2023-10-29 08:11:29 CST W44D0

examples/resampling-audio/main.go

@@ -1,5 +1,186 @@
 package main
 
-func main() {
+import (
+	"fmt"
+	"math"
+	"os"
+	"syscall"
+	"unsafe"
 
+	ffmpeg "github.com/qrtc/ffmpeg-dev-go"
+)
+
+func getFormatFromSampleFmt(sampleFmt ffmpeg.AVSampleFormat) (string, int32) {
+	sampleFmtEntry := []struct {
+		sampleFmt ffmpeg.AVSampleFormat
+		fmtBe     string
+		fmtLe     string
+	}{
+		{ffmpeg.AV_SAMPLE_FMT_U8, "u8", "u8"},
+		{ffmpeg.AV_SAMPLE_FMT_S16, "s16be", "s16le"},
+		{ffmpeg.AV_SAMPLE_FMT_S32, "s32be", "s32le"},
+		{ffmpeg.AV_SAMPLE_FMT_FLT, "f32be", "f32le"},
+		{ffmpeg.AV_SAMPLE_FMT_DBL, "f64be", "f64le"},
+	}
+
+	for _, entry := range sampleFmtEntry {
+		if sampleFmt == entry.sampleFmt {
+			return ffmpeg.AV_NE(entry.fmtBe, entry.fmtLe), 0
+		}
+	}
+
+	fmt.Fprintf(os.Stderr, "sample format %s is not supported as output format\n",
+		ffmpeg.AvGetSampleFmtName(sampleFmt))
+	return ffmpeg.NIL, -1
+}
+
+// Fill dst buffer with nb_samples, generated starting from t.
+func fileSamples(dst []float64, nbSamples, nbChannels, sampleRate int32, t *float64) {
+	var (
+		tincr = 1.0 / float64(sampleRate)
+		c     = 2 * math.Pi * 440.0
+	)
+
+	// generate sin tone with 440Hz frequency and duplicated channels
+	for i := int32(0); i < nbSamples; i++ {
+		dst[i*nbChannels] = math.Sin(c * (*t))
+		for j := int32(1); j < nbChannels; j++ {
+			dst[i*nbChannels+j] = dst[i*nbChannels]
+		}
+		*t += tincr
+	}
+}
+
+func main() {
+	var (
+		srcRate, dstRate             int32 = 48000, 44100
+		srcData, dstData             **uint8
+		srcNbChannels, dstNbChannels int32
+		srcLinesize, dstLinesize     int32
+		srcNbSamples, dstNbSamples   int32 = 1024, 0
+		maxDstNbSamples              int32
+		srcChLayout, dstChLayout     = ffmpeg.AV_CH_LAYOUT_STEREO, ffmpeg.AV_CH_LAYOUT_SURROUND
+		srcSampleFmt, dstSampleFmt   = ffmpeg.AV_SAMPLE_FMT_DBL, ffmpeg.AV_SAMPLE_FMT_S16
+		swrCtx                       *ffmpeg.SwrContext
+		ret                          int32
+		t                            float64
+		dstBufsize                   int32
+		_fmt                         string
+	)
+
+	if len(os.Args) != 2 {
+		fmt.Fprintf(os.Stdout, "Usage: %s output_file\n"+
+			"API example program to show how to resample an audio stream with libswresample.\n"+
+			"This program generates a series of audio frames, resamples them to a specified "+
+			"output format and rate and saves them to an output file named output_file.\n",
+			os.Args[0])
+		os.Exit(1)
+	}
+	dstFilename := os.Args[1]
+
+	dstFile, err := os.OpenFile(dstFilename, os.O_WRONLY|os.O_CREATE|os.O_TRUNC, 0755)
+	if err != nil {
+		fmt.Fprintf(os.Stderr, "Could not open destination file %s\n", dstFilename)
+		os.Exit(1)
+	}
+
+	// create resampler context
+	if swrCtx = ffmpeg.SwrAlloc(); swrCtx == nil {
+		fmt.Fprintf(os.Stderr, "Could not allocate resampler context\n")
+		ret = ffmpeg.AVERROR(syscall.ENOMEM)
+		goto end
+	}
+
+	// set options
+	ffmpeg.AvOptSetInt(swrCtx, "in_channel_layout", srcChLayout, 0)
+	ffmpeg.AvOptSetInt(swrCtx, "in_sample_rate", srcRate, 0)
+	ffmpeg.AvOptSetSampleFmt(swrCtx, "in_sample_fmt", srcSampleFmt, 0)
+
+	ffmpeg.AvOptSetInt(swrCtx, "out_channel_layout", dstChLayout, 0)
+	ffmpeg.AvOptSetInt(swrCtx, "out_sample_rate", dstRate, 0)
+	ffmpeg.AvOptSetSampleFmt(swrCtx, "out_sample_fmt", dstSampleFmt, 0)
+
+	// initialize the resampling context
+	if ret = ffmpeg.SwrInit(swrCtx); ret < 0 {
+		fmt.Fprintf(os.Stderr, "Failed to initialize the resampling context\n")
+		goto end
+	}
+
+	// allocate source and destination samples buffers
+	srcNbChannels = ffmpeg.AvGetChannelLayoutNbChannels(srcChLayout)
+	if ret = ffmpeg.AvSamplesAllocArrayAndSamples(&srcData, &srcLinesize, srcNbChannels,
+		srcNbSamples, srcSampleFmt, 0); ret < 0 {
+		fmt.Fprintf(os.Stderr, "Could not allocate source samples\n")
+		goto end
+	}
+
+	// compute the number of converted samples: buffering is avoided
+	// ensuring that the output buffer will contain at least all the
+	// converted input samples
+	dstNbSamples = ffmpeg.AvRescaleRnd(srcNbSamples, dstRate, srcRate, ffmpeg.AV_ROUND_UP)
+	maxDstNbSamples = dstNbSamples
+
+	// buffer is going to be directly written to a rawaudio file, no alignment
+	dstNbChannels = ffmpeg.AvGetChannelLayoutNbChannels(dstChLayout)
+	if ret = ffmpeg.AvSamplesAllocArrayAndSamples(&dstData, &dstLinesize, dstNbChannels,
+		dstNbSamples, dstSampleFmt, 0); ret < 0 {
+		fmt.Fprintf(os.Stderr, "Could not allocate destination samples\n")
+		goto end
+	}
+
+	for ok := true; ok; ok = (t < 10) {
+		// generate synthetic audio
+		fileSamples(unsafe.Slice((*float64)(unsafe.Pointer(*srcData)), srcNbSamples*srcNbChannels),
+			srcNbSamples, srcNbChannels, srcRate, &t)
+
+		//compute destination number of samples
+		dstNbSamples = ffmpeg.AvRescaleRnd(ffmpeg.SwrGetDelay(swrCtx, srcRate)+srcNbSamples,
+			dstRate, srcRate, ffmpeg.AV_ROUND_UP)
+		if dstNbSamples > maxDstNbSamples {
+			ffmpeg.AvFreep(dstData)
+			if ret = ffmpeg.AvSamplesAlloc(dstData, &dstLinesize, dstNbChannels,
+				dstNbSamples, dstSampleFmt, 1); ret < 0 {
+				break
+			}
+			maxDstNbSamples = dstNbSamples
+		}
+
+		// convert to destination format
+		if ret = ffmpeg.SwrConvert(swrCtx, dstData, dstNbSamples, srcData, srcNbSamples); ret < 0 {
+			fmt.Fprintf(os.Stderr, "Error while converting\n")
+			goto end
+		}
+		if dstBufsize = ffmpeg.AvSamplesGetBufferSize(&dstLinesize, dstNbChannels,
+			ret, dstSampleFmt, 1); dstBufsize < 0 {
+			fmt.Fprintf(os.Stderr, "Could not get sample buffer size\n")
+			goto end
+		}
+		fmt.Fprintf(os.Stdout, "t:%f in:%d out:%d\n", t, srcNbSamples, ret)
+		dstFile.Write(ffmpeg.SliceSlice(dstData, 1, dstBufsize)[0])
+	}
+
+	if _fmt, ret = getFormatFromSampleFmt(dstSampleFmt); ret < 0 {
+		goto end
+	}
+	fmt.Fprintf(os.Stderr, "Resampling succeeded. Play the output file with the command:\n"+
+		"ffplay -f %s -channel_layout %d -channels %d -ar %d %s\n",
+		_fmt, dstChLayout, dstNbChannels, dstRate, dstFilename)
+
+end:
+	dstFile.Close()
+
+	if srcData != nil {
+		ffmpeg.AvFreep(srcData)
+	}
+	ffmpeg.AvFreep(&srcData)
+
+	if dstData != nil {
+		ffmpeg.AvFreep(dstData)
+	}
+	ffmpeg.AvFreep(&dstData)
+
+	ffmpeg.SwrFree(&swrCtx)
+	if ret < 0 {
+		os.Exit(1)
+	}
 }