[MPlayer-dev-eng] [PATCH v2 4/8] Switch to FFmpeg AVChannelLayout
Alexander Strasser
eclipse7 at gmx.net
Sun Apr 7 17:57:35 EEST 2024
The old way to specify channel layouts is removed on lavu major bump to 59.
---
av_helpers.c | 19 ++++++++++---------
libaf/af_lavcac3enc.c | 4 ++--
libmpcodecs/ad_ffmpeg.c | 16 ++++++++--------
libmpcodecs/ae_lavc.c | 8 ++++----
libmpdemux/demux_lavf.c | 6 +++---
libmpdemux/muxer_lavf.c | 2 +-
6 files changed, 28 insertions(+), 27 deletions(-)
diff --git a/av_helpers.c b/av_helpers.c
index 2e5bc3262..9639692a8 100644
--- a/av_helpers.c
+++ b/av_helpers.c
@@ -116,38 +116,39 @@ int lavc_encode_audio(AVCodecContext *ctx, void *src, int src_len, void *dst, in
{
void *orig_src = src;
int bps = av_get_bytes_per_sample(ctx->sample_fmt);
- int planar = ctx->channels > 1 && av_sample_fmt_is_planar(ctx->sample_fmt);
+ int channels = ctx->ch_layout.nb_channels;
+ int planar = channels > 1 && av_sample_fmt_is_planar(ctx->sample_fmt);
int isac3 = ctx->codec->id == AV_CODEC_ID_AC3;
int n;
int got;
AVPacket pkt;
AVFrame *frame = av_frame_alloc();
- if ((ctx->channels == 6 || ctx->channels == 5) &&
+ if ((channels == 6 || channels == 5) &&
(isac3 || !strcmp(ctx->codec->name,"libfaac"))) {
reorder_channel_nch(src, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
isac3 ? AF_CHANNEL_LAYOUT_LAVC_DEFAULT : AF_CHANNEL_LAYOUT_AAC_DEFAULT,
- ctx->channels,
+ channels,
src_len / bps, bps);
}
- frame->nb_samples = src_len / ctx->channels / bps;
+ frame->nb_samples = src_len / channels / bps;
if (planar) {
// TODO: this is horribly inefficient.
int ch;
src = av_mallocz(src_len);
- for (ch = 0; ch < ctx->channels; ch++) {
+ for (ch = 0; ch < channels; ch++) {
uint8_t *tmps = (uint8_t *)orig_src + ch*bps;
- uint8_t *tmpd = (uint8_t *)src + ch*src_len/ctx->channels;
+ uint8_t *tmpd = (uint8_t *)src + ch*src_len/channels;
int s;
for (s = 0; s < frame->nb_samples; s++) {
memcpy(tmpd, tmps, bps);
- tmps += ctx->channels * bps;
+ tmps += channels * bps;
tmpd += bps;
}
}
}
frame->format = ctx->sample_fmt;
- frame->channels = ctx->channels;
- n = avcodec_fill_audio_frame(frame, ctx->channels, ctx->sample_fmt, src, src_len, 1);
+ frame->ch_layout.nb_channels = channels;
+ n = avcodec_fill_audio_frame(frame, channels, ctx->sample_fmt, src, src_len, 1);
if (n < 0) return 0;
n = avcodec_send_frame(ctx, frame);
av_init_packet(&pkt);
diff --git a/libaf/af_lavcac3enc.c b/libaf/af_lavcac3enc.c
index 030aebd87..9c8b6eec8 100644
--- a/libaf/af_lavcac3enc.c
+++ b/libaf/af_lavcac3enc.c
@@ -96,7 +96,7 @@ static int control(struct af_instance_s *af, int cmd, void *arg)
bit_rate = s->bit_rate ? s->bit_rate : default_bit_rate[af->data->nch];
- if (s->lavc_actx->channels != af->data->nch ||
+ if (s->lavc_actx->ch_layout.nb_channels != af->data->nch ||
s->lavc_actx->sample_rate != af->data->rate ||
s->lavc_actx->bit_rate != bit_rate) {
@@ -104,7 +104,7 @@ static int control(struct af_instance_s *af, int cmd, void *arg)
avcodec_close(s->lavc_actx);
// Put sample parameters
- s->lavc_actx->channels = af->data->nch;
+ s->lavc_actx->ch_layout.nb_channels = af->data->nch;
s->lavc_actx->sample_rate = af->data->rate;
s->lavc_actx->sample_fmt = AV_SAMPLE_FMT_S16P;
s->lavc_actx->bit_rate = bit_rate;
diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
index daa9c43ea..866379946 100644
--- a/libmpcodecs/ad_ffmpeg.c
+++ b/libmpcodecs/ad_ffmpeg.c
@@ -79,10 +79,10 @@ static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context
} else if (sh_audio->wf->nSamplesPerSec && !c->srate_changed)
samplerate=sh_audio->wf->nSamplesPerSec;
}
- if (lavc_context->channels != sh_audio->channels ||
+ if (lavc_context->ch_layout.nb_channels != sh_audio->channels ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
- sh_audio->channels=lavc_context->channels;
+ sh_audio->channels=lavc_context->ch_layout.nb_channels;
sh_audio->samplerate=samplerate;
sh_audio->sample_format = sample_format;
sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
@@ -121,14 +121,14 @@ static int init(sh_audio_t *sh_audio)
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
if(sh_audio->wf){
- lavc_context->channels = sh_audio->wf->nChannels;
+ lavc_context->ch_layout.nb_channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
+ } else {
+ lavc_context->ch_layout.nb_channels = sh_audio->channels;
}
- lavc_context->channel_layout = sh_audio->channel_layout;
- lavc_context->request_channel_layout = av_get_default_channel_layout(audio_output_channels);
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
@@ -274,7 +274,7 @@ static av_always_inline void copy_samples_planar(size_t bps,
static int copy_samples(AVCodecContext *avc, AVFrame *frame,
unsigned char *buf, int max_size)
{
- int channels = avc->channels;
+ int channels = avc->ch_layout.nb_channels;
int sample_size = av_get_bytes_per_sample(avc->sample_fmt);
int size = channels * sample_size * frame->nb_samples;
@@ -371,12 +371,12 @@ static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m
if (len2 < 0)
return len2;
if(len2>0){
- if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
+ if (((AVCodecContext *)sh_audio->context)->ch_layout.nb_channels >= 5) {
int samplesize = av_get_bytes_per_sample(((AVCodecContext *)
sh_audio->context)->sample_fmt);
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
- ((AVCodecContext *)sh_audio->context)->channels,
+ ((AVCodecContext *)sh_audio->context)->ch_layout.nb_channels,
len2 / samplesize, samplesize);
}
//len=len2;break;
diff --git a/libmpcodecs/ae_lavc.c b/libmpcodecs/ae_lavc.c
index bf147b32f..39a0d28cf 100644
--- a/libmpcodecs/ae_lavc.c
+++ b/libmpcodecs/ae_lavc.c
@@ -50,7 +50,7 @@ static int bind_lavc(audio_encoder_t *encoder, muxer_stream_t *mux_a)
{
mux_a->wf = malloc(sizeof(WAVEFORMATEX)+lavc_actx->extradata_size+256);
mux_a->wf->wFormatTag = lavc_param_atag;
- mux_a->wf->nChannels = lavc_actx->channels;
+ mux_a->wf->nChannels = lavc_actx->ch_layout.nb_channels;
mux_a->wf->nSamplesPerSec = lavc_actx->sample_rate;
mux_a->wf->nAvgBytesPerSec = (lavc_actx->bit_rate / 8);
mux_a->avg_rate= lavc_actx->bit_rate;
@@ -192,7 +192,7 @@ int mpae_init_lavc(audio_encoder_t *encoder)
mp_msg(MSGT_MENCODER,MSGL_ERR, "Audio encoder requires unknown or unsupported input format\n");
return 0;
}
- lavc_actx->channels = encoder->params.channels;
+ lavc_actx->ch_layout.nb_channels = encoder->params.channels;
lavc_actx->sample_rate = encoder->params.sample_rate;
lavc_actx->time_base.num = 1;
lavc_actx->time_base.den = encoder->params.sample_rate;
@@ -219,7 +219,7 @@ int mpae_init_lavc(audio_encoder_t *encoder)
*/
if(lavc_param_atag == 0x11) {
int blkalign = 2048;
- int framesize = (blkalign - 4 * lavc_actx->channels) * 8 / (4 * lavc_actx->channels) + 1;
+ int framesize = (blkalign - 4 * lavc_actx->ch_layout.nb_channels) * 8 / (4 * lavc_actx->ch_layout.nb_channels) + 1;
lavc_actx->bit_rate = lavc_actx->sample_rate*8*blkalign/framesize;
}
if((lavc_param_audio_global_header&1)
@@ -238,7 +238,7 @@ int mpae_init_lavc(audio_encoder_t *encoder)
if(lavc_param_atag == 0x11) {
lavc_actx->block_align = 2048;
- lavc_actx->frame_size = (lavc_actx->block_align - 4 * lavc_actx->channels) * 8 / (4 * lavc_actx->channels) + 1;
+ lavc_actx->frame_size = (lavc_actx->block_align - 4 * lavc_actx->ch_layout.nb_channels) * 8 / (4 * lavc_actx->ch_layout.nb_channels) + 1;
}
encoder->decode_buffer_size = lavc_actx->frame_size *
diff --git a/libmpdemux/demux_lavf.c b/libmpdemux/demux_lavf.c
index caa254ebc..06a9e15cc 100644
--- a/libmpdemux/demux_lavf.c
+++ b/libmpdemux/demux_lavf.c
@@ -309,8 +309,8 @@ static void handle_stream(demuxer_t *demuxer, AVFormatContext *avfc, int i) {
wf= calloc(sizeof(*wf) + codec->extradata_size, 1);
codec->codec_tag = mp_codec_id2tag(codec->codec_id, codec->codec_tag, 1);
wf->wFormatTag= codec->codec_tag;
- wf->nChannels= codec->channels;
- sh_audio->channel_layout = codec->channel_layout;
+ wf->nChannels= codec->ch_layout.nb_channels;
+ sh_audio->channel_layout = codec->ch_layout.u.mask;
wf->nSamplesPerSec= codec->sample_rate;
wf->nAvgBytesPerSec= codec->bit_rate/8;
wf->nBlockAlign= codec->block_align ? codec->block_align : 1;
@@ -332,7 +332,7 @@ static void handle_stream(demuxer_t *demuxer, AVFormatContext *avfc, int i) {
sh_audio->audio.dwRate /= g;
// printf("sca:%d rat:%d fs:%d sr:%d ba:%d\n", sh_audio->audio.dwScale, sh_audio->audio.dwRate, codec->frame_size, codec->sample_rate, codec->block_align);
sh_audio->format= codec->codec_tag;
- sh_audio->channels= codec->channels;
+ sh_audio->channels= codec->ch_layout.nb_channels;
sh_audio->samplerate= codec->sample_rate;
sh_audio->i_bps= codec->bit_rate/8;
switch (codec->codec_id) {
diff --git a/libmpdemux/muxer_lavf.c b/libmpdemux/muxer_lavf.c
index 11ca05a1b..19fa9e956 100644
--- a/libmpdemux/muxer_lavf.c
+++ b/libmpdemux/muxer_lavf.c
@@ -192,7 +192,7 @@ static void fix_parameters(muxer_stream_t *stream)
mp_msg(MSGT_MUXER, MSGL_INFO, "AUDIO CODEC ID: %x, TAG: %x\n", ctx->codec_id, (uint32_t) ctx->codec_tag);
ctx->sample_rate = stream->wf->nSamplesPerSec;
// mp_msg(MSGT_MUXER, MSGL_INFO, "stream->h.dwSampleSize: %d\n", stream->h.dwSampleSize);
- ctx->channels = stream->wf->nChannels;
+ ctx->ch_layout.nb_channels = stream->wf->nChannels;
if(stream->h.dwRate && (stream->h.dwScale * (int64_t)ctx->sample_rate) % stream->h.dwRate == 0)
ctx->frame_size= (stream->h.dwScale * (int64_t)ctx->sample_rate) / stream->h.dwRate;
mp_msg(MSGT_MUXER, MSGL_V, "MUXER_LAVF(audio stream) frame_size: %d, scale: %u, sps: %u, rate: %u, ctx->block_align = stream->wf->nBlockAlign; %d=%d stream->wf->nAvgBytesPerSec:%d\n",
--
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