audio: SDL_GetAudioStreamQueued now returns bytes, not frames.

Reference #8266.
This commit is contained in:
Ryan C. Gordon 2023-09-20 14:15:19 -04:00
parent 703aefbce0
commit a4541a255e
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GPG Key ID: FA148B892AB48044
3 changed files with 13 additions and 12 deletions

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@ -860,12 +860,13 @@ extern DECLSPEC int SDLCALL SDL_GetAudioStreamAvailable(SDL_AudioStream *stream)
/**
* Get the number of sample frames currently queued.
* Get the number of bytes 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.
* Note that audio streams can change their input format at any time, even if
* there is still data queued in a different format, so the returned byte
* count will not necessarily match the number of _sample frames_ available.
* Users of this API should be aware of format changes they make when feeding
* a stream and plan accordingly.
*
* Queued data is not converted until it is consumed by
* SDL_GetAudioStreamData, so this value should be representative of the exact
@ -878,7 +879,7 @@ extern DECLSPEC int SDLCALL SDL_GetAudioStreamAvailable(SDL_AudioStream *stream)
* clamped.
*
* \param stream The audio stream to query
* \returns the number of sample frames queued.
* \returns the number of bytes queued.
*
* \threadsafety It is safe to call this function from any thread.
*

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@ -660,7 +660,7 @@ int SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len)
}
if (retval == 0) {
stream->total_frames_queued += len / SDL_AUDIO_FRAMESIZE(stream->src_spec);
stream->total_bytes_queued += len;
if (stream->put_callback) {
const int newavail = SDL_GetAudioStreamAvailable(stream) - prev_available;
stream->put_callback(stream->put_callback_userdata, stream, newavail, newavail);
@ -863,7 +863,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int ou
SDL_assert(!"Not enough data in queue (read)");
}
stream->total_frames_queued -= output_frames;
stream->total_bytes_queued -= input_bytes;
// Even if we aren't currently resampling, we always need to update the history buffer
UpdateAudioStreamHistoryBuffer(stream, input_buffer, input_bytes, NULL, 0);
@ -953,7 +953,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int ou
if (SDL_ReadFromAudioQueue(stream->queue, input_buffer, input_bytes) != 0) {
SDL_assert(!"Not enough data in queue (resample read)");
}
stream->total_frames_queued -= input_frames;
stream->total_bytes_queued -= input_bytes;
// Update the history buffer and fill in the left padding
UpdateAudioStreamHistoryBuffer(stream, input_buffer, input_bytes, left_padding, padding_bytes);
@ -1124,7 +1124,7 @@ int SDL_GetAudioStreamQueued(SDL_AudioStream *stream)
}
SDL_LockMutex(stream->lock);
const Uint64 total = stream->total_frames_queued;
const Uint64 total = stream->total_bytes_queued;
SDL_UnlockMutex(stream->lock);
// if this overflows an int, just clamp it to a maximum.
@ -1142,7 +1142,7 @@ int SDL_ClearAudioStream(SDL_AudioStream *stream)
SDL_ClearAudioQueue(stream->queue);
SDL_zero(stream->input_spec);
stream->resample_offset = 0;
stream->total_frames_queued = 0;
stream->total_bytes_queued = 0;
SDL_UnlockMutex(stream->lock);
return 0;

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@ -179,7 +179,7 @@ struct SDL_AudioStream
float freq_ratio;
struct SDL_AudioQueue* queue;
Uint64 total_frames_queued;
Uint64 total_bytes_queued;
SDL_AudioSpec input_spec; // The spec of input data currently being processed
Sint64 resample_offset;