audio: Added SDL_GetAudioStreamQueued

main
Ryan C. Gordon 2023-09-19 12:37:21 -04:00
parent 23206b9e3f
commit 34b931f7eb
6 changed files with 63 additions and 4 deletions

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@ -858,6 +858,39 @@ extern DECLSPEC int SDLCALL SDL_GetAudioStreamData(SDL_AudioStream *stream, void
*/ */
extern DECLSPEC int SDLCALL SDL_GetAudioStreamAvailable(SDL_AudioStream *stream); extern DECLSPEC int SDLCALL SDL_GetAudioStreamAvailable(SDL_AudioStream *stream);
/**
* Get the number of sample frames currently queued.
*
* Since audio streams can change their input format at any time, even if there
* is still data queued in a different format, this reports the queued _sample
* frames_, so if you queue two stereo samples in float32 format and then
* queue five mono samples in Sint16 format, this will return 6.
*
* Queued data is not converted until it is consumed by SDL_GetAudioStreamData,
* so this value should be representative of the exact data that was put into
* the stream.
*
* If the stream has so much data that it would overflow an int, the return
* value is clamped to a maximum value, but no queued data is lost; if there
* are gigabytes of data queued, the app might need to read some of it with
* SDL_GetAudioStreamData before this function's return value is no longer
* clamped.
*
* \param stream The audio stream to query
* \returns the number of sample frames queued.
*
* \threadsafety It is safe to call this function from any thread.
*
* \since This function is available since SDL 3.0.0.
*
* \sa SDL_PutAudioStreamData
* \sa SDL_GetAudioStreamData
* \sa SDL_ClearAudioStream
*/
extern DECLSPEC int SDLCALL SDL_GetAudioStreamQueued(SDL_AudioStream *stream);
/** /**
* Tell the stream that you're done sending data, and anything being buffered * Tell the stream that you're done sending data, and anything being buffered
* should be converted/resampled and made available immediately. * should be converted/resampled and made available immediately.

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@ -657,10 +657,13 @@ int SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len)
retval = SDL_WriteToAudioQueue(stream->queue, &stream->src_spec, buf, len); retval = SDL_WriteToAudioQueue(stream->queue, &stream->src_spec, buf, len);
} }
if ((retval == 0) && stream->put_callback) { if (retval == 0) {
stream->total_frames_queued += len / SDL_AUDIO_FRAMESIZE(stream->src_spec);
if (stream->put_callback) {
const int newavail = SDL_GetAudioStreamAvailable(stream) - prev_available; const int newavail = SDL_GetAudioStreamAvailable(stream) - prev_available;
stream->put_callback(stream->put_callback_userdata, stream, newavail, newavail); stream->put_callback(stream->put_callback_userdata, stream, newavail, newavail);
} }
}
SDL_UnlockMutex(stream->lock); SDL_UnlockMutex(stream->lock);
@ -858,6 +861,8 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int ou
SDL_assert(!"Not enough data in queue (read)"); SDL_assert(!"Not enough data in queue (read)");
} }
stream->total_frames_queued -= output_frames;
// Even if we aren't currently resampling, we always need to update the history buffer // Even if we aren't currently resampling, we always need to update the history buffer
UpdateAudioStreamHistoryBuffer(stream, input_buffer, input_bytes, NULL, 0); UpdateAudioStreamHistoryBuffer(stream, input_buffer, input_bytes, NULL, 0);
@ -946,6 +951,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int ou
if (SDL_ReadFromAudioQueue(stream->queue, input_buffer, input_bytes) != 0) { if (SDL_ReadFromAudioQueue(stream->queue, input_buffer, input_bytes) != 0) {
SDL_assert(!"Not enough data in queue (resample read)"); SDL_assert(!"Not enough data in queue (resample read)");
} }
stream->total_frames_queued -= input_frames;
// Update the history buffer and fill in the left padding // Update the history buffer and fill in the left padding
UpdateAudioStreamHistoryBuffer(stream, input_buffer, input_bytes, left_padding, padding_bytes); UpdateAudioStreamHistoryBuffer(stream, input_buffer, input_bytes, left_padding, padding_bytes);
@ -1083,7 +1089,7 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len)
return total; return total;
} }
// number of converted/resampled bytes available // number of converted/resampled bytes available for output
int SDL_GetAudioStreamAvailable(SDL_AudioStream *stream) int SDL_GetAudioStreamAvailable(SDL_AudioStream *stream)
{ {
if (!stream) { if (!stream) {
@ -1108,6 +1114,21 @@ int SDL_GetAudioStreamAvailable(SDL_AudioStream *stream)
return (int) SDL_min(count, SDL_INT_MAX); return (int) SDL_min(count, SDL_INT_MAX);
} }
// number of sample frames that are currently queued as input.
int SDL_GetAudioStreamQueued(SDL_AudioStream *stream)
{
if (!stream) {
return SDL_InvalidParamError("stream");
}
SDL_LockMutex(stream->lock);
const Uint64 total = stream->total_frames_queued;
SDL_UnlockMutex(stream->lock);
// if this overflows an int, just clamp it to a maximum.
return (int) SDL_min(total, SDL_INT_MAX);
}
int SDL_ClearAudioStream(SDL_AudioStream *stream) int SDL_ClearAudioStream(SDL_AudioStream *stream)
{ {
if (stream == NULL) { if (stream == NULL) {
@ -1119,6 +1140,7 @@ int SDL_ClearAudioStream(SDL_AudioStream *stream)
SDL_ClearAudioQueue(stream->queue); SDL_ClearAudioQueue(stream->queue);
SDL_zero(stream->input_spec); SDL_zero(stream->input_spec);
stream->resample_offset = 0; stream->resample_offset = 0;
stream->total_frames_queued = 0;
SDL_UnlockMutex(stream->lock); SDL_UnlockMutex(stream->lock);
return 0; return 0;

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@ -175,6 +175,7 @@ struct SDL_AudioStream
float freq_ratio; float freq_ratio;
struct SDL_AudioQueue* queue; struct SDL_AudioQueue* queue;
Uint64 total_frames_queued;
SDL_AudioSpec input_spec; // The spec of input data currently being processed SDL_AudioSpec input_spec; // The spec of input data currently being processed
Sint64 resample_offset; Sint64 resample_offset;

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@ -906,6 +906,7 @@ SDL3_0.0.0 {
SDL_GetAudioStreamFrequencyRatio; SDL_GetAudioStreamFrequencyRatio;
SDL_SetAudioStreamFrequencyRatio; SDL_SetAudioStreamFrequencyRatio;
SDL_SetAudioPostmixCallback; SDL_SetAudioPostmixCallback;
SDL_GetAudioStreamQueued;
# extra symbols go here (don't modify this line) # extra symbols go here (don't modify this line)
local: *; local: *;
}; };

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@ -931,3 +931,4 @@
#define SDL_GetAudioStreamFrequencyRatio SDL_GetAudioStreamFrequencyRatio_REAL #define SDL_GetAudioStreamFrequencyRatio SDL_GetAudioStreamFrequencyRatio_REAL
#define SDL_SetAudioStreamFrequencyRatio SDL_SetAudioStreamFrequencyRatio_REAL #define SDL_SetAudioStreamFrequencyRatio SDL_SetAudioStreamFrequencyRatio_REAL
#define SDL_SetAudioPostmixCallback SDL_SetAudioPostmixCallback_REAL #define SDL_SetAudioPostmixCallback SDL_SetAudioPostmixCallback_REAL
#define SDL_GetAudioStreamQueued SDL_GetAudioStreamQueued_REAL

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@ -977,3 +977,4 @@ SDL_DYNAPI_PROC(int,SDL_SetWindowFocusable,(SDL_Window *a, SDL_bool b),(a,b),ret
SDL_DYNAPI_PROC(float,SDL_GetAudioStreamFrequencyRatio,(SDL_AudioStream *a),(a),return) SDL_DYNAPI_PROC(float,SDL_GetAudioStreamFrequencyRatio,(SDL_AudioStream *a),(a),return)
SDL_DYNAPI_PROC(int,SDL_SetAudioStreamFrequencyRatio,(SDL_AudioStream *a, float b),(a,b),return) SDL_DYNAPI_PROC(int,SDL_SetAudioStreamFrequencyRatio,(SDL_AudioStream *a, float b),(a,b),return)
SDL_DYNAPI_PROC(int,SDL_SetAudioPostmixCallback,(SDL_AudioDeviceID a, SDL_AudioPostmixCallback b, void *c),(a,b,c),return) SDL_DYNAPI_PROC(int,SDL_SetAudioPostmixCallback,(SDL_AudioDeviceID a, SDL_AudioPostmixCallback b, void *c),(a,b,c),return)
SDL_DYNAPI_PROC(int,SDL_GetAudioStreamQueued,(SDL_AudioStream *a),(a),return)