[FFmpeg-devel] [PATCH 167/281] wav: convert to new channel layout API
James Almer
jamrial at gmail.com
Thu Jan 13 04:00:14 EET 2022
From: Vittorio Giovara <vittorio.giovara at gmail.com>
Signed-off-by: Vittorio Giovara <vittorio.giovara at gmail.com>
Signed-off-by: Anton Khirnov <anton at khirnov.net>
Signed-off-by: James Almer <jamrial at gmail.com>
---
libavformat/wavdec.c | 32 +++++++++++++++++---------------
libavformat/wavenc.c | 14 +++++++-------
2 files changed, 24 insertions(+), 22 deletions(-)
diff --git a/libavformat/wavdec.c b/libavformat/wavdec.c
index 01090b6947..74f7a4d9aa 100644
--- a/libavformat/wavdec.c
+++ b/libavformat/wavdec.c
@@ -221,9 +221,11 @@ static int wav_parse_xma2_tag(AVFormatContext *s, int64_t size, AVStream *st)
channels += avio_r8(pb);
avio_skip(pb, 3);
}
- st->codecpar->channels = channels;
+ av_channel_layout_uninit(&st->codecpar->ch_layout);
+ st->codecpar->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
+ st->codecpar->ch_layout.nb_channels = channels;
- if (st->codecpar->channels <= 0 || st->codecpar->sample_rate <= 0)
+ if (st->codecpar->ch_layout.nb_channels <= 0 || st->codecpar->sample_rate <= 0)
return AVERROR_INVALIDDATA;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
@@ -611,15 +613,15 @@ break_loop:
if ( st->codecpar->bit_rate > 0 && data_size > 0
&& st->codecpar->sample_rate > 0
- && sample_count > 0 && st->codecpar->channels > 1
- && sample_count % st->codecpar->channels == 0) {
- if (fabs(8.0 * data_size * st->codecpar->channels * st->codecpar->sample_rate /
+ && sample_count > 0 && st->codecpar->ch_layout.nb_channels > 1
+ && sample_count % st->codecpar->ch_layout.nb_channels == 0) {
+ if (fabs(8.0 * data_size * st->codecpar->ch_layout.nb_channels * st->codecpar->sample_rate /
sample_count /st->codecpar->bit_rate - 1.0) < 0.3)
- sample_count /= st->codecpar->channels;
+ sample_count /= st->codecpar->ch_layout.nb_channels;
}
- if ( data_size > 0 && sample_count && st->codecpar->channels
- && (data_size << 3) / sample_count / st->codecpar->channels > st->codecpar->bits_per_coded_sample + 1) {
+ if (data_size > 0 && sample_count && st->codecpar->ch_layout.nb_channels &&
+ (data_size << 3) / sample_count / st->codecpar->ch_layout.nb_channels > st->codecpar->bits_per_coded_sample + 1) {
av_log(s, AV_LOG_WARNING, "ignoring wrong sample_count %"PRId64"\n", sample_count);
sample_count = 0;
}
@@ -632,34 +634,34 @@ break_loop:
}
if (!sample_count || av_get_exact_bits_per_sample(st->codecpar->codec_id) > 0)
- if ( st->codecpar->channels
+ if ( st->codecpar->ch_layout.nb_channels
&& data_size
&& av_get_bits_per_sample(st->codecpar->codec_id)
&& wav->data_end <= avio_size(pb))
sample_count = (data_size << 3)
/
- (st->codecpar->channels * (uint64_t)av_get_bits_per_sample(st->codecpar->codec_id));
+ (st->codecpar->ch_layout.nb_channels * (uint64_t)av_get_bits_per_sample(st->codecpar->codec_id));
if (sample_count)
st->duration = sample_count;
if (st->codecpar->codec_id == AV_CODEC_ID_PCM_S32LE &&
- st->codecpar->block_align == st->codecpar->channels * 4 &&
+ st->codecpar->block_align == st->codecpar->ch_layout.nb_channels * 4 &&
st->codecpar->bits_per_coded_sample == 32 &&
st->codecpar->extradata_size == 2 &&
AV_RL16(st->codecpar->extradata) == 1) {
st->codecpar->codec_id = AV_CODEC_ID_PCM_F16LE;
st->codecpar->bits_per_coded_sample = 16;
} else if (st->codecpar->codec_id == AV_CODEC_ID_PCM_S24LE &&
- st->codecpar->block_align == st->codecpar->channels * 4 &&
+ st->codecpar->block_align == st->codecpar->ch_layout.nb_channels * 4 &&
st->codecpar->bits_per_coded_sample == 24) {
st->codecpar->codec_id = AV_CODEC_ID_PCM_F24LE;
} else if (st->codecpar->codec_id == AV_CODEC_ID_XMA1 ||
st->codecpar->codec_id == AV_CODEC_ID_XMA2) {
st->codecpar->block_align = 2048;
- } else if (st->codecpar->codec_id == AV_CODEC_ID_ADPCM_MS && st->codecpar->channels > 2 &&
- st->codecpar->block_align < INT_MAX / st->codecpar->channels) {
- st->codecpar->block_align *= st->codecpar->channels;
+ } else if (st->codecpar->codec_id == AV_CODEC_ID_ADPCM_MS && st->codecpar->ch_layout.nb_channels > 2 &&
+ st->codecpar->block_align < INT_MAX / st->codecpar->ch_layout.nb_channels) {
+ st->codecpar->block_align *= st->codecpar->ch_layout.nb_channels;
}
ff_metadata_conv_ctx(s, NULL, wav_metadata_conv);
diff --git a/libavformat/wavenc.c b/libavformat/wavenc.c
index 2317700be1..5629bd487d 100644
--- a/libavformat/wavenc.c
+++ b/libavformat/wavenc.c
@@ -170,12 +170,12 @@ static av_cold int peak_init_writer(AVFormatContext *s)
"Writing 16 bit peak for 8 bit audio does not make sense\n");
return AVERROR(EINVAL);
}
- if (par->channels > INT_MAX / (wav->peak_bps * wav->peak_ppv))
+ if (par->ch_layout.nb_channels > INT_MAX / (wav->peak_bps * wav->peak_ppv))
return AVERROR(ERANGE);
- wav->size_increment = par->channels * wav->peak_bps * wav->peak_ppv;
+ wav->size_increment = par->ch_layout.nb_channels * wav->peak_bps * wav->peak_ppv;
- wav->peak_maxpos = av_calloc(par->channels, sizeof(*wav->peak_maxpos));
- wav->peak_maxneg = av_calloc(par->channels, sizeof(*wav->peak_maxneg));
+ wav->peak_maxpos = av_calloc(par->ch_layout.nb_channels, sizeof(*wav->peak_maxpos));
+ wav->peak_maxneg = av_calloc(par->ch_layout.nb_channels, sizeof(*wav->peak_maxneg));
if (!wav->peak_maxpos || !wav->peak_maxneg)
goto nomem;
@@ -205,7 +205,7 @@ static int peak_write_frame(AVFormatContext *s)
}
wav->peak_output = tmp;
- for (c = 0; c < par->channels; c++) {
+ for (c = 0; c < par->ch_layout.nb_channels; c++) {
wav->peak_maxneg[c] = -wav->peak_maxneg[c];
if (wav->peak_bps == 2 && wav->peak_format == PEAK_FORMAT_UINT8) {
@@ -277,7 +277,7 @@ static int peak_write_chunk(AVFormatContext *s)
avio_wl32(pb, wav->peak_format); /* 8 or 16 bit */
avio_wl32(pb, wav->peak_ppv); /* positive and negative */
avio_wl32(pb, wav->peak_block_size); /* frames per value */
- avio_wl32(pb, par->channels); /* number of channels */
+ avio_wl32(pb, par->ch_layout.nb_channels); /* number of channels */
avio_wl32(pb, wav->peak_num_frames); /* number of peak frames */
avio_wl32(pb, -1); /* audio sample frame position (not implemented) */
avio_wl32(pb, 128); /* equal to size of header */
@@ -384,7 +384,7 @@ static int wav_write_packet(AVFormatContext *s, AVPacket *pkt)
wav->peak_maxpos[c] = FFMAX(wav->peak_maxpos[c], (int16_t)AV_RL16(pkt->data + i));
wav->peak_maxneg[c] = FFMIN(wav->peak_maxneg[c], (int16_t)AV_RL16(pkt->data + i));
}
- if (++c == s->streams[0]->codecpar->channels) {
+ if (++c == s->streams[0]->codecpar->ch_layout.nb_channels) {
c = 0;
if (++wav->peak_block_pos == wav->peak_block_size) {
int ret = peak_write_frame(s);
--
2.34.1
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