static void gst_flac_dec_metadata_cb (const FLAC__StreamDecoder * decoder, const FLAC__StreamMetadata * metadata, void *client_data) { GstFlacDec *flacdec = GST_FLAC_DEC (client_data); GST_LOG_OBJECT (flacdec, "metadata type: %d", metadata->type); switch (metadata->type) { case FLAC__METADATA_TYPE_STREAMINFO:{ gint64 samples; guint depth, width, gdepth; samples = metadata->data.stream_info.total_samples; flacdec->min_blocksize = metadata->data.stream_info.min_blocksize; flacdec->max_blocksize = metadata->data.stream_info.max_blocksize; flacdec->depth = depth = metadata->data.stream_info.bits_per_sample; if (depth < 9) { gdepth = width = 8; } else if (depth < 17) { gdepth = width = 16; } else if (depth < 25) { gdepth = 24; width = 32; } else { gdepth = width = 32; } gst_audio_info_set_format (&flacdec->info, gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth), metadata->data.stream_info.sample_rate, metadata->data.stream_info.channels, NULL); memcpy (flacdec->info.position, channel_positions[flacdec->info.channels - 1], sizeof (GstAudioChannelPosition) * flacdec->info.channels); gst_audio_channel_positions_to_valid_order (flacdec->info.position, flacdec->info.channels); /* Note: we create the inverse reordering map here */ gst_audio_get_channel_reorder_map (flacdec->info.channels, flacdec->info.position, channel_positions[flacdec->info.channels - 1], flacdec->channel_reorder_map); GST_DEBUG_OBJECT (flacdec, "blocksize: min=%u, max=%u", flacdec->min_blocksize, flacdec->max_blocksize); GST_DEBUG_OBJECT (flacdec, "sample rate: %u, channels: %u", flacdec->info.rate, flacdec->info.channels); GST_DEBUG_OBJECT (flacdec, "depth: %u, width: %u", flacdec->depth, flacdec->info.finfo->width); GST_DEBUG_OBJECT (flacdec, "total samples = %" G_GINT64_FORMAT, samples); break; } default: break; } }
/* Converts an AudioStreamBasicDescription to preferred caps. * * These caps will indicate the AU element's canonical format, which won't * make Core Audio resample nor convert. * * NOTE ON MULTI-CHANNEL AUDIO: * * If layout is not NULL, resulting caps will only include the subset * of channels supported by GStreamer. If the Core Audio layout contained * ANY positioned channels, then ONLY positioned channels will be included * in the resulting caps. Otherwise, resulting caps will be unpositioned, * and include only unpositioned channels. * (Channels with unsupported AudioChannelLabel will be skipped either way.) * * Naturally, the number of channels indicated by 'channels' can be lower * than the AU element's total number of channels. */ GstCaps * gst_core_audio_asbd_to_caps (AudioStreamBasicDescription * asbd, AudioChannelLayout * layout) { GstAudioInfo info; GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN; guint rate, channels, bps, endianness; guint64 channel_mask; gboolean sign, interleaved; GstAudioChannelPosition pos[GST_OSX_AUDIO_MAX_CHANNEL]; if (asbd->mFormatID != kAudioFormatLinearPCM) { GST_WARNING ("Only linear PCM is supported"); goto error; } if (!(asbd->mFormatFlags & kAudioFormatFlagIsPacked)) { GST_WARNING ("Only packed formats supported"); goto error; } if (asbd->mFormatFlags & kLinearPCMFormatFlagsSampleFractionMask) { GST_WARNING ("Fixed point audio is unsupported"); goto error; } rate = asbd->mSampleRate; if (rate == kAudioStreamAnyRate) { GST_WARNING ("No sample rate"); goto error; } bps = asbd->mBitsPerChannel; endianness = asbd->mFormatFlags & kAudioFormatFlagIsBigEndian ? G_BIG_ENDIAN : G_LITTLE_ENDIAN; sign = asbd->mFormatID & kAudioFormatFlagIsSignedInteger ? TRUE : FALSE; interleaved = asbd->mFormatFlags & kAudioFormatFlagIsNonInterleaved ? TRUE : FALSE; if (asbd->mFormatFlags & kAudioFormatFlagIsFloat) { if (bps == 32) { if (endianness == G_LITTLE_ENDIAN) format = GST_AUDIO_FORMAT_F32LE; else format = GST_AUDIO_FORMAT_F32BE; } else if (bps == 64) { if (endianness == G_LITTLE_ENDIAN) format = GST_AUDIO_FORMAT_F64LE; else format = GST_AUDIO_FORMAT_F64BE; } } else { format = gst_audio_format_build_integer (sign, endianness, bps, bps); } if (format == GST_AUDIO_FORMAT_UNKNOWN) { GST_WARNING ("Unsupported sample format"); goto error; } if (layout) { if (!gst_core_audio_parse_channel_layout (layout, &channels, &channel_mask, pos)) { GST_WARNING ("Failed to parse channel layout, best effort channels layout mapping will be used"); layout = NULL; } } if (layout) { /* The AU can have arbitrary channel order, but we're using GstAudioInfo * which supports only the GStreamer channel order. * Also, we're eventually producing caps, which only have channel-mask * (whose implied order is the GStreamer channel order). */ gst_audio_channel_positions_to_valid_order (pos, channels); gst_audio_info_set_format (&info, format, rate, channels, pos); } else { channels = MIN (asbd->mChannelsPerFrame, GST_OSX_AUDIO_MAX_CHANNEL); gst_audio_info_set_format (&info, format, rate, channels, NULL); } return gst_audio_info_to_caps (&info); error: return NULL; }
static HRESULT WINAPI Gstreamer_AudioConvert_SetMediaType(TransformFilter *tf, PIN_DIRECTION dir, const AM_MEDIA_TYPE *amt) { GstTfImpl *This = (GstTfImpl*)tf; GstCaps *capsin, *capsout; AM_MEDIA_TYPE *outpmt = &This->tf.pmt; WAVEFORMATEX *inwfe; WAVEFORMATEX *outwfe; WAVEFORMATEXTENSIBLE *outwfx; GstAudioFormat format; HRESULT hr; BOOL inisfloat = FALSE; int indepth; TRACE("%p 0x%x %p\n", This, dir, amt); mark_wine_thread(); if (dir != PINDIR_INPUT) return S_OK; if (Gstreamer_AudioConvert_QueryConnect(&This->tf, amt) == S_FALSE || !amt->pbFormat) return E_FAIL; FreeMediaType(outpmt); *outpmt = *amt; outpmt->pUnk = NULL; outpmt->cbFormat = sizeof(WAVEFORMATEXTENSIBLE); outpmt->pbFormat = CoTaskMemAlloc(outpmt->cbFormat); inwfe = (WAVEFORMATEX*)amt->pbFormat; indepth = inwfe->wBitsPerSample; if (inwfe->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { WAVEFORMATEXTENSIBLE *inwfx = (WAVEFORMATEXTENSIBLE*)inwfe; inisfloat = IsEqualGUID(&inwfx->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT); if (inwfx->Samples.wValidBitsPerSample) indepth = inwfx->Samples.wValidBitsPerSample; } else if (inwfe->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) inisfloat = TRUE; if (inisfloat) format = inwfe->wBitsPerSample == 64 ? GST_AUDIO_FORMAT_F64LE : GST_AUDIO_FORMAT_F32LE; else format = gst_audio_format_build_integer(inwfe->wBitsPerSample != 8, G_LITTLE_ENDIAN, inwfe->wBitsPerSample, indepth); capsin = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING, gst_audio_format_to_string(format), "channels", G_TYPE_INT, inwfe->nChannels, "rate", G_TYPE_INT, inwfe->nSamplesPerSec, NULL); outwfe = (WAVEFORMATEX*)outpmt->pbFormat; outwfx = (WAVEFORMATEXTENSIBLE*)outwfe; outwfe->wFormatTag = WAVE_FORMAT_EXTENSIBLE; outwfe->nChannels = 2; outwfe->nSamplesPerSec = inwfe->nSamplesPerSec; outwfe->wBitsPerSample = 16; outwfe->nBlockAlign = outwfe->nChannels * outwfe->wBitsPerSample / 8; outwfe->nAvgBytesPerSec = outwfe->nBlockAlign * outwfe->nSamplesPerSec; outwfe->cbSize = sizeof(*outwfx) - sizeof(*outwfe); outwfx->Samples.wValidBitsPerSample = outwfe->wBitsPerSample; outwfx->dwChannelMask = SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT; outwfx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM; capsout = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING, "S16LE", "channels", G_TYPE_INT, outwfe->nChannels, "rate", G_TYPE_INT, outwfe->nSamplesPerSec, NULL); hr = Gstreamer_transform_ConnectInput(This, amt, capsin, capsout); gst_caps_unref(capsin); gst_caps_unref(capsout); This->cbBuffer = inwfe->nAvgBytesPerSec; return hr; }
static FLAC__StreamDecoderWriteStatus gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame, const FLAC__int32 * const buffer[]) { GstFlowReturn ret = GST_FLOW_OK; GstBuffer *outbuf; guint depth = frame->header.bits_per_sample; guint width, gdepth; guint sample_rate = frame->header.sample_rate; guint channels = frame->header.channels; guint samples = frame->header.blocksize; guint j, i; GstMapInfo map; gboolean caps_changed; GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples); if (depth == 0) { if (flacdec->depth < 4 || flacdec->depth > 32) { GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO", flacdec->depth); ret = GST_FLOW_ERROR; goto done; } depth = flacdec->depth; } switch (depth) { case 8: gdepth = width = 8; break; case 12: case 16: gdepth = width = 16; break; case 20: case 24: gdepth = 24; width = 32; break; case 32: gdepth = width = 32; break; default: GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth); ret = GST_FLOW_ERROR; goto done; } if (sample_rate == 0) { if (flacdec->info.rate != 0) { sample_rate = flacdec->info.rate; } else { GST_ERROR_OBJECT (flacdec, "unknown sample rate"); ret = GST_FLOW_ERROR; goto done; } } caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info)) || (width != GST_AUDIO_INFO_WIDTH (&flacdec->info)) || (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info)) || (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info)); if (caps_changed || !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) { GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate, channels); gst_audio_info_set_format (&flacdec->info, gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth), sample_rate, channels, NULL); memcpy (flacdec->info.position, channel_positions[flacdec->info.channels - 1], sizeof (GstAudioChannelPosition) * flacdec->info.channels); gst_audio_channel_positions_to_valid_order (flacdec->info.position, flacdec->info.channels); /* Note: we create the inverse reordering map here */ gst_audio_get_channel_reorder_map (flacdec->info.channels, flacdec->info.position, channel_positions[flacdec->info.channels - 1], flacdec->channel_reorder_map); flacdec->depth = depth; gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec), &flacdec->info); } outbuf = gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL); gst_buffer_map (outbuf, &map, GST_MAP_WRITE); if (width == 8) { gint8 *outbuffer = (gint8 *) map.data; gint *reorder_map = flacdec->channel_reorder_map; if (gdepth != depth) { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint8) (buffer[reorder_map[j]][i] << (gdepth - depth)); } } } else { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint8) buffer[reorder_map[j]][i]; } } } } else if (width == 16) { gint16 *outbuffer = (gint16 *) map.data; gint *reorder_map = flacdec->channel_reorder_map; if (gdepth != depth) { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint16) (buffer[reorder_map[j]][i] << (gdepth - depth)); } } } else { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint16) buffer[reorder_map[j]][i]; } } } } else if (width == 32) { gint32 *outbuffer = (gint32 *) map.data; gint *reorder_map = flacdec->channel_reorder_map; if (gdepth != depth) { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint32) (buffer[reorder_map[j]][i] << (gdepth - depth)); } } } else { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint32) buffer[reorder_map[j]][i]; } } } } else { g_assert_not_reached (); } gst_buffer_unmap (outbuf, &map); GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples); if (flacdec->error_count) flacdec->error_count--; ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1); if (G_UNLIKELY (ret != GST_FLOW_OK)) { GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret)); } done: /* we act on the flow return value later in the handle_frame function, as we * don't want to mess up the internal decoder state by returning ABORT when * the error is in fact non-fatal (like a pad in flushing mode) and we want * to continue later. So just pretend everything's dandy and act later. */ flacdec->last_flow = ret; return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; }
static GstCaps * gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc * src, IPin * pin, IAMStreamConfig * streamcaps) { GstCaps *caps = NULL; HRESULT hres = S_OK; int icount = 0; int isize = 0; AUDIO_STREAM_CONFIG_CAPS ascc; int i = 0; if (!streamcaps) return NULL; streamcaps->GetNumberOfCapabilities (&icount, &isize); if (isize != sizeof (ascc)) return NULL; for (; i < icount; i++) { GstCapturePinMediaType *pin_mediatype = g_new0 (GstCapturePinMediaType, 1); pin->AddRef (); pin_mediatype->capture_pin = pin; hres = streamcaps->GetStreamCaps (i, &pin_mediatype->mediatype, (BYTE *) & ascc); if (hres == S_OK && pin_mediatype->mediatype) { GstCaps *mediacaps = NULL; if (!caps) caps = gst_caps_new_empty (); if (gst_dshow_check_mediatype (pin_mediatype->mediatype, MEDIASUBTYPE_PCM, FORMAT_WaveFormatEx)) { GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN; WAVEFORMATEX *wavformat = (WAVEFORMATEX *) pin_mediatype->mediatype->pbFormat; switch (wavformat->wFormatTag) { case WAVE_FORMAT_PCM: format = gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, wavformat->wBitsPerSample, wavformat->wBitsPerSample); break; default: break; } if (format != GST_AUDIO_FORMAT_UNKNOWN) { GstAudioInfo info; gst_audio_info_init(&info); gst_audio_info_set_format(&info, format, wavformat->nSamplesPerSec, wavformat->nChannels, NULL); mediacaps = gst_audio_info_to_caps(&info); } if (mediacaps) { src->pins_mediatypes = g_list_append (src->pins_mediatypes, pin_mediatype); gst_caps_append (caps, mediacaps); } else { gst_dshow_free_pin_mediatype (pin_mediatype); } } else { gst_dshow_free_pin_mediatype (pin_mediatype); } } else { gst_dshow_free_pin_mediatype (pin_mediatype); } } if (caps && gst_caps_is_empty (caps)) { gst_caps_unref (caps); caps = NULL; } return caps; }
static void gst_omx_audio_dec_loop (GstOMXAudioDec * self) { GstOMXPort *port = self->dec_out_port; GstOMXBuffer *buf = NULL; GstFlowReturn flow_ret = GST_FLOW_OK; GstOMXAcquireBufferReturn acq_return; OMX_ERRORTYPE err; acq_return = gst_omx_port_acquire_buffer (port, &buf); if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) { goto component_error; } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) { goto flushing; } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) { goto eos; } if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (self)) || acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) { OMX_PARAM_PORTDEFINITIONTYPE port_def; OMX_AUDIO_PARAM_PCMMODETYPE pcm_param; GstAudioChannelPosition omx_position[OMX_AUDIO_MAXCHANNELS]; GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self); gint i; GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps"); /* Reallocate all buffers */ if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE && gst_omx_port_is_enabled (port)) { err = gst_omx_port_set_enabled (port, FALSE); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_deallocate_buffers (port); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND); if (err != OMX_ErrorNone) goto reconfigure_error; } /* Just update caps */ GST_AUDIO_DECODER_STREAM_LOCK (self); gst_omx_port_get_port_definition (port, &port_def); g_assert (port_def.format.audio.eEncoding == OMX_AUDIO_CodingPCM); GST_OMX_INIT_STRUCT (&pcm_param); pcm_param.nPortIndex = self->dec_out_port->index; err = gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioPcm, &pcm_param); if (err != OMX_ErrorNone) { GST_ERROR_OBJECT (self, "Failed to get PCM parameters: %s (0x%08x)", gst_omx_error_to_string (err), err); goto caps_failed; } g_assert (pcm_param.ePCMMode == OMX_AUDIO_PCMModeLinear); g_assert (pcm_param.bInterleaved == OMX_TRUE); gst_audio_info_init (&self->info); for (i = 0; i < pcm_param.nChannels; i++) { switch (pcm_param.eChannelMapping[i]) { case OMX_AUDIO_ChannelLF: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; break; case OMX_AUDIO_ChannelRF: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; break; case OMX_AUDIO_ChannelCF: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; break; case OMX_AUDIO_ChannelLS: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT; break; case OMX_AUDIO_ChannelRS: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT; break; case OMX_AUDIO_ChannelLFE: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_LFE1; break; case OMX_AUDIO_ChannelCS: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; break; case OMX_AUDIO_ChannelLR: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; break; case OMX_AUDIO_ChannelRR: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; break; case OMX_AUDIO_ChannelNone: default: /* This will break the outer loop too as the * i == pcm_param.nChannels afterwards */ for (i = 0; i < pcm_param.nChannels; i++) omx_position[i] = GST_AUDIO_CHANNEL_POSITION_NONE; break; } } if (pcm_param.nChannels == 1 && omx_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER) omx_position[0] = GST_AUDIO_CHANNEL_POSITION_MONO; if (omx_position[0] == GST_AUDIO_CHANNEL_POSITION_NONE && klass->get_channel_positions) { GST_WARNING_OBJECT (self, "Failed to get a valid channel layout, trying fallback"); klass->get_channel_positions (self, self->dec_out_port, omx_position); } memcpy (self->position, omx_position, sizeof (omx_position)); gst_audio_channel_positions_to_valid_order (self->position, pcm_param.nChannels); self->needs_reorder = (memcmp (self->position, omx_position, sizeof (GstAudioChannelPosition) * pcm_param.nChannels) != 0); if (self->needs_reorder) gst_audio_get_channel_reorder_map (pcm_param.nChannels, self->position, omx_position, self->reorder_map); gst_audio_info_set_format (&self->info, gst_audio_format_build_integer (pcm_param.eNumData == OMX_NumericalDataSigned, pcm_param.eEndian == OMX_EndianLittle ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, pcm_param.nBitPerSample, pcm_param.nBitPerSample), pcm_param.nSamplingRate, pcm_param.nChannels, self->position); GST_DEBUG_OBJECT (self, "Setting output state: format %s, rate %u, channels %u", gst_audio_format_to_string (self->info.finfo->format), (guint) pcm_param.nSamplingRate, (guint) pcm_param.nChannels); if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self), &self->info) || !gst_audio_decoder_negotiate (GST_AUDIO_DECODER (self))) { if (buf) gst_omx_port_release_buffer (port, buf); goto caps_failed; } GST_AUDIO_DECODER_STREAM_UNLOCK (self); if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) { err = gst_omx_port_set_enabled (port, TRUE); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_allocate_buffers (port); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_populate (port); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_mark_reconfigured (port); if (err != OMX_ErrorNone) goto reconfigure_error; } /* Now get a buffer */ if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) { return; } } g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK); if (!buf) { g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)); GST_AUDIO_DECODER_STREAM_LOCK (self); goto eos; } /* This prevents a deadlock between the srcpad stream * lock and the audiocodec stream lock, if ::reset() * is called at the wrong time */ if (gst_omx_port_is_flushing (port)) { GST_DEBUG_OBJECT (self, "Flushing"); gst_omx_port_release_buffer (port, buf); goto flushing; } GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT, (guint) buf->omx_buf->nFlags, (guint64) buf->omx_buf->nTimeStamp); GST_AUDIO_DECODER_STREAM_LOCK (self); if (buf->omx_buf->nFilledLen > 0) { GstBuffer *outbuf; gint nframes, spf; GstMapInfo minfo; GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self); GST_DEBUG_OBJECT (self, "Handling output data"); if (buf->omx_buf->nFilledLen % self->info.bpf != 0) { gst_omx_port_release_buffer (port, buf); goto invalid_buffer; } outbuf = gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self), buf->omx_buf->nFilledLen); gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE); if (self->needs_reorder) { gint i, n_samples, c, n_channels; gint *reorder_map = self->reorder_map; gint16 *dest, *source; dest = (gint16 *) minfo.data; source = (gint16 *) (buf->omx_buf->pBuffer + buf->omx_buf->nOffset); n_samples = buf->omx_buf->nFilledLen / self->info.bpf; n_channels = self->info.channels; for (i = 0; i < n_samples; i++) { for (c = 0; c < n_channels; c++) { dest[i * n_channels + reorder_map[c]] = source[i * n_channels + c]; } } } else { memcpy (minfo.data, buf->omx_buf->pBuffer + buf->omx_buf->nOffset, buf->omx_buf->nFilledLen); } gst_buffer_unmap (outbuf, &minfo); nframes = 1; spf = klass->get_samples_per_frame (self, self->dec_out_port); if (spf != -1) { nframes = buf->omx_buf->nFilledLen / self->info.bpf; if (nframes % spf != 0) GST_WARNING_OBJECT (self, "Output buffer does not contain an integer " "number of input frames (frames: %d, spf: %d)", nframes, spf); nframes = (nframes + spf - 1) / spf; } GST_BUFFER_TIMESTAMP (outbuf) = gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND, OMX_TICKS_PER_SECOND); if (buf->omx_buf->nTickCount != 0) GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND, OMX_TICKS_PER_SECOND); flow_ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, nframes); } GST_DEBUG_OBJECT (self, "Read frame from component"); GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret)); if (buf) { err = gst_omx_port_release_buffer (port, buf); if (err != OMX_ErrorNone) goto release_error; } self->downstream_flow_ret = flow_ret; if (flow_ret != GST_FLOW_OK) goto flow_error; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; component_error: { GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("OpenMAX component in error state %s (0x%08x)", gst_omx_component_get_last_error_string (self->dec), gst_omx_component_get_last_error (self->dec))); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; self->started = FALSE; return; } flushing: { GST_DEBUG_OBJECT (self, "Flushing -- stopping task"); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_FLUSHING; self->started = FALSE; return; } eos: { g_mutex_lock (&self->drain_lock); if (self->draining) { GST_DEBUG_OBJECT (self, "Drained"); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); flow_ret = GST_FLOW_OK; gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); } else { GST_DEBUG_OBJECT (self, "Component signalled EOS"); flow_ret = GST_FLOW_EOS; } g_mutex_unlock (&self->drain_lock); GST_AUDIO_DECODER_STREAM_LOCK (self); self->downstream_flow_ret = flow_ret; /* Here we fallback and pause the task for the EOS case */ if (flow_ret != GST_FLOW_OK) goto flow_error; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; } flow_error: { if (flow_ret == GST_FLOW_EOS) { GST_DEBUG_OBJECT (self, "EOS"); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->started = FALSE; } else if (flow_ret < GST_FLOW_EOS) { GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Internal data stream error."), ("stream stopped, reason %s", gst_flow_get_name (flow_ret))); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->started = FALSE; } else if (flow_ret == GST_FLOW_FLUSHING) { GST_DEBUG_OBJECT (self, "Flushing -- stopping task"); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->started = FALSE; } GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; } reconfigure_error: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Unable to reconfigure output port")); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; self->started = FALSE; return; } invalid_buffer: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Invalid sized input buffer")); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED; self->started = FALSE; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; } caps_failed: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps")); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); GST_AUDIO_DECODER_STREAM_UNLOCK (self); self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED; self->started = FALSE; return; } release_error: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to relase output buffer to component: %s (0x%08x)", gst_omx_error_to_string (err), err)); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; self->started = FALSE; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; } }
static GstFlowReturn gst_imx_audio_uniaudio_dec_handle_frame(GstAudioDecoder *dec, GstBuffer *buffer) { GstMapInfo in_map; GstBuffer *out_buffer; gsize avail_out_size; GstImxAudioUniaudioDec *imx_audio_uniaudio_dec = GST_IMX_AUDIO_UNIAUDIO_DEC(dec); int32 dec_ret; uint32 offset = 0; uint8 *in_buf = NULL; uint32 in_size = 0; gboolean dec_loop = TRUE, flow_error = FALSE; /* With some formats such as Vorbis, the first few buffers are actually redundant, * since they contain codec data that was already specified in codec_data or * streamheader caps earlier. If this is the case, skip these buffers. */ if (imx_audio_uniaudio_dec->skip_header_counter < imx_audio_uniaudio_dec->num_vorbis_headers) { GST_TRACE_OBJECT(dec, "skipping header buffer #%u", imx_audio_uniaudio_dec->skip_header_counter); ++imx_audio_uniaudio_dec->skip_header_counter; return gst_audio_decoder_finish_frame(dec, NULL, 1); } if (buffer != NULL) { gst_buffer_map(buffer, &in_map, GST_MAP_READ); in_buf = in_map.data; in_size = in_map.size; } while (dec_loop) { GstBuffer *tmp_buf; uint8 *out_buf = NULL; uint32 out_size = 0; if (buffer != NULL) GST_TRACE_OBJECT(dec, "feeding %" G_GUINT32_FORMAT " bytes to the decoder", (guint32)in_size); else GST_TRACE_OBJECT(dec, "draining decoder"); dec_ret = imx_audio_uniaudio_dec->codec->decode_frame( imx_audio_uniaudio_dec->handle, in_buf, in_size, &offset, &out_buf, &out_size ); GST_TRACE_OBJECT(dec, "decode_frame: return 0x%x offset %" G_GUINT32_FORMAT " out_size %" G_GUINT32_FORMAT, (unsigned int)dec_ret, (guint32)offset, (guint32)out_size); if ((out_buf != NULL) && (out_size > 0)) { tmp_buf = gst_audio_decoder_allocate_output_buffer(dec, out_size); tmp_buf = gst_buffer_make_writable(tmp_buf); gst_buffer_fill(tmp_buf, 0, out_buf, out_size); gst_adapter_push(imx_audio_uniaudio_dec->out_adapter, tmp_buf); } if (out_buf != NULL) { gst_imx_audio_uniaudio_dec_free(out_buf); } if ((buffer != NULL) && (offset == in_map.size)) { dec_loop = FALSE; } switch (dec_ret) { case ACODEC_SUCCESS: break; case ACODEC_END_OF_STREAM: dec_loop = FALSE; break; case ACODEC_NOT_ENOUGH_DATA: break; case ACODEC_CAPIBILITY_CHANGE: break; default: { dec_loop = FALSE; flow_error = TRUE; GST_ELEMENT_ERROR(dec, STREAM, DECODE, ("could not decode"), ("error message: %s", imx_audio_uniaudio_dec->codec->get_last_error(imx_audio_uniaudio_dec->handle))); } } } if (buffer != NULL) gst_buffer_unmap(buffer, &in_map); if (flow_error) return GST_FLOW_ERROR; if (!(imx_audio_uniaudio_dec->has_audioinfo_set)) { UniACodecParameter parameter; GstAudioFormat pcm_fmt; GstAudioInfo audio_info; imx_audio_uniaudio_dec->codec->get_parameter(imx_audio_uniaudio_dec->handle, UNIA_OUTPUT_PCM_FORMAT, ¶meter); if ((parameter.outputFormat.width == 0) || (parameter.outputFormat.depth == 0)) { GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "no output format available yet"); return gst_audio_decoder_finish_frame(dec, NULL, 1); } GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "output sample width: %" G_GUINT32_FORMAT " depth: %" G_GUINT32_FORMAT, (guint32)(parameter.outputFormat.width), (guint32)(parameter.outputFormat.depth)); pcm_fmt = gst_audio_format_build_integer(TRUE, G_BYTE_ORDER, parameter.outputFormat.width, parameter.outputFormat.depth); GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "setting output format to: %s %d Hz %d channels", gst_audio_format_to_string(pcm_fmt), (gint)(parameter.outputFormat.samplerate), (gint)(parameter.outputFormat.channels)); gst_imx_audio_uniaudio_dec_clear_channel_positions(imx_audio_uniaudio_dec); gst_imx_audio_uniaudio_dec_fill_channel_positions(imx_audio_uniaudio_dec, parameter.outputFormat.layout, parameter.outputFormat.channels); imx_audio_uniaudio_dec->pcm_format = pcm_fmt; imx_audio_uniaudio_dec->num_channels = parameter.outputFormat.channels; gst_audio_info_set_format( &audio_info, pcm_fmt, parameter.outputFormat.samplerate, parameter.outputFormat.channels, imx_audio_uniaudio_dec->reordered_channel_positions ); gst_audio_decoder_set_output_format(dec, &audio_info); imx_audio_uniaudio_dec->has_audioinfo_set = TRUE; } avail_out_size = gst_adapter_available(imx_audio_uniaudio_dec->out_adapter); if (avail_out_size > 0) { out_buffer = gst_adapter_take_buffer(imx_audio_uniaudio_dec->out_adapter, avail_out_size); if (imx_audio_uniaudio_dec->original_channel_positions != imx_audio_uniaudio_dec->reordered_channel_positions) { gst_audio_buffer_reorder_channels( out_buffer, imx_audio_uniaudio_dec->pcm_format, imx_audio_uniaudio_dec->num_channels, imx_audio_uniaudio_dec->original_channel_positions, imx_audio_uniaudio_dec->reordered_channel_positions ); } return gst_audio_decoder_finish_frame(dec, out_buffer, 1); } else { return gst_audio_decoder_finish_frame(dec, NULL, 1); } }