int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
if(pts == INT64_MIN)
return s->outpts;
if (s->firstpts == AV_NOPTS_VALUE)
s->outpts = s->firstpts = pts;
if(s->min_compensation >= FLT_MAX) {
return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
} else {
int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
if(fabs(fdelta) > s->min_compensation) {
if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
int ret;
if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
if(ret<0){
av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
}
} else if(s->soft_compensation_duration && s->max_soft_compensation) {
int duration = s->out_sample_rate * s->soft_compensation_duration;
double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
swr_set_compensation(s, comp, duration);
}
}
return s->outpts;
}
}
int AudioController::filter(af_instance *af, mp_audio *data, int /*flags*/) {
auto ac = priv(af); auto d = ac->d;
d->af->delay = 0.0;
Q_ASSERT(d->mixer != nullptr);
if (d->dirty) {
if (d->dirty & Amp)
d->mixer->setAmp(d->amp);
if (d->dirty & Normalizer)
d->mixer->setNormalizer(d->normalizerActivated, d->normalizerOption);
if (d->dirty & ChMap)
d->mixer->setChannelLayoutMap(d->map);
if (d->dirty & Scale)
d->mixer->setScaler(d->tempoScalerActivated, d->scale);
d->dirty = 0;
}
const mp_audio *in = data;
if (d->resample) {
const int frames_delay = swr_get_delay(d->swr, data->rate);
const int frames = av_rescale_rnd(frames_delay + data->samples, d->resampled->rate, data->rate, AV_ROUND_UP);
mp_audio_realloc_min(d->resampled, frames);
d->af->delay = (double)frames/data->rate;
if ((d->resampled->samples = frames))
d->resampled->samples = swr_convert(d->swr, (uchar**)d->resampled->planes, d->resampled->samples, (const uchar**)data->planes, data->samples);
in = d->resampled;
}
d->mixer->apply(in);
*data = *d->af->data;
af->delay += d->mixer->delay();
return 0;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->st->codec;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
return (frame || got_packet) ? 0 : 1;
}
AudioPlayerLoader::ReadResult ChildFFMpegLoader::readFromReadyFrame(QByteArray &result, int64 &samplesAdded) {
int res = 0;
if (_dstSamplesData) { // convert needed
int64_t dstSamples = av_rescale_rnd(swr_get_delay(_swrContext, _srcRate) + _frame->nb_samples, _dstRate, _srcRate, AV_ROUND_UP);
if (dstSamples > _maxResampleSamples) {
_maxResampleSamples = dstSamples;
av_free(_dstSamplesData[0]);
if ((res = av_samples_alloc(_dstSamplesData, 0, AudioToChannels, _maxResampleSamples, AudioToFormat, 1)) < 0) {
_dstSamplesData[0] = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to av_samples_alloc for file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return ReadResult::Error;
}
}
if ((res = swr_convert(_swrContext, _dstSamplesData, dstSamples, (const uint8_t**)_frame->extended_data, _frame->nb_samples)) < 0) {
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to swr_convert for file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return ReadResult::Error;
}
int32 resultLen = av_samples_get_buffer_size(0, AudioToChannels, res, AudioToFormat, 1);
result.append((const char*)_dstSamplesData[0], resultLen);
samplesAdded += resultLen / _sampleSize;
} else {
result.append((const char*)_frame->extended_data[0], _frame->nb_samples * _sampleSize);
samplesAdded += _frame->nb_samples;
}
return ReadResult::Ok;
}
static int get_out_samples(struct af_resample *s, int in_samples)
{
#if LIBSWRESAMPLE_VERSION_MAJOR > 1 || LIBSWRESAMPLE_VERSION_MINOR >= 2
return swr_get_out_samples(s->avrctx, in_samples);
#else
return av_rescale_rnd(in_samples, s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP)
+ swr_get_delay(s->avrctx, s->ctx.out_rate);
#endif
}
boost::int64_t AudioResampleImpl::resultSamples(boost::int64_t sourceSamples) const
{
if (!m_resampler)
return 0;
return av_rescale_rnd(
swr_get_delay(m_resampler.get(), m_from.sampleRate()) + sourceSamples,
m_to.sampleRate(), m_from.sampleRate(), AV_ROUND_UP);
}
static int WriteAudioFrame(AVFormatContext *oc, OutputStream *ost, AVI6 *avi)
{
AVCodecContext *c = NULL;
AVPacket pkt = { 0 };
AVFrame *frame = NULL;
int ret = 0;
int got_packet = 0;
int dst_nb_samples = 0;
av_init_packet(&pkt);
c = ost->st->codec;
frame = GetAudioFrame(ost, avi);
if (frame) {
// フォーマット変換後のサンプル数を決定
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, frame->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
//av_assert0(dst_nb_samples == frame->nb_samples);
// フレームを書き込み可能にする
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
// 音声フォーマットを変換
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
return 0;
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", MakeErrorString(ret));
return 0;
}
if (got_packet) {
ret = WriteFrame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
MakeErrorString(ret));
return 0;
}
}
}
return (frame || got_packet) ? 0 : 1;
}
int MpegLoader::readMore(QByteArray &result, qint64 &samplesAdded)
{
int res;
if ((res = av_read_frame(FmtContext_, &Avpkt_)) < 0)
return -1;
if (Avpkt_.stream_index == StreamId_)
{
av_frame_unref(Frame_);
int got_frame = 0;
if ((res = avcodec_decode_audio4(CodecContext_, Frame_, &got_frame, &Avpkt_)) < 0)
{
av_packet_unref(&Avpkt_);
if (res == AVERROR_INVALIDDATA)
return 0;
return -1;
}
if (got_frame)
{
if (OutSamplesData_)
{
int64_t dstSamples = av_rescale_rnd(swr_get_delay(SwrContext_, SrcRate_) + Frame_->nb_samples, DstRate_, SrcRate_, AV_ROUND_UP);
if (dstSamples > MaxResampleSamples_)
{
MaxResampleSamples_ = dstSamples;
av_free(OutSamplesData_[0]);
if ((res = av_samples_alloc(OutSamplesData_, 0, OutChannels, MaxResampleSamples_, OutFormat, 1)) < 0)
{
OutSamplesData_[0] = 0;
av_packet_unref(&Avpkt_);
return -1;
}
}
if ((res = swr_convert(SwrContext_, OutSamplesData_, dstSamples, (const uint8_t**)Frame_->extended_data, Frame_->nb_samples)) < 0)
{
av_packet_unref(&Avpkt_);
return -1;
}
qint32 resultLen = av_samples_get_buffer_size(0, OutChannels, res, OutFormat, 1);
result.append((const char*)OutSamplesData_[0], resultLen);
samplesAdded += resultLen / SampleSize_;
}
else
{
result.append((const char*)Frame_->extended_data[0], Frame_->nb_samples * SampleSize_);
samplesAdded += Frame_->nb_samples;
}
}
}
av_packet_unref(&Avpkt_);
return 1;
}
static int ffmpeg_decode(struct ffmpeg_file *file)
{
if (file->pkt_decoded >= file->pkt->size) {
int e = av_read_frame(file->format, file->pkt);
if (e < 0) {
return e;
}
file->pkt_decoded = 0;
}
int got_frame = 0;
int e = avcodec_decode_audio4(file->codec, file->frame, &got_frame,
file->pkt);
if (e < 0) {
return e;
}
if (!got_frame) {
return 0;
}
file->pkt_decoded += e;
AVFrame *frame = file->frame;
int delay_nsamples = swr_get_delay(file->swr, file->codec->sample_rate);
int dst_nsamples = av_rescale_rnd(delay_nsamples + frame->nb_samples,
file->sample_rate, file->codec->sample_rate, AV_ROUND_UP);
if (file->buf_nsamples < dst_nsamples) {
if (file->buf) {
av_freep(&file->buf[0]);
}
av_freep(&file->buf);
int e = av_samples_alloc_array_and_samples(&file->buf, NULL,
file->channels, file->sample_rate, file->sample_fmt, 0);
if (e < 0) {
return e;
}
file->buf_nsamples = dst_nsamples;
}
int ns = swr_convert(file->swr, file->buf, dst_nsamples,
(const uint8_t**) frame->data, frame->nb_samples);
int nb = av_samples_get_buffer_size(NULL, file->channels,
ns, file->sample_fmt, 1);
if (nb < 0) {
return nb;
}
file->buf_len = nb;
file->buf_offset = 0;
if (file->frame->pts > 0) {
file->time = file->frame->pts;
}
return nb;
}
BOOL CVideoLivRecord::write_audio_frame(AVStream *st, void* pBuffer, LONG len)
{
AVCodecContext* avcc = st->codec;
AVPacket pkt = {0};
av_init_packet(&pkt);
AVFrame* frame = get_audio_frame(st, pBuffer, len);
int ret = 0;
int dst_nb_samples = 0;
if (frame){
dst_nb_samples = (int)av_rescale_rnd(swr_get_delay(m_pAudioSwrctx, avcc->sample_rate) + frame->nb_samples,
avcc->sample_rate, avcc->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
ret = av_frame_make_writable(m_pAudioFrame);
if (ret < 0){
log("[CVideoLivRecord::write_audio_frame] -- av_frame_make_writable() error");
return FALSE;
}
ret = swr_convert(m_pAudioSwrctx, m_pAudioFrame->data, dst_nb_samples,
(const uint8_t**)frame->data, frame->nb_samples);
if (ret < 0){
log("[CVideoLivRecord::write_audio_frame] -- av_frame_make_writable() error");
return FALSE;
}
frame = m_pAudioFrame;
AVRational tmp = {1, avcc->sample_rate};
frame->pts = av_rescale_q(m_AudioSamplesCount, tmp, avcc->time_base);
m_AudioSamplesCount += dst_nb_samples;
}
int got_packet = 0;
ret = avcodec_encode_audio2(avcc, &pkt, frame, &got_packet);
if (ret < 0){
log("[CVideoLivRecord::write_audio_frame] -- avcodec_encode_audio2() error");
return FALSE;
}
if(got_packet){
av_packet_rescale_ts(&pkt, avcc->time_base, st->time_base);
pkt.stream_index = st->index;
ret = av_interleaved_write_frame(m_pAVFormatContext, &pkt);
//ret = write_audio_frame(m_pAudioStream, pBuffer, len);
if (ret < 0){
log("[CVideoLivRecord::write_audio_frame] -- write_audio_frame() error");
return FALSE;
}
}
return (frame || got_packet)? FALSE : TRUE;
}
int create_channel_data( AVFrame *pFrame )
{
//printf( "In Create Channel data : Thread 1\n" );
int dst_nb_channels = av_get_channel_layout_nb_channels
( AV_CH_LAYOUT_STEREO );
int dst_linesize;
int delay = swr_get_delay( gMedia->pSwrContext,
pFrame->sample_rate );
int dst_nb_samples = av_rescale_rnd( pFrame->nb_samples + delay,
pFrame->sample_rate,
pFrame->sample_rate,
AV_ROUND_UP );
//printf("Destination channels = %d\n",dst_nb_channels);
//printf("Destination samples = %d\n",dst_nb_samples);
int error_check = av_samples_alloc_array_and_samples
( &gMedia->audio_buf,
&dst_linesize,
dst_nb_channels,
dst_nb_samples,
AV_SAMPLE_FMT_FLT,
1 );
if ( error_check < 0 ) {
fprintf( stderr, "Could not allocate destination samples\n" );
}
int data_size = av_samples_get_buffer_size
( NULL,
pFrame->channels,
pFrame->nb_samples,
pFrame->format,
0 );
/* printf("Number of samples = %d\n",pFrame->nb_samples);
printf("Number of bytes = %d\n",pFrame->nb_samples*2*4);
printf("Linesize per channel is %d\n",pFrame->linesize[0]);
printf("Calculated datasize is %d\n",data_size);
*/
swr_convert( gMedia->pSwrContext,
( uint8_t ** )( gMedia->audio_buf ) ,
pFrame->nb_samples,
( const uint8_t ** )pFrame->data,
pFrame->nb_samples );
return data_size;
}
int submitAudioFrame(char *data, int length, long frameTimestamp) {
AVPacket pkt;
int ret;
AVFrame *frame;
int got_packet;
int sampleCount;
// If no stream header has been written yet, don't do anything
if (hasWrittenStreamHeader == 0) {
return 0;
}
frame = avcodec_alloc_frame();
if (frame == NULL) {
fprintf(stderr, "Failed to allocate frame\n");
return -1;
}
av_init_packet(&pkt);
// Copy our data in
memcpy(srcSamplesData[0], data, length);
srcSamplesCount = length / 4;
// Resample to floating point
sampleCount = av_rescale_rnd(swr_get_delay(swrContext, audioCodecCtx->sample_rate) + srcSamplesCount,
audioCodecCtx->sample_rate, audioCodecCtx->sample_rate, AV_ROUND_UP);
if (sampleCount > maxDstSamplesCount) {
// Need to resize the buffer
av_free(dstSamplesData[0]);
ret = av_samples_alloc(dstSamplesData, &dstSamplesLinesize, audioCodecCtx->channels,
sampleCount, audioCodecCtx->sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "av_samples_alloc() failed: %d\n", ret);
return ret;
}
maxDstSamplesCount = sampleCount;
dstSamplesSize = av_samples_get_buffer_size(NULL, audioCodecCtx->channels, sampleCount,
audioCodecCtx->sample_fmt, 0);
}
ret = swr_convert(swrContext, dstSamplesData, sampleCount,
(const unsigned char **)srcSamplesData, srcSamplesCount);
if (ret < 0) {
fprintf(stderr, "swr_convert() failed: %d\n", ret);
return ret;
}
frame->nb_samples = sampleCount;
ret = avcodec_fill_audio_frame(frame, audioCodecCtx->channels,
audioCodecCtx->sample_fmt, dstSamplesData[0], dstSamplesSize, 0);
if (ret < 0) {
fprintf(stderr, "avcodec_fill_audio_frame() failed: %d\n", ret);
avcodec_free_frame(&frame);
return ret;
}
// pkt is freed on failure or !got_packet
pkt.data = NULL;
pkt.size = 0;
ret = avcodec_encode_audio2(audioCodecCtx, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "avcodec_encode_audio2() failed: %d\n", ret);
avcodec_free_frame(&frame);
return ret;
}
if (!got_packet) {
// Nothing to do here
return 0;
}
pkt.stream_index = audioStream->index;
pkt.pts = frameTimestamp;
pthread_mutex_lock(&streamLock);
ret = av_interleaved_write_frame(formatContext, &pkt);
pthread_mutex_unlock(&streamLock);
avcodec_free_frame(&frame);
av_free_packet(&pkt);
if (ret != 0) {
fprintf(stderr, "av_interleaved_write_frame() failed: %d\n", ret);
return ret;
}
return 0;
}
status_t FFMPEGer::encodeAudio(MediaBuffer* src, MediaBuffer* dst){
AVCodecContext *c;
AVFrame *frame = NULL;
int ret;
int got_packet;
int dst_nb_samples;
OutputStream* ost = &audio_st;
unsigned char* srcData = (unsigned char*)src->data() + src->range_offset();
int copySize = getAudioEncodeBufferSize();
while(srcData < ((unsigned char*)src->data() + src->range_offset() + src->range_length())){
AVPacket pkt = { 0 }; // data and size must be 0;
av_init_packet(&pkt);
c = ost->st->codec;
frame = audio_st.tmp_frame;
memcpy(frame->data[0], srcData, copySize);
srcData += copySize;
frame->pts = audio_st.next_pts;
audio_st.next_pts += frame->nb_samples;
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
return UNKNOWN_ERROR;
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
ALOGE("Error while converting");
return UNKNOWN_ERROR;
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
ALOGE("Error encoding audio frame: %s", av_err2str(ret));
return UNKNOWN_ERROR;
}
pkt.pts = frame->pts;//
#if 0
static int count = 0;
char a[50] = {0};
sprintf(a, "/sdcard/pcm%d", count++);
FILE* f1 = fopen(a, "ab");
if(f1 != NULL){
size_t res = fwrite(pkt.data, 1, pkt.size, f1);
fclose(f1);
ALOGV("fwrite %d of %d to /sdcard/pcm!", res, pkt.size);
}else
ALOGE("can not fopen /sdcard/pcm!!");
#endif
if (got_packet) {
ret = write_frame(fmt_ctx, &c->time_base, ost->st, &pkt);
if (ret < 0) {
ALOGE("Error while writing audio frame: %s", av_err2str(ret));
return UNKNOWN_ERROR;
}
}
}
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "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",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(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 */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt) < 0))
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
if (dst_file)
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}
AkPacket ConvertAudioFFmpeg::convert(const AkAudioPacket &packet,
const AkCaps &oCaps)
{
AkAudioCaps oAudioCaps(oCaps);
int64_t iSampleLayout = channelLayouts->value(packet.caps().layout(), 0);
AVSampleFormat iSampleFormat =
av_get_sample_fmt(AkAudioCaps::sampleFormatToString(packet.caps().format())
.toStdString().c_str());
int iSampleRate = packet.caps().rate();
int iNChannels = packet.caps().channels();
int iNSamples = packet.caps().samples();
int64_t oSampleLayout = channelLayouts->value(oAudioCaps.layout(),
AV_CH_LAYOUT_STEREO);
AVSampleFormat oSampleFormat =
av_get_sample_fmt(AkAudioCaps::sampleFormatToString(oAudioCaps.format())
.toStdString().c_str());
int oSampleRate = oAudioCaps.rate();
int oNChannels = oAudioCaps.channels();
this->m_resampleContext =
swr_alloc_set_opts(this->m_resampleContext,
oSampleLayout,
oSampleFormat,
oSampleRate,
iSampleLayout,
iSampleFormat,
iSampleRate,
0,
NULL);
if (!this->m_resampleContext)
return AkPacket();
// Create input audio frame.
static AVFrame iFrame;
memset(&iFrame, 0, sizeof(AVFrame));
iFrame.format = iSampleFormat;
iFrame.channels = iNChannels;
iFrame.channel_layout = uint64_t(iSampleLayout);
iFrame.sample_rate = iSampleRate;
iFrame.nb_samples = iNSamples;
iFrame.pts = iFrame.pkt_pts = packet.pts();
if (avcodec_fill_audio_frame(&iFrame,
iFrame.channels,
iSampleFormat,
reinterpret_cast<const uint8_t *>(packet.buffer().constData()),
packet.buffer().size(),
1) < 0) {
return AkPacket();
}
// Fill output audio frame.
AVFrame oFrame;
memset(&oFrame, 0, sizeof(AVFrame));
oFrame.format = oSampleFormat;
oFrame.channels = oNChannels;
oFrame.channel_layout = uint64_t(oSampleLayout);
oFrame.sample_rate = oSampleRate;
oFrame.nb_samples = int(swr_get_delay(this->m_resampleContext, oSampleRate))
+ iFrame.nb_samples
* oSampleRate
/ iSampleRate
+ 3;
oFrame.pts = oFrame.pkt_pts = iFrame.pts * oSampleRate / iSampleRate;
// Calculate the size of the audio buffer.
int frameSize = av_samples_get_buffer_size(oFrame.linesize,
oFrame.channels,
oFrame.nb_samples,
oSampleFormat,
1);
QByteArray oBuffer(frameSize, Qt::Uninitialized);
if (avcodec_fill_audio_frame(&oFrame,
oFrame.channels,
oSampleFormat,
reinterpret_cast<const uint8_t *>(oBuffer.constData()),
oBuffer.size(),
1) < 0) {
return AkPacket();
}
// convert to destination format
if (swr_convert_frame(this->m_resampleContext,
&oFrame,
&iFrame) < 0)
return AkPacket();
frameSize = av_samples_get_buffer_size(oFrame.linesize,
oFrame.channels,
oFrame.nb_samples,
oSampleFormat,
1);
oBuffer.resize(frameSize);
AkAudioPacket oAudioPacket;
oAudioPacket.caps() = oAudioCaps;
oAudioPacket.caps().samples() = oFrame.nb_samples;
oAudioPacket.buffer() = oBuffer;
oAudioPacket.pts() = oFrame.pts;
oAudioPacket.timeBase() = AkFrac(1, oAudioCaps.rate());
oAudioPacket.index() = packet.index();
oAudioPacket.id() = packet.id();
return oAudioPacket.toPacket();
}
CBaseDec::RetCode CFfmpegDec::Decoder(FILE *_in, int /*OutputFd*/, State* state, CAudioMetaData* _meta_data, time_t* time_played, unsigned int* secondsToSkip)
{
in = _in;
RetCode Status=OK;
is_stream = fseek((FILE *)in, 0, SEEK_SET);
if (!SetMetaData((FILE *)in, _meta_data, true)) {
DeInit();
Status=DATA_ERR;
return Status;
}
AVCodecContext *c = avc->streams[best_stream]->codec;
mutex.lock();
int r = avcodec_open2(c, codec, NULL);
mutex.unlock();
if (r)
{
DeInit();
Status=DATA_ERR;
return Status;
}
SwrContext *swr = swr_alloc();
if (!swr) {
mutex.lock();
avcodec_close(c);
mutex.unlock();
DeInit();
Status=DATA_ERR;
return Status;
}
mSampleRate = samplerate;
mChannels = av_get_channel_layout_nb_channels(AV_CH_LAYOUT_STEREO);
#if __BYTE_ORDER == __LITTLE_ENDIAN
audioDecoder->PrepareClipPlay(mChannels, mSampleRate, 16, 1);
#else
audioDecoder->PrepareClipPlay(mChannels, mSampleRate, 16, 0);
#endif
AVFrame *frame = NULL;
AVPacket rpacket;
av_init_packet(&rpacket);
av_opt_set_int(swr, "in_channel_layout", c->channel_layout, 0);
//av_opt_set_int(swr, "out_channel_layout", c->channel_layout, 0);
av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(swr, "in_sample_rate", c->sample_rate, 0);
av_opt_set_int(swr, "out_sample_rate", c->sample_rate, 0);
av_opt_set_int(swr, "in_sample_fmt", c->sample_fmt, 0);
av_opt_set_int(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
swr_init(swr);
uint8_t *outbuf = NULL;
int outsamples = 0;
int outsamples_max = 0;
int64_t pts = 0, start_pts = 0, next_skip_pts = 0;
uint64_t skip = 0;
int seek_flags = 0;
do
{
int actSecsToSkip = *secondsToSkip;
if (!is_stream && (actSecsToSkip || *state==FF || *state==REV) && avc->streams[best_stream]->time_base.num) {
if (!next_skip_pts || pts >= next_skip_pts) {
skip = avc->streams[best_stream]->time_base.den / avc->streams[best_stream]->time_base.num;
if (actSecsToSkip)
skip *= actSecsToSkip;
if (*state == REV) {
next_skip_pts = pts - skip;
pts = next_skip_pts - skip/4;
seek_flags = AVSEEK_FLAG_BACKWARD;
if (pts < start_pts) {
pts = start_pts;
*state = PAUSE;
}
} else {
pts += skip;
next_skip_pts = pts + skip/4;
seek_flags = 0;
}
av_seek_frame(avc, best_stream, pts, seek_flags);
skip = 0;
// if a custom value was set we only jump once
if (actSecsToSkip != 0) {
*state=PLAY;
*secondsToSkip = 0;
}
}
}
while(*state==PAUSE && !is_stream)
usleep(10000);
if (av_read_frame(avc, &rpacket)) {
Status=DATA_ERR;
break;
}
if (rpacket.stream_index != best_stream) {
av_free_packet(&rpacket);
continue;
}
AVPacket packet = rpacket;
while (packet.size > 0) {
int got_frame = 0;
if (!frame) {
if (!(frame = avcodec_alloc_frame())) {
Status=DATA_ERR;
break;
}
} else
avcodec_get_frame_defaults(frame);
int len = avcodec_decode_audio4(c, frame, &got_frame, &packet);
if (len < 0) {
// skip frame
packet.size = 0;
avcodec_flush_buffers(c);
mutex.lock();
avcodec_close(c);
avcodec_open2(c, codec, NULL);
mutex.unlock();
continue;
}
if (got_frame && *state!=PAUSE) {
int out_samples;
outsamples = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (outsamples > outsamples_max) {
av_free(outbuf);
if (av_samples_alloc(&outbuf, &out_samples, mChannels, //c->channels,
frame->nb_samples, AV_SAMPLE_FMT_S16, 1) < 0) {
Status=WRITE_ERR;
packet.size = 0;
break;
}
outsamples_max = outsamples;
}
outsamples = swr_convert(swr, &outbuf, outsamples,
(const uint8_t **) &frame->data[0], frame->nb_samples);
int outbuf_size = av_samples_get_buffer_size(&out_samples, mChannels, //c->channels,
outsamples, AV_SAMPLE_FMT_S16, 1);
if(audioDecoder->WriteClip((unsigned char*) outbuf, outbuf_size) != outbuf_size)
{
fprintf(stderr,"%s: PCM write error (%s).\n", ProgName, strerror(errno));
Status=WRITE_ERR;
}
pts = av_frame_get_best_effort_timestamp(frame);
if (!start_pts)
start_pts = pts;
}
packet.size -= len;
packet.data += len;
}
if (time_played && avc->streams[best_stream]->time_base.den)
*time_played = (pts - start_pts) * avc->streams[best_stream]->time_base.num / avc->streams[best_stream]->time_base.den;
av_free_packet(&rpacket);
} while (*state!=STOP_REQ && Status==OK);
audioDecoder->StopClip();
meta_data_valid = false;
swr_free(&swr);
av_free(outbuf);
av_free_packet(&rpacket);
avcodec_free_frame(&frame);
avcodec_close(c);
//av_free(avcc);
DeInit();
if (_meta_data->cover_temporary && !_meta_data->cover.empty()) {
_meta_data->cover_temporary = false;
unlink(_meta_data->cover.c_str());
}
return Status;
}
static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
int got_packet, ret, dst_nb_samples;
av_init_packet(&pkt);
c = st->codec;
get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_samples_data[0]);
ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
dst_nb_samples, c->sample_fmt, 0);
if (ret < 0)
exit(1);
max_dst_nb_samples = dst_nb_samples;
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
c->sample_fmt, 0);
}
/* convert to destination format */
ret = swr_convert(swr_ctx,
dst_samples_data, dst_nb_samples,
(const uint8_t **)src_samples_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
} else {
dst_nb_samples = src_nb_samples;
}
audio_frame->nb_samples = dst_nb_samples;
audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
dst_samples_data[0], dst_samples_size, 0);
samples_count += dst_nb_samples;
ret = avcodec_encode_audio2(c, &pkt, audio_frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (!got_packet)
return;
/* rescale output packet timestamp values from codec to stream timebase */
pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base);
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(oc, &pkt);
if (ret != 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
void ACapsConvertElement::iStream(const QbPacket &packet)
{
if (!packet.caps().isValid() ||
packet.caps().mimeType() != "audio/x-raw" ||
this->state() != ElementStatePlaying)
return;
// Input Format
AVSampleFormat iSampleFormat = av_get_sample_fmt(packet.caps().property("format").toString().toStdString().c_str());
int iNChannels = packet.caps().property("channels").toInt();
int64_t iChannelLayout = av_get_channel_layout(packet.caps().property("layout").toString().toStdString().c_str());
int iNPlanes = av_sample_fmt_is_planar(iSampleFormat)? iNChannels: 1;
int iSampleRate = packet.caps().property("rate").toInt();
int iNSamples = packet.caps().property("samples").toInt();
if (iNSamples < 1)
iNSamples = 1024;
bool sameMimeType = packet.caps().mimeType() == this->m_caps.mimeType();
// Output Format
AVSampleFormat oSampleFormat = (sameMimeType && this->m_caps.dynamicPropertyNames().contains("format"))?
av_get_sample_fmt(this->m_caps.property("format").toString().toStdString().c_str()):
iSampleFormat;
int oNChannels = (sameMimeType && this->m_caps.dynamicPropertyNames().contains("channels"))?
this->m_caps.property("channels").toInt():
iNChannels;
int64_t oChannelLayout = (sameMimeType && this->m_caps.dynamicPropertyNames().contains("layout"))?
av_get_channel_layout(this->m_caps.property("layout").toString().toStdString().c_str()):
iChannelLayout;
int oSampleRate = (sameMimeType && this->m_caps.dynamicPropertyNames().contains("rate"))?
this->m_caps.property("rate").toInt():
iSampleRate;
QVector<uint8_t *> iData(iNPlanes);
int iLineSize;
if (av_samples_fill_arrays(&iData.data()[0],
&iLineSize,
(const uint8_t *) packet.buffer().data(),
iNChannels,
iNSamples,
iSampleFormat,
1) < 0)
return;
QbCaps caps1(packet.caps());
QbCaps caps2(this->m_curInputCaps);
caps1.setProperty("samples", QVariant());
caps2.setProperty("samples", QVariant());
if (caps1 != caps2)
{
// create resampler context
this->m_resampleContext = SwrContextPtr(swr_alloc(), this->deleteSwrContext);
if (!this->m_resampleContext)
return;
// set options
av_opt_set_int(this->m_resampleContext.data(), "in_channel_layout", iChannelLayout, 0);
av_opt_set_int(this->m_resampleContext.data(), "in_sample_rate", iSampleRate, 0);
av_opt_set_sample_fmt(this->m_resampleContext.data(), "in_sample_fmt", iSampleFormat, 0);
av_opt_set_int(this->m_resampleContext.data(), "out_channel_layout", oChannelLayout, 0);
av_opt_set_int(this->m_resampleContext.data(), "out_sample_rate", oSampleRate, 0);
av_opt_set_sample_fmt(this->m_resampleContext.data(), "out_sample_fmt", oSampleFormat, 0);
// initialize the resampling context
if (swr_init(this->m_resampleContext.data()) < 0)
return;
this->m_curInputCaps = packet.caps();
}
// compute destination number of samples
int oNSamples = av_rescale_rnd(swr_get_delay(this->m_resampleContext.data(),
iSampleRate) +
iNSamples,
oSampleRate,
iSampleRate,
AV_ROUND_UP);
// buffer is going to be directly written to a rawaudio file, no alignment
int oNPlanes = av_sample_fmt_is_planar(oSampleFormat)? oNChannels: 1;
QVector<uint8_t *> oData(oNPlanes);
int oLineSize;
int oBufferSize = av_samples_get_buffer_size(&oLineSize,
oNChannels,
oNSamples,
oSampleFormat,
1);
QSharedPointer<uchar> oBuffer(new uchar[oBufferSize]);
if (!oBuffer)
return;
if (av_samples_fill_arrays(&oData.data()[0],
&oLineSize,
(const uint8_t *) oBuffer.data(),
oNChannels,
oNSamples,
oSampleFormat,
1) < 0)
return;
// convert to destination format
if (swr_convert(this->m_resampleContext.data(),
oData.data(),
oNSamples,
(const uint8_t **) iData.data(),
iNSamples) < 0)
return;
const char *format = av_get_sample_fmt_name(oSampleFormat);
char layout[256];
av_get_channel_layout_string(layout,
sizeof(layout),
oNChannels,
oChannelLayout);
QString caps = QString("audio/x-raw,"
"format=%1,"
"channels=%2,"
"rate=%3,"
"layout=%4,"
"samples=%5").arg(format)
.arg(oNChannels)
.arg(oSampleRate)
.arg(layout)
.arg(oNSamples);
QbPacket oPacket(caps,
oBuffer,
oBufferSize);
oPacket.setPts(packet.pts());
oPacket.setDuration(packet.duration());
oPacket.setTimeBase(packet.timeBase());
oPacket.setIndex(packet.index());
emit this->oStream(oPacket);
}
/**
* Decode a frame to a packet, run the result through SwrContext, if desired, encode it via an appropriate
* encoder, and write the results to the Java-side native buffer.
*
* @param aio FFAudio context
* @param cached true or false
* @return number of bytes placed into java buffer or a negative value, if something went wrong
*/
static int decode_packet(FFAudioIO *aio, int cached) {
int res = 0;
uint8_t **resample_buf = NULL;
jobject byte_buffer = NULL;
uint8_t *javaBuffer = NULL;
uint32_t out_buf_size = 0;
int out_buf_samples = 0;
int64_t out_channel_count;
int64_t out_sample_rate;
int flush = aio->got_frame;
enum AVSampleFormat out;
int bytesConsumed = 0;
init_ids(aio->env, aio->java_instance);
av_opt_get_int(aio->swr_context, "out_channel_count", 0, &out_channel_count);
av_opt_get_int(aio->swr_context, "out_sample_rate", 0, &out_sample_rate);
av_opt_get_sample_fmt(aio->swr_context, "out_sample_fmt", 0, &out);
resample_buf = av_mallocz(sizeof(uint8_t *) * 1); // one plane!
// make sure we really have an audio packet
if (aio->decode_packet.stream_index == aio->stream_index) {
// decode frame
// got_frame indicates whether we got a frame
bytesConsumed = avcodec_decode_audio4(aio->decode_context, aio->decode_frame, &aio->got_frame, &aio->decode_packet);
if (bytesConsumed < 0) {
throwUnsupportedAudioFileExceptionIfError(aio->env, bytesConsumed, "Failed to decode audio frame.");
return bytesConsumed;
}
if (aio->got_frame) {
aio->decoded_samples += aio->decode_frame->nb_samples;
out_buf_samples = aio->decode_frame->nb_samples;
#ifdef DEBUG
fprintf(stderr, "samples%s n:%" PRIu64 " nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
aio->decoded_samples, aio->decode_frame->nb_samples,
av_ts2timestr(aio->decode_frame->pts, &aio->decode_context->time_base));
#endif
// adjust out sample number for a different sample rate
// this is an estimate!!
out_buf_samples = av_rescale_rnd(
swr_get_delay(aio->swr_context, aio->stream->codecpar->sample_rate) + aio->decode_frame->nb_samples,
out_sample_rate,
aio->stream->codecpar->sample_rate,
AV_ROUND_UP
);
// allocate new aio->audio_data buffers
res = av_samples_alloc(aio->audio_data, NULL, av_frame_get_channels(aio->decode_frame),
aio->decode_frame->nb_samples, aio->decode_frame->format, 1);
if (res < 0) {
throwIOExceptionIfError(aio->env, res, "Could not allocate audio buffer.");
return AVERROR(ENOMEM);
}
// copy audio data to aio->audio_data
av_samples_copy(aio->audio_data, aio->decode_frame->data, 0, 0,
aio->decode_frame->nb_samples, av_frame_get_channels(aio->decode_frame), aio->decode_frame->format);
res = resample(aio, resample_buf, out_buf_samples, (const uint8_t **)aio->audio_data, aio->decode_frame->nb_samples);
if (res < 0) goto bail;
else out_buf_samples = res;
} else if (flush && swr_get_delay(aio->swr_context, aio->stream->codecpar->sample_rate)) {
res = resample(aio, resample_buf, swr_get_delay(aio->swr_context, aio->stream->codecpar->sample_rate), NULL, 0);
if (res < 0) goto bail;
else out_buf_samples = res;
} else {
#ifdef DEBUG
fprintf(stderr, "Got no frame.\n");
#endif
}
if (out_buf_samples > 0) {
res = av_samples_get_buffer_size(NULL, (int)out_channel_count, out_buf_samples, out, 1);
if (res < 0) goto bail;
else out_buf_size = res;
// ensure native buffer capacity
if (aio->java_buffer_capacity < out_buf_size) {
aio->java_buffer_capacity = (*aio->env)->CallIntMethod(aio->env, aio->java_instance, setNativeBufferCapacity_MID, (jint)out_buf_size);
}
// get java-managed byte buffer reference
byte_buffer = (*aio->env)->GetObjectField(aio->env, aio->java_instance, nativeBuffer_FID);
if (!byte_buffer) {
res = -1;
throwIOExceptionIfError(aio->env, 1, "Failed to get native buffer.");
goto bail;
}
// we have some samples, let's copy them to the java buffer, using the desired encoding
javaBuffer = (uint8_t *)(*aio->env)->GetDirectBufferAddress(aio->env, byte_buffer);
if (!javaBuffer) {
throwIOExceptionIfError(aio->env, 1, "Failed to get address for native buffer.");
goto bail;
}
if (aio->encode_context) {
aio->encode_frame->nb_samples = out_buf_samples;
res = encode_buffer(aio, resample_buf[0], out_buf_size, javaBuffer);
if (res < 0) {
out_buf_size = 0;
goto bail;
}
out_buf_size = res;
} else {
memcpy(javaBuffer, resample_buf[0], out_buf_size);
}
// we already wrote to the buffer, now we still need to
// set new bytebuffer limit and position to 0.
(*aio->env)->CallObjectMethod(aio->env, byte_buffer, rewind_MID);
(*aio->env)->CallObjectMethod(aio->env, byte_buffer, limit_MID, out_buf_size);
}
}
aio->resampled_samples += out_buf_size;
bail:
if (resample_buf) {
if (resample_buf[0]) av_freep(&resample_buf[0]);
av_free(resample_buf);
}
if (aio->audio_data[0]) av_freep(&aio->audio_data[0]);
return res;
}
std::int64_t Swr::GetDelay(std::int64_t base)
{
assert(this->context != nullptr);
return swr_get_delay(this->context, base);
}
static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
int got_packet, ret, dst_nb_samples;
av_init_packet(&pkt);
c = st->codec;
if (!flush) {
get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_samples_data[0]);
ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
dst_nb_samples, c->sample_fmt, 0);
if (ret < 0)
exit(1);
max_dst_nb_samples = dst_nb_samples;
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
c->sample_fmt, 0);
}
/* convert to destination format */
ret = swr_convert(swr_ctx,
dst_samples_data, dst_nb_samples,
(const uint8_t **)src_samples_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
} else {
dst_nb_samples = src_nb_samples;
}
audio_frame->nb_samples = dst_nb_samples;
audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
dst_samples_data[0], dst_samples_size, 0);
samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (!got_packet) {
if (flush)
audio_is_eof = 1;
return;
}
ret = write_frame(oc, &c->time_base, st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
static int get_delay(struct af_resample *s)
{
return swr_get_delay(s->avrctx, s->ctx.in_rate);
}
int decode_frame_from_packet(State *state, AVPacket *aPacket, int *frame_size_ptr, int from_thread)
{
int n;
int16_t *samples;
AVPacket *pkt = aPacket;
AVFrame *decoded_frame = NULL;
int got_frame = 0;
int64_t src_ch_layout, dst_ch_layout;
int src_rate, dst_rate;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt, dst_sample_fmt;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (aPacket->stream_index == state->audio_stream) {
if (!decoded_frame) {
if (!(decoded_frame = avcodec_alloc_frame())) {
__android_log_print(ANDROID_LOG_INFO, "TAG", "Could not allocate audio frame\n");
return -2;
}
}
if (avcodec_decode_audio4(state->audio_st->codec, decoded_frame, &got_frame, aPacket) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "TAG", "avcodec_decode_audio4() decoded no frame");
return -2;
}
int data_size = 0;
if (got_frame) {
/* if a frame has been decoded, output it */
data_size = av_samples_get_buffer_size(NULL, state->audio_st->codec->channels,
decoded_frame->nb_samples,
state->audio_st->codec->sample_fmt, 1);
} else {
*frame_size_ptr = 0;
return 0;
}
if (decoded_frame->format != AV_SAMPLE_FMT_S16) {
src_nb_samples = decoded_frame->nb_samples;
src_linesize = (int) decoded_frame->linesize;
src_data = decoded_frame->data;
if (decoded_frame->channel_layout == 0) {
decoded_frame->channel_layout = av_get_default_channel_layout(decoded_frame->channels);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
//fprintf(stderr, "Could not allocate resampler context\n");
//ret = AVERROR(ENOMEM);
//goto end;
}
src_rate = decoded_frame->sample_rate;
dst_rate = decoded_frame->sample_rate;
src_ch_layout = decoded_frame->channel_layout;
dst_ch_layout = decoded_frame->channel_layout;
src_sample_fmt = decoded_frame->format;
dst_sample_fmt = AV_SAMPLE_FMT_S16;
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
__android_log_print(ANDROID_LOG_INFO, "TAG", "Failed to initialize the resampling context\n");
//goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
__android_log_print(ANDROID_LOG_INFO, "TAG", "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 */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
__android_log_print(ANDROID_LOG_INFO, "TAG", "Could not allocate destination samples\n");
//goto end;
}
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)decoded_frame->data, src_nb_samples);
if (ret < 0) {
__android_log_print(ANDROID_LOG_INFO, "TAG", "Error while converting\n");
//goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
//goto end;
}
samples = malloc(dst_bufsize);
memcpy(samples, dst_data[0], dst_bufsize);
data_size = dst_bufsize;
swr_free(&swr_ctx);
} else {
/* if a frame has been decoded, output it */
data_size = av_samples_get_buffer_size(NULL, state->audio_st->codec->channels,
decoded_frame->nb_samples, state->audio_st->codec->sample_fmt, 1);
samples = malloc(data_size);
memcpy(samples, decoded_frame->data[0], data_size);
}
*frame_size_ptr = data_size;
// TODO add this call back!
//*pts_ptr = pts;
n = 2 * state->audio_st->codec->channels;
state->audio_clock += (double)*frame_size_ptr /
(double)(n * state->audio_st->codec->sample_rate);
/* if update, update the audio clock w/pts */
if(pkt->pts != AV_NOPTS_VALUE) {
state->audio_clock = av_q2d(state->audio_st->time_base) * pkt->pts;
}
//*frame_size_ptr = data_size;
state->write_audio_callback(state->clazz, samples, data_size, from_thread);
avcodec_free_frame(&decoded_frame);
free(samples);
return AUDIO_DATA_ID;
}
return 0;
}
void cAudio::run()
{
lt_info("====================== start decoder thread ================================\n");
/* libavcodec & friends */
av_register_all();
AVCodec *codec;
AVFormatContext *avfc = NULL;
AVInputFormat *inp;
AVFrame *frame;
uint8_t *inbuf = (uint8_t *)av_malloc(INBUF_SIZE);
AVPacket avpkt;
int ret, driver;
/* libao */
ao_info *ai;
// ao_device *adevice;
// ao_sample_format sformat;
/* resample */
SwrContext *swr = NULL;
uint8_t *obuf = NULL;
int obuf_sz = 0; /* in samples */
int obuf_sz_max = 0;
int o_ch, o_sr; /* output channels and sample rate */
uint64_t o_layout; /* output channels layout */
char tmp[64] = "unknown";
curr_pts = 0;
av_init_packet(&avpkt);
inp = av_find_input_format("mpegts");
AVIOContext *pIOCtx = avio_alloc_context(inbuf, INBUF_SIZE, // internal Buffer and its size
0, // bWriteable (1=true,0=false)
NULL, // user data; will be passed to our callback functions
_my_read, // read callback
NULL, // write callback
NULL); // seek callback
avfc = avformat_alloc_context();
avfc->pb = pIOCtx;
avfc->iformat = inp;
avfc->probesize = 188*5;
thread_started = true;
if (avformat_open_input(&avfc, NULL, inp, NULL) < 0) {
lt_info("%s: avformat_open_input() failed.\n", __func__);
goto out;
}
ret = avformat_find_stream_info(avfc, NULL);
lt_debug("%s: avformat_find_stream_info: %d\n", __func__, ret);
if (avfc->nb_streams != 1)
{
lt_info("%s: nb_streams: %d, should be 1!\n", __func__, avfc->nb_streams);
goto out;
}
if (avfc->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO)
lt_info("%s: stream 0 no audio codec? 0x%x\n", __func__, avfc->streams[0]->codec->codec_type);
c = avfc->streams[0]->codec;
codec = avcodec_find_decoder(c->codec_id);
if (!codec) {
lt_info("%s: Codec for %s not found\n", __func__, avcodec_get_name(c->codec_id));
goto out;
}
if (avcodec_open2(c, codec, NULL) < 0) {
lt_info("%s: avcodec_open2() failed\n", __func__);
goto out;
}
frame = av_frame_alloc();
if (!frame) {
lt_info("%s: avcodec_alloc_frame failed\n", __func__);
goto out2;
}
/* output sample rate, channels, layout could be set here if necessary */
o_ch = c->channels; /* 2 */
o_sr = c->sample_rate; /* 48000 */
o_layout = c->channel_layout; /* AV_CH_LAYOUT_STEREO */
if (sformat.channels != o_ch || sformat.rate != o_sr ||
sformat.byte_format != AO_FMT_NATIVE || sformat.bits != 16 || adevice == NULL)
{
driver = ao_default_driver_id();
sformat.bits = 16;
sformat.channels = o_ch;
sformat.rate = o_sr;
sformat.byte_format = AO_FMT_NATIVE;
sformat.matrix = 0;
if (adevice)
ao_close(adevice);
adevice = ao_open_live(driver, &sformat, NULL);
ai = ao_driver_info(driver);
lt_info("%s: changed params ch %d srate %d bits %d adevice %p\n",
__func__, o_ch, o_sr, 16, adevice);;
lt_info("libao driver: %d name '%s' short '%s' author '%s'\n",
driver, ai->name, ai->short_name, ai->author);
}
#if 0
lt_info(" driver options:");
for (int i = 0; i < ai->option_count; ++i)
fprintf(stderr, " %s", ai->options[i]);
fprintf(stderr, "\n");
#endif
av_get_sample_fmt_string(tmp, sizeof(tmp), c->sample_fmt);
lt_info("decoding %s, sample_fmt %d (%s) sample_rate %d channels %d\n",
avcodec_get_name(c->codec_id), c->sample_fmt, tmp, c->sample_rate, c->channels);
swr = swr_alloc_set_opts(swr,
o_layout, AV_SAMPLE_FMT_S16, o_sr, /* output */
c->channel_layout, c->sample_fmt, c->sample_rate, /* input */
0, NULL);
if (! swr) {
lt_info("could not alloc resample context\n");
goto out3;
}
swr_init(swr);
while (thread_started) {
int gotframe = 0;
if (av_read_frame(avfc, &avpkt) < 0)
break;
avcodec_decode_audio4(c, frame, &gotframe, &avpkt);
if (gotframe && thread_started) {
int out_linesize;
obuf_sz = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) +
frame->nb_samples, o_sr, c->sample_rate, AV_ROUND_UP);
if (obuf_sz > obuf_sz_max) {
lt_info("obuf_sz: %d old: %d\n", obuf_sz, obuf_sz_max);
av_free(obuf);
if (av_samples_alloc(&obuf, &out_linesize, o_ch,
frame->nb_samples, AV_SAMPLE_FMT_S16, 1) < 0) {
lt_info("av_samples_alloc failed\n");
av_free_packet(&avpkt);
break; /* while (thread_started) */
}
obuf_sz_max = obuf_sz;
}
obuf_sz = swr_convert(swr, &obuf, obuf_sz,
(const uint8_t **)frame->extended_data, frame->nb_samples);
curr_pts = av_frame_get_best_effort_timestamp(frame);
lt_debug("%s: pts 0x%" PRIx64 " %3f\n", __func__, curr_pts, curr_pts/90000.0);
int o_buf_sz = av_samples_get_buffer_size(&out_linesize, o_ch,
obuf_sz, AV_SAMPLE_FMT_S16, 1);
ao_play(adevice, (char *)obuf, o_buf_sz);
}
av_free_packet(&avpkt);
}
// ao_close(adevice); /* can take long :-( */
av_free(obuf);
swr_free(&swr);
out3:
av_frame_free(&frame);
out2:
avcodec_close(c);
c = NULL;
out:
avformat_close_input(&avfc);
av_free(pIOCtx->buffer);
av_free(pIOCtx);
lt_info("======================== end decoder thread ================================\n");
}
long audio_tutorial_resample(VideoState *is, struct AVFrame *inframe) {
#ifdef __RESAMPLER__
#ifdef __LIBAVRESAMPLE__
// There is pre 1.0 libavresample and then there is above..
#if LIBAVRESAMPLE_VERSION_MAJOR == 0
void **resample_input_bytes = (void **)inframe->extended_data;
#else
uint8_t **resample_input_bytes = (uint8_t **)inframe->extended_data;
#endif
#else
uint8_t **resample_input_bytes = (uint8_t **)inframe->extended_data;
#endif
int resample_nblen = 0;
long resample_long_bytes = 0;
if( is->pResampledOut == NULL || inframe->nb_samples > is->resample_size) {
#if __LIBAVRESAMPLE__
is->resample_size = av_rescale_rnd(avresample_get_delay(is->pSwrCtx) +
inframe->nb_samples,
44100,
44100,
AV_ROUND_UP);
#else
is->resample_size = av_rescale_rnd(swr_get_delay(is->pSwrCtx,
44100) +
inframe->nb_samples,
44100,
44100,
AV_ROUND_UP);
#endif
if(is->pResampledOut != NULL) {
av_free(is->pResampledOut);
is->pResampledOut = NULL;
}
av_samples_alloc(&is->pResampledOut, &is->resample_lines, 2, is->resample_size,
AV_SAMPLE_FMT_S16, 0);
}
#ifdef __LIBAVRESAMPLE__
// OLD API (0.0.3) ... still NEW API (1.0.0 and above).. very frustrating..
// USED IN FFMPEG 1.0 (LibAV SOMETHING!). New in FFMPEG 1.1 and libav 9
#if LIBAVRESAMPLE_VERSION_INT <= 3
// AVResample OLD
resample_nblen = avresample_convert(is->pSwrCtx, (void **)&is->pResampledOut, 0,
is->resample_size,
(void **)resample_input_bytes, 0, inframe->nb_samples);
#else
//AVResample NEW
resample_nblen = avresample_convert(is->pSwrCtx, (uint8_t **)&is->pResampledOut,
0, is->resample_size,
(uint8_t **)resample_input_bytes, 0, inframe->nb_samples);
#endif
#else
// SWResample
resample_nblen = swr_convert(is->pSwrCtx, (uint8_t **)&is->pResampledOut,
is->resample_size,
(const uint8_t **)resample_input_bytes, inframe->nb_samples);
#endif
resample_long_bytes = av_samples_get_buffer_size(NULL, 2, resample_nblen,
AV_SAMPLE_FMT_S16, 1);
if (resample_nblen < 0) {
fprintf(stderr, "reSample to another sample format failed!\n");
return -1;
}
return resample_long_bytes;
#else
return -1;
#endif
}
int procAudioResampling(AVCodecContext * audio_dec_ctx,
AVFrame * pAudioDecodeFrame,
int out_sample_fmt,
int out_channels,
int out_sample_rate,
uint8_t * audio_chunk)
{
SwrContext * swr_ctx = NULL;
int data_size = 0;
int ret = 0;
int64_t src_ch_layout = audio_dec_ctx->channel_layout;
int64_t dst_ch_layout = AV_CH_LAYOUT_STEREO;
int dst_nb_channels = 0;
int dst_linesize = 0;
int src_nb_samples = 0;
int dst_nb_samples = 0;
int max_dst_nb_samples = 0;
uint8_t **dst_data = NULL;
int resampled_data_size = 0;
swr_ctx = swr_alloc();
if (!swr_ctx)
{
LOGD("swr_alloc error \n");
return -1;
}
src_ch_layout = (audio_dec_ctx->channels ==
av_get_channel_layout_nb_channels(audio_dec_ctx->channel_layout)) ?
audio_dec_ctx->channel_layout :
av_get_default_channel_layout(audio_dec_ctx->channels);
if (out_channels == 1)
{
dst_ch_layout = AV_CH_LAYOUT_MONO;
//LOGD("dst_ch_layout: AV_CH_LAYOUT_MONO\n");
}
else if (out_channels == 2)
{
dst_ch_layout = AV_CH_LAYOUT_STEREO;
//LOGD("dst_ch_layout: AV_CH_LAYOUT_STEREO\n");
}
else
{
dst_ch_layout = AV_CH_LAYOUT_SURROUND;
//LOGD("dst_ch_layout: AV_CH_LAYOUT_SURROUND\n");
}
if (src_ch_layout <= 0)
{
LOGD("src_ch_layout error \n");
return -1;
}
src_nb_samples = pAudioDecodeFrame->nb_samples;
if (src_nb_samples <= 0)
{
LOGD("src_nb_samples error \n");
return -1;
}
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", audio_dec_ctx->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", audio_dec_ctx->sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", out_sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", (AVSampleFormat)out_sample_fmt, 0);
if ((ret = swr_init(swr_ctx)) < 0) {
LOGD("Failed to initialize the resampling context\n");
return -1;
}
max_dst_nb_samples = dst_nb_samples = av_rescale_rnd(src_nb_samples,
out_sample_rate, audio_dec_ctx->sample_rate, AV_ROUND_UP);
if (max_dst_nb_samples <= 0)
{
LOGD("av_rescale_rnd error \n");
return -1;
}
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, (AVSampleFormat)out_sample_fmt, 0);
if (ret < 0)
{
LOGD("av_samples_alloc_array_and_samples error \n");
return -1;
}
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, audio_dec_ctx->sample_rate) +
src_nb_samples, out_sample_rate, audio_dec_ctx->sample_rate, AV_ROUND_UP);
if (dst_nb_samples <= 0)
{
LOGD("av_rescale_rnd error \n");
return -1;
}
if (dst_nb_samples > max_dst_nb_samples)
{
av_free(dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, (AVSampleFormat)out_sample_fmt, 1);
max_dst_nb_samples = dst_nb_samples;
}
if (swr_ctx)
{
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples,
(const uint8_t **)pAudioDecodeFrame->data, pAudioDecodeFrame->nb_samples);
if (ret < 0)
{
LOGD("swr_convert error \n");
return -1;
}
resampled_data_size = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, (AVSampleFormat)out_sample_fmt, 1);
if (resampled_data_size < 0)
{
LOGD("av_samples_get_buffer_size error \n");
return -1;
}
}
else
{
LOGD("swr_ctx null error \n");
return -1;
}
//LOGD("resampled_data_size:%d",resampled_data_size);
memcpy(audio_chunk, dst_data[0], resampled_data_size);
if (dst_data)
{
av_freep(&dst_data[0]);
}
av_freep(&dst_data);
dst_data = NULL;
if (swr_ctx)
{
swr_free(&swr_ctx);
}
return resampled_data_size;
}
int audioResampling(AVCodecContext *audio_dec_ctx, AVFrame *pAudioDecodeFrame,
int out_sample_fmt, int out_channels, int out_sample_rate, char *out_buf) {
__android_log_print(ANDROID_LOG_DEBUG, TAG, "stream12");
SwrContext *swr_ctx = NULL;
int data_size = 0;
int ret = 0;
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO; //初始化这样根据不同文件做调整
int64_t dst_ch_layout = AV_CH_LAYOUT_STEREO; //这里设定ok
int dst_nb_channels = 0;
int dst_linesize = 0;
int src_nb_samples = 0;
int dst_nb_samples = 0;
int max_dst_nb_samples = 0;
uint8_t **dst_data = NULL;
int resampled_data_size = 0;
//重新采样
if (swr_ctx) {
swr_free(&swr_ctx);
}
swr_ctx = swr_alloc();
__android_log_print(ANDROID_LOG_DEBUG, TAG, "stream12-1");
if (!swr_ctx) {
printf("swr_alloc error \n");
__android_log_print(ANDROID_LOG_DEBUG, TAG, "stream13");
return -1;
}
src_ch_layout = (audio_dec_ctx->channel_layout &&
audio_dec_ctx->channels ==
av_get_channel_layout_nb_channels(audio_dec_ctx->channel_layout)) ?
audio_dec_ctx->channel_layout :
av_get_default_channel_layout(audio_dec_ctx->channels);
if (out_channels == 1) {
dst_ch_layout = AV_CH_LAYOUT_MONO;
}
else if (out_channels == 2) {
dst_ch_layout = AV_CH_LAYOUT_STEREO;
}
else {
//可扩展
}
__android_log_print(ANDROID_LOG_DEBUG, TAG, "stream12-2");
if (src_ch_layout <= 0) {
printf("src_ch_layout error \n");
__android_log_print(ANDROID_LOG_DEBUG, TAG, "stream14");
return -1;
}
__android_log_print(ANDROID_LOG_DEBUG, TAG, "stream12-3");
src_nb_samples = pAudioDecodeFrame->nb_samples;
if (src_nb_samples <= 0) {
printf("src_nb_samples error \n");
__android_log_print(ANDROID_LOG_DEBUG, TAG, "stream15");
return -1;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", audio_dec_ctx->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", audio_dec_ctx->sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", out_sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", (AVSampleFormat) out_sample_fmt, 0);
swr_init(swr_ctx);
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, out_sample_rate, audio_dec_ctx->sample_rate,
AV_ROUND_UP);
if (max_dst_nb_samples <= 0) {
printf("av_rescale_rnd error \n");
return -1;
}
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, (AVSampleFormat) out_sample_fmt, 0);
if (ret < 0) {
printf("av_samples_alloc_array_and_samples error \n");
return -1;
}
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, audio_dec_ctx->sample_rate) +
src_nb_samples, out_sample_rate, audio_dec_ctx->sample_rate,
AV_ROUND_UP);
if (dst_nb_samples <= 0) {
return -1;
}
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, (AVSampleFormat) out_sample_fmt, 1);
max_dst_nb_samples = dst_nb_samples;
}
data_size = av_samples_get_buffer_size(NULL, audio_dec_ctx->channels,
pAudioDecodeFrame->nb_samples,
audio_dec_ctx->sample_fmt, 1);
if (data_size <= 0) {
return -1;
}
resampled_data_size = data_size;
if (swr_ctx) {
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples,
(const uint8_t **) pAudioDecodeFrame->data,
pAudioDecodeFrame->nb_samples);
if (ret <= 0) {
return -1;
}
resampled_data_size = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, (AVSampleFormat) out_sample_fmt, 1);
if (resampled_data_size <= 0) {
return -1;
}
}
else {
return -1;
}
//将值返回去
memcpy(out_buf, dst_data[0], resampled_data_size);
if (dst_data) {
av_freep(&dst_data[0]);
}
av_freep(&dst_data);
dst_data = NULL;
if (swr_ctx) {
swr_free(&swr_ctx);
}
return resampled_data_size;
}
int decode_frame_from_packet(VideoState *is, AVFrame decoded_frame)
{
int64_t src_ch_layout, dst_ch_layout;
int src_rate, dst_rate;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt, dst_sample_fmt;
int dst_bufsize;
int ret;
src_nb_samples = decoded_frame.nb_samples;
src_linesize = (int) decoded_frame.linesize;
src_data = decoded_frame.data;
if (decoded_frame.channel_layout == 0) {
decoded_frame.channel_layout = av_get_default_channel_layout(decoded_frame.channels);
}
src_rate = decoded_frame.sample_rate;
dst_rate = decoded_frame.sample_rate;
src_ch_layout = decoded_frame.channel_layout;
dst_ch_layout = decoded_frame.channel_layout;
src_sample_fmt = decoded_frame.format;
dst_sample_fmt = AV_SAMPLE_FMT_S16;
av_opt_set_int(is->sws_ctx_audio, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(is->sws_ctx_audio, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(is->sws_ctx_audio, "in_sample_rate", src_rate, 0);
av_opt_set_int(is->sws_ctx_audio, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(is->sws_ctx_audio, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_sample_fmt(is->sws_ctx_audio, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(is->sws_ctx_audio)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
return -1;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels, src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
return -1;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples = av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels, dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
return -1;
}
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(is->sws_ctx_audio, src_rate) + src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* convert to destination format */
ret = swr_convert(is->sws_ctx_audio, dst_data, dst_nb_samples, (const uint8_t **)decoded_frame.data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
return -1;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels, ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
return -1;
}
memcpy(is->audio_buf, dst_data[0], dst_bufsize);
if (src_data) {
av_freep(&src_data[0]);
}
av_freep(&src_data);
if (dst_data) {
av_freep(&dst_data[0]);
}
av_freep(&dst_data);
return dst_bufsize;
}
static double get_delay(struct af_resample *s)
{
int64_t base = s->in_rate * (int64_t)s->out_rate;
return swr_get_delay(s->avrctx, base) / (double)base;
}
QbPacket ConvertAudio::convert(const QbAudioPacket &packet,
const QbCaps &oCaps)
{
QbAudioCaps oAudioCaps(oCaps);
int64_t iSampleLayout = channelLayouts->value(packet.caps().layout(), 0);
AVSampleFormat iSampleFormat = sampleFormats->value(packet.caps().format(),
AV_SAMPLE_FMT_NONE);
int iSampleRate = packet.caps().rate();
int iNChannels = packet.caps().channels();
int iNSamples = packet.caps().samples();
int64_t oSampleLayout = channelLayouts->value(oAudioCaps.layout(),
AV_CH_LAYOUT_STEREO);
AVSampleFormat oSampleFormat = sampleFormats->value(oAudioCaps.format(),
AV_SAMPLE_FMT_FLT);
int oSampleRate = oAudioCaps.rate();
int oNChannels = oAudioCaps.channels();
this->m_resampleContext =
swr_alloc_set_opts(this->m_resampleContext,
oSampleLayout,
oSampleFormat,
oSampleRate,
iSampleLayout,
iSampleFormat,
iSampleRate,
0,
NULL);
if (!this->m_resampleContext)
return QbPacket();
if (!swr_is_initialized(this->m_resampleContext))
if (swr_init(this->m_resampleContext) < 0)
return QbPacket();
// Create input audio frame.
static AVFrame iFrame;
memset(&iFrame, 0, sizeof(AVFrame));
if (av_samples_fill_arrays(iFrame.data,
iFrame.linesize,
(const uint8_t *) packet.buffer().data(),
iNChannels,
iNSamples,
iSampleFormat,
1) < 0)
return QbPacket();
iFrame.channels = iNChannels;
iFrame.channel_layout = iSampleLayout;
iFrame.format = iSampleFormat;
iFrame.sample_rate = iSampleRate;
iFrame.nb_samples = iNSamples;
iFrame.pts = iFrame.pkt_pts = packet.pts();
// Create output audio packet.
int oNSamples = swr_get_delay(this->m_resampleContext, oSampleRate)
+ iFrame.nb_samples
* (int64_t) oSampleRate
/ iSampleRate
+ 3;
int oLineSize;
int oBufferSize = av_samples_get_buffer_size(&oLineSize,
oNChannels,
oNSamples,
oSampleFormat,
1);
QByteArray oBuffer(oBufferSize, Qt::Uninitialized);
int oNPlanes = av_sample_fmt_is_planar(oSampleFormat)? oNChannels: 1;
QVector<uint8_t *> oData(oNPlanes);
if (av_samples_fill_arrays(&oData.data()[0],
&oLineSize,
(const uint8_t *) oBuffer.data(),
oNChannels,
oNSamples,
oSampleFormat,
1) < 0)
return QbPacket();
int64_t oPts = swr_next_pts(this->m_resampleContext, iFrame.pts);
// convert to destination format
int outputSamples = swr_convert(this->m_resampleContext,
oData.data(),
oNSamples,
(const uint8_t **) iFrame.data,
iFrame.nb_samples);
if (outputSamples < 1)
return QbPacket();
oBufferSize = oBufferSize * outputSamples / oNSamples;
QbBufferPtr buffer(new char[oBufferSize]);
memcpy(buffer.data(), oBuffer.data(), oBufferSize);
QbAudioPacket oAudioPacket;
oAudioPacket.caps() = oAudioCaps;
oAudioPacket.caps().samples() = outputSamples;
oAudioPacket.buffer() = buffer;
oAudioPacket.bufferSize() = oBufferSize;
oAudioPacket.pts() = oPts;
oAudioPacket.timeBase() = QbFrac(1, oAudioCaps.rate());
oAudioPacket.index() = packet.index();
oAudioPacket.id() = packet.id();
return oAudioPacket.toPacket();
}
