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bh=MuimzO/d8VJ2BRE0OUsGJPQw9elHxlAH19Orsoeh4xM=; b=ZhVZen/HcJIzZwwemFhFM0HcQ8fgH/HTGARib4A1tLHY87Rn3haeoNGCdQ4j8w2s0AJ/jn KQKzCN0BTKGiLbtETpiiRGbnmJoPumNZTUz2DRtJ7Q22QQrfbdbFhlKYCxnL8ST5ZhtBt0 WCay2yrMr2BSqrFC7UG869gvw4SXXvY= X-MC-Unique: VTnorYBjNfGnapAUZ3X_cQ-1 From: marcandre.lureau@redhat.com To: qemu-devel@nongnu.org Cc: Gerd Hoffmann , =?UTF-8?q?Marc-Andr=C3=A9=20Lureau?= , Christian Schoenebeck , =?UTF-8?q?Volker=20R=C3=BCmelin?= Subject: [PULL 25/27] audio: handle leftover audio frame from upsampling Date: Mon, 6 Mar 2023 10:52:00 +0400 Message-Id: <20230306065202.2160066-26-marcandre.lureau@redhat.com> In-Reply-To: <20230306065202.2160066-1-marcandre.lureau@redhat.com> References: <20230306065202.2160066-1-marcandre.lureau@redhat.com> MIME-Version: 1.0 Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: quoted-printable X-Scanned-By: MIMEDefang 3.1 on 10.11.54.2 Received-SPF: pass (zohomail.com: domain of gnu.org designates 209.51.188.17 as permitted sender) client-ip=209.51.188.17; envelope-from=qemu-devel-bounces+importer2=patchew.org@nongnu.org; helo=lists.gnu.org; Received-SPF: pass client-ip=170.10.133.124; envelope-from=marcandre.lureau@redhat.com; helo=us-smtp-delivery-124.mimecast.com X-Spam_score_int: -20 X-Spam_score: -2.1 X-Spam_bar: -- X-Spam_report: (-2.1 / 5.0 requ) BAYES_00=-1.9, DKIMWL_WL_HIGH=-0.001, DKIM_SIGNED=0.1, DKIM_VALID=-0.1, DKIM_VALID_AU=-0.1, DKIM_VALID_EF=-0.1, RCVD_IN_DNSWL_NONE=-0.0001, RCVD_IN_MSPIKE_H2=-0.001, SPF_HELO_NONE=0.001, SPF_PASS=-0.001 autolearn=ham autolearn_force=no X-Spam_action: no action X-BeenThere: qemu-devel@nongnu.org X-Mailman-Version: 2.1.29 Precedence: list List-Id: List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: qemu-devel-bounces+importer2=patchew.org@nongnu.org Sender: qemu-devel-bounces+importer2=patchew.org@nongnu.org X-ZohoMail-DKIM: pass (identity @redhat.com) X-ZM-MESSAGEID: 1678085817183100007 From: Volker R=C3=BCmelin Upsampling may leave one remaining audio frame in the input buffer. The emulated audio playback devices are currently resposible to write this audio frame again in the next write cycle. Push that task down to audio_pcm_sw_write. This is another step towards an audio callback interface that guarantees that when audio frontends are told they can write n audio frames, they can actually do so. Acked-by: Mark Cave-Ayland Acked-by: Marc-Andr=C3=A9 Lureau Signed-off-by: Volker R=C3=BCmelin Message-Id: <20230224190555.7409-13-vr_qemu@t-online.de> --- audio/audio_template.h | 6 ++++++ audio/audio.c | 34 ++++++++++++++++++++++++++++------ 2 files changed, 34 insertions(+), 6 deletions(-) diff --git a/audio/audio_template.h b/audio/audio_template.h index a0b653f52c..0d8aab6fad 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -138,6 +138,12 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) = (SW *sw) return -1; } =20 + /* + * Allocate one additional audio frame that is needed for upsampling + * if the resample buffer size is small. For large buffer sizes take + * care of overflows. + */ + samples =3D samples < INT_MAX ? samples + 1 : INT_MAX; sw->resample_buf.buffer =3D g_new0(st_sample, samples); sw->resample_buf.size =3D samples; sw->resample_buf.pos =3D 0; diff --git a/audio/audio.c b/audio/audio.c index dad17e59b8..4836ab8ca8 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -731,16 +731,21 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void= *buf, size_t buf_len) hw_free =3D hw_free > live ? hw_free - live : 0; frames_out_max =3D MIN(dead, hw_free); sw_max =3D st_rate_frames_in(sw->rate, frames_out_max); - fe_max =3D MIN(buf_len / sw->info.bytes_per_frame, sw->resample_buf.si= ze); + fe_max =3D MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.p= os, + sw->resample_buf.size); frames_in_max =3D MIN(sw_max, fe_max); =20 if (!frames_in_max) { return 0; } =20 - sw->conv(sw->resample_buf.buffer, buf, frames_in_max); - if (!sw->hw->pcm_ops->volume_out) { - mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol); + if (frames_in_max > sw->resample_buf.pos) { + sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos, + buf, frames_in_max - sw->resample_buf.pos); + if (!sw->hw->pcm_ops->volume_out) { + mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos, + frames_in_max - sw->resample_buf.pos, &sw->vol); + } } =20 audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max, @@ -749,6 +754,22 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void = *buf, size_t buf_len) sw->total_hw_samples_mixed +=3D total_out; sw->empty =3D sw->total_hw_samples_mixed =3D=3D 0; =20 + /* + * Upsampling may leave one audio frame in the resample buffer. Decrem= ent + * total_in by one if there was a leftover frame from the previous res= ample + * pass in the resample buffer. Increment total_in by one if the curre= nt + * resample pass left one frame in the resample buffer. + */ + if (frames_in_max - total_in =3D=3D 1) { + /* copy one leftover audio frame to the beginning of the buffer */ + *sw->resample_buf.buffer =3D *(sw->resample_buf.buffer + total_in); + total_in +=3D 1 - sw->resample_buf.pos; + sw->resample_buf.pos =3D 1; + } else if (total_in >=3D sw->resample_buf.pos) { + total_in -=3D sw->resample_buf.pos; + sw->resample_buf.pos =3D 0; + } + #ifdef DEBUG_OUT dolog ( "%s: write size %zu written %zu total mixed %zu\n", @@ -1155,8 +1176,9 @@ static void audio_run_out (AudioState *s) } else { free =3D 0; } - if (free > 0) { - free =3D MIN(free, sw->resample_buf.size); + if (free > sw->resample_buf.pos) { + free =3D MIN(free, sw->resample_buf.size) + - sw->resample_buf.pos; sw->callback.fn(sw->callback.opaque, free * sw->info.bytes_per_frame); } --=20 2.39.2