static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame = avcodec_alloc_frame();
int got_packet;
av_init_packet(&pkt);
c = st->codec;
get_audio_frame(samples, audio_input_frame_size, c->channels);
frame->nb_samples = audio_input_frame_size;
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(uint8_t *)samples,
audio_input_frame_size *
av_get_bytes_per_sample(c->sample_fmt) *
c->channels, 1);
avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (!got_packet)
return;
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
if (av_interleaved_write_frame(oc, &pkt) != 0) {
fprintf(stderr, "Error while writing audio frame\n");
exit(1);
}
avcodec_free_frame(&frame);
}
/*
* 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;
}
/*
* Class: com_chris_yanlong_player
* Method: get_audio_frame
* Signature: (I)[B
*/
JNIEXPORT jbyteArray JNICALL Java_com_chris_yanlong_PlayerView_get_1audio_1frame
(JNIEnv *env, jobject obj, jint handle , jobject audio_pts , jobject audio_size ,jobject byteBuffer){
jbyte *directBuffer = (*env)->GetDirectBufferAddress(env, byteBuffer);
media_handle_union_t * media_handle = (media_handle_union_t *) handle;
jclass c;
jfieldID id;
c = (*env)->FindClass(env ,"java/lang/Double");
if (c == NULL)
{
__android_log_print(ANDROID_LOG_INFO,"chris_magic", "jclass is null");
// return -1;
exit(NULL_POINT);
}
id = (*env)->GetFieldID( env ,c, "value", "D");
if (id==NULL)
{
__android_log_print(ANDROID_LOG_INFO,"chris_magic", "id is null");
// return -1;
exit(NULL_POINT);
}
double audio_frame_pts = 0;
int audio_frame_size = 0;
get_audio_frame((int)handle ,&audio_frame_size ,&audio_frame_pts ,directBuffer);
(*env)->SetDoubleField(env ,audio_pts, id, audio_frame_pts);
jclass c1;
jfieldID id1;
c1 = (*env)->FindClass(env ,"java/lang/Integer");
if (c1 == NULL)
{
__android_log_print(ANDROID_LOG_INFO,"chris_magic", "jclass is null");
// return -1;
exit(NULL_POINT);
}
id1 = (*env)->GetFieldID( env ,c1, "value", "I");
if (id1==NULL)
{
__android_log_print(ANDROID_LOG_INFO,"chris_magic", "id is null");
// return -1;
exit(NULL_POINT);
}
(*env)->SetIntField(env ,audio_size, id1, audio_frame_size);
jbyteArray result=NULL;
// result= (*env)->NewByteArray(env ,media_handle->audio_size);
//
// (*env)->SetByteArrayRegion(env ,result ,0 ,media_handle->audio_size ,media_handle->audio_buf );
return result;
}
bool AV::addTestAudioFrame() {
if(!use_audio) {
printf("Cannot add audio stream, we're not using audio.\n");
return false;
}
AVCodecContext* c = ct.as->codec;
AVPacket packet = {0}; // data and size must be '0' (allocation is done for you :> )
AVFrame* frame = avcodec_alloc_frame();
int got_packet = 0;
av_init_packet(&packet);
get_audio_frame(ct.atest_samples, ct.atest_frame_size);
frame->nb_samples = ct.atest_frame_size;
avcodec_fill_audio_frame(
frame
,c->channels
,c->sample_fmt
,(uint8_t*)ct.atest_samples
,ct.atest_frame_size * av_get_bytes_per_sample(c->sample_fmt) * c->channels
,1
);
avcodec_encode_audio2(c, &packet, frame, &got_packet);
if(!got_packet) {
return false;
}
AVRational stream_time_base = ct.as->time_base; // stream time base
AVRational codec_time_base = ct.as->codec->time_base; // codec time base
packet.stream_index = ct.as->index;
packet.pts = av_rescale_q(packet.pts, codec_time_base, stream_time_base);
packet.dts = av_rescale_q(packet.dts, codec_time_base, stream_time_base);
packet.duration = av_rescale_q(packet.duration, codec_time_base, stream_time_base);
printf("Audio (test): stream: %d\n", packet.stream_index);
printf("Audio (test): ct.acounter: %d\n", ct.acounter);
printf("Audio (test): packet.duration: %d\n", packet.duration);
printf("Audio (test): stream.time_base, num=%d, den=%d\n", stream_time_base.num, stream_time_base.den);
printf("Audio (test): codec.time_base, num=%d, den=%d\n", codec_time_base.num, codec_time_base.den);
printf("Audio (test): coded_frame.pts: %lld\n", ct.as->codec->coded_frame->pts);
printf("Audio (test): packet.pts: %lld\n" ,packet.pts);
printf("-------------------\n");
if(av_interleaved_write_frame(ct.c, &packet) != 0) {
printf("Cannot write frame.\n");
return false;
}
return true;
}
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;
}
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 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);
}
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int process_audio_stream(AVFormatContext *oc, OutputStream *ost)
{
AVFrame *frame;
int got_output = 0;
int ret;
frame = get_audio_frame(ost);
got_output |= !!frame;
/* feed the data to lavr */
if (frame) {
ret = avresample_convert(ost->avr, NULL, 0, 0,
frame->extended_data, frame->linesize[0],
frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error feeding audio data to the resampler\n");
exit(1);
}
}
while ((frame && avresample_available(ost->avr) >= ost->frame->nb_samples) ||
(!frame && avresample_get_out_samples(ost->avr, 0))) {
/* 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);
/* the difference between the two avresample calls here is that the
* first one just reads the already converted data that is buffered in
* the lavr output buffer, while the second one also flushes the
* resampler */
if (frame) {
ret = avresample_read(ost->avr, ost->frame->extended_data,
ost->frame->nb_samples);
} else {
ret = avresample_convert(ost->avr, ost->frame->extended_data,
ost->frame->linesize[0], ost->frame->nb_samples,
NULL, 0, 0);
}
if (ret < 0) {
fprintf(stderr, "Error while resampling\n");
exit(1);
} else if (frame && ret != ost->frame->nb_samples) {
fprintf(stderr, "Too few samples returned from lavr\n");
exit(1);
}
ost->frame->nb_samples = ret;
ost->frame->pts = ost->next_pts;
ost->next_pts += ost->frame->nb_samples;
got_output |= encode_audio_frame(oc, ost, ret ? ost->frame : NULL);
}
return !got_output;
}