| Message ID | 1474988473-15177-1-git-send-email-fu.qiuping@hotmail.com |
|---|---|
| State | Superseded |
| Headers | show |
2016-09-27 23:01 GMT+08:00 <fu.qiuping@hotmail.com>: > From: frankos2 <frankqpfu@localhost.localdomain> > > --- > libavformat/flvenc.c | 8 ++++++++ > 1 files changed, 8 insertions(+), 0 deletions(-) > > diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c > index 99903f5..296426a 100644 > --- a/libavformat/flvenc.c > +++ b/libavformat/flvenc.c > @@ -107,6 +107,13 @@ static int get_audio_flags(AVFormatContext *s, > AVCodecParameters *par) > return FLV_CODECID_SPEEX | FLV_SAMPLERATE_11025HZ | > FLV_SAMPLESSIZE_16BIT; > } else { > switch (par->sample_rate) { > + case 48000: > + if (par->codec_id == AV_CODEC_ID_MP3) { > + flags |= FLV_SAMPLERATE_44100HZ; > maybe you want support 48000HZ for mp3? > + break; > + } else { > + goto error; > + } > case 44100: > flags |= FLV_SAMPLERATE_44100HZ; > break; > @@ -124,6 +131,7 @@ static int get_audio_flags(AVFormatContext *s, > AVCodecParameters *par) > break; > } > default: > +error: > av_log(s, AV_LOG_ERROR, > "FLV does not support sample rate %d, " > "choose from (44100, 22050, 11025)\n", > par->sample_rate); > -- > 1.7.1 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel >
2016-09-27 23:01 GMT+08:00 <fu.qiuping@hotmail.com>: > From: frankos2 <frankqpfu@localhost.localdomain> > > --- > libavformat/flvenc.c | 8 ++++++++ > 1 files changed, 8 insertions(+), 0 deletions(-) > > diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c > index 99903f5..296426a 100644 > --- a/libavformat/flvenc.c > +++ b/libavformat/flvenc.c > @@ -107,6 +107,13 @@ static int get_audio_flags(AVFormatContext *s, > AVCodecParameters *par) > return FLV_CODECID_SPEEX | FLV_SAMPLERATE_11025HZ | > FLV_SAMPLESSIZE_16BIT; > } else { > switch (par->sample_rate) { > + case 48000: > + if (par->codec_id == AV_CODEC_ID_MP3) { > + flags |= FLV_SAMPLERATE_44100HZ; > and you can support aac too. > + break; > + } else { > + goto error; > + } > case 44100: > flags |= FLV_SAMPLERATE_44100HZ; > break; > @@ -124,6 +131,7 @@ static int get_audio_flags(AVFormatContext *s, > AVCodecParameters *par) > break; > } > default: > +error: > av_log(s, AV_LOG_ERROR, > "FLV does not support sample rate %d, " > "choose from (44100, 22050, 11025)\n", > par->sample_rate); > -- > 1.7.1 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel >
AAC 48khz has alreay been supported in flv_enc.c file, when the flv with audio format is aac
it just set it into:
if (enc->codec_id == AV_CODEC_ID_AAC) // specs force these parameters
return FLV_CODECID_AAC | FLV_SAMPLERATE_44100HZ |
FLV_SAMPLESSIZE_16BIT | FLV_STEREO;
when i use ffmpeg in the live video stream project, some user publish the rtmp stream with (h264 and mp3 48khz)
then i need to use ffmpeg to convert the "rtmp" live stream to "flv" live stream, ffmpeg can not support and make a warning.
so i change this code to make it support this.
________________________________
发件人: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> 代表 Steven Liu <lingjiujianke@gmail.com>
发送时间: 2016年9月27日 15:15
收件人: FFmpeg development discussions and patches
主题: Re: [FFmpeg-devel] [PATCH] flv format support mp3 audio with 48khz
2016-09-27 23:01 GMT+08:00 <fu.qiuping@hotmail.com>:
> From: frankos2 <frankqpfu@localhost.localdomain>
>
> ---
> libavformat/flvenc.c | 8 ++++++++
> 1 files changed, 8 insertions(+), 0 deletions(-)
>
> diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c
> index 99903f5..296426a 100644
> --- a/libavformat/flvenc.c
> +++ b/libavformat/flvenc.c
> @@ -107,6 +107,13 @@ static int get_audio_flags(AVFormatContext *s,
> AVCodecParameters *par)
> return FLV_CODECID_SPEEX | FLV_SAMPLERATE_11025HZ |
> FLV_SAMPLESSIZE_16BIT;
> } else {
> switch (par->sample_rate) {
> + case 48000:
> + if (par->codec_id == AV_CODEC_ID_MP3) {
> + flags |= FLV_SAMPLERATE_44100HZ;
>
and you can support aac too.
> + break;
> + } else {
> + goto error;
> + }
> case 44100:
> flags |= FLV_SAMPLERATE_44100HZ;
> break;
> @@ -124,6 +131,7 @@ static int get_audio_flags(AVFormatContext *s,
> AVCodecParameters *par)
> break;
> }
> default:
> +error:
> av_log(s, AV_LOG_ERROR,
> "FLV does not support sample rate %d, "
> "choose from (44100, 22050, 11025)\n",
> par->sample_rate);
> --
> 1.7.1
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
ffmpeg-devel Info Page<http://ffmpeg.org/mailman/listinfo/ffmpeg-devel>
ffmpeg.org
This list is about FFmpeg development discussions and patches; but not for bug-reports. Please read the Code-of-conduct. To see the collection of prior postings to ...
>
_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
2016-09-27 23:30 GMT+08:00 付 秋平 <fu.qiuping@hotmail.com>: > AAC 48khz has alreay been supported in flv_enc.c file, when the flv with > audio format is aac > > it just set it into: > > if (enc->codec_id == AV_CODEC_ID_AAC) // specs force these parameters > return FLV_CODECID_AAC | FLV_SAMPLERATE_44100HZ | > FLV_SAMPLESSIZE_16BIT | FLV_STEREO; > > when i use ffmpeg in the live video stream project, some user publish the > rtmp stream with (h264 and mp3 48khz) > then i need to use ffmpeg to convert the "rtmp" live stream to "flv" live > stream, ffmpeg can not support and make a warning. > so i change this code to make it support this. > ________________________________ > 发件人: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> 代表 Steven Liu < > lingjiujianke@gmail.com> > 发送时间: 2016年9月27日 15:15 > 收件人: FFmpeg development discussions and patches > 主题: Re: [FFmpeg-devel] [PATCH] flv format support mp3 audio with 48khz > > 2016-09-27 23:01 GMT+08:00 <fu.qiuping@hotmail.com>: > > > From: frankos2 <frankqpfu@localhost.localdomain> > > > > --- > > libavformat/flvenc.c | 8 ++++++++ > > 1 files changed, 8 insertions(+), 0 deletions(-) > > > > diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c > > index 99903f5..296426a 100644 > > --- a/libavformat/flvenc.c > > +++ b/libavformat/flvenc.c > > @@ -107,6 +107,13 @@ static int get_audio_flags(AVFormatContext *s, > > AVCodecParameters *par) > > return FLV_CODECID_SPEEX | FLV_SAMPLERATE_11025HZ | > > FLV_SAMPLESSIZE_16BIT; > > } else { > > switch (par->sample_rate) { > > + case 48000: > > + if (par->codec_id == AV_CODEC_ID_MP3) { > > + flags |= FLV_SAMPLERATE_44100HZ; > > > and you can support aac too. > > > + break; > > + } else { > > + goto error; > > + } > > case 44100: > > flags |= FLV_SAMPLERATE_44100HZ; > > break; > > @@ -124,6 +131,7 @@ static int get_audio_flags(AVFormatContext *s, > > AVCodecParameters *par) > > break; > > } > > default: > > +error: > > av_log(s, AV_LOG_ERROR, > > "FLV does not support sample rate %d, " > > "choose from (44100, 22050, 11025)\n", > > par->sample_rate); > > -- > > 1.7.1 > > > > _______________________________________________ > > ffmpeg-devel mailing list > > ffmpeg-devel@ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > ffmpeg-devel Info Page<http://ffmpeg.org/mailman/listinfo/ffmpeg-devel> > ffmpeg.org > This list is about FFmpeg development discussions and patches; but not for > bug-reports. Please read the Code-of-conduct. To see the collection of > prior postings to ... > > > > > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > Ok, LGTM, thanks!
On Tue, Sep 27, 2016 at 08:01:13AM -0700, fu.qiuping@hotmail.com wrote:
> From: frankos2 <frankqpfu@localhost.localdomain>
This is not a name and email address
is this intended as "Author" information for git ?
Or is this a mistake ?
If its a mistake please correct it and resubmit the patch
thx
[...]
2016-09-28 0:01 GMT+09:00 <fu.qiuping@hotmail.com>: > From: frankos2 <frankqpfu@localhost.localdomain> > > --- > libavformat/flvenc.c | 8 ++++++++ > 1 files changed, 8 insertions(+), 0 deletions(-) > > diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c > index 99903f5..296426a 100644 > --- a/libavformat/flvenc.c > +++ b/libavformat/flvenc.c > @@ -107,6 +107,13 @@ static int get_audio_flags(AVFormatContext *s, > AVCodecParameters *par) > return FLV_CODECID_SPEEX | FLV_SAMPLERATE_11025HZ | > FLV_SAMPLESSIZE_16BIT; > } else { > switch (par->sample_rate) { > + case 48000: > + if (par->codec_id == AV_CODEC_ID_MP3) { > + flags |= FLV_SAMPLERATE_44100HZ; > + break; > + } else { > + goto error; > + } > case 44100: > flags |= FLV_SAMPLERATE_44100HZ; > break; > @@ -124,6 +131,7 @@ static int get_audio_flags(AVFormatContext *s, > AVCodecParameters *par) > break; > } > default: > +error: > av_log(s, AV_LOG_ERROR, > "FLV does not support sample rate %d, " > "choose from (44100, 22050, 11025)\n", > par->sample_rate); > -- > 1.7.1 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > Out of the spec I think. Did you confirm that Adobe (Flash Player) allows 48kHz mp3? About AAC, the spec (Adobe Flash Video File Format Specification) says the Flash Player ignores sample rate in AudioTagHeader, which is a dummy, and extracts the actual sample rate from AAC bitstream. But the spec does not say about MP3 in the same way.
I have already tried the "mp3_audio_48khz" flv live stream and rtmp live stream in the adobe flash player. It supports the format and works well. ________________________________ 发件人: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> 代表 Yusuke Nakamura <muken.the.vfrmaniac@gmail.com> 发送时间: 2016年9月27日 21:15 收件人: FFmpeg development discussions and patches 主题: Re: [FFmpeg-devel] [PATCH] flv format support mp3 audio with 48khz 2016-09-28 0:01 GMT+09:00 <fu.qiuping@hotmail.com>: > From: frankos2 <frankqpfu@localhost.localdomain> > > --- > libavformat/flvenc.c | 8 ++++++++ > 1 files changed, 8 insertions(+), 0 deletions(-) > > diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c > index 99903f5..296426a 100644 > --- a/libavformat/flvenc.c > +++ b/libavformat/flvenc.c > @@ -107,6 +107,13 @@ static int get_audio_flags(AVFormatContext *s, > AVCodecParameters *par) > return FLV_CODECID_SPEEX | FLV_SAMPLERATE_11025HZ | > FLV_SAMPLESSIZE_16BIT; > } else { > switch (par->sample_rate) { > + case 48000: > + if (par->codec_id == AV_CODEC_ID_MP3) { > + flags |= FLV_SAMPLERATE_44100HZ; > + break; > + } else { > + goto error; > + } > case 44100: > flags |= FLV_SAMPLERATE_44100HZ; > break; > @@ -124,6 +131,7 @@ static int get_audio_flags(AVFormatContext *s, > AVCodecParameters *par) > break; > } > default: > +error: > av_log(s, AV_LOG_ERROR, > "FLV does not support sample rate %d, " > "choose from (44100, 22050, 11025)\n", > par->sample_rate); > -- > 1.7.1 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel ffmpeg-devel Info Page<http://ffmpeg.org/mailman/listinfo/ffmpeg-devel> ffmpeg.org This list is about FFmpeg development discussions and patches; but not for bug-reports. Please read the Code-of-conduct. To see the collection of prior postings to ... > Out of the spec I think. Did you confirm that Adobe (Flash Player) allows 48kHz mp3? About AAC, the spec (Adobe Flash Video File Format Specification) says the Flash Player ignores sample rate in AudioTagHeader, which is a dummy, and extracts the actual sample rate from AAC bitstream. But the spec does not say about MP3 in the same way. _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Sorry, this is a mistake.
I will change it and resubmit the patch.
________________________________
发件人: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> 代表 Michael Niedermayer <michael@niedermayer.cc>
发送时间: 2016年9月27日 19:57
收件人: FFmpeg development discussions and patches
主题: Re: [FFmpeg-devel] [PATCH] flv format support mp3 audio with 48khz
On Tue, Sep 27, 2016 at 08:01:13AM -0700, fu.qiuping@hotmail.com wrote:
> From: frankos2 <frankqpfu@localhost.localdomain>
This is not a name and email address
is this intended as "Author" information for git ?
Or is this a mistake ?
If its a mistake please correct it and resubmit the patch
thx
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
There will always be a question for which you do not know the correct answer.
diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c index 99903f5..296426a 100644 --- a/libavformat/flvenc.c +++ b/libavformat/flvenc.c @@ -107,6 +107,13 @@ static int get_audio_flags(AVFormatContext *s, AVCodecParameters *par) return FLV_CODECID_SPEEX | FLV_SAMPLERATE_11025HZ | FLV_SAMPLESSIZE_16BIT; } else { switch (par->sample_rate) { + case 48000: + if (par->codec_id == AV_CODEC_ID_MP3) { + flags |= FLV_SAMPLERATE_44100HZ; + break; + } else { + goto error; + } case 44100: flags |= FLV_SAMPLERATE_44100HZ; break; @@ -124,6 +131,7 @@ static int get_audio_flags(AVFormatContext *s, AVCodecParameters *par) break; } default: +error: av_log(s, AV_LOG_ERROR, "FLV does not support sample rate %d, " "choose from (44100, 22050, 11025)\n", par->sample_rate);
From: frankos2 <frankqpfu@localhost.localdomain> --- libavformat/flvenc.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-)