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authorTuomas Virtanen <katajakasa@gmail.com>2018-10-07 21:00:45 +0300
committerTuomas Virtanen <katajakasa@gmail.com>2018-10-07 21:00:45 +0300
commit76bc8c49d46be38f6d4a65a810f8ef64923daef9 (patch)
tree4b24e74fcb5a65f24e59dcf97d9afd8200109c2b /src/internal/audio/kitaudio.c
parent25747441205f7973ea8815f1014372378ff34858 (diff)
Some initial work on supporting new ffmpeg decoder API
Diffstat (limited to 'src/internal/audio/kitaudio.c')
-rw-r--r--src/internal/audio/kitaudio.c82
1 files changed, 78 insertions, 4 deletions
diff --git a/src/internal/audio/kitaudio.c b/src/internal/audio/kitaudio.c
index 3a534bb..4bc284d 100644
--- a/src/internal/audio/kitaudio.c
+++ b/src/internal/audio/kitaudio.c
@@ -104,9 +104,13 @@ static void free_out_audio_packet_cb(void *packet) {
free(p);
}
-static void dec_decode_audio_cb(Kit_Decoder *dec, AVPacket *in_packet) {
+#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(57, 48, 101)
+static int dec_decode_audio_cb(Kit_Decoder *dec, AVPacket *in_packet) {
assert(dec != NULL);
- assert(in_packet != NULL);
+
+ if(in_packet == NULL) {
+ return 0;
+ }
Kit_AudioDecoder *audio_dec = dec->userdata;
int frame_finished;
@@ -117,13 +121,13 @@ static void dec_decode_audio_cb(Kit_Decoder *dec, AVPacket *in_packet) {
int dst_bufsize;
double pts;
unsigned char **dst_data;
- Kit_AudioPacket *out_packet;
+ Kit_AudioPacket *out_packet = NULL;
// Decode as long as there is data
while(in_packet->size > 0) {
len = avcodec_decode_audio4(dec->codec_ctx, audio_dec->scratch_frame, &frame_finished, in_packet);
if(len < 0) {
- return;
+ return 0;
}
if(frame_finished) {
@@ -175,7 +179,77 @@ static void dec_decode_audio_cb(Kit_Decoder *dec, AVPacket *in_packet) {
in_packet->size -= len;
in_packet->data += len;
}
+ return 0;
+}
+#else
+static int dec_decode_audio_cb(Kit_Decoder *dec, AVPacket *in_packet) {
+ assert(dec != NULL);
+ assert(in_packet != NULL);
+
+ Kit_AudioDecoder *audio_dec = dec->userdata;
+ int ret;
+ int len;
+ int dst_linesize;
+ int dst_nb_samples;
+ int dst_bufsize;
+ double pts;
+ unsigned char **dst_data;
+ Kit_AudioPacket *out_packet = NULL;
+
+ // Write packet to the decoder for handling.
+ ret = avcodec_send_packet(dec->codec_ctx, in_packet);
+ if(ret < 0) {
+ return 1;
+ }
+
+ // Pull decoded frames out when ready and if we have room in decoder output buffer
+ while(!ret && Kit_CanWriteDecoderOutput(dec)) {
+ ret = avcodec_receive_frame(dec->codec_ctx, audio_dec->scratch_frame);
+ if(!ret) {
+ dst_nb_samples = av_rescale_rnd(
+ audio_dec->scratch_frame->nb_samples,
+ dec->output.samplerate, // Target samplerate
+ dec->codec_ctx->sample_rate, // Source samplerate
+ AV_ROUND_UP);
+
+ av_samples_alloc_array_and_samples(
+ &dst_data,
+ &dst_linesize,
+ dec->output.channels,
+ dst_nb_samples,
+ _FindAVSampleFormat(dec->output.format),
+ 0);
+
+ len = swr_convert(
+ audio_dec->swr,
+ dst_data,
+ audio_dec->scratch_frame->nb_samples,
+ (const unsigned char **)audio_dec->scratch_frame->extended_data,
+ audio_dec->scratch_frame->nb_samples);
+
+ dst_bufsize = av_samples_get_buffer_size(
+ &dst_linesize,
+ dec->output.channels,
+ len,
+ _FindAVSampleFormat(dec->output.format), 1);
+
+ // Get presentation timestamp
+ pts = audio_dec->scratch_frame->best_effort_timestamp;
+ pts *= av_q2d(dec->format_ctx->streams[dec->stream_index]->time_base);
+
+ // Lock, write to audio buffer, unlock
+ out_packet = _CreateAudioPacket(
+ (char*)dst_data[0], (size_t)dst_bufsize, pts);
+ Kit_WriteDecoderOutput(dec, out_packet);
+
+ // Free temps
+ av_freep(&dst_data[0]);
+ av_freep(&dst_data);
+ }
+ }
+ return 0;
}
+#endif
static void dec_close_audio_cb(Kit_Decoder *dec) {
if(dec == NULL) return;