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-rw-r--r--src/audio.c44
-rw-r--r--src/external/dr_mp3.h372
2 files changed, 312 insertions, 104 deletions
diff --git a/src/audio.c b/src/audio.c
index 51023280..5ec23bb2 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -214,7 +214,7 @@ typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioB
// Audio buffer structure
// NOTE: Slightly different logic is used when feeding data to the playback device depending on whether or not data is streamed
-typedef struct AudioBuffer AudioBuffer;
+typedef struct AudioBuffer AudioBuffer;
struct AudioBuffer {
mal_dsp dsp; // Required for format conversion
float volume;
@@ -1130,13 +1130,12 @@ Music LoadMusicStream(const char *fileName)
TraceLog(LOG_INFO, "[%s] MP3 sample rate: %i", fileName, music->ctxMp3.sampleRate);
TraceLog(LOG_INFO, "[%s] MP3 bits per sample: %i", fileName, 32);
TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, music->ctxMp3.channels);
- TraceLog(LOG_INFO, "[%s] MP3 frames remaining: %i", fileName, (unsigned int)music->ctxMp3.framesRemaining);
music->stream = InitAudioStream(music->ctxMp3.sampleRate, 32, music->ctxMp3.channels);
// TODO: There is not an easy way to compute the total number of samples available
// in an MP3, frames size could be variable... we tried with a 60 seconds music... but crashes...
- music->totalSamples = 60*music->ctxMp3.sampleRate*music->ctxMp3.channels;
+ music->totalSamples = drmp3_get_pcm_frame_count(&music->ctxMp3)*music->ctxMp3.channels;
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_MP3;
music->loopCount = -1; // Infinite loop by default
@@ -1161,8 +1160,8 @@ Music LoadMusicStream(const char *fileName)
music->ctxType = MUSIC_MODULE_XM;
music->loopCount = -1; // Infinite loop by default
- TraceLog(LOG_DEBUG, "[%s] XM number of samples: %i", fileName, music->totalSamples);
- TraceLog(LOG_DEBUG, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
+ TraceLog(LOG_INFO, "[%s] XM number of samples: %i", fileName, music->totalSamples);
+ TraceLog(LOG_INFO, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
else musicLoaded = false;
}
@@ -1283,7 +1282,7 @@ void StopMusicStream(Music music)
case MUSIC_AUDIO_FLAC: /* TODO: Restart FLAC context */ break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
- case MUSIC_AUDIO_MP3: /* TODO: Restart MP3 context */ break;
+ case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame(&music->ctxMp3, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM: /* TODO: Restart XM context */ break;
@@ -1306,13 +1305,13 @@ void UpdateMusicStream(Music music)
unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.audioBuffer)->bufferSizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
- void *pcm = calloc(subBufferSizeInFrames*music->stream.sampleSize/8*music->stream.channels, 1);
+ void *pcm = calloc(subBufferSizeInFrames*music->stream.channels*music->stream.sampleSize/8, 1);
int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
while (IsAudioBufferProcessed(music->stream))
{
- if (music->samplesLeft >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames;
+ if ((music->samplesLeft/music->stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music->stream.channels;
else samplesCount = music->samplesLeft;
// TODO: Really don't like ctxType thingy...
@@ -1321,27 +1320,31 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
- stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels);
+ stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount);
} break;
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process
- unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm);
+ unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
- // NOTE: Returns the number of samples to process
- unsigned int numSamplesMp3 = (unsigned int)drmp3_read_f32(&music->ctxMp3, samplesCount*music->stream.channels, (float *)pcm);
+ // NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
+ unsigned int numFramesMp3 = (unsigned int)drmp3_read_pcm_frames_f32(&music->ctxMp3, samplesCount/music->stream.channels, (float *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
- case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break;
+ case MUSIC_MODULE_XM:
+ {
+ // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 --> WEIRD
+ jar_xm_generate_samples_16bit(music->ctxXm, (short *)pcm, samplesCount/2);
+ } break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break;
@@ -1414,7 +1417,7 @@ void SetMusicLoopCount(Music music, int count)
// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
- float totalSeconds = (float)music->totalSamples/music->stream.sampleRate;
+ float totalSeconds = (float)music->totalSamples/(music->stream.sampleRate*music->stream.channels);
return totalSeconds;
}
@@ -1425,12 +1428,11 @@ float GetMusicTimePlayed(Music music)
float secondsPlayed = 0.0f;
unsigned int samplesPlayed = music->totalSamples - music->samplesLeft;
- secondsPlayed = (float)samplesPlayed/music->stream.sampleRate;
+ secondsPlayed = (float)samplesPlayed/(music->stream.sampleRate*music->stream.channels);
return secondsPlayed;
}
-
// Init audio stream (to stream audio pcm data)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
{
@@ -1509,11 +1511,11 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
// Does this API expect a whole buffer to be updated in one go? Assuming so, but if not will need to change this logic.
- if (subBufferSizeInFrames >= (mal_uint32)samplesCount)
+ if (subBufferSizeInFrames >= (mal_uint32)samplesCount/stream.channels)
{
mal_uint32 framesToWrite = subBufferSizeInFrames;
- if (framesToWrite > (mal_uint32)samplesCount) framesToWrite = (mal_uint32)samplesCount;
+ if (framesToWrite > ((mal_uint32)samplesCount/stream.channels)) framesToWrite = (mal_uint32)samplesCount/stream.channels;
mal_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
memcpy(subBuffer, data, bytesToWrite);
@@ -1867,13 +1869,13 @@ static Wave LoadMP3(const char *fileName)
Wave wave = { 0 };
// Decode an entire MP3 file in one go
- uint64_t totalSampleCount = 0;
+ uint64_t totalFrameCount = 0;
drmp3_config config = { 0 };
- wave.data = drmp3_open_and_decode_file_f32(fileName, &config, &totalSampleCount);
+ wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);
wave.channels = config.outputChannels;
wave.sampleRate = config.outputSampleRate;
- wave.sampleCount = (int)totalSampleCount;
+ wave.sampleCount = (int)totalFrameCount*wave.channels;
wave.sampleSize = 32;
// NOTE: Only support up to 2 channels (mono, stereo)
diff --git a/src/external/dr_mp3.h b/src/external/dr_mp3.h
index cd8920a1..070f0c15 100644
--- a/src/external/dr_mp3.h
+++ b/src/external/dr_mp3.h
@@ -1,5 +1,5 @@
// MP3 audio decoder. Public domain. See "unlicense" statement at the end of this file.
-// dr_mp3 - v0.3.2 - 2018-09-11
+// dr_mp3 - v0.4.0 - 2018-xx-xx
//
// David Reid - [email protected]
//
@@ -52,11 +52,6 @@
//
// #define DR_MP3_NO_SIMD
// Disable SIMD optimizations.
-//
-//
-// LIMITATIONS
-// ===========
-// - Seeking is extremely inefficient.
#ifndef dr_mp3_h
#define dr_mp3_h
@@ -92,7 +87,8 @@ typedef drmp3_uint32 drmp3_bool32;
#define DRMP3_TRUE 1
#define DRMP3_FALSE 0
-#define DRMP3_MAX_SAMPLES_PER_FRAME (1152*2)
+#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152
+#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2)
// Low Level Push API
@@ -214,11 +210,12 @@ typedef struct
drmp3_read_proc onRead;
drmp3_seek_proc onSeek;
void* pUserData;
- drmp3_uint32 frameChannels; // The number of channels in the currently loaded MP3 frame. Internal use only.
- drmp3_uint32 frameSampleRate; // The sample rate of the currently loaded MP3 frame. Internal use only.
- drmp3_uint32 framesConsumed;
- drmp3_uint32 framesRemaining;
- drmp3_uint8 frames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME]; // <-- Multipled by sizeof(float) to ensure there's enough room for DR_MP3_FLOAT_OUTPUT.
+ drmp3_uint32 mp3FrameChannels; // The number of channels in the currently loaded MP3 frame. Internal use only.
+ drmp3_uint32 mp3FrameSampleRate; // The sample rate of the currently loaded MP3 frame. Internal use only.
+ drmp3_uint32 pcmFramesConsumedInMP3Frame;
+ drmp3_uint32 pcmFramesRemainingInMP3Frame;
+ drmp3_uint8 pcmFrames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME]; // <-- Multipled by sizeof(float) to ensure there's enough room for DR_MP3_FLOAT_OUTPUT.
+ drmp3_uint64 currentPCMFrame; // The current PCM frame, globally, based on the output sample rate. Mainly used for seeking.
drmp3_src src;
size_t dataSize;
size_t dataCapacity;
@@ -268,12 +265,21 @@ void drmp3_uninit(drmp3* pMP3);
// Reads PCM frames as interleaved 32-bit IEEE floating point PCM.
//
// Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames.
-drmp3_uint64 drmp3_read_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut);
+drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut);
// Seeks to a specific frame.
//
// Note that this is _not_ an MP3 frame, but rather a PCM frame.
-drmp3_bool32 drmp3_seek_to_frame(drmp3* pMP3, drmp3_uint64 frameIndex);
+drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex);
+
+// Calculates the total number of PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet
+// radio. Runs in linear time. Returns 0 on error.
+drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3);
+
+// Calculates the total number of MP3 frames in the MP3 stream. Cannot be used for infinite streams such as internet
+// radio. Runs in linear time. Returns 0 on error.
+drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3);
+
// Opens an decodes an entire MP3 stream as a single operation.
@@ -281,10 +287,10 @@ drmp3_bool32 drmp3_seek_to_frame(drmp3* pMP3, drmp3_uint64 frameIndex);
// pConfig is both an input and output. On input it contains what you want. On output it contains what you got.
//
// Free the returned pointer with drmp3_free().
-float* drmp3_open_and_decode_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount);
-float* drmp3_open_and_decode_memory_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount);
+float* drmp3_open_and_read_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount);
+float* drmp3_open_memory_and_read_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount);
#ifndef DR_MP3_NO_STDIO
-float* drmp3_open_and_decode_file_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount);
+float* drmp3_open_file_and_read_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount);
#endif
// Frees any memory that was allocated by a public drmp3 API.
@@ -2376,35 +2382,46 @@ drmp3_uint64 drmp3_src_read_frames_linear(drmp3_src* pSRC, drmp3_uint64 frameCou
}
+static size_t drmp3__on_read(drmp3* pMP3, void* pBufferOut, size_t bytesToRead)
+{
+ return pMP3->onRead(pMP3->pUserData, pBufferOut, bytesToRead);
+}
-static drmp3_bool32 drmp3_decode_next_frame(drmp3* pMP3)
+static drmp3_bool32 drmp3__on_seek(drmp3* pMP3, int offset, drmp3_seek_origin origin)
+{
+ drmp3_assert(offset >= 0);
+ return pMP3->onSeek(pMP3->pUserData, offset, origin);
+}
+
+
+static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames)
{
drmp3_assert(pMP3 != NULL);
drmp3_assert(pMP3->onRead != NULL);
if (pMP3->atEnd) {
- return DRMP3_FALSE;
+ return 0;
}
- do
- {
+ drmp3_uint32 pcmFramesRead = 0;
+ do {
// minimp3 recommends doing data submission in 16K chunks. If we don't have at least 16K bytes available, get more.
if (pMP3->dataSize < DRMP3_DATA_CHUNK_SIZE) {
if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) {
pMP3->dataCapacity = DRMP3_DATA_CHUNK_SIZE;
drmp3_uint8* pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity);
if (pNewData == NULL) {
- return DRMP3_FALSE; // Out of memory.
+ return 0; // Out of memory.
}
pMP3->pData = pNewData;
}
- size_t bytesRead = pMP3->onRead(pMP3->pUserData, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize));
+ size_t bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize));
if (bytesRead == 0) {
if (pMP3->dataSize == 0) {
pMP3->atEnd = DRMP3_TRUE;
- return DRMP3_FALSE; // No data.
+ return 0; // No data.
}
}
@@ -2413,23 +2430,23 @@ static drmp3_bool32 drmp3_decode_next_frame(drmp3* pMP3)
if (pMP3->dataSize > INT_MAX) {
pMP3->atEnd = DRMP3_TRUE;
- return DRMP3_FALSE; // File too big.
+ return 0; // File too big.
}
drmp3dec_frame_info info;
- drmp3_uint32 samplesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData, (int)pMP3->dataSize, (drmp3d_sample_t*)pMP3->frames, &info); // <-- Safe size_t -> int conversion thanks to the check above.
- if (samplesRead != 0) {
+ pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData, (int)pMP3->dataSize, pPCMFrames, &info); // <-- Safe size_t -> int conversion thanks to the check above.
+ if (pcmFramesRead != 0) {
size_t leftoverDataSize = (pMP3->dataSize - (size_t)info.frame_bytes);
for (size_t i = 0; i < leftoverDataSize; ++i) {
pMP3->pData[i] = pMP3->pData[i + (size_t)info.frame_bytes];
}
pMP3->dataSize = leftoverDataSize;
- pMP3->framesConsumed = 0;
- pMP3->framesRemaining = samplesRead;
- pMP3->frameChannels = info.channels;
- pMP3->frameSampleRate = info.hz;
- drmp3_src_set_input_sample_rate(&pMP3->src, pMP3->frameSampleRate);
+ pMP3->pcmFramesConsumedInMP3Frame = 0;
+ pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead;
+ pMP3->mp3FrameChannels = info.channels;
+ pMP3->mp3FrameSampleRate = info.hz;
+ drmp3_src_set_input_sample_rate(&pMP3->src, pMP3->mp3FrameSampleRate);
break;
} else {
// Need more data. minimp3 recommends doing data submission in 16K chunks.
@@ -2438,24 +2455,47 @@ static drmp3_bool32 drmp3_decode_next_frame(drmp3* pMP3)
pMP3->dataCapacity += DRMP3_DATA_CHUNK_SIZE;
drmp3_uint8* pNewData = (drmp3_uint8*)drmp3_realloc(pMP3->pData, pMP3->dataCapacity);
if (pNewData == NULL) {
- return DRMP3_FALSE; // Out of memory.
+ return 0; // Out of memory.
}
pMP3->pData = pNewData;
}
// Fill in a chunk.
- size_t bytesRead = pMP3->onRead(pMP3->pUserData, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize));
+ size_t bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize));
if (bytesRead == 0) {
pMP3->atEnd = DRMP3_TRUE;
- return DRMP3_FALSE; // Error reading more data.
+ return 0; // Error reading more data.
}
pMP3->dataSize += bytesRead;
}
} while (DRMP3_TRUE);
- return DRMP3_TRUE;
+ return pcmFramesRead;
+}
+
+static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3)
+{
+ drmp3_assert(pMP3 != NULL);
+ return drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames);
+}
+
+static drmp3_uint32 drmp3_seek_next_frame(drmp3* pMP3)
+{
+ drmp3_assert(pMP3 != NULL);
+
+ drmp3_uint32 pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, NULL);
+ if (pcmFrameCount == 0) {
+ return 0;
+ }
+
+ // We have essentially just skipped past the frame, so just set the remaining samples to 0.
+ pMP3->currentPCMFrame += pcmFrameCount;
+ pMP3->pcmFramesConsumedInMP3Frame = pcmFrameCount;
+ pMP3->pcmFramesRemainingInMP3Frame = 0;
+
+ return pcmFrameCount;
}
static drmp3_uint64 drmp3_read_src(drmp3_src* pSRC, drmp3_uint64 frameCount, void* pFramesOut, void* pUserData)
@@ -2465,64 +2505,64 @@ static drmp3_uint64 drmp3_read_src(drmp3_src* pSRC, drmp3_uint64 frameCount, voi
drmp3_assert(pMP3->onRead != NULL);
float* pFramesOutF = (float*)pFramesOut;
- drmp3_uint32 totalFramesRead = 0;
+ drmp3_uint64 totalFramesRead = 0;
while (frameCount > 0) {
// Read from the in-memory buffer first.
- while (pMP3->framesRemaining > 0 && frameCount > 0) {
- drmp3d_sample_t* frames = (drmp3d_sample_t*)pMP3->frames;
+ while (pMP3->pcmFramesRemainingInMP3Frame > 0 && frameCount > 0) {
+ drmp3d_sample_t* frames = (drmp3d_sample_t*)pMP3->pcmFrames;
#ifndef DR_MP3_FLOAT_OUTPUT
- if (pMP3->frameChannels == 1) {
+ if (pMP3->mp3FrameChannels == 1) {
if (pMP3->channels == 1) {
// Mono -> Mono.
- pFramesOutF[0] = frames[pMP3->framesConsumed] / 32768.0f;
+ pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f;
} else {
// Mono -> Stereo.
- pFramesOutF[0] = frames[pMP3->framesConsumed] / 32768.0f;
- pFramesOutF[1] = frames[pMP3->framesConsumed] / 32768.0f;
+ pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f;
+ pFramesOutF[1] = frames[pMP3->pcmFramesConsumedInMP3Frame] / 32768.0f;
}
} else {
if (pMP3->channels == 1) {
// Stereo -> Mono
float sample = 0;
- sample += frames[(pMP3->framesConsumed*pMP3->frameChannels)+0] / 32768.0f;
- sample += frames[(pMP3->framesConsumed*pMP3->frameChannels)+1] / 32768.0f;
+ sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0] / 32768.0f;
+ sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1] / 32768.0f;
pFramesOutF[0] = sample * 0.5f;
} else {
// Stereo -> Stereo
- pFramesOutF[0] = frames[(pMP3->framesConsumed*pMP3->frameChannels)+0] / 32768.0f;
- pFramesOutF[1] = frames[(pMP3->framesConsumed*pMP3->frameChannels)+1] / 32768.0f;
+ pFramesOutF[0] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0] / 32768.0f;
+ pFramesOutF[1] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1] / 32768.0f;
}
}
#else
- if (pMP3->frameChannels == 1) {
+ if (pMP3->mp3FrameChannels == 1) {
if (pMP3->channels == 1) {
// Mono -> Mono.
- pFramesOutF[0] = frames[pMP3->framesConsumed];
+ pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame];
} else {
// Mono -> Stereo.
- pFramesOutF[0] = frames[pMP3->framesConsumed];
- pFramesOutF[1] = frames[pMP3->framesConsumed];
+ pFramesOutF[0] = frames[pMP3->pcmFramesConsumedInMP3Frame];
+ pFramesOutF[1] = frames[pMP3->pcmFramesConsumedInMP3Frame];
}
} else {
if (pMP3->channels == 1) {
// Stereo -> Mono
float sample = 0;
- sample += frames[(pMP3->framesConsumed*pMP3->frameChannels)+0];
- sample += frames[(pMP3->framesConsumed*pMP3->frameChannels)+1];
+ sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0];
+ sample += frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1];
pFramesOutF[0] = sample * 0.5f;
} else {
// Stereo -> Stereo
- pFramesOutF[0] = frames[(pMP3->framesConsumed*pMP3->frameChannels)+0];
- pFramesOutF[1] = frames[(pMP3->framesConsumed*pMP3->frameChannels)+1];
+ pFramesOutF[0] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+0];
+ pFramesOutF[1] = frames[(pMP3->pcmFramesConsumedInMP3Frame*pMP3->mp3FrameChannels)+1];
}
}
#endif
- pMP3->framesConsumed += 1;
- pMP3->framesRemaining -= 1;
- frameCount -= 1;
+ pMP3->pcmFramesConsumedInMP3Frame += 1;
+ pMP3->pcmFramesRemainingInMP3Frame -= 1;
totalFramesRead += 1;
+ frameCount -= 1;
pFramesOutF += pSRC->config.channels;
}
@@ -2530,11 +2570,11 @@ static drmp3_uint64 drmp3_read_src(drmp3_src* pSRC, drmp3_uint64 frameCount, voi
break;
}
- drmp3_assert(pMP3->framesRemaining == 0);
+ drmp3_assert(pMP3->pcmFramesRemainingInMP3Frame == 0);
// At this point we have exhausted our in-memory buffer so we need to re-fill. Note that the sample rate may have changed
// at this point which means we'll also need to update our sample rate conversion pipeline.
- if (!drmp3_decode_next_frame(pMP3)) {
+ if (drmp3_decode_next_frame(pMP3) == 0) {
break;
}
}
@@ -2710,7 +2750,9 @@ drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* filePath, const drmp3_conf
void drmp3_uninit(drmp3* pMP3)
{
- if (pMP3 == NULL) return;
+ if (pMP3 == NULL) {
+ return;
+ }
#ifndef DR_MP3_NO_STDIO
if (pMP3->onRead == drmp3__on_read_stdio) {
@@ -2721,9 +2763,11 @@ void drmp3_uninit(drmp3* pMP3)
drmp3_free(pMP3->pData);
}
-drmp3_uint64 drmp3_read_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut)
+drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut)
{
- if (pMP3 == NULL || pMP3->onRead == NULL) return 0;
+ if (pMP3 == NULL || pMP3->onRead == NULL) {
+ return 0;
+ }
drmp3_uint64 totalFramesRead = 0;
@@ -2735,7 +2779,7 @@ drmp3_uint64 drmp3_read_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBuff
framesToReadRightNow = framesToRead;
}
- drmp3_uint64 framesJustRead = drmp3_read_f32(pMP3, framesToReadRightNow, temp);
+ drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp);
if (framesJustRead == 0) {
break;
}
@@ -2745,40 +2789,191 @@ drmp3_uint64 drmp3_read_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBuff
}
} else {
totalFramesRead = drmp3_src_read_frames_ex(&pMP3->src, framesToRead, pBufferOut, DRMP3_TRUE);
+ pMP3->currentPCMFrame += totalFramesRead;
}
return totalFramesRead;
}
-drmp3_bool32 drmp3_seek_to_frame(drmp3* pMP3, drmp3_uint64 frameIndex)
+drmp3_bool32 drmp3_seek_to_start_of_stream(drmp3* pMP3)
{
- if (pMP3 == NULL || pMP3->onSeek == NULL) return DRMP3_FALSE;
+ drmp3_assert(pMP3 != NULL);
+ drmp3_assert(pMP3->onSeek != NULL);
// Seek to the start of the stream to begin with.
- if (!pMP3->onSeek(pMP3->pUserData, 0, drmp3_seek_origin_start)) {
+ if (!drmp3__on_seek(pMP3, 0, drmp3_seek_origin_start)) {
return DRMP3_FALSE;
}
// Clear any cached data.
- pMP3->framesConsumed = 0;
- pMP3->framesRemaining = 0;
+ pMP3->pcmFramesConsumedInMP3Frame = 0;
+ pMP3->pcmFramesRemainingInMP3Frame = 0;
+ pMP3->currentPCMFrame = 0;
pMP3->dataSize = 0;
pMP3->atEnd = DRMP3_FALSE;
- // TODO: Optimize.
- //
- // This is inefficient. We simply read frames from the start of the stream.
- drmp3_uint64 framesRead = drmp3_read_f32(pMP3, frameIndex, NULL);
- if (framesRead != frameIndex) {
+ return DRMP3_TRUE;
+}
+
+drmp3_bool32 drmp3_seek_to_pcm_frame__brute_force(drmp3* pMP3, drmp3_uint64 frameIndex)
+{
+ drmp3_assert(pMP3 != NULL);
+
+ if (frameIndex == pMP3->currentPCMFrame) {
+ return DRMP3_TRUE;
+ }
+
+ // If we're moving foward we just read from where we're at. Otherwise we need to move back to the start of
+ // the stream and read from the beginning.
+ drmp3_uint64 framesToReadAndDiscard;
+ if (frameIndex >= pMP3->currentPCMFrame) {
+ // Moving foward.
+ framesToReadAndDiscard = frameIndex - pMP3->currentPCMFrame;
+ } else {
+ // Moving backward. Move to the start of the stream and then move forward.
+ framesToReadAndDiscard = frameIndex;
+ if (!drmp3_seek_to_start_of_stream(pMP3)) {
+ return DRMP3_FALSE;
+ }
+ }
+
+ // MP3 is a bit annoying when it comes to seeking because of the bit reservoir. It basically means that an MP3 frame can possibly
+ // depend on some of the data of prior frames. This means it's not as simple as seeking to the first byte of the MP3 frame that
+ // contains the sample because that MP3 frame will need the data from the previous MP3 frame (which we just seeked past!). To
+ // resolve this we seek past a number of MP3 frames up to a point, and then read-and-discard the remainder.
+ drmp3_uint64 maxFramesToReadAndDiscard = DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME * 3;
+
+ // First get rid of anything that's still sitting in the buffer.
+ if (framesToReadAndDiscard > maxFramesToReadAndDiscard && framesToReadAndDiscard > pMP3->pcmFramesRemainingInMP3Frame) {
+ framesToReadAndDiscard -= pMP3->pcmFramesRemainingInMP3Frame;
+ pMP3->currentPCMFrame += pMP3->pcmFramesRemainingInMP3Frame;
+ pMP3->pcmFramesConsumedInMP3Frame += pMP3->pcmFramesRemainingInMP3Frame;
+ pMP3->pcmFramesRemainingInMP3Frame = 0;
+ }
+
+ // Now get rid of leading whole frames.
+ while (framesToReadAndDiscard > maxFramesToReadAndDiscard) {
+ drmp3_uint32 pcmFramesSeeked = drmp3_seek_next_frame(pMP3);
+ if (pcmFramesSeeked == 0) {
+ break;
+ }
+
+ framesToReadAndDiscard -= pcmFramesSeeked;
+ }
+
+ // The last step is to read-and-discard any remaining PCM frames to make it sample-exact.
+ drmp3_uint64 framesRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadAndDiscard, NULL);
+ if (framesRead != framesToReadAndDiscard) {
return DRMP3_FALSE;
}
return DRMP3_TRUE;
}
+drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex)
+{
+ if (pMP3 == NULL || pMP3->onSeek == NULL) {
+ return DRMP3_FALSE;
+ }
+
+ // We currently only support brute force seeking.
+ return drmp3_seek_to_pcm_frame__brute_force(pMP3, frameIndex);
+}
+
+drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3)
+{
+ if (pMP3 == NULL) {
+ return 0;
+ }
+
+ // The way this works is we move back to the start of the stream, iterate over each MP3 frame and calculate the frame count based
+ // on our output sample rate, the seek back to the PCM frame we were sitting on before calling this function.
+
+ // The stream must support seeking for this to work.
+ if (pMP3->onSeek == NULL) {
+ return 0;
+ }
+
+ // We'll need to seek back to where we were, so grab the PCM frame we're currently sitting on so we can restore later.
+ drmp3_uint64 currentPCMFrame = pMP3->currentPCMFrame;
+
+ if (!drmp3_seek_to_start_of_stream(pMP3)) {
+ return 0;
+ }
+
+ drmp3_uint64 totalPCMFrameCount = 0;
+ float totalPCMFrameCountFractionalPart = 0; // <-- With resampling there will be a fractional part to each MP3 frame that we need to accumulate.
+ for (;;) {
+ drmp3_uint32 pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL); // <-- Passing in NULL here will prevent decoding of the MP3 frame which should save time.
+ if (pcmFramesInCurrentMP3FrameIn == 0) {
+ break;
+ }
+
+ float srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate;
+ drmp3_assert(srcRatio > 0);
+
+ float pcmFramesInCurrentMP3FrameOutF = totalPCMFrameCountFractionalPart + (pcmFramesInCurrentMP3FrameIn / srcRatio);
+ drmp3_uint32 pcmFramesInCurrentMP3FrameOut = (drmp3_uint32)pcmFramesInCurrentMP3FrameOutF;
+ totalPCMFrameCountFractionalPart = pcmFramesInCurrentMP3FrameOutF - pcmFramesInCurrentMP3FrameOut;
+ totalPCMFrameCount += pcmFramesInCurrentMP3FrameOut;
+ }
+ // Finally, we need to seek back to where we were.
+ if (!drmp3_seek_to_start_of_stream(pMP3)) {
+ return 0;
+ }
+
+ if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) {
+ return 0;
+ }
+
+ return totalPCMFrameCount;
+}
-float* drmp3__full_decode_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
+drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3)
+{
+ if (pMP3 == NULL) {
+ return 0;
+ }
+
+ // This works the same way as drmp3_get_pcm_frame_count() - move to the start, count MP3 frames, move back to the previous position.
+
+ // The stream must support seeking for this to work.
+ if (pMP3->onSeek == NULL) {
+ return 0;
+ }
+
+ // We'll need to seek back to where we were, so grab the PCM frame we're currently sitting on so we can restore later.
+ drmp3_uint64 currentPCMFrame = pMP3->currentPCMFrame;
+
+ if (!drmp3_seek_to_start_of_stream(pMP3)) {
+ return 0;
+ }
+
+ drmp3_uint64 totalMP3FrameCount = 0;
+ for (;;) {
+ drmp3_uint32 pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL);
+ if (pcmFramesInCurrentMP3FrameIn == 0) {
+ break;
+ }
+
+ totalMP3FrameCount += 1;
+ }
+
+ // Finally, we need to seek back to where we were.
+ if (!drmp3_seek_to_start_of_stream(pMP3)) {
+ return 0;
+ }
+
+ if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) {
+ return 0;
+ }
+
+ return totalMP3FrameCount;
+}
+
+
+float* drmp3__full_read_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
{
drmp3_assert(pMP3 != NULL);
@@ -2789,7 +2984,7 @@ float* drmp3__full_decode_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp
float temp[4096];
for (;;) {
drmp3_uint64 framesToReadRightNow = drmp3_countof(temp) / pMP3->channels;
- drmp3_uint64 framesJustRead = drmp3_read_f32(pMP3, framesToReadRightNow, temp);
+ drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp);
if (framesJustRead == 0) {
break;
}
@@ -2835,35 +3030,35 @@ float* drmp3__full_decode_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp
return pFrames;
}
-float* drmp3_open_and_decode_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
+float* drmp3_open_and_read_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
{
drmp3 mp3;
if (!drmp3_init(&mp3, onRead, onSeek, pUserData, pConfig)) {
return NULL;
}
- return drmp3__full_decode_and_close_f32(&mp3, pConfig, pTotalFrameCount);
+ return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
}
-float* drmp3_open_and_decode_memory_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
+float* drmp3_open_memory_and_read_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
{
drmp3 mp3;
if (!drmp3_init_memory(&mp3, pData, dataSize, pConfig)) {
return NULL;
}
- return drmp3__full_decode_and_close_f32(&mp3, pConfig, pTotalFrameCount);
+ return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
}
#ifndef DR_MP3_NO_STDIO
-float* drmp3_open_and_decode_file_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
+float* drmp3_open_file_and_read_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
{
drmp3 mp3;
if (!drmp3_init_file(&mp3, filePath, pConfig)) {
return NULL;
}
- return drmp3__full_decode_and_close_f32(&mp3, pConfig, pTotalFrameCount);
+ return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
}
#endif
@@ -2890,7 +3085,18 @@ void drmp3_free(void* p)
// REVISION HISTORY
-// ===============
+// ================
+//
+// v0.4.0 - 2018-xx-xx
+// - API CHANGE: Rename some APIs:
+// - drmp3_read_f32 -> to drmp3_read_pcm_frames_f32
+// - drmp3_seek_to_frame -> drmp3_seek_to_pcm_frame
+// - drmp3_open_and_decode_f32 -> drmp3_open_and_read_f32
+// - drmp3_open_and_decode_memory_f32 -> drmp3_open_memory_and_read_f32
+// - drmp3_open_and_decode_file_f32 -> drmp3_open_file_and_read_f32
+// - Add drmp3_get_pcm_frame_count().
+// - Add drmp3_get_mp3_frame_count().
+// - Improve seeking performance.
//
// v0.3.2 - 2018-09-11
// - Fix a couple of memory leaks.
generated by cgit v1.2.3 (git 2.39.1) at 2026-05-13 17:15:39 +0000