Re: [FFmpeg-devel] [PATCH v2 1/2] doc/examples/transcode_aac: Don't ignore last encoded frame

[Andreas Unterweger <dustsigns@gmail.com>]

Am 24.04.2022 um 14:38 schrieb James Almer:
> On 4/24/2022 9:36 AM, Andreas Unterweger wrote:
>> Am 11.04.2022 um 08:46 schrieb Paul B Mahol:
>>> On Tue, Mar 22, 2022 at 6:47 PM Andreas Unterweger <dustsigns@gmail.com>
>>> wrote:
>>>
>>>> The last encoded frame is now fetched on EOF. It was previously left in
>>>> the encoder and caused a "1 frame left in queue" warning.
>>>>
>>>> Signed-off-by: Andreas Unterweger <dustsigns@gmail.com>
>>>> ---
>>>>    doc/examples/transcode_aac.c | 22 +++++++++++-----------
>>>>    1 file changed, 11 insertions(+), 11 deletions(-)
>>>>
>>>> diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
>>>> index 9102e55f16..c9b93f6439 100644
>>>> --- a/doc/examples/transcode_aac.c
>>>> +++ b/doc/examples/transcode_aac.c
>>>> @@ -377,6 +377,8 @@ static int decode_audio_frame(AVFrame *frame,
>>>>        if (error < 0)
>>>>            return error;
>>>>
>>>> +    *data_present = 0;
>>>> +    *finished = 0;
>>>>        /* Read one audio frame from the input file into a temporary packet.
>>>> */
>>>>        if ((error = av_read_frame(input_format_context, input_packet)) < 0)
>>>> {
>>>>            /* If we are at the end of the file, flush the decoder below. */
>>>> @@ -555,7 +557,7 @@ static int read_decode_convert_and_store(AVAudioFifo
>>>> *fifo,
>>>>        AVFrame *input_frame = NULL;
>>>>        /* Temporary storage for the converted input samples. */
>>>>        uint8_t **converted_input_samples = NULL;
>>>> -    int data_present = 0;
>>>> +    int data_present;
>>>>        int ret = AVERROR_EXIT;
>>>>
>>>>        /* Initialize temporary storage for one input frame. */
>>>> @@ -675,18 +677,17 @@ static int encode_audio_frame(AVFrame *frame,
>>>>            frame->pts = pts;
>>>>            pts += frame->nb_samples;
>>>>        }
>>>> -
>>>> +
>>>> +    *data_present = 0;
>>>>        /* Send the audio frame stored in the temporary packet to the
>>>> encoder.
>>>>         * The output audio stream encoder is used to do this. */
>>>>        error = avcodec_send_frame(output_codec_context, frame);
>>>> -    /* The encoder signals that it has nothing more to encode. */
>>>> -    if (error == AVERROR_EOF) {
>>>> -        error = 0;
>>>> -        goto cleanup;
>>>> -    } else if (error < 0) {
>>>> -        fprintf(stderr, "Could not send packet for encoding (error
>>>> '%s')\n",
>>>> -                av_err2str(error));
>>>> -        goto cleanup;
>>>> +    /* Check for errors, but proceed with fetching encoded samples if the
>>>> +     *  encoder signals that it has nothing more to encode. */
>>>> +    if (error < 0 && error != AVERROR_EOF) {
>>>> +      fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
>>>> +              av_err2str(error));
>>>> +      goto cleanup;
>>>>        }
>>>>
>>>>        /* Receive one encoded frame from the encoder. */
>>>> @@ -857,7 +858,6 @@ int main(int argc, char **argv)
>>>>                int data_written;
>>>>                /* Flush the encoder as it may have delayed frames. */
>>>>                do {
>>>> -                data_written = 0;
>>>>                    if (encode_audio_frame(NULL, output_format_context,
>>>>                                           output_codec_context,
>>>> &data_written))
>>>>                        goto cleanup;
>>>> -- 
>>>> 2.30.2
>>>>
>>>>
>>> LGTM. can someone apply this?
>>
>> Ping again
> 
> See the reply about the patch no longer applying.

I did already rebase the patch in v2. There have been no changes to the file since.

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