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[Gnash-commit] /srv/bzr/gnash/trunk r9955: Change some methods and membe
From: |
Sandro Santilli |
Subject: |
[Gnash-commit] /srv/bzr/gnash/trunk r9955: Change some methods and member names in sound stuff (readability) |
Date: |
Wed, 08 Oct 2008 10:25:38 +0200 |
User-agent: |
Bazaar (1.5) |
------------------------------------------------------------
revno: 9955
committer: Sandro Santilli <address@hidden>
branch nick: trunk
timestamp: Wed 2008-10-08 10:25:38 +0200
message:
Change some methods and member names in sound stuff (readability)
modified:
libmedia/ffmpeg/sound_handler_sdl.cpp
libmedia/ffmpeg/sound_handler_sdl.h
=== modified file 'libmedia/ffmpeg/sound_handler_sdl.cpp'
--- a/libmedia/ffmpeg/sound_handler_sdl.cpp 2008-10-08 07:31:25 +0000
+++ b/libmedia/ffmpeg/sound_handler_sdl.cpp 2008-10-08 08:25:38 +0000
@@ -237,7 +237,7 @@
// Play the index'd sample.
void
-SDL_sound_handler::play_sound(int sound_handle, int loop_count, int offset,
long start_position, const std::vector<sound_envelope>* envelopes)
+SDL_sound_handler::play_sound(int sound_handle, int loopCount, int offset,
long start_position, const std::vector<sound_envelope>* envelopes)
{
boost::mutex::scoped_lock lock(_mutex);
@@ -267,15 +267,16 @@
return;
}
- // Make a "active_sound" for this sound which is later placed on the
vector of instances of this sound being played
+ // Make a "active_sound" for this sound which is later placed
+ // on the vector of instances of this sound being played
std::auto_ptr<active_sound> sound ( new active_sound() );
// Set source data to the active_sound
sound->set_data(sounddata);
// Set the given options of the sound
- if (start_position < 0) sound->position = 0;
- else sound->position = start_position;
+ if (start_position < 0) sound->decodingPosition = 0;
+ else sound->decodingPosition = start_position;
if (offset < 0) sound->offset = 0;
else sound->offset = (sounddata->soundinfo->isStereo() ? offset :
offset*2); // offset is stored as stereo
@@ -283,7 +284,7 @@
sound->envelopes = envelopes;
// Set number of loop we should do. -1 is infinte loop, 0 plays it
once, 1 twice etc.
- sound->loop_count = loop_count;
+ sound->loopCount = loopCount;
sound->decoder = NULL;
@@ -601,7 +602,7 @@
// Pointer handling and checking functions
boost::uint8_t*
-active_sound::get_raw_data_ptr(unsigned long int pos)
+active_sound::getDecodedData(unsigned long int pos)
{
if ( _decodedData.get() )
{
@@ -612,15 +613,15 @@
}
boost::uint8_t*
-active_sound::get_data_ptr(unsigned long int pos)
+active_sound::getEncodedData(unsigned long int pos)
{
- assert(_undecodedData);
- return _undecodedData->data(pos);
+ assert(_encodedData);
+ return _encodedData->data(pos);
}
void active_sound::set_data(sound_data* idata)
{
- _undecodedData = idata;
+ _encodedData = idata;
}
void active_sound::deleteDecodedData()
@@ -667,12 +668,12 @@
unsigned int startpos = 0;
// Make sure we start adjusting at the right sample
if (sound.current_env == 0 &&
(*sound.envelopes)[sound.current_env].m_mark44 > sound.samples_played) {
- startpos = sound.raw_position +
((*sound.envelopes)[sound.current_env].m_mark44 - sound.samples_played)*2;
+ startpos = sound.playbackPosition +
((*sound.envelopes)[sound.current_env].m_mark44 - sound.samples_played)*2;
} else {
- startpos = sound.raw_position;
+ startpos = sound.playbackPosition;
}
- boost::int16_t* data =
reinterpret_cast<boost::int16_t*>(sound.get_raw_data_ptr(startpos));
+ boost::int16_t* data =
reinterpret_cast<boost::int16_t*>(sound.getDecodedData(startpos));
for (unsigned int i=0; i < length/2; i+=2) {
float left =
static_cast<float>((*sound.envelopes)[sound.current_env].m_level0 / 32768.0);
@@ -710,7 +711,7 @@
SDL_MixAudio(static_cast<Uint8*>(stream),static_cast<const
Uint8*>(data), mix_length, SDL_MIX_MAXVOLUME);
// Update sound info
- sound.raw_position += mix_length;
+ sound.playbackPosition += mix_length;
// Sample size is always 2
sound.samples_played += mix_length / 2;
@@ -875,28 +876,30 @@
/*private*/
void
-SDL_sound_handler::mixActiveSound(active_sound& sound, sound_data& sounddata,
Uint8* buffer, unsigned int buffer_length)
+SDL_sound_handler::mixActiveSound(active_sound& sound, sound_data& sounddata,
+ Uint8* buffer, unsigned int buffer_length)
{
// If there exist no decoder, then we can't decode!
if (sound.decoder == NULL) return;
- int volume = int(sounddata.volume*getFinalVolume()/100.0); //
concatenate global volume
+ // concatenate global volume
+ int volume = int(sounddata.volume*getFinalVolume()/100.0);
// When the current sound don't have enough decoded data to fill the
buffer,
// we first mix what is already decoded, then decode some more data, and
// mix some more until the buffer is full. If a sound loops the magic
// happens here ;)
//
- if (sound.rawDataSize() - sound.raw_position < buffer_length
- && (sound.position < sound.dataSize() || sound.loop_count != 0))
+ if (sound.decodedDataSize() - sound.playbackPosition < buffer_length
+ && (sound.decodingPosition < sound.encodedDataSize() ||
sound.loopCount != 0))
{
// First we mix what is decoded
unsigned int index = 0;
- if (sound.rawDataSize() - sound.raw_position > 0)
+ if (sound.decodedDataSize() - sound.playbackPosition > 0)
{
- index = sound.rawDataSize() - sound.raw_position;
+ index = sound.decodedDataSize() -
sound.playbackPosition;
- do_mixing(buffer, sound,
sound.get_raw_data_ptr(sound.raw_position),
+ do_mixing(buffer, sound,
sound.getDecodedData(sound.playbackPosition),
index, volume);
}
@@ -913,14 +916,14 @@
{
// If we need to loop, we reset the data pointer
- if (sound.dataSize() == sound.position &&
sound.loop_count != 0) {
- sound.loop_count--;
- sound.position = 0;
+ if (sound.encodedDataSize() == sound.decodingPosition
&& sound.loopCount != 0) {
+ sound.loopCount--;
+ sound.decodingPosition = 0;
sound.samples_played = 0;
}
// Test if we will get problems... Should not happen...
- assert(sound.dataSize() > sound.position);
+ assert(sound.encodedDataSize() >
sound.decodingPosition);
// temp raw buffer
Uint8* tmp_raw_buffer=0;
@@ -931,16 +934,20 @@
bool parse = true;
if (sounddata.soundinfo->getFormat() ==
AUDIO_CODEC_ADPCM) {
parse = false;
- if (sounddata.m_frames_size.size() > 0)
inputSize = sounddata.m_frames_size[sound.position];
- else inputSize = sound.dataSize() -
sound.position;
+ if (sounddata.m_frames_size.size() > 0) {
+ inputSize =
sounddata.m_frames_size[sound.decodingPosition];
+ }
+ else {
+ inputSize = sound.encodedDataSize() -
sound.decodingPosition;
+ }
} else {
- inputSize = sound.dataSize() - sound.position;
+ inputSize = sound.encodedDataSize() -
sound.decodingPosition;
}
- tmp_raw_buffer =
sound.decoder->decode(sound.get_data_ptr(sound.position),
+ tmp_raw_buffer =
sound.decoder->decode(sound.getEncodedData(sound.decodingPosition),
inputSize, tmp_raw_buffer_size,
decodedBytes, parse);
- sound.position += decodedBytes;
+ sound.decodingPosition += decodedBytes;
// tmp_raw_buffer ownership transferred here
sound.appendDecodedData(tmp_raw_buffer,
tmp_raw_buffer_size);
@@ -948,15 +955,15 @@
decoded_size += tmp_raw_buffer_size;
// no more to decode from this sound, so we break the
loop
- if ((sound.dataSize() <= sound.position &&
sound.loop_count == 0)
+ if ((sound.encodedDataSize() <= sound.decodingPosition
&& sound.loopCount == 0)
|| (tmp_raw_buffer_size == 0 && decodedBytes == 0)) {
- sound.position = sound.dataSize();
+ sound.decodingPosition =
sound.encodedDataSize();
break;
}
} // end of "decode min. bufferlength data" while loop
- sound.raw_position = 0;
+ sound.playbackPosition = 0;
// Determine how much should be mixed
unsigned int mix_length = 0;
@@ -965,21 +972,21 @@
} else {
mix_length = decoded_size;
}
- if (sound.rawDataSize() < 2)
+ if (sound.decodedDataSize() < 2)
{
log_error("Something went terribly wrong during mixing
of an active sound");
return; // something went terrible wrong
}
- do_mixing(buffer+index, sound, sound.get_raw_data_ptr(0),
mix_length, volume);
+ do_mixing(buffer+index, sound, sound.getDecodedData(0),
mix_length, volume);
}
// When the current sound has enough decoded data to fill
// the buffer, we do just that.
- else if (sound.rawDataSize() > sound.raw_position &&
sound.rawDataSize() - sound.raw_position > buffer_length )
+ else if (sound.decodedDataSize() > sound.playbackPosition &&
sound.decodedDataSize() - sound.playbackPosition > buffer_length )
{
- do_mixing(buffer, sound,
sound.get_raw_data_ptr(sound.raw_position),
+ do_mixing(buffer, sound,
sound.getDecodedData(sound.playbackPosition),
buffer_length, volume);
}
@@ -987,14 +994,14 @@
// When the current sound doesn't have anymore data to decode,
// and doesn't loop (anymore), but still got unplayed data,
// we put the last data on the stream
- else if (sound.rawDataSize() - sound.raw_position <= buffer_length &&
sound.rawDataSize() > sound.raw_position+1)
+ else if (sound.decodedDataSize() - sound.playbackPosition <=
buffer_length && sound.decodedDataSize() > sound.playbackPosition+1)
{
- do_mixing(buffer, sound,
sound.get_raw_data_ptr(sound.raw_position),
- sound.rawDataSize() - sound.raw_position, volume);
+ do_mixing(buffer, sound,
sound.getDecodedData(sound.playbackPosition),
+ sound.decodedDataSize() - sound.playbackPosition,
volume);
- sound.raw_position = sound.rawDataSize();
+ sound.playbackPosition = sound.decodedDataSize();
}
}
@@ -1014,7 +1021,7 @@
mixActiveSound(*sound, sounddata, buffer, buffer_length);
// Sound is done, remove it from the active list
- if (sound->position == sound->dataSize() && sound->raw_position
== sound->rawDataSize() && sound->loop_count == 0)
+ if (sound->decodingPosition == sound->encodedDataSize() &&
sound->playbackPosition == sound->decodedDataSize() && sound->loopCount == 0)
{
i = sounddata.eraseActiveSound(i);
=== modified file 'libmedia/ffmpeg/sound_handler_sdl.h'
--- a/libmedia/ffmpeg/sound_handler_sdl.h 2008-10-06 13:44:15 +0000
+++ b/libmedia/ffmpeg/sound_handler_sdl.h 2008-10-08 08:25:38 +0000
@@ -150,13 +150,13 @@
active_sound()
:
decoder(0),
- position(0),
- raw_position(0),
- loop_count(0),
+ decodingPosition(0),
+ playbackPosition(0),
+ loopCount(0),
offset(0),
current_env(0),
samples_played(0),
- _undecodedData(0)
+ _encodedData(0)
{}
~active_sound()
@@ -168,15 +168,15 @@
/// The decoder object used to convert the data into the playable format
AudioDecoder* decoder;
- /// Current decoding position in the stream
- unsigned long position;
+ /// Current decoding position in the encoded stream
+ unsigned long decodingPosition;
- /// Current playing position in the decoded stream
- unsigned long raw_position;
+ /// Current playback position in the decoded stream
+ unsigned long playbackPosition;
/// Numbers of loops: -1 means loop forever, 0 means play once.
/// For every loop completed, it is decremented.
- long loop_count;
+ long loopCount;
/// Offset to make playback start in-sync, only used with mp3 streams.
unsigned int offset;
@@ -199,13 +199,13 @@
///
void set_data(sound_data* newUndecodedData);
- /// Returns the data pointer in the undecoded datastream
+ /// Returns the data pointer in the encoded datastream
/// for the given position. Boundaries are checked.
- boost::uint8_t* get_data_ptr(unsigned long int pos);
+ boost::uint8_t* getEncodedData(unsigned long int pos);
/// Returns the data pointer in the decoded datastream
/// for the given position. Boundaries are checked.
- boost::uint8_t* get_raw_data_ptr(unsigned long int pos);
+ boost::uint8_t* getDecodedData(unsigned long int pos);
/// Release resources associated with decoded data, if any.
//
@@ -254,7 +254,7 @@
delete [] data; // ownership transferred...
}
- size_t rawDataSize() const
+ size_t decodedDataSize() const
{
if ( _decodedData.get() )
{
@@ -263,19 +263,19 @@
else return 0;
}
- size_t dataSize() const
+ size_t encodedDataSize() const
{
- return _undecodedData ? _undecodedData->size() : 0;
+ return _encodedData ? _encodedData->size() : 0;
}
private:
- /// The undecoded data
- sound_data* _undecodedData;
+ /// The encoded data
+ sound_data* _encodedData;
/// The decoded buffer
//
- /// If NULL, the _undecodedData will be considered
+ /// If NULL, the _encodedData will be considered
/// decoded instead
///
std::auto_ptr<SimpleBuffer> _decodedData;
@@ -350,7 +350,7 @@
unsigned int sample_count, int
handle_id);
/// Play the index'd sample.
- virtual void play_sound(int sound_handle, int loop_count, int offset,
+ virtual void play_sound(int sound_handle, int loopCount, int offset,
long start_position, const
std::vector<sound_envelope>* envelopes);
/// Stop the index'd sample.
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