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Try and avoid overflow when handling very large audio streams

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Brick
2023-08-31 20:21:51 +01:00
committed by Ryan C. Gordon
parent 5394a805f4
commit a59152688a
1 changed files with 24 additions and 21 deletions
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45
src/audio/SDL_audiocvt.c
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@@ -470,18 +470,23 @@ static Sint64 GetResampleRate(const int src_rate, const int dst_rate)
return sample_rate; return sample_rate;
} }
static size_t GetResamplerAvailableOutputFrames(const size_t input_frames, const Sint64 resample_rate, const Sint64 resample_offset) // !!! FIXME: This will blow up on weird processors.
#ifndef SDL_INT_MAX
#define SDL_INT_MAX 0x7FFFFFFF
#endif
static int GetResamplerAvailableOutputFrames(const size_t input_frames, const Sint64 resample_rate, const Sint64 resample_offset)
{ {
const Sint64 output_frames = (((Sint64)input_frames << 32) - resample_offset + resample_rate - 1) / resample_rate; const Sint64 output_frames = (((Sint64)input_frames << 32) - resample_offset + resample_rate - 1) / resample_rate;
return (size_t) SDL_max(output_frames, 0); return (int) SDL_clamp(output_frames, 0, SDL_INT_MAX);
} }
static int GetResamplerNeededInputFrames(const int output_frames, const Sint64 resample_rate, const Sint64 resample_offset) static int GetResamplerNeededInputFrames(const int output_frames, const Sint64 resample_rate, const Sint64 resample_offset)
{ {
const Sint32 input_frames = (Sint32)((((output_frames - 1) * resample_rate) + resample_offset) >> 32) + 1; const Sint32 input_frames = (Sint32)((((output_frames - 1) * resample_rate) + resample_offset) >> 32) + 1;
return (int) SDL_max(input_frames, 0); return (int) SDL_clamp(input_frames, 0, SDL_INT_MAX);
} }
static int GetResamplerPaddingFrames(const Sint64 resample_rate) static int GetResamplerPaddingFrames(const Sint64 resample_rate)
@@ -1411,28 +1416,28 @@ static void UpdateStreamHistoryBuffer(SDL_AudioStream* stream, const SDL_AudioSp
} }
} }
static size_t GetAudioStreamTrackAvailableFrames(SDL_AudioStream* stream, SDL_AudioTrack* track, Sint64 resample_offset) static Sint64 GetAudioStreamTrackAvailableFrames(SDL_AudioStream* stream, SDL_AudioTrack* track, Sint64 resample_offset)
{ {
size_t frames = track->queued_bytes / GetAudioSpecFrameSize(&track->spec); size_t input_frames = track->queued_bytes / GetAudioSpecFrameSize(&track->spec);
Sint64 resample_rate = GetStreamResampleRate(stream, track->spec.freq); Sint64 resample_rate = GetStreamResampleRate(stream, track->spec.freq);
Sint64 output_frames = (Sint64) input_frames;
if (resample_rate) { if (resample_rate) {
if (!track->flushed) { if (!track->flushed) {
SDL_assert(track->next == NULL); SDL_assert(track->next == NULL);
const int history_buffer_frames = GetHistoryBufferSampleFrames(); const int history_buffer_frames = GetHistoryBufferSampleFrames();
frames -= SDL_min(frames, (size_t)history_buffer_frames); input_frames -= SDL_min(input_frames, (size_t) history_buffer_frames);
} }
frames = GetResamplerAvailableOutputFrames(frames, resample_rate, resample_offset); output_frames = GetResamplerAvailableOutputFrames(input_frames, resample_rate, resample_offset);
} }
return frames; return output_frames;
} }
static size_t GetAudioStreamAvailableFrames(SDL_AudioStream *stream) static Sint64 GetAudioStreamAvailableFrames(SDL_AudioStream *stream)
{ {
size_t total = 0; Sint64 total = 0;
Sint64 resample_offset = stream->resample_offset; Sint64 resample_offset = stream->resample_offset;
SDL_AudioTrack* track; SDL_AudioTrack* track;
@@ -1647,22 +1652,21 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len)
// give the callback a chance to fill in more stream data if it wants. // give the callback a chance to fill in more stream data if it wants.
if (stream->get_callback) { if (stream->get_callback) {
int approx_request = len / dst_frame_size; // start with sample frames desired Sint64 approx_request = len / dst_frame_size; // start with sample frames desired
const int available_frames = (int) GetAudioStreamAvailableFrames(stream); const Sint64 available_frames = GetAudioStreamAvailableFrames(stream);
approx_request -= SDL_min(available_frames, approx_request); approx_request -= SDL_min(available_frames, approx_request);
const Sint64 resample_rate = GetStreamResampleRate(stream, stream->src_spec.freq); const Sint64 resample_rate = GetStreamResampleRate(stream, stream->src_spec.freq);
// FIXME: Is this correct?
if (resample_rate) { if (resample_rate) {
approx_request = GetResamplerNeededInputFrames(approx_request, resample_rate, 0); approx_request = GetResamplerNeededInputFrames((int) approx_request, resample_rate, 0);
} }
approx_request *= GetAudioSpecFrameSize(&stream->src_spec); // convert sample frames to bytes. approx_request *= GetAudioSpecFrameSize(&stream->src_spec); // convert sample frames to bytes.
if (approx_request > 0) { // don't call the callback if we can satisfy this request with existing data. if (approx_request > 0) { // don't call the callback if we can satisfy this request with existing data.
stream->get_callback(stream->get_callback_userdata, stream, approx_request); stream->get_callback(stream->get_callback_userdata, stream, (int) SDL_min(approx_request, SDL_INT_MAX));
} }
} }
@@ -1677,7 +1681,7 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len)
break; break;
} }
const int max_frames = (int) GetAudioStreamTrackAvailableFrames(stream, track, stream->resample_offset); const Sint64 max_frames = GetAudioStreamTrackAvailableFrames(stream, track, stream->resample_offset);
if (max_frames == 0) { if (max_frames == 0) {
if (track->flushed) { if (track->flushed) {
@@ -1703,7 +1707,7 @@ int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len)
// GetAudioStreamDataInternal assumes enough input data is available. // GetAudioStreamDataInternal assumes enough input data is available.
int output_frames = len / dst_frame_size; int output_frames = len / dst_frame_size;
output_frames = SDL_min(output_frames, chunk_size); output_frames = SDL_min(output_frames, chunk_size);
output_frames = SDL_min(output_frames, max_frames); output_frames = (int) SDL_min(output_frames, max_frames);
if (GetAudioStreamDataInternal(stream, buf, output_frames) != 0) { if (GetAudioStreamDataInternal(stream, buf, output_frames) != 0) {
if (retval == 0) { if (retval == 0) {
@@ -1742,7 +1746,7 @@ int SDL_GetAudioStreamAvailable(SDL_AudioStream *stream)
return 0; return 0;
} }
size_t count = GetAudioStreamAvailableFrames(stream); Sint64 count = GetAudioStreamAvailableFrames(stream);
// convert from sample frames to bytes in destination format. // convert from sample frames to bytes in destination format.
count *= GetAudioSpecFrameSize(&stream->dst_spec); count *= GetAudioSpecFrameSize(&stream->dst_spec);
@@ -1750,8 +1754,7 @@ int SDL_GetAudioStreamAvailable(SDL_AudioStream *stream)
SDL_UnlockMutex(stream->lock); SDL_UnlockMutex(stream->lock);
// if this overflows an int, just clamp it to a maximum. // if this overflows an int, just clamp it to a maximum.
const int max_int = 0x7FFFFFFF; // !!! FIXME: This will blow up on weird processors. Is there an SDL_INT_MAX? return (int) SDL_min(count, 0x7FFFFFFF);
return (count >= ((size_t) max_int)) ? max_int : ((int) count);
} }
int SDL_ClearAudioStream(SDL_AudioStream *stream) int SDL_ClearAudioStream(SDL_AudioStream *stream)