Initial working microphone redirection for android.
Added debug messages to alsa and pulse microphone backends.
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bd7845e656
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c2d3f60ad0
@ -165,7 +165,12 @@ static BOOL audin_alsa_thread_receive(AudinALSADevice* alsa, BYTE* src, int size
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if (WaitForSingleObject(alsa->stopEvent, 0) == WAIT_OBJECT_0)
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break;
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else
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{
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DEBUG_DVC("encoded %d [%d] to %d [%X]", alsa->buffer_frames,
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tbytes_per_frame, encoded_size,
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alsa->wformat);
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ret = alsa->receive(encoded_data, encoded_size, alsa->user_data);
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}
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alsa->buffer_frames = 0;
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@ -53,7 +53,6 @@ typedef struct _AudinOpenSLESDevice
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UINT32 bytes_per_channel;
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UINT32 wformat;
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UINT32 format;
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UINT32 block_size;
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@ -84,7 +83,7 @@ static void* audin_opensles_thread_func(void* arg)
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assert(opensles->stopEvent);
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assert(opensles->stream);
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buffer.v = calloc(sizeof(short), opensles->frames_per_packet);
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buffer.v = calloc(opensles->bytes_per_channel, opensles->frames_per_packet);
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ZeroMemory(buffer.v, opensles->frames_per_packet);
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freerdp_dsp_context_reset_adpcm(opensles->dsp_context);
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@ -98,23 +97,22 @@ static void* audin_opensles_thread_func(void* arg)
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if (rc < 0)
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{
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DEBUG_WARN("android_RecIn %d", rc);
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continue;
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}
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DEBUG_DVC("Got %d frames from microphone", opensles->frames_per_packet);
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if (opensles->format == WAVE_FORMAT_ADPCM)
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{
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opensles->dsp_context->encode_ms_adpcm(opensles->dsp_context,
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buffer.b, opensles->frames_per_packet * sizeof(short),
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buffer.b, rc * opensles->bytes_per_channel,
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opensles->channels, opensles->block_size);
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encoded_data = opensles->dsp_context->adpcm_buffer;
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encoded_size = opensles->dsp_context->adpcm_size;
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}
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else if (opensles->format == WAVE_FORMAT_DVI_ADPCM)
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{
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opensles->dsp_context->encode_ima_adpcm(opensles->dsp_context,
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buffer.b, opensles->frames_per_packet * sizeof(short),
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buffer.b, rc * opensles->bytes_per_channel,
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opensles->channels, opensles->block_size);
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encoded_data = opensles->dsp_context->adpcm_buffer;
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@ -123,9 +121,11 @@ static void* audin_opensles_thread_func(void* arg)
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else
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{
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encoded_data = buffer.v;
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encoded_size = opensles->frames_per_packet;
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encoded_size = rc * opensles->bytes_per_channel;
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}
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DEBUG_DVC("encoded %d [%d] to %d [%X]", rc,
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opensles->bytes_per_channel, encoded_size, opensles->format);
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rc = opensles->receive(encoded_data, encoded_size, opensles->user_data);
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if (!rc)
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break;
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@ -172,19 +172,17 @@ static BOOL audin_opensles_format_supported(IAudinDevice* device, audinFormat* f
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switch (format->wFormatTag)
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{
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/*
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case WAVE_FORMAT_PCM:
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case WAVE_FORMAT_PCM: /* PCM */
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if (format->cbSize == 0 &&
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(format->nSamplesPerSec <= 48000) &&
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(format->wBitsPerSample == 8 || format->wBitsPerSample == 16) &&
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(format->nChannels == 1 || format->nChannels == 2))
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(format->nChannels >= 1 && format->nChannels <= 2))
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{
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return TRUE;
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}
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break;
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*/
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case WAVE_FORMAT_ADPCM:
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// case WAVE_FORMAT_DVI_ADPCM:
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// case WAVE_FORMAT_ADPCM: /* IMA ADPCM */
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case WAVE_FORMAT_DVI_ADPCM:
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if ((format->nSamplesPerSec <= 48000) &&
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(format->wBitsPerSample == 4) &&
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(format->nChannels == 1 || format->nChannels == 2))
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@ -192,6 +190,11 @@ static BOOL audin_opensles_format_supported(IAudinDevice* device, audinFormat* f
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return TRUE;
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}
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break;
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default:
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DEBUG_DVC("Encoding '%s' [%08X] not supported",
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rdpsnd_get_audio_tag_string(format->wFormatTag),
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format->wFormatTag);
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break;
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}
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return FALSE;
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@ -220,31 +223,18 @@ static void audin_opensles_set_format(IAudinDevice* device,
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switch (format->wBitsPerSample)
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{
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case 4:
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opensles->format = WAVE_FORMAT_ADPCM;
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opensles->bytes_per_channel = 1;
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break;
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case 8:
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opensles->format = WAVE_FORMAT_PCM;
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opensles->bytes_per_channel = 1;
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break;
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case 16:
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opensles->format = WAVE_FORMAT_ADPCM;
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opensles->bytes_per_channel = 2;
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break;
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}
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break;
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case WAVE_FORMAT_ADPCM:
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opensles->format = WAVE_FORMAT_ADPCM;
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opensles->bytes_per_channel = 2;
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bs = (format->nBlockAlign - 4 * format->nChannels) * 4;
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opensles->frames_per_packet =
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(FramesPerPacket * format->nChannels * 2 /
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bs + 1) * bs / (format->nChannels * 2);
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break;
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case WAVE_FORMAT_DVI_ADPCM:
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opensles->format = WAVE_FORMAT_DVI_ADPCM;
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opensles->bytes_per_channel = 2;
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bs = (format->nBlockAlign - 4 * format->nChannels) * 4;
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@ -252,8 +242,14 @@ static void audin_opensles_set_format(IAudinDevice* device,
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(FramesPerPacket * format->nChannels * 2 /
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bs + 1) * bs / (format->nChannels * 2);
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break;
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case WAVE_FORMAT_ALAW:
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case WAVE_FORMAT_MULAW:
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opensles->frames_per_packet = FramesPerPacket;
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break;
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default:
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DEBUG_WARN("Unsupported fromat %08X requested, ignoring.",
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DEBUG_WARN("Encoding '%s' [%08X] not supported",
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rdpsnd_get_audio_tag_string(format->wFormatTag),
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format->wFormatTag);
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return;
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}
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@ -261,17 +257,11 @@ static void audin_opensles_set_format(IAudinDevice* device,
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opensles->rate = format->nSamplesPerSec;
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opensles->channels = format->nChannels;
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opensles->wformat = format->wFormatTag;
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opensles->format = format->wFormatTag;
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opensles->block_size = format->nBlockAlign;
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if (opensles->stream)
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{
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android_CloseRecDevice(opensles->stream);
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opensles->stream = android_OpenRecDevice(opensles->device_name,
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opensles->rate, opensles->channels,
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opensles->frames_per_packet);
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}
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DEBUG_DVC("aligned frames_per_packet=%d, block_size=%d",
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opensles->frames_per_packet, opensles->block_size);
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}
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static void audin_opensles_open(IAudinDevice* device, AudinReceive receive,
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@ -291,7 +281,9 @@ static void audin_opensles_open(IAudinDevice* device, AudinReceive receive,
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opensles->stream = android_OpenRecDevice(
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opensles->device_name,
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opensles->rate,
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opensles->channels, opensles->frames_per_packet);
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opensles->channels,
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opensles->frames_per_packet,
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opensles->bytes_per_channel * 8);
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assert(opensles->stream);
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opensles->receive = receive;
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@ -314,7 +306,10 @@ static void audin_opensles_close(IAudinDevice* device)
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/* The function may have been called out of order,
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* ignore duplicate requests. */
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if (!opensles->stopEvent)
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{
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DEBUG_WARN("[ERROR] function called without matching open.");
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return;
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}
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assert(opensles->stopEvent);
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assert(opensles->thread);
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@ -134,9 +134,27 @@ static SLresult openSLRecOpen(OPENSL_STREAM *p){
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else
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speakers = SL_SPEAKER_FRONT_CENTER;
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SLDataLocator_AndroidSimpleBufferQueue loc_bq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
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SLDataFormat_PCM format_pcm = {SL_DATAFORMAT_PCM, channels, sr,
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SL_PCMSAMPLEFORMAT_FIXED_16, SL_PCMSAMPLEFORMAT_FIXED_16,
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speakers, SL_BYTEORDER_LITTLEENDIAN};
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SLDataFormat_PCM format_pcm;
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format_pcm.formatType = SL_DATAFORMAT_PCM;
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format_pcm.numChannels = channels;
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format_pcm.samplesPerSec = sr;
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format_pcm.channelMask = speakers;
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format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
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if (16 == p->bits_per_sample)
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{
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format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
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format_pcm.containerSize = 16;
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}
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else if (8 == p->bits_per_sample)
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{
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format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_8;
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format_pcm.containerSize = 8;
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}
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else
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assert(0);
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SLDataSink audioSnk = {&loc_bq, &format_pcm};
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// create audio recorder
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@ -211,7 +229,7 @@ static void openSLDestroyEngine(OPENSL_STREAM *p)
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// open the android audio device for input
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OPENSL_STREAM *android_OpenRecDevice(char *name, int sr, int inchannels,
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int bufferframes)
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int bufferframes, int bits_per_sample)
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{
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OPENSL_STREAM *p;
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@ -221,18 +239,26 @@ OPENSL_STREAM *android_OpenRecDevice(char *name, int sr, int inchannels,
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p->inchannels = inchannels;
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p->sr = sr;
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p->queue = Queue_New(TRUE, -1, -1);
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if(openSLCreateEngine(p) != SL_RESULT_SUCCESS) {
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android_CloseRecDevice(p);
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return NULL;
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}
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if(openSLRecOpen(p) != SL_RESULT_SUCCESS) {
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android_CloseRecDevice(p);
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return NULL;
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}
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p->buffersize = bufferframes;
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p->bits_per_sample = bits_per_sample;
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if ((p->bits_per_sample != 8) && (p->bits_per_sample != 16))
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{
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android_CloseRecDevice(p);
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return NULL;
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}
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if(openSLCreateEngine(p) != SL_RESULT_SUCCESS)
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{
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android_CloseRecDevice(p);
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return NULL;
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}
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if(openSLRecOpen(p) != SL_RESULT_SUCCESS)
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{
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android_CloseRecDevice(p);
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return NULL;
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}
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return p;
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}
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@ -245,12 +271,16 @@ void android_CloseRecDevice(OPENSL_STREAM *p)
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if (p == NULL)
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return;
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if (p->queue)
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{
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while (Queue_Count(p->queue) > 0)
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{
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queue_element *e = Queue_Dequeue(p->queue);
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free(e->data);
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free(e);
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}
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Queue_Free(p->queue);
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}
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if (p->next)
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{
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@ -258,7 +288,6 @@ void android_CloseRecDevice(OPENSL_STREAM *p)
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free(p->next);
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}
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Queue_Free(p->queue);
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openSLDestroyEngine(p);
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free(p);
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@ -276,9 +305,9 @@ void bqRecorderCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
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assert(p);
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assert(p->queue);
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DEBUG_DVC("Signalled");
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printf("Signalled");
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e = calloc(1, sizeof(queue_element));
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e->data = calloc(p->buffersize, sizeof(short));
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e->data = calloc(p->buffersize, p->bits_per_sample / 8);
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e->size = p->buffersize;
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if (p->next)
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@ -306,19 +335,21 @@ int android_RecIn(OPENSL_STREAM *p,short *buffer,int size)
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bqRecorderCallback(p->recorderBufferQueue, p);
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}
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WaitForSingleObject(p->queue->event, INFINITE);
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e = Queue_Dequeue(p->queue);
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if (!e)
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{
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DEBUG_WARN("[ERROR] got e=%p from queue", e);
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return -1;
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}
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rc = e->size;
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rc = (e->size < size) ? e->size : size;
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assert(p->buffersize == e->size);
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memcpy(buffer, e->data, size > e->size ? e->size : size);
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memcpy(buffer, e->data, rc * sizeof(short));
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free(e->data);
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free(e);
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DEBUG_DVC("yay!");
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return size;
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return rc;
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}
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@ -64,6 +64,7 @@ typedef struct opensl_stream {
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unsigned int inchannels;
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unsigned int sr;
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unsigned int buffersize;
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unsigned int bits_per_sample;
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wQueue *queue;
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queue_element *next;
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@ -74,7 +75,7 @@ typedef struct opensl_stream {
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in frames. Returns a handle to the OpenSL stream
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*/
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OPENSL_STREAM* android_OpenRecDevice(char *name, int sr, int inchannels,
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int bufferframes);
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int bufferframes, int bits_per_sample);
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/*
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Close the audio device
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*/
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@ -324,8 +324,6 @@ static void audin_pulse_stream_request_callback(pa_stream* stream, size_t length
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pulse->sample_spec.channels, pulse->block_size);
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encoded_data = pulse->dsp_context->adpcm_buffer;
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encoded_size = pulse->dsp_context->adpcm_size;
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DEBUG_DVC("encoded %d to %d",
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pulse->buffer_frames * pulse->bytes_per_frame, encoded_size);
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}
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else
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{
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@ -333,6 +331,9 @@ static void audin_pulse_stream_request_callback(pa_stream* stream, size_t length
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encoded_size = pulse->buffer_frames * pulse->bytes_per_frame;
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}
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DEBUG_DVC("encoded %d [%d] to %d [%X]",
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pulse->buffer_frames, pulse->bytes_per_frame, encoded_size,
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pulse->format);
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ret = pulse->receive(encoded_data, encoded_size, pulse->user_data);
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pulse->buffer_frames = 0;
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if (!ret)
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