* [FFmpeg-devel] [PATCH] avformat/cafenc: derive Opus frame size from the relevant stream parameters
@ 2022-09-21 12:21 James Almer
2022-09-22 23:14 ` [FFmpeg-devel] [PATCH v2] " James Almer
0 siblings, 1 reply; 3+ messages in thread
From: James Almer @ 2022-09-21 12:21 UTC (permalink / raw)
To: ffmpeg-devel
Use the stream duration as last resort, as an off-by-one result of the
"st->duration / (caf->packets - 1)" calculation can break playback on some
devices.
Fixes ticket #9930.
Signed-off-by: James Almer <jamrial@gmail.com>
---
libavformat/cafenc.c | 12 +++++++++---
1 file changed, 9 insertions(+), 3 deletions(-)
diff --git a/libavformat/cafenc.c b/libavformat/cafenc.c
index fedb430b17..c203747b15 100644
--- a/libavformat/cafenc.c
+++ b/libavformat/cafenc.c
@@ -53,7 +53,11 @@ static uint32_t codec_flags(enum AVCodecID codec_id) {
}
}
-static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int block_align) {
+static uint32_t samples_per_packet(AVCodecParameters *par) {
+ enum AVCodecID codec_id = par->codec_id;
+ int channels = par->ch_layout.nb_channels, block_align = par->block_align;
+ int frame_size = par->frame_size, sample_rate = par->sample_rate;
+
switch (codec_id) {
case AV_CODEC_ID_PCM_S8:
case AV_CODEC_ID_PCM_S16LE:
@@ -83,6 +87,8 @@ static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int bl
return 320;
case AV_CODEC_ID_MP1:
return 384;
+ case AV_CODEC_ID_OPUS:
+ return frame_size * 48000 / sample_rate;
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
return 1152;
@@ -139,7 +145,7 @@ static int caf_write_header(AVFormatContext *s)
}
if (par->codec_id != AV_CODEC_ID_MP3 || frame_size != 576)
- frame_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
+ frame_size = samples_per_packet(par);
ffio_wfourcc(pb, "caff"); //< mFileType
avio_wb16(pb, 1); //< mFileVersion
@@ -248,7 +254,7 @@ static int caf_write_trailer(AVFormatContext *s)
avio_seek(pb, caf->data, SEEK_SET);
avio_wb64(pb, file_size - caf->data - 8);
if (!par->block_align) {
- int packet_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
+ int packet_size = samples_per_packet(par);
if (!packet_size) {
packet_size = st->duration / (caf->packets - 1);
avio_seek(pb, FRAME_SIZE_OFFSET, SEEK_SET);
--
2.37.3
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^ permalink raw reply [flat|nested] 3+ messages in thread
* [FFmpeg-devel] [PATCH v2] avformat/cafenc: derive Opus frame size from the relevant stream parameters
2022-09-21 12:21 [FFmpeg-devel] [PATCH] avformat/cafenc: derive Opus frame size from the relevant stream parameters James Almer
@ 2022-09-22 23:14 ` James Almer
2022-09-23 21:20 ` James Almer
0 siblings, 1 reply; 3+ messages in thread
From: James Almer @ 2022-09-22 23:14 UTC (permalink / raw)
To: ffmpeg-devel
Use the stream duration as last resort, as an off-by-one result of the
"st->duration / (caf->packets - 1)" calculation can break playback on some
devices.
Also, don't write the sample_rate value propagated by encoders like libopus.
The sample rate of the audio fed to it is irrelevant for the container after
being encoded.
Fixes ticket #9930.
Signed-off-by: James Almer <jamrial@gmail.com>
---
libavformat/cafenc.c | 19 ++++++++++++++-----
1 file changed, 14 insertions(+), 5 deletions(-)
diff --git a/libavformat/cafenc.c b/libavformat/cafenc.c
index fedb430b17..b90811d46f 100644
--- a/libavformat/cafenc.c
+++ b/libavformat/cafenc.c
@@ -53,7 +53,11 @@ static uint32_t codec_flags(enum AVCodecID codec_id) {
}
}
-static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int block_align) {
+static uint32_t samples_per_packet(const AVCodecParameters *par) {
+ enum AVCodecID codec_id = par->codec_id;
+ int channels = par->ch_layout.nb_channels, block_align = par->block_align;
+ int frame_size = par->frame_size, sample_rate = par->sample_rate;
+
switch (codec_id) {
case AV_CODEC_ID_PCM_S8:
case AV_CODEC_ID_PCM_S16LE:
@@ -83,6 +87,8 @@ static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int bl
return 320;
case AV_CODEC_ID_MP1:
return 384;
+ case AV_CODEC_ID_OPUS:
+ return frame_size * 48000 / sample_rate;
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
return 1152;
@@ -110,7 +116,7 @@ static int caf_write_header(AVFormatContext *s)
AVDictionaryEntry *t = NULL;
unsigned int codec_tag = ff_codec_get_tag(ff_codec_caf_tags, par->codec_id);
int64_t chunk_size = 0;
- int frame_size = par->frame_size;
+ int frame_size = par->frame_size, sample_rate = par->sample_rate;
if (s->nb_streams != 1) {
av_log(s, AV_LOG_ERROR, "CAF files have exactly one stream\n");
@@ -139,7 +145,10 @@ static int caf_write_header(AVFormatContext *s)
}
if (par->codec_id != AV_CODEC_ID_MP3 || frame_size != 576)
- frame_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
+ frame_size = samples_per_packet(par);
+
+ if (par->codec_id == AV_CODEC_ID_OPUS)
+ sample_rate = 48000;
ffio_wfourcc(pb, "caff"); //< mFileType
avio_wb16(pb, 1); //< mFileVersion
@@ -147,7 +156,7 @@ static int caf_write_header(AVFormatContext *s)
ffio_wfourcc(pb, "desc"); //< Audio Description chunk
avio_wb64(pb, 32); //< mChunkSize
- avio_wb64(pb, av_double2int(par->sample_rate)); //< mSampleRate
+ avio_wb64(pb, av_double2int(sample_rate)); //< mSampleRate
avio_wl32(pb, codec_tag); //< mFormatID
avio_wb32(pb, codec_flags(par->codec_id)); //< mFormatFlags
avio_wb32(pb, par->block_align); //< mBytesPerPacket
@@ -248,7 +257,7 @@ static int caf_write_trailer(AVFormatContext *s)
avio_seek(pb, caf->data, SEEK_SET);
avio_wb64(pb, file_size - caf->data - 8);
if (!par->block_align) {
- int packet_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
+ int packet_size = samples_per_packet(par);
if (!packet_size) {
packet_size = st->duration / (caf->packets - 1);
avio_seek(pb, FRAME_SIZE_OFFSET, SEEK_SET);
--
2.37.3
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^ permalink raw reply [flat|nested] 3+ messages in thread
* Re: [FFmpeg-devel] [PATCH v2] avformat/cafenc: derive Opus frame size from the relevant stream parameters
2022-09-22 23:14 ` [FFmpeg-devel] [PATCH v2] " James Almer
@ 2022-09-23 21:20 ` James Almer
0 siblings, 0 replies; 3+ messages in thread
From: James Almer @ 2022-09-23 21:20 UTC (permalink / raw)
To: ffmpeg-devel
On 9/22/2022 8:14 PM, James Almer wrote:
> Use the stream duration as last resort, as an off-by-one result of the
> "st->duration / (caf->packets - 1)" calculation can break playback on some
> devices.
> Also, don't write the sample_rate value propagated by encoders like libopus.
> The sample rate of the audio fed to it is irrelevant for the container after
> being encoded.
>
> Fixes ticket #9930.
>
> Signed-off-by: James Almer <jamrial@gmail.com>
> ---
> libavformat/cafenc.c | 19 ++++++++++++++-----
> 1 file changed, 14 insertions(+), 5 deletions(-)
>
> diff --git a/libavformat/cafenc.c b/libavformat/cafenc.c
> index fedb430b17..b90811d46f 100644
> --- a/libavformat/cafenc.c
> +++ b/libavformat/cafenc.c
> @@ -53,7 +53,11 @@ static uint32_t codec_flags(enum AVCodecID codec_id) {
> }
> }
>
> -static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int block_align) {
> +static uint32_t samples_per_packet(const AVCodecParameters *par) {
> + enum AVCodecID codec_id = par->codec_id;
> + int channels = par->ch_layout.nb_channels, block_align = par->block_align;
> + int frame_size = par->frame_size, sample_rate = par->sample_rate;
> +
> switch (codec_id) {
> case AV_CODEC_ID_PCM_S8:
> case AV_CODEC_ID_PCM_S16LE:
> @@ -83,6 +87,8 @@ static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int bl
> return 320;
> case AV_CODEC_ID_MP1:
> return 384;
> + case AV_CODEC_ID_OPUS:
> + return frame_size * 48000 / sample_rate;
> case AV_CODEC_ID_MP2:
> case AV_CODEC_ID_MP3:
> return 1152;
> @@ -110,7 +116,7 @@ static int caf_write_header(AVFormatContext *s)
> AVDictionaryEntry *t = NULL;
> unsigned int codec_tag = ff_codec_get_tag(ff_codec_caf_tags, par->codec_id);
> int64_t chunk_size = 0;
> - int frame_size = par->frame_size;
> + int frame_size = par->frame_size, sample_rate = par->sample_rate;
>
> if (s->nb_streams != 1) {
> av_log(s, AV_LOG_ERROR, "CAF files have exactly one stream\n");
> @@ -139,7 +145,10 @@ static int caf_write_header(AVFormatContext *s)
> }
>
> if (par->codec_id != AV_CODEC_ID_MP3 || frame_size != 576)
> - frame_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
> + frame_size = samples_per_packet(par);
> +
> + if (par->codec_id == AV_CODEC_ID_OPUS)
> + sample_rate = 48000;
>
> ffio_wfourcc(pb, "caff"); //< mFileType
> avio_wb16(pb, 1); //< mFileVersion
> @@ -147,7 +156,7 @@ static int caf_write_header(AVFormatContext *s)
>
> ffio_wfourcc(pb, "desc"); //< Audio Description chunk
> avio_wb64(pb, 32); //< mChunkSize
> - avio_wb64(pb, av_double2int(par->sample_rate)); //< mSampleRate
> + avio_wb64(pb, av_double2int(sample_rate)); //< mSampleRate
> avio_wl32(pb, codec_tag); //< mFormatID
> avio_wb32(pb, codec_flags(par->codec_id)); //< mFormatFlags
> avio_wb32(pb, par->block_align); //< mBytesPerPacket
> @@ -248,7 +257,7 @@ static int caf_write_trailer(AVFormatContext *s)
> avio_seek(pb, caf->data, SEEK_SET);
> avio_wb64(pb, file_size - caf->data - 8);
> if (!par->block_align) {
> - int packet_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
> + int packet_size = samples_per_packet(par);
> if (!packet_size) {
> packet_size = st->duration / (caf->packets - 1);
> avio_seek(pb, FRAME_SIZE_OFFSET, SEEK_SET);
Will apply.
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^ permalink raw reply [flat|nested] 3+ messages in thread
end of thread, other threads:[~2022-09-23 21:20 UTC | newest]
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2022-09-21 12:21 [FFmpeg-devel] [PATCH] avformat/cafenc: derive Opus frame size from the relevant stream parameters James Almer
2022-09-22 23:14 ` [FFmpeg-devel] [PATCH v2] " James Almer
2022-09-23 21:20 ` James Almer
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