if any, ensuring that all audio
* previously written via guac_audio_stream_write_pcm() has been encoded and
* sent to the client.
*
* @param stream The guac_audio_stream to flush.
* @param stream
* The guac_audio_stream whose audio buffers should be flushed.
*/
void guac_audio_stream_flush(guac_audio_stream* stream);
/**
* Appends arbitrarily-encoded data to the encoded_data buffer within the given
* audio stream. This data must be encoded in the output format of the encoder
* used by the stream. This function is mainly for use by encoder
* implementations.
*
* @param audio The guac_audio_stream to write data through.
* @param data Arbitrary encoded data to write through the audio stream.
* @param length The number of bytes of data provided.
*/
void guac_audio_stream_write_encoded(guac_audio_stream* audio,
const unsigned char* data, int length);
#endif

21
src/libguac/guacamole/protocol.h
View File

@ -295,15 +295,20 @@ int guac_protocol_send_undefine(guac_socket* socket,
* If an error occurs sending the instruction, a non-zero value is
* returned, and guac_error is set appropriately.
*
* @param socket The guac_socket connection to use.
* @param stream The stream to use.
* @param channel The index of the audio channel to use.
* @param mimetype The mimetype of the data being sent.
* @param duration The duration of the sound being sent, in milliseconds.
* @return Zero on success, non-zero on error.
* @param socket
* The guac_socket connection to use when sending the audio instruction.
*
* @param stream
* The stream to use for future audio data.
*
* @param mimetype
* The mimetype of the audio data which will be sent over the given stream.
*
* @return
* Zero on success, non-zero on error.
*/
int guac_protocol_send_audio(guac_socket* socket, const guac_stream* stream,
int channel, const char* mimetype, double duration);
const char* mimetype);
/**
* Sends a file instruction over the given guac_socket connection.
@ -350,7 +355,7 @@ int guac_protocol_send_pipe(guac_socket* socket, const guac_stream* stream,
* @return Zero on success, non-zero on error.
*/
int guac_protocol_send_blob(guac_socket* socket, const guac_stream* stream,
void* data, int count);
const void* data, int count);
/**
* Sends an end instruction over the given guac_socket connection.

195
src/libguac/ogg_encoder.c
View File

@ -1,195 +0,0 @@
/*
* Copyright (C) 2013 Glyptodon LLC
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "config.h"
#include "audio.h"
#include "ogg_encoder.h"
#include <vorbis/vorbisenc.h>
#include <stdlib.h>
void ogg_encoder_begin_handler(guac_audio_stream* audio) {
/* Allocate stream state */
ogg_encoder_state* state = (ogg_encoder_state*)
malloc(sizeof(ogg_encoder_state));
/* Init state */
vorbis_info_init(&(state->info));
vorbis_encode_init_vbr(&(state->info), audio->channels, audio->rate, 0.4);
vorbis_analysis_init(&(state->vorbis_state), &(state->info));
vorbis_block_init(&(state->vorbis_state), &(state->vorbis_block));
vorbis_comment_init(&(state->comment));
vorbis_comment_add_tag(&(state->comment), "ENCODER", "libguac");
ogg_stream_init(&(state->ogg_state), rand());
/* Write headers */
{
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout(
&(state->vorbis_state),
&(state->comment),
&header, &header_comm, &header_code);
ogg_stream_packetin(&(state->ogg_state), &header);
ogg_stream_packetin(&(state->ogg_state), &header_comm);
ogg_stream_packetin(&(state->ogg_state), &header_code);
/* For each packet */
while (ogg_stream_flush(&(state->ogg_state), &(state->ogg_page)) != 0) {
/* Write packet header */
guac_audio_stream_write_encoded(audio,
state->ogg_page.header,
state->ogg_page.header_len);
/* Write packet body */
guac_audio_stream_write_encoded(audio,
state->ogg_page.body,
state->ogg_page.body_len);
}
}
audio->data = state;
}
void ogg_encoder_write_blocks(guac_audio_stream* audio) {
/* Get state */
ogg_encoder_state* state = (ogg_encoder_state*) audio->data;
while (vorbis_analysis_blockout(&(state->vorbis_state),
&(state->vorbis_block)) == 1) {
/* Analyze */
vorbis_analysis(&(state->vorbis_block), NULL);
vorbis_bitrate_addblock(&(state->vorbis_block));
/* Flush Ogg pages */
while (vorbis_bitrate_flushpacket(&(state->vorbis_state),
&(state->ogg_packet))) {
/* Weld packet into bitstream */
ogg_stream_packetin(&(state->ogg_state), &(state->ogg_packet));
/* Write out pages */
while (ogg_stream_pageout(&(state->ogg_state),
&(state->ogg_page)) != 0) {
/* Write packet header */
guac_audio_stream_write_encoded(audio,
state->ogg_page.header,
state->ogg_page.header_len);
/* Write packet body */
guac_audio_stream_write_encoded(audio,
state->ogg_page.body,
state->ogg_page.body_len);
if (ogg_page_eos(&(state->ogg_page)))
break;
}
}
}
}
void ogg_encoder_end_handler(guac_audio_stream* audio) {
/* Get state */
ogg_encoder_state* state = (ogg_encoder_state*) audio->data;
/* Write end-of-stream */
vorbis_analysis_wrote(&(state->vorbis_state), 0);
ogg_encoder_write_blocks(audio);
/* Clean up encoder */
ogg_stream_clear(&(state->ogg_state));
vorbis_block_clear(&(state->vorbis_block));
vorbis_dsp_clear(&(state->vorbis_state));
vorbis_comment_clear(&(state->comment));
vorbis_info_clear(&(state->info));
/* Free stream state */
free(audio->data);
}
void ogg_encoder_write_handler(guac_audio_stream* audio,
const unsigned char* pcm_data, int length) {
/* Get state */
ogg_encoder_state* state = (ogg_encoder_state*) audio->data;
/* Calculate samples */
int samples = length / audio->channels * 8 / audio->bps;
int i;
/* Get buffer */
float** buffer = vorbis_analysis_buffer(&(state->vorbis_state), samples);
signed char* readbuffer = (signed char*) pcm_data;
for (i=0; i<samples; i++) {
/* FIXME: For now, assume 2 channels, 16-bit */
int left = ((readbuffer[i*4+1]<<8)|(0x00ff&(int)readbuffer[i*4]));
int right = ((readbuffer[i*4+3]<<8)|(0x00ff&(int)readbuffer[i*4+2]));
/* Store sample in buffer */
buffer[0][i] = left / 32768.f;
buffer[1][i] = right / 32768.f;
}
/* Submit data */
vorbis_analysis_wrote(&(state->vorbis_state), samples);
/* Write data */
ogg_encoder_write_blocks(audio);
}
/* Encoder handlers */
guac_audio_encoder _ogg_encoder = {
.mimetype = "audio/ogg",
.begin_handler = ogg_encoder_begin_handler,
.write_handler = ogg_encoder_write_handler,
.end_handler = ogg_encoder_end_handler
};
/* Actual encoder */
guac_audio_encoder* ogg_encoder = &_ogg_encoder;

8
src/libguac/protocol.c
View File

@ -153,7 +153,7 @@ int guac_protocol_send_arc(guac_socket* socket, const guac_layer* layer,
}
int guac_protocol_send_audio(guac_socket* socket, const guac_stream* stream,
int channel, const char* mimetype, double duration) {
const char* mimetype) {
int ret_val;
@ -162,11 +162,7 @@ int guac_protocol_send_audio(guac_socket* socket, const guac_stream* stream,
guac_socket_write_string(socket, "5.audio,")
|| __guac_socket_write_length_int(socket, stream->index)
|| guac_socket_write_string(socket, ",")
|| __guac_socket_write_length_int(socket, channel)
|| guac_socket_write_string(socket, ",")
|| __guac_socket_write_length_string(socket, mimetype)
|| guac_socket_write_string(socket, ",")
|| __guac_socket_write_length_double(socket, duration)
|| guac_socket_write_string(socket, ";");
guac_socket_instruction_end(socket);
@ -175,7 +171,7 @@ int guac_protocol_send_audio(guac_socket* socket, const guac_stream* stream,
}
int guac_protocol_send_blob(guac_socket* socket, const guac_stream* stream,
void* data, int count) {
const void* data, int count) {
int base64_length = (count + 2) / 3 * 4;

161
src/libguac/raw_encoder.c Normal file
View File

@ -0,0 +1,161 @@
/*
* Copyright (C) 2015 Glyptodon LLC
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "config.h"
#include "audio.h"
#include "raw_encoder.h"
#include <guacamole/audio.h>
#include <guacamole/client.h>
#include <guacamole/protocol.h>
#include <guacamole/socket.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
static void raw_encoder_begin_handler(guac_audio_stream* audio) {
raw_encoder_state* state;
char mimetype[256];
/* Produce mimetype string from format info */
snprintf(mimetype, sizeof(mimetype), "audio/L%i;rate=%i,channels=%i",
audio->bps, audio->rate, audio->channels);
/* Associate stream */
guac_protocol_send_audio(audio->client->socket, audio->stream, mimetype);
/* Allocate and init encoder state */
audio->data = state = malloc(sizeof(raw_encoder_state));
state->written = 0;
state->length = GUAC_RAW_ENCODER_BUFFER_SIZE
* audio->rate * audio->channels * audio->bps
/ 8 / 1000;
state->buffer = malloc(state->length);
guac_client_log(audio->client, GUAC_LOG_DEBUG,
"Using raw encoder (%s) with a %i byte buffer.",
mimetype, state->length);
}
static void raw_encoder_end_handler(guac_audio_stream* audio) {
raw_encoder_state* state = (raw_encoder_state*) audio->data;
/* Send end of stream */
guac_protocol_send_end(audio->client->socket, audio->stream);
/* Free state information */
free(state->buffer);
free(state);
}
static void raw_encoder_write_handler(guac_audio_stream* audio,
const unsigned char* pcm_data, int length) {
raw_encoder_state* state = (raw_encoder_state*) audio->data;
while (length > 0) {
/* Prefer to copy a chunk of equal size to available buffer space */
int chunk_size = state->length - state->written;
/* If no space remains, flush and retry */
if (chunk_size == 0) {
guac_audio_stream_flush(audio);
continue;
}
/* Do not copy more data than is available in source PCM */
if (chunk_size > length)
chunk_size = length;
/* Copy block of PCM data into buffer */
memcpy(state->buffer + state->written, pcm_data, chunk_size);
/* Advance to next block */
state->written += chunk_size;
pcm_data += chunk_size;
length -= chunk_size;
}
}
static void raw_encoder_flush_handler(guac_audio_stream* audio) {
raw_encoder_state* state = (raw_encoder_state*) audio->data;
guac_socket* socket = audio->client->socket;
guac_stream* stream = audio->stream;
unsigned char* current = state->buffer;
int remaining = state->written;
/* Flush all data in buffer as blobs */
while (remaining > 0) {
/* Determine size of blob to be written */
int chunk_size = remaining;
if (chunk_size > 6048)
chunk_size = 6048;
/* Send audio data */
guac_protocol_send_blob(socket, stream, current, chunk_size);
/* Advance to next blob */
current += chunk_size;
remaining -= chunk_size;
}
/* All data has been flushed */
state->written = 0;
}
/* 8-bit raw encoder handlers */
guac_audio_encoder _raw8_encoder = {
.mimetype = "audio/L8",
.begin_handler = raw_encoder_begin_handler,
.write_handler = raw_encoder_write_handler,
.flush_handler = raw_encoder_flush_handler,
.end_handler = raw_encoder_end_handler
};
/* 16-bit raw encoder handlers */
guac_audio_encoder _raw16_encoder = {
.mimetype = "audio/L16",
.begin_handler = raw_encoder_begin_handler,
.write_handler = raw_encoder_write_handler,
.flush_handler = raw_encoder_flush_handler,
.end_handler = raw_encoder_end_handler
};
/* Actual encoder definitions */
guac_audio_encoder* raw8_encoder = &_raw8_encoder;
guac_audio_encoder* raw16_encoder = &_raw16_encoder;

55
src/libguac/ogg_encoder.h → src/libguac/raw_encoder.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (C) 2013 Glyptodon LLC
* Copyright (C) 2015 Glyptodon LLC
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
@ -21,35 +21,58 @@
*/
#ifndef __GUAC_OGG_ENCODER_H
#define __GUAC_OGG_ENCODER_H
#ifndef GUAC_RAW_ENCODER_H
#define GUAC_RAW_ENCODER_H
#include "config.h"
#include "audio.h"
#include <vorbis/vorbisenc.h>
/**
* The number of bytes to send in each audio blob.
*/
#define GUAC_RAW_ENCODER_BLOB_SIZE 6048
typedef struct ogg_encoder_state {
/**
* The size of the raw encoder output PCM buffer, in milliseconds. The
* equivalent size in bytes will vary by PCM rate, number of channels, and bits
* per sample.
*/
#define GUAC_RAW_ENCODER_BUFFER_SIZE 250
/**
* The current state of the raw encoder. The raw encoder performs very minimal
* processing, buffering provided PCM data only as necessary to ensure audio
* packet sizes are reasonable.
*/
typedef struct raw_encoder_state {
/**
* Ogg state
* Buffer of not-yet-written raw PCM data.
*/
ogg_stream_state ogg_state;
ogg_page ogg_page;
ogg_packet ogg_packet;
unsigned char* buffer;
/**
* Vorbis state
* Size of the PCM buffer, in bytes.
*/
vorbis_info info;
vorbis_comment comment;
vorbis_dsp_state vorbis_state;
vorbis_block vorbis_block;
int length;
} ogg_encoder_state;
/**
* The current number of bytes stored within the PCM buffer.
*/
int written;
extern guac_audio_encoder* ogg_encoder;
} raw_encoder_state;
/**
* Audio encoder which writes 8-bit raw PCM (one byte per sample).
*/
extern guac_audio_encoder* raw8_encoder;
/**
* Audio encoder which writes 16-bit raw PCM (two bytes per sample).
*/
extern guac_audio_encoder* raw16_encoder;
#endif

185
src/libguac/wav_encoder.c
View File

@ -1,185 +0,0 @@
/*
* Copyright (C) 2013 Glyptodon LLC
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "config.h"
#include "audio.h"
#include "wav_encoder.h"
#include <stdlib.h>
#include <string.h>
#define WAV_BUFFER_SIZE 0x4000
void wav_encoder_begin_handler(guac_audio_stream* audio) {
/* Allocate stream state */
wav_encoder_state* state = (wav_encoder_state*)
malloc(sizeof(wav_encoder_state));
/* Initialize buffer */
state->length = WAV_BUFFER_SIZE;
state->used = 0;
state->data_buffer = (unsigned char*) malloc(state->length);
audio->data = state;
}
void _wav_encoder_write_le(unsigned char* buffer, int value, int length) {
int offset;
/* Write all bytes in the given value in little-endian byte order */
for (offset=0; offset<length; offset++) {
/* Store byte */
*buffer = value & 0xFF;
/* Move to next byte */
value >>= 8;
buffer++;
}
}
void wav_encoder_end_handler(guac_audio_stream* audio) {
/*
* Static header init
*/
wav_encoder_riff_header riff_header = {
.chunk_id = "RIFF",
.chunk_format = "WAVE"
};
wav_encoder_fmt_header fmt_header = {
.subchunk_id = "fmt ",
.subchunk_size = {0x10, 0x00, 0x00, 0x00}, /* 16 */
.subchunk_format = {0x01, 0x00} /* 1 = PCM */
};
wav_encoder_data_header data_header = {
.subchunk_id = "data"
};
/* Get state */
wav_encoder_state* state = (wav_encoder_state*) audio->data;
/*
* RIFF HEADER
*/
/* Chunk size */
_wav_encoder_write_le(riff_header.chunk_size,
4 + sizeof(fmt_header) + sizeof(data_header) + state->used,
sizeof(riff_header.chunk_size));
guac_audio_stream_write_encoded(audio,
(unsigned char*) &riff_header,
sizeof(riff_header));
/*
* FMT HEADER
*/
/* Channels */
_wav_encoder_write_le(fmt_header.subchunk_channels,
audio->channels, sizeof(fmt_header.subchunk_channels));
/* Sample rate */
_wav_encoder_write_le(fmt_header.subchunk_sample_rate,
audio->rate, sizeof(fmt_header.subchunk_sample_rate));
/* Byte rate */
_wav_encoder_write_le(fmt_header.subchunk_byte_rate,
audio->rate * audio->channels * audio->bps / 8,
sizeof(fmt_header.subchunk_byte_rate));
/* Block align */
_wav_encoder_write_le(fmt_header.subchunk_block_align,
audio->channels * audio->bps / 8,
sizeof(fmt_header.subchunk_block_align));
/* Bits per second */
_wav_encoder_write_le(fmt_header.subchunk_bps,
audio->bps, sizeof(fmt_header.subchunk_bps));
guac_audio_stream_write_encoded(audio,
(unsigned char*) &fmt_header,
sizeof(fmt_header));
/*
* DATA HEADER
*/
/* PCM data size */
_wav_encoder_write_le(data_header.subchunk_size,
state->used, sizeof(data_header.subchunk_size));
guac_audio_stream_write_encoded(audio,
(unsigned char*) &data_header,
sizeof(data_header));
/* Write .wav data */
guac_audio_stream_write_encoded(audio, state->data_buffer, state->used);
/* Free stream state */
free(state);
}
void wav_encoder_write_handler(guac_audio_stream* audio,
const unsigned char* pcm_data, int length) {
/* Get state */
wav_encoder_state* state = (wav_encoder_state*) audio->data;
/* Increase size of buffer if necessary */
if (state->used + length > state->length) {
/* Increase to double concatenated size to accomodate */
state->length = (state->length + length)*2;
state->data_buffer = realloc(state->data_buffer,
state->length);
}
/* Append to buffer */
memcpy(&(state->data_buffer[state->used]), pcm_data, length);
state->used += length;
}
/* Encoder handlers */
guac_audio_encoder _wav_encoder = {
.mimetype = "audio/wav",
.begin_handler = wav_encoder_begin_handler,
.write_handler = wav_encoder_write_handler,
.end_handler = wav_encoder_end_handler
};
/* Actual encoder */
guac_audio_encoder* wav_encoder = &_wav_encoder;

132
src/libguac/wav_encoder.h
View File

@ -1,132 +0,0 @@
/*
* Copyright (C) 2013 Glyptodon LLC
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#ifndef __GUAC_WAV_ENCODER_H
#define __GUAC_WAV_ENCODER_H
#include "config.h"
#include "audio.h"
typedef struct wav_encoder_riff_header {
/**
* The RIFF chunk header, normally the string "RIFF".
*/
unsigned char chunk_id[4];
/**
* Size of the entire file, not including chunk_id or chunk_size.
*/
unsigned char chunk_size[4];
/**
* The format of this file, normally the string "WAVE".
*/
unsigned char chunk_format[4];
} wav_encoder_riff_header;
typedef struct wav_encoder_fmt_header {
/**
* ID of this subchunk. For the fmt subchunk, this should be "fmt ".
*/
unsigned char subchunk_id[4];
/**
* The size of the rest of this subchunk. For PCM, this will be 16.
*/
unsigned char subchunk_size[4];
/**
* Format of this subchunk. For PCM, this will be 1.
*/
unsigned char subchunk_format[2];
/**
* The number of channels in the PCM data.
*/
unsigned char subchunk_channels[2];
/**
* The sample rate of the PCM data.
*/
unsigned char subchunk_sample_rate[4];
/**
* The sample rate of the PCM data in bytes per second.
*/
unsigned char subchunk_byte_rate[4];
/**
* The number of bytes per sample.
*/
unsigned char subchunk_block_align[2];
/**
* The number of bits per sample.
*/
unsigned char subchunk_bps[2];
} wav_encoder_fmt_header;
typedef struct wav_encoder_state {
/**
* Arbitrary PCM data available for writing when the overall WAV is
* flushed.
*/
unsigned char* data_buffer;
/**
* The number of bytes currently present in the data buffer.
*/
int used;
/**
* The total number of bytes that can be written into the data buffer
* without requiring resizing.
*/
int length;
} wav_encoder_state;
typedef struct wav_encoder_data_header {
/**
* ID of this subchunk. For the data subchunk, this should be "data".
*/
unsigned char subchunk_id[4];
/**
* The number of bytes in the PCM data.
*/
unsigned char subchunk_size[4];
} wav_encoder_data_header;
extern guac_audio_encoder* wav_encoder;
#endif

5
src/protocols/rdp/client.c
View File

@ -286,7 +286,10 @@ BOOL rdp_freerdp_pre_connect(freerdp* instance) {
/* If audio enabled, choose an encoder */
if (guac_client_data->settings.audio_enabled) {
guac_client_data->audio = guac_audio_stream_alloc(client, NULL);
guac_client_data->audio = guac_audio_stream_alloc(client, NULL,
GUAC_RDP_AUDIO_RATE,
GUAC_RDP_AUDIO_CHANNELS,
GUAC_RDP_AUDIO_BPS);
/* If an encoding is available, load the sound plugin */
if (guac_client_data->audio != NULL) {

21
src/protocols/rdp/client.h
View File

@ -87,6 +87,27 @@
*/
#define GUAC_RDP_CLIPBOARD_MAX_LENGTH 262144
/**
* Initial rate of audio to stream, in Hz. If the RDP server uses a different
* value, the Guacamole audio stream will simply be reset appropriately.
*/
#define GUAC_RDP_AUDIO_RATE 44100
/**
* The number of channels to stream for audio. If the RDP server uses a
* different value, the Guacamole audio stream will simply be reset
* appropriately.
*/
#define GUAC_RDP_AUDIO_CHANNELS 2
/**
* The number of bits per sample within the audio stream. If the RDP server
* uses a different value, the Guacamole audio stream will simply be reset
* appropriately.
*/
#define GUAC_RDP_AUDIO_BPS 16
/**
* Client data that will remain accessible through the guac_client.
* This should generally include data commonly used by Guacamole handlers.

22
src/protocols/rdp/guac_rdpsnd/rdpsnd_messages.c
View File

@ -133,6 +133,9 @@ void guac_rdpsnd_formats_handler(guac_rdpsndPlugin* rdpsnd,
"%i Hz",
bps, channels, rate);
/* Ensure audio stream is configured to use accepted format */
guac_audio_stream_reset(audio, NULL, rate, channels, bps);
/* Queue format for sending as accepted */
Stream_EnsureRemainingCapacity(output_stream, 18 + body_size);
Stream_Write(output_stream, format_start, 18 + body_size);
@ -225,7 +228,6 @@ void guac_rdpsnd_wave_info_handler(guac_rdpsndPlugin* rdpsnd,
guac_audio_stream* audio, wStream* input_stream,
guac_rdpsnd_pdu_header* header) {
unsigned char buffer[4];
int format;
/* Read wave information */
@ -233,7 +235,7 @@ void guac_rdpsnd_wave_info_handler(guac_rdpsndPlugin* rdpsnd,
Stream_Read_UINT16(input_stream, format);
Stream_Read_UINT8(input_stream, rdpsnd->waveinfo_block_number);
Stream_Seek(input_stream, 3);
Stream_Read(input_stream, buffer, 4);
Stream_Read(input_stream, rdpsnd->initial_wave_data, 4);
/*
* Size of incoming wave data is equal to the body size field of this
@ -245,15 +247,12 @@ void guac_rdpsnd_wave_info_handler(guac_rdpsndPlugin* rdpsnd,
/* Read wave in next iteration */
rdpsnd->next_pdu_is_wave = TRUE;
/* Init stream with requested format */
guac_audio_stream_begin(audio,
/* Reset audio stream if format has changed */
guac_audio_stream_reset(audio, NULL,
rdpsnd->formats[format].rate,
rdpsnd->formats[format].channels,
rdpsnd->formats[format].bps);
/* Write initial 4 bytes of data */
guac_audio_stream_write_pcm(audio, buffer, 4);
}
void guac_rdpsnd_wave_handler(guac_rdpsndPlugin* rdpsnd,
@ -269,11 +268,14 @@ void guac_rdpsnd_wave_handler(guac_rdpsndPlugin* rdpsnd,
wStream* output_stream = Stream_New(NULL, 8);
/* Get wave data */
unsigned char* buffer = Stream_Buffer(input_stream) + 4;
unsigned char* buffer = Stream_Buffer(input_stream);
/* Copy over first four bytes */
memcpy(buffer, rdpsnd->initial_wave_data, 4);
/* Write rest of audio packet */
guac_audio_stream_write_pcm(audio, buffer, rdpsnd->incoming_wave_size);
guac_audio_stream_end(audio);
guac_audio_stream_write_pcm(audio, buffer, rdpsnd->incoming_wave_size + 4);
guac_audio_stream_flush(audio);
/* Write Wave Confirmation PDU */
Stream_Write_UINT8(output_stream, SNDC_WAVECONFIRM);

8
src/protocols/rdp/guac_rdpsnd/rdpsnd_service.h
View File

@ -95,8 +95,16 @@ typedef struct guac_rdpsndPlugin {
*/
int next_pdu_is_wave;
/**
* The wave data received within the last SNDC_WAVE (WaveInfo) PDU.
*/
unsigned char initial_wave_data[4];
/**
* The size, in bytes, of the wave data in the coming Wave PDU, if any.
* This does not include the initial wave data received within the last
* SNDC_WAVE (WaveInfo) PDU, which is always the first four bytes of the
* actual wave data block.
*/
int incoming_wave_size;

5
src/protocols/vnc/client.c
View File

@ -401,7 +401,10 @@ int guac_client_init(guac_client* client, int argc, char** argv) {
/* If an encoding is available, load an audio stream */
if (guac_client_data->audio_enabled) {
guac_client_data->audio = guac_audio_stream_alloc(client, NULL);
guac_client_data->audio = guac_audio_stream_alloc(client, NULL,
GUAC_VNC_AUDIO_RATE,
GUAC_VNC_AUDIO_CHANNELS,
GUAC_VNC_AUDIO_BPS);
/* Load servername if specified */
if (argv[IDX_AUDIO_SERVERNAME][0] != '\0')

24
src/protocols/vnc/pulse.c
View File

@ -74,23 +74,13 @@ static void __stream_read_callback(pa_stream* stream, size_t length,
/* Read data */
pa_stream_peek(stream, &buffer, &length);
/* Avoid sending silence unless data is waiting to be flushed */
if (audio->pcm_bytes_written != 0 || !guac_pa_is_silence(buffer, length)) {
/* Write data */
/* Continuously write received PCM data */
if (!guac_pa_is_silence(buffer, length))
guac_audio_stream_write_pcm(audio, buffer, length);
/* Flush occasionally */
if (audio->pcm_bytes_written > GUAC_VNC_PCM_WRITE_RATE) {
guac_audio_stream_end(audio);
guac_audio_stream_begin(client_data->audio,
GUAC_VNC_AUDIO_RATE,
GUAC_VNC_AUDIO_CHANNELS,
GUAC_VNC_AUDIO_BPS);
guac_socket_flush(client->socket);
}
}
/* Flush upon silence */
else
guac_audio_stream_flush(audio);
/* Advance buffer */
pa_stream_drop(stream);
@ -245,10 +235,6 @@ void guac_pa_start_stream(guac_client* client) {
pa_context* context;
guac_client_log(client, GUAC_LOG_INFO, "Starting audio stream");
guac_audio_stream_begin(client_data->audio,
GUAC_VNC_AUDIO_RATE,
GUAC_VNC_AUDIO_CHANNELS,
GUAC_VNC_AUDIO_BPS);
/* Init main loop */
client_data->pa_mainloop = pa_threaded_mainloop_new();