static int libopus_decode(AVCodecContext *avc, void *frame, int *got_frame_ptr, AVPacket *pkt) { struct libopus_context *opus = avc->priv_data; int ret, nb_samples; opus->frame.nb_samples = MAX_FRAME_SIZE; ret = avc->get_buffer(avc, &opus->frame); if (ret < 0) { av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } if (avc->sample_fmt == AV_SAMPLE_FMT_S16) nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size, (opus_int16 *)opus->frame.data[0], opus->frame.nb_samples, 0); else nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size, (float *)opus->frame.data[0], opus->frame.nb_samples, 0); if (nb_samples < 0) { av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n", opus_strerror(nb_samples)); return ff_opus_error_to_averror(nb_samples); } opus->frame.nb_samples = nb_samples; *(AVFrame *)frame = opus->frame; *got_frame_ptr = 1; return pkt->size; }
int32 Decode(const uint8* FrameData, uint16 FrameSize, int16* OutPCMData, int32 SampleSize) { #if WITH_OPUS return opus_multistream_decode(Decoder, FrameData, FrameSize, OutPCMData, SampleSize, 0); #else return -1; #endif }
// packets must be decoded in order // a packet loss must call this function with NULL indata and 0 inlen // returns the number of decoded samples int nv_opus_decode(unsigned char* indata, int inlen, short* outpcmdata, int framesize) { int err; // Decoding to 16-bit PCM with FEC off // Maximum length assuming 48KHz sample rate err = opus_multistream_decode(decoder, indata, inlen, outpcmdata, framesize, 0); return err; }
static int libopus_decode(AVCodecContext *avc, void *data, int *got_frame_ptr, AVPacket *pkt) { struct libopus_context *opus = avc->priv_data; AVFrame *frame = data; int ret, nb_samples; frame->nb_samples = MAX_FRAME_SIZE; ret = ff_get_buffer(avc, frame); if (ret < 0) { av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } if (avc->sample_fmt == AV_SAMPLE_FMT_S16) nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size, (opus_int16 *)frame->data[0], frame->nb_samples, 0); else nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size, (float *)frame->data[0], frame->nb_samples, 0); if (nb_samples < 0) { av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n", opus_strerror(nb_samples)); return ff_opus_error_to_averror(nb_samples); } #ifndef OPUS_SET_GAIN { int i = avc->channels * nb_samples; if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) { float *pcm = (float *)frame->data[0]; for (; i > 0; i--, pcm++) *pcm = av_clipf(*pcm * opus->gain.d, -1, 1); } else { int16_t *pcm = (int16_t *)frame->data[0]; for (; i > 0; i--, pcm++) *pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16); } } #endif frame->nb_samples = nb_samples; *got_frame_ptr = 1; return pkt->size; }
static void omx_renderer_decode_and_play_sample(char* data, int length) { int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0); if (decodeLen > 0) { buf = ilclient_get_input_buffer(component, 100, 1); buf->nOffset = 0; buf->nFlags = OMX_BUFFERFLAG_TIME_UNKNOWN; int bufLength = decodeLen * sizeof(short) * channelCount; memcpy(buf->pBuffer, pcmBuffer, bufLength); buf->nFilledLen = bufLength; int r = OMX_EmptyThisBuffer(ilclient_get_handle(component), buf); if (r != OMX_ErrorNone) { fprintf(stderr, "Empty buffer error\n"); } } else { printf("Opus error from decode: %d\n", decodeLen); } }
void MoonlightInstance::AudDecDecodeAndPlaySample(char* sampleData, int sampleLength) { int decodeLen; // Check if there is space for this sample in the buffer. Again, this can race // but in the worst case, we'll not see the sample callback having consumed a sample. if (((s_WriteIndex + 1) % CIRCULAR_BUFFER_SIZE) == s_ReadIndex) { return; } decodeLen = opus_multistream_decode(g_Instance->m_OpusDecoder, (unsigned char *)sampleData, sampleLength, s_CircularBuffer[s_WriteIndex], FRAME_SIZE, 0); if (decodeLen > 0) { // Use a full memory barrier to ensure the circular buffer is written before incrementing the index __sync_synchronize(); // This can race with the reader in the sample callback, however this is a benign // race since we'll either read the original value of s_WriteIndex (which is safe, // we just won't consider this sample) or the new value of s_WriteIndex s_WriteIndex = (s_WriteIndex + 1) % CIRCULAR_BUFFER_SIZE; } }
static GstFlowReturn opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer) { GstFlowReturn res = GST_FLOW_OK; gsize size; guint8 *data; GstBuffer *outbuf; gint16 *out_data; int n, err; int samples; unsigned int packet_size; GstBuffer *buf; GstMapInfo map, omap; if (dec->state == NULL) { /* If we did not get any headers, default to 2 channels */ if (dec->n_channels == 0) { GST_INFO_OBJECT (dec, "No header, assuming single stream"); dec->n_channels = 2; dec->sample_rate = 48000; /* default stereo mapping */ dec->channel_mapping_family = 0; dec->channel_mapping[0] = 0; dec->channel_mapping[1] = 1; dec->n_streams = 1; dec->n_stereo_streams = 1; gst_opus_dec_negotiate (dec, NULL); } GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz", dec->n_channels, dec->sample_rate); #ifndef GST_DISABLE_GST_DEBUG gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug, "Mapping table", dec->n_channels, dec->channel_mapping); #endif GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams, dec->n_stereo_streams); dec->state = opus_multistream_decoder_create (dec->sample_rate, dec->n_channels, dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err); if (!dec->state || err != OPUS_OK) goto creation_failed; } if (buffer) { GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT, gst_buffer_get_size (buffer)); } else { GST_DEBUG_OBJECT (dec, "Received missing buffer"); } /* if using in-band FEC, we introdude one extra frame's delay as we need to potentially wait for next buffer to decode a missing buffer */ if (dec->use_inband_fec && !dec->primed) { GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out"); gst_buffer_replace (&dec->last_buffer, buffer); dec->primed = TRUE; goto done; } /* That's the buffer we'll be sending to the opus decoder. */ buf = (dec->use_inband_fec && gst_buffer_get_size (dec->last_buffer) > 0) ? dec->last_buffer : buffer; if (buf && gst_buffer_get_size (buf) > 0) { gst_buffer_map (buf, &map, GST_MAP_READ); data = map.data; size = map.size; GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size); } else { /* concealment data, pass NULL as the bits parameters */ GST_DEBUG_OBJECT (dec, "Using NULL buffer"); data = NULL; size = 0; } /* use maximum size (120 ms) as the number of returned samples is not constant over the stream. */ samples = 120 * dec->sample_rate / 1000; packet_size = samples * dec->n_channels * 2; outbuf = gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec), packet_size); if (!outbuf) { goto buffer_failed; } gst_buffer_map (outbuf, &omap, GST_MAP_WRITE); out_data = (gint16 *) omap.data; if (dec->use_inband_fec) { if (dec->last_buffer) { /* normal delayed decode */ GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer"); n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0); } else { /* FEC reconstruction decode */ GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer"); n = opus_multistream_decode (dec->state, data, size, out_data, samples, 1); } } else { /* normal decode */ GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer"); n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0); } gst_buffer_unmap (outbuf, &omap); if (data != NULL) gst_buffer_unmap (buf, &map); if (n < 0) { GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL)); gst_buffer_unref (outbuf); return GST_FLOW_ERROR; } GST_DEBUG_OBJECT (dec, "decoded %d samples", n); gst_buffer_set_size (outbuf, n * 2 * dec->n_channels); /* Skip any samples that need skipping */ if (dec->pre_skip > 0) { guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000; guint skip = scaled_pre_skip > n ? n : scaled_pre_skip; guint scaled_skip = skip * 48000 / dec->sample_rate; gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1); dec->pre_skip -= scaled_skip; GST_INFO_OBJECT (dec, "Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip, scaled_skip, dec->pre_skip); } if (gst_buffer_get_size (outbuf) == 0) { gst_buffer_unref (outbuf); outbuf = NULL; } else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) { gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16, dec->n_channels, dec->opus_pos, dec->info.position); } /* Apply gain */ /* Would be better off leaving this to a volume element, as this is a naive conversion that does too many int/float conversions. However, we don't have control over the pipeline... So make it optional if the user program wants to use a volume, but do it by default so the correct volume goes out by default */ if (dec->apply_gain && outbuf && dec->r128_gain) { gsize rsize; unsigned int i, nsamples; double volume = dec->r128_gain_volume; gint16 *samples; gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE); samples = (gint16 *) omap.data; rsize = omap.size; GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume); nsamples = rsize / 2; for (i = 0; i < nsamples; ++i) { int sample = (int) (samples[i] * volume + 0.5); samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample; } gst_buffer_unmap (outbuf, &omap); } if (dec->use_inband_fec) { gst_buffer_replace (&dec->last_buffer, buffer); } res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1); if (res != GST_FLOW_OK) GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res)); done: return res; creation_failed: GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err); return GST_FLOW_ERROR; buffer_failed: GST_ERROR_OBJECT (dec, "Failed to create %u byte buffer", packet_size); return GST_FLOW_ERROR; }
static GstFlowReturn opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer) { GstFlowReturn res = GST_FLOW_OK; gsize size; guint8 *data; GstBuffer *outbuf, *bufd; gint16 *out_data; int n, err; int samples; unsigned int packet_size; GstBuffer *buf; GstMapInfo map, omap; GstAudioClippingMeta *cmeta = NULL; if (dec->state == NULL) { /* If we did not get any headers, default to 2 channels */ if (dec->n_channels == 0) { GST_INFO_OBJECT (dec, "No header, assuming single stream"); dec->n_channels = 2; dec->sample_rate = 48000; /* default stereo mapping */ dec->channel_mapping_family = 0; dec->channel_mapping[0] = 0; dec->channel_mapping[1] = 1; dec->n_streams = 1; dec->n_stereo_streams = 1; if (!gst_opus_dec_negotiate (dec, NULL)) return GST_FLOW_NOT_NEGOTIATED; } if (dec->n_channels == 2 && dec->n_streams == 1 && dec->n_stereo_streams == 0) { /* if we are automatically decoding 2 channels, but only have a single encoded one, direct both channels to it */ dec->channel_mapping[1] = 0; } GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz", dec->n_channels, dec->sample_rate); #ifndef GST_DISABLE_GST_DEBUG gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug, "Mapping table", dec->n_channels, dec->channel_mapping); #endif GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams, dec->n_stereo_streams); dec->state = opus_multistream_decoder_create (dec->sample_rate, dec->n_channels, dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err); if (!dec->state || err != OPUS_OK) goto creation_failed; } if (buffer) { GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT, gst_buffer_get_size (buffer)); } else { GST_DEBUG_OBJECT (dec, "Received missing buffer"); } /* if using in-band FEC, we introdude one extra frame's delay as we need to potentially wait for next buffer to decode a missing buffer */ if (dec->use_inband_fec && !dec->primed) { GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out"); gst_buffer_replace (&dec->last_buffer, buffer); dec->primed = TRUE; goto done; } /* That's the buffer we'll be sending to the opus decoder. */ buf = (dec->use_inband_fec && gst_buffer_get_size (dec->last_buffer) > 0) ? dec->last_buffer : buffer; /* That's the buffer we get duration from */ bufd = dec->use_inband_fec ? dec->last_buffer : buffer; if (buf && gst_buffer_get_size (buf) > 0) { gst_buffer_map (buf, &map, GST_MAP_READ); data = map.data; size = map.size; GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size); } else { /* concealment data, pass NULL as the bits parameters */ GST_DEBUG_OBJECT (dec, "Using NULL buffer"); data = NULL; size = 0; } if (gst_buffer_get_size (bufd) == 0) { GstClockTime const opus_plc_alignment = 2500 * GST_USECOND; GstClockTime aligned_missing_duration; GstClockTime missing_duration = GST_BUFFER_DURATION (bufd); if (!GST_CLOCK_TIME_IS_VALID (missing_duration) || missing_duration == 0) { if (GST_CLOCK_TIME_IS_VALID (dec->last_known_buffer_duration)) { missing_duration = dec->last_known_buffer_duration; GST_WARNING_OBJECT (dec, "Missing duration, using last duration %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration)); } else { GST_WARNING_OBJECT (dec, "Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms"); missing_duration = 20 * GST_MSECOND; } } GST_DEBUG_OBJECT (dec, "missing buffer, doing PLC duration %" GST_TIME_FORMAT " plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration), GST_TIME_ARGS (dec->leftover_plc_duration)); /* add the leftover PLC duration to that of the buffer */ missing_duration += dec->leftover_plc_duration; /* align the combined buffer and leftover PLC duration to multiples * of 2.5ms, rounding to nearest, and store excess duration for later */ aligned_missing_duration = ((missing_duration + opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment; dec->leftover_plc_duration = missing_duration - aligned_missing_duration; /* Opus' PLC cannot operate with less than 2.5ms; skip PLC * and accumulate the missing duration in the leftover_plc_duration * for the next PLC attempt */ if (aligned_missing_duration < opus_plc_alignment) { GST_DEBUG_OBJECT (dec, "current duration %" GST_TIME_FORMAT " of missing data not enough for PLC (minimum needed: %" GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration), GST_TIME_ARGS (opus_plc_alignment)); goto done; } /* convert the duration (in nanoseconds) to sample count */ samples = gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate, GST_SECOND); GST_DEBUG_OBJECT (dec, "calculated PLC frame length: %" GST_TIME_FORMAT " num frame samples: %d new leftover: %" GST_TIME_FORMAT, GST_TIME_ARGS (aligned_missing_duration), samples, GST_TIME_ARGS (dec->leftover_plc_duration)); } else { /* use maximum size (120 ms) as the number of returned samples is not constant over the stream. */ samples = 120 * dec->sample_rate / 1000; } packet_size = samples * dec->n_channels * 2; outbuf = gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec), packet_size); if (!outbuf) { goto buffer_failed; } if (size > 0) dec->last_known_buffer_duration = packet_duration_opus (data, size); gst_buffer_map (outbuf, &omap, GST_MAP_WRITE); out_data = (gint16 *) omap.data; do { if (dec->use_inband_fec) { if (gst_buffer_get_size (dec->last_buffer) > 0) { /* normal delayed decode */ GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer"); n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0); } else { /* FEC reconstruction decode */ GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer"); n = opus_multistream_decode (dec->state, data, size, out_data, samples, 1); } } else { /* normal decode */ GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer"); n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0); } if (n == OPUS_BUFFER_TOO_SMALL) { /* if too small, add 2.5 milliseconds and try again, up to the * Opus max size of 120 milliseconds */ if (samples >= 120 * dec->sample_rate / 1000) break; samples += 25 * dec->sample_rate / 10000; packet_size = samples * dec->n_channels * 2; gst_buffer_unmap (outbuf, &omap); gst_buffer_unref (outbuf); outbuf = gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec), packet_size); if (!outbuf) { goto buffer_failed; } gst_buffer_map (outbuf, &omap, GST_MAP_WRITE); out_data = (gint16 *) omap.data; } } while (n == OPUS_BUFFER_TOO_SMALL); gst_buffer_unmap (outbuf, &omap); if (data != NULL) gst_buffer_unmap (buf, &map); if (n < 0) { GstFlowReturn ret = GST_FLOW_ERROR; gst_buffer_unref (outbuf); GST_AUDIO_DECODER_ERROR (dec, 1, STREAM, DECODE, (NULL), ("Decoding error (%d): %s", n, opus_strerror (n)), ret); return ret; } GST_DEBUG_OBJECT (dec, "decoded %d samples", n); gst_buffer_set_size (outbuf, n * 2 * dec->n_channels); GST_BUFFER_DURATION (outbuf) = samples * GST_SECOND / dec->sample_rate; samples = n; cmeta = gst_buffer_get_audio_clipping_meta (buf); g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT); /* Skip any samples that need skipping */ if (cmeta && cmeta->start) { guint pre_skip = cmeta->start; guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000; guint skip = scaled_pre_skip > n ? n : scaled_pre_skip; guint scaled_skip = skip * 48000 / dec->sample_rate; gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1); GST_INFO_OBJECT (dec, "Skipping %u samples at the beginning (%u at 48000 Hz)", skip, scaled_skip); } if (cmeta && cmeta->end) { guint post_skip = cmeta->end; guint scaled_post_skip = post_skip * dec->sample_rate / 48000; guint skip = scaled_post_skip > n ? n : scaled_post_skip; guint scaled_skip = skip * 48000 / dec->sample_rate; guint outsize = gst_buffer_get_size (outbuf); guint skip_bytes = skip * 2 * dec->n_channels; if (outsize > skip_bytes) outsize -= skip_bytes; else outsize = 0; gst_buffer_resize (outbuf, 0, outsize); GST_INFO_OBJECT (dec, "Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip); } if (gst_buffer_get_size (outbuf) == 0) { gst_buffer_unref (outbuf); outbuf = NULL; } else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) { gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16, dec->n_channels, dec->opus_pos, dec->info.position); } /* Apply gain */ /* Would be better off leaving this to a volume element, as this is a naive conversion that does too many int/float conversions. However, we don't have control over the pipeline... So make it optional if the user program wants to use a volume, but do it by default so the correct volume goes out by default */ if (dec->apply_gain && outbuf && dec->r128_gain) { gsize rsize; unsigned int i, nsamples; double volume = dec->r128_gain_volume; gint16 *samples; gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE); samples = (gint16 *) omap.data; rsize = omap.size; GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume); nsamples = rsize / 2; for (i = 0; i < nsamples; ++i) { int sample = (int) (samples[i] * volume + 0.5); samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample; } gst_buffer_unmap (outbuf, &omap); } if (dec->use_inband_fec) { gst_buffer_replace (&dec->last_buffer, buffer); } res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1); if (res != GST_FLOW_OK) GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res)); done: return res; creation_failed: GST_ELEMENT_ERROR (dec, LIBRARY, INIT, ("Failed to create Opus decoder"), ("Failed to create Opus decoder (%d): %s", err, opus_strerror (err))); return GST_FLOW_ERROR; buffer_failed: GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Failed to create %u byte buffer", packet_size)); return GST_FLOW_ERROR; }
void SoftOpus::onQueueFilled(OMX_U32 /* portIndex */) { List<BufferInfo *> &inQueue = getPortQueue(0); List<BufferInfo *> &outQueue = getPortQueue(1); if (mOutputPortSettingsChange != NONE) { return; } while (!mHaveEOS && !inQueue.empty() && !outQueue.empty()) { BufferInfo *inInfo = *inQueue.begin(); OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; if (mInputBufferCount < 3) { const uint8_t *data = inHeader->pBuffer + inHeader->nOffset; size_t size = inHeader->nFilledLen; if ((inHeader->nFlags & OMX_BUFFERFLAG_EOS) && size == 0) { handleEOS(); return; } if (mInputBufferCount == 0) { CHECK(mHeader == NULL); mHeader = new OpusHeader(); memset(mHeader, 0, sizeof(*mHeader)); if (!ParseOpusHeader(data, size, mHeader)) { ALOGV("Parsing Opus Header failed."); notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); return; } uint8_t channel_mapping[kMaxChannels] = {0}; if (mHeader->channels <= kMaxChannelsWithDefaultLayout) { memcpy(&channel_mapping, kDefaultOpusChannelLayout, kMaxChannelsWithDefaultLayout); } else { memcpy(&channel_mapping, mHeader->stream_map, mHeader->channels); } int status = OPUS_INVALID_STATE; mDecoder = opus_multistream_decoder_create(kRate, mHeader->channels, mHeader->num_streams, mHeader->num_coupled, channel_mapping, &status); if (!mDecoder || status != OPUS_OK) { ALOGV("opus_multistream_decoder_create failed status=%s", opus_strerror(status)); notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); return; } status = opus_multistream_decoder_ctl(mDecoder, OPUS_SET_GAIN(mHeader->gain_db)); if (status != OPUS_OK) { ALOGV("Failed to set OPUS header gain; status=%s", opus_strerror(status)); notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); return; } } else if (mInputBufferCount == 1) { mCodecDelay = ns_to_samples( *(reinterpret_cast<int64_t*>(inHeader->pBuffer + inHeader->nOffset)), kRate); mSamplesToDiscard = mCodecDelay; } else { mSeekPreRoll = ns_to_samples( *(reinterpret_cast<int64_t*>(inHeader->pBuffer + inHeader->nOffset)), kRate); notify(OMX_EventPortSettingsChanged, 1, 0, NULL); mOutputPortSettingsChange = AWAITING_DISABLED; } if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { handleEOS(); return; } inQueue.erase(inQueue.begin()); inInfo->mOwnedByUs = false; notifyEmptyBufferDone(inHeader); ++mInputBufferCount; continue; } // Ignore CSD re-submissions. if (mInputBufferCount >= 3 && (inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) { if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { handleEOS(); return; } inQueue.erase(inQueue.begin()); inInfo->mOwnedByUs = false; notifyEmptyBufferDone(inHeader); continue; } BufferInfo *outInfo = *outQueue.begin(); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; if ((inHeader->nFlags & OMX_BUFFERFLAG_EOS) && inHeader->nFilledLen == 0) { handleEOS(); return; } if (inHeader->nOffset == 0) { mAnchorTimeUs = inHeader->nTimeStamp; mNumFramesOutput = 0; } // When seeking to zero, |mCodecDelay| samples has to be discarded // instead of |mSeekPreRoll| samples (as we would when seeking to any // other timestamp). if (inHeader->nTimeStamp == 0) { mSamplesToDiscard = mCodecDelay; } const uint8_t *data = inHeader->pBuffer + inHeader->nOffset; const uint32_t size = inHeader->nFilledLen; size_t frameSize = kMaxOpusOutputPacketSizeSamples; if (frameSize > outHeader->nAllocLen / sizeof(int16_t) / mHeader->channels) { frameSize = outHeader->nAllocLen / sizeof(int16_t) / mHeader->channels; android_errorWriteLog(0x534e4554, "27833616"); } int numFrames = opus_multistream_decode(mDecoder, data, size, (int16_t *)outHeader->pBuffer, frameSize, 0); if (numFrames < 0) { ALOGE("opus_multistream_decode returned %d", numFrames); notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); return; } outHeader->nOffset = 0; if (mSamplesToDiscard > 0) { if (mSamplesToDiscard > numFrames) { mSamplesToDiscard -= numFrames; numFrames = 0; } else { numFrames -= mSamplesToDiscard; outHeader->nOffset = mSamplesToDiscard * sizeof(int16_t) * mHeader->channels; mSamplesToDiscard = 0; } } outHeader->nFilledLen = numFrames * sizeof(int16_t) * mHeader->channels; outHeader->nTimeStamp = mAnchorTimeUs + (mNumFramesOutput * 1000000ll) / kRate; mNumFramesOutput += numFrames; if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { outHeader->nFlags = OMX_BUFFERFLAG_EOS; mHaveEOS = true; } else { outHeader->nFlags = 0; } inInfo->mOwnedByUs = false; inQueue.erase(inQueue.begin()); notifyEmptyBufferDone(inHeader); ++mInputBufferCount; outInfo->mOwnedByUs = false; outQueue.erase(outQueue.begin()); notifyFillBufferDone(outHeader); } }