| Message ID | 20210301222049.15456-1-cus@passwd.hu |
|---|---|
| State | Accepted |
| Commit | 1a90cf4410a3464c8d749242e23629776f310ee0 |
| Headers | show |
| Series | None | expand |
On Mon, 1 Mar 2021, Marton Balint wrote: > Otherwise we might return 1-2 samples per packet if av_read_frame() call rate is > only sligthly less than the stream sample rate. Ping for this and the rest of the series. Note that the approach in this patch makes 1/5 unneeded because constant frame size is now indeed guaranteed. Regards, Marton > > Signed-off-by: Marton Balint <cus@passwd.hu> > --- > libavdevice/alsa.c | 5 +++++ > libavdevice/alsa.h | 1 + > libavdevice/alsa_dec.c | 22 ++++++++++++---------- > 3 files changed, 18 insertions(+), 10 deletions(-) > > diff --git a/libavdevice/alsa.c b/libavdevice/alsa.c > index 117b2ea144..ee282fac16 100644 > --- a/libavdevice/alsa.c > +++ b/libavdevice/alsa.c > @@ -286,6 +286,10 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode, > } > } > > + s->pkt = av_packet_alloc(); > + if (!s->pkt) > + goto fail1; > + > s->h = h; > return 0; > > @@ -308,6 +312,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1) > if (CONFIG_ALSA_INDEV) > ff_timefilter_destroy(s->timefilter); > snd_pcm_close(s->h); > + av_packet_free(&s->pkt); > return 0; > } > > diff --git a/libavdevice/alsa.h b/libavdevice/alsa.h > index 1ed8c82199..07783c983a 100644 > --- a/libavdevice/alsa.h > +++ b/libavdevice/alsa.h > @@ -58,6 +58,7 @@ typedef struct AlsaData { > void *reorder_buf; > int reorder_buf_size; ///< in frames > int64_t timestamp; ///< current timestamp, without latency applied. > + AVPacket *pkt; > } AlsaData; > > /** > diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c > index 6d568737b3..88bf32d25f 100644 > --- a/libavdevice/alsa_dec.c > +++ b/libavdevice/alsa_dec.c > @@ -104,34 +104,36 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) > int64_t dts; > snd_pcm_sframes_t delay = 0; > > - if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) { > - return AVERROR(EIO); > + if (!s->pkt->data) { > + int ret = av_new_packet(s->pkt, s->period_size * s->frame_size); > + if (ret < 0) > + return ret; > + s->pkt->size = 0; > } > > - while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) { > + do { > + while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, s->period_size - s->pkt->size / s->frame_size)) < 0) { > if (res == -EAGAIN) { > - av_packet_unref(pkt); > - > return AVERROR(EAGAIN); > } > + s->pkt->size = 0; > if (ff_alsa_xrun_recover(s1, res) < 0) { > av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", > snd_strerror(res)); > - av_packet_unref(pkt); > - > return AVERROR(EIO); > } > ff_timefilter_reset(s->timefilter); > - } > + } > + s->pkt->size += res * s->frame_size; > + } while (s->pkt->size < s->period_size * s->frame_size); > > + av_packet_move_ref(pkt, s->pkt); > dts = av_gettime(); > snd_pcm_delay(s->h, &delay); > dts -= av_rescale(delay + res, 1000000, s->sample_rate); > pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period); > s->last_period = res; > > - pkt->size = res * s->frame_size; > - > return 0; > } > > -- > 2.26.2 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
On Sat, 6 Mar 2021, Marton Balint wrote: > > > On Mon, 1 Mar 2021, Marton Balint wrote: > >> Otherwise we might return 1-2 samples per packet if av_read_frame() call > rate is >> only sligthly less than the stream sample rate. > > Ping for this and the rest of the series. Note that the approach in > this patch makes 1/5 unneeded because constant frame size is now indeed > guaranteed. Will apply the series soon. Regards, Marton >> >> Signed-off-by: Marton Balint <cus@passwd.hu> >> --- >> libavdevice/alsa.c | 5 +++++ >> libavdevice/alsa.h | 1 + >> libavdevice/alsa_dec.c | 22 ++++++++++++---------- >> 3 files changed, 18 insertions(+), 10 deletions(-) >> >> diff --git a/libavdevice/alsa.c b/libavdevice/alsa.c >> index 117b2ea144..ee282fac16 100644 >> --- a/libavdevice/alsa.c >> +++ b/libavdevice/alsa.c >> @@ -286,6 +286,10 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, > snd_pcm_stream_t mode, >> } >> } >> >> + s->pkt = av_packet_alloc(); >> + if (!s->pkt) >> + goto fail1; >> + >> s->h = h; >> return 0; >> >> @@ -308,6 +312,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1) >> if (CONFIG_ALSA_INDEV) >> ff_timefilter_destroy(s->timefilter); >> snd_pcm_close(s->h); >> + av_packet_free(&s->pkt); >> return 0; >> } >> >> diff --git a/libavdevice/alsa.h b/libavdevice/alsa.h >> index 1ed8c82199..07783c983a 100644 >> --- a/libavdevice/alsa.h >> +++ b/libavdevice/alsa.h >> @@ -58,6 +58,7 @@ typedef struct AlsaData { >> void *reorder_buf; >> int reorder_buf_size; ///< in frames >> int64_t timestamp; ///< current timestamp, without latency applied. >> + AVPacket *pkt; >> } AlsaData; >> >> /** >> diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c >> index 6d568737b3..88bf32d25f 100644 >> --- a/libavdevice/alsa_dec.c >> +++ b/libavdevice/alsa_dec.c >> @@ -104,34 +104,36 @@ static int audio_read_packet(AVFormatContext *s1, > AVPacket *pkt) >> int64_t dts; >> snd_pcm_sframes_t delay = 0; >> >> - if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) { >> - return AVERROR(EIO); >> + if (!s->pkt->data) { >> + int ret = av_new_packet(s->pkt, s->period_size * s->frame_size); >> + if (ret < 0) >> + return ret; >> + s->pkt->size = 0; >> } >> >> - while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) { >> + do { >> + while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, > s->period_size - s->pkt->size / s->frame_size)) < 0) { >> if (res == -EAGAIN) { >> - av_packet_unref(pkt); >> - >> return AVERROR(EAGAIN); >> } >> + s->pkt->size = 0; >> if (ff_alsa_xrun_recover(s1, res) < 0) { >> av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", >> snd_strerror(res)); >> - av_packet_unref(pkt); >> - >> return AVERROR(EIO); >> } >> ff_timefilter_reset(s->timefilter); >> - } >> + } >> + s->pkt->size += res * s->frame_size; >> + } while (s->pkt->size < s->period_size * s->frame_size); >> >> + av_packet_move_ref(pkt, s->pkt); >> dts = av_gettime(); >> snd_pcm_delay(s->h, &delay); >> dts -= av_rescale(delay + res, 1000000, s->sample_rate); >> pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period); >> s->last_period = res; >> >> - pkt->size = res * s->frame_size; >> - >> return 0; >> } >> >> -- >> 2.26.2 >> >> _______________________________________________ >> ffmpeg-devel mailing list >> ffmpeg-devel@ffmpeg.org >> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel >> >> To unsubscribe, visit link above, or email >> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
On Wed, 10 Mar 2021, Marton Balint wrote: > > > On Sat, 6 Mar 2021, Marton Balint wrote: > >> >> >> On Mon, 1 Mar 2021, Marton Balint wrote: >> >>> Otherwise we might return 1-2 samples per packet if av_read_frame() call >> rate is >>> only sligthly less than the stream sample rate. >> >> Ping for this and the rest of the series. Note that the approach in >> this patch makes 1/5 unneeded because constant frame size is now indeed >> guaranteed. > > Will apply the series soon. Applied. Regards, Marton > > Regards, > Marton > >>> >>> Signed-off-by: Marton Balint <cus@passwd.hu> >>> --- >>> libavdevice/alsa.c | 5 +++++ >>> libavdevice/alsa.h | 1 + >>> libavdevice/alsa_dec.c | 22 ++++++++++++---------- >>> 3 files changed, 18 insertions(+), 10 deletions(-) >>> >>> diff --git a/libavdevice/alsa.c b/libavdevice/alsa.c >>> index 117b2ea144..ee282fac16 100644 >>> --- a/libavdevice/alsa.c >>> +++ b/libavdevice/alsa.c >>> @@ -286,6 +286,10 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, >> snd_pcm_stream_t mode, >>> } >>> } >>> >>> + s->pkt = av_packet_alloc(); >>> + if (!s->pkt) >>> + goto fail1; >>> + >>> s->h = h; >>> return 0; >>> >>> @@ -308,6 +312,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1) >>> if (CONFIG_ALSA_INDEV) >>> ff_timefilter_destroy(s->timefilter); >>> snd_pcm_close(s->h); >>> + av_packet_free(&s->pkt); >>> return 0; >>> } >>> >>> diff --git a/libavdevice/alsa.h b/libavdevice/alsa.h >>> index 1ed8c82199..07783c983a 100644 >>> --- a/libavdevice/alsa.h >>> +++ b/libavdevice/alsa.h >>> @@ -58,6 +58,7 @@ typedef struct AlsaData { >>> void *reorder_buf; >>> int reorder_buf_size; ///< in frames >>> int64_t timestamp; ///< current timestamp, without latency applied. >>> + AVPacket *pkt; >>> } AlsaData; >>> >>> /** >>> diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c >>> index 6d568737b3..88bf32d25f 100644 >>> --- a/libavdevice/alsa_dec.c >>> +++ b/libavdevice/alsa_dec.c >>> @@ -104,34 +104,36 @@ static int audio_read_packet(AVFormatContext *s1, >> AVPacket *pkt) >>> int64_t dts; >>> snd_pcm_sframes_t delay = 0; >>> >>> - if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) { >>> - return AVERROR(EIO); >>> + if (!s->pkt->data) { >>> + int ret = av_new_packet(s->pkt, s->period_size * s->frame_size); >>> + if (ret < 0) >>> + return ret; >>> + s->pkt->size = 0; >>> } >>> >>> - while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) { >>> + do { >>> + while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, >> s->period_size - s->pkt->size / s->frame_size)) < 0) { >>> if (res == -EAGAIN) { >>> - av_packet_unref(pkt); >>> - >>> return AVERROR(EAGAIN); >>> } >>> + s->pkt->size = 0; >>> if (ff_alsa_xrun_recover(s1, res) < 0) { >>> av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", >>> snd_strerror(res)); >>> - av_packet_unref(pkt); >>> - >>> return AVERROR(EIO); >>> } >>> ff_timefilter_reset(s->timefilter); >>> - } >>> + } >>> + s->pkt->size += res * s->frame_size; >>> + } while (s->pkt->size < s->period_size * s->frame_size); >>> >>> + av_packet_move_ref(pkt, s->pkt); >>> dts = av_gettime(); >>> snd_pcm_delay(s->h, &delay); >>> dts -= av_rescale(delay + res, 1000000, s->sample_rate); >>> pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period); >>> s->last_period = res; >>> >>> - pkt->size = res * s->frame_size; >>> - >>> return 0; >>> } >>> >>> -- >>> 2.26.2 >>> >>> _______________________________________________ >>> ffmpeg-devel mailing list >>> ffmpeg-devel@ffmpeg.org >>> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel >>> >>> To unsubscribe, visit link above, or email >>> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". >> _______________________________________________ >> ffmpeg-devel mailing list >> ffmpeg-devel@ffmpeg.org >> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel >> >> To unsubscribe, visit link above, or email >> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
diff --git a/libavdevice/alsa.c b/libavdevice/alsa.c index 117b2ea144..ee282fac16 100644 --- a/libavdevice/alsa.c +++ b/libavdevice/alsa.c @@ -286,6 +286,10 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode, } } + s->pkt = av_packet_alloc(); + if (!s->pkt) + goto fail1; + s->h = h; return 0; @@ -308,6 +312,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1) if (CONFIG_ALSA_INDEV) ff_timefilter_destroy(s->timefilter); snd_pcm_close(s->h); + av_packet_free(&s->pkt); return 0; } diff --git a/libavdevice/alsa.h b/libavdevice/alsa.h index 1ed8c82199..07783c983a 100644 --- a/libavdevice/alsa.h +++ b/libavdevice/alsa.h @@ -58,6 +58,7 @@ typedef struct AlsaData { void *reorder_buf; int reorder_buf_size; ///< in frames int64_t timestamp; ///< current timestamp, without latency applied. + AVPacket *pkt; } AlsaData; /** diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c index 6d568737b3..88bf32d25f 100644 --- a/libavdevice/alsa_dec.c +++ b/libavdevice/alsa_dec.c @@ -104,34 +104,36 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) int64_t dts; snd_pcm_sframes_t delay = 0; - if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) { - return AVERROR(EIO); + if (!s->pkt->data) { + int ret = av_new_packet(s->pkt, s->period_size * s->frame_size); + if (ret < 0) + return ret; + s->pkt->size = 0; } - while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) { + do { + while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, s->period_size - s->pkt->size / s->frame_size)) < 0) { if (res == -EAGAIN) { - av_packet_unref(pkt); - return AVERROR(EAGAIN); } + s->pkt->size = 0; if (ff_alsa_xrun_recover(s1, res) < 0) { av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", snd_strerror(res)); - av_packet_unref(pkt); - return AVERROR(EIO); } ff_timefilter_reset(s->timefilter); - } + } + s->pkt->size += res * s->frame_size; + } while (s->pkt->size < s->period_size * s->frame_size); + av_packet_move_ref(pkt, s->pkt); dts = av_gettime(); snd_pcm_delay(s->h, &delay); dts -= av_rescale(delay + res, 1000000, s->sample_rate); pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period); s->last_period = res; - pkt->size = res * s->frame_size; - return 0; }
Otherwise we might return 1-2 samples per packet if av_read_frame() call rate is only sligthly less than the stream sample rate. Signed-off-by: Marton Balint <cus@passwd.hu> --- libavdevice/alsa.c | 5 +++++ libavdevice/alsa.h | 1 + libavdevice/alsa_dec.c | 22 ++++++++++++---------- 3 files changed, 18 insertions(+), 10 deletions(-)