diff options
| -rw-r--r-- | audio/out/ao_coreaudio.c | 649 |
1 files changed, 339 insertions, 310 deletions
diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c index 6389cbec6f..bd2bc34377 100644 --- a/audio/out/ao_coreaudio.c +++ b/audio/out/ao_coreaudio.c @@ -30,9 +30,6 @@ * simpler frameworks do. This is due to the fact that CoreAudio pulls * audio samples rather than having them pushed at it (which is nice * when you are wanting to do good buffering of audio). - * - * AC-3 and MPEG audio passthrough is possible, but has never been tested - * due to lack of a soundcard that supports it. */ #include <CoreServices/CoreServices.h> @@ -49,41 +46,25 @@ #include "core/mp_msg.h" #include "ao.h" -#include "audio_out_internal.h" #include "audio/format.h" #include "osdep/timer.h" #include "libavutil/fifo.h" #include "core/subopt-helper.h" -static const ao_info_t info = -{ - "Darwin/Mac OS X native audio output", - "coreaudio", - "Timothy J. Wood & Dan Christiansen & Chris Roccati", - "" -}; - -LIBAO_EXTERN(coreaudio) +#define ca_msg(a, b, c ...) mp_msg(a, b, "AO: [coreaudio] " c) -/* Prefix for all mp_msg() calls */ -#define ao_msg(a, b, c ...) mp_msg(a, b, "AO: [coreaudio] " c) +static void audio_pause(struct ao *ao); +static void audio_resume(struct ao *ao); +static void reset(struct ao *ao); -#if MAC_OS_X_VERSION_MAX_ALLOWED <= 1040 -/* AudioDeviceIOProcID does not exist in Mac OS X 10.4. We can emulate - * this by using AudioDeviceAddIOProc() and AudioDeviceRemoveIOProc(). */ -#define AudioDeviceIOProcID AudioDeviceIOProc -#define AudioDeviceDestroyIOProcID AudioDeviceRemoveIOProc -static OSStatus AudioDeviceCreateIOProcID(AudioDeviceID dev, - AudioDeviceIOProc proc, - void *data, - AudioDeviceIOProcID *procid) -{ - *procid = proc; - return AudioDeviceAddIOProc(dev, proc, data); -} -#endif +struct ca_ringbuffer { + AVFifoBuffer *fifo; + int len; + int chunks; + int chunk_size; +}; -typedef struct ao_coreaudio_s +struct priv { AudioDeviceID i_selected_dev; /* Keeps DeviceID of the selected device. */ int b_supports_digital; /* Does the currently selected device support digital mode? */ @@ -109,79 +90,103 @@ typedef struct ao_coreaudio_s int packetSize; int paused; - /* Ring-buffer */ - AVFifoBuffer *buffer; - unsigned int buffer_len; ///< must always be num_chunks * chunk_size - unsigned int num_chunks; - unsigned int chunk_size; -} ao_coreaudio_t; + struct ca_ringbuffer *buffer; +}; -static ao_coreaudio_t *ao = NULL; +static struct ca_ringbuffer *new_ca_ringbuffer(void *talloc_ctx, size_t chunks, + size_t chunk_size) +{ + struct ca_ringbuffer *buffer = + talloc_zero(talloc_ctx, struct ca_ringbuffer); + + *buffer = (struct ca_ringbuffer) { + .fifo = av_fifo_alloc(chunks * chunk_size), + .len = chunks * chunk_size, + .chunks = chunks, + .chunk_size = chunk_size, + }; -/** - * \brief add data to ringbuffer - */ -static int write_buffer(unsigned char *data, int len) + return buffer; +} + +static int ca_ringbuffer_buffered(struct ca_ringbuffer *buffer) { - int free = ao->buffer_len - av_fifo_size(ao->buffer); - if (len > free) - len = free; - return av_fifo_generic_write(ao->buffer, data, len, NULL); + return av_fifo_size(buffer->fifo); } -/** - * \brief remove data from ringbuffer - */ -static int read_buffer(unsigned char *data, int len) +static void ca_ringbuffer_reset(struct ca_ringbuffer *buffer) +{ + av_fifo_reset(buffer->fifo); +} + +static int ca_ringbuffer_read(struct ca_ringbuffer *buffer, + unsigned char *data, int len) { - int buffered = av_fifo_size(ao->buffer); + int buffered = ca_ringbuffer_buffered(buffer); if (len > buffered) len = buffered; if (data) - av_fifo_generic_read(ao->buffer, data, len, NULL); + av_fifo_generic_read(buffer->fifo, data, len, NULL); else - av_fifo_drain(ao->buffer, len); + av_fifo_drain(buffer->fifo, len); return len; } +static int ca_ringbuffer_write(struct ca_ringbuffer *buffer, + unsigned char *data, int len) +{ + int free = buffer->len - av_fifo_size(buffer->fifo); + if (len > free) + len = free; + return av_fifo_generic_write(buffer->fifo, data, len, NULL); +} + static OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData) { - int amt = av_fifo_size(ao->buffer); - int req = (inNumFrames) * ao->packetSize; + struct ao *ao = inRefCon; + struct priv *p = ao->priv; - if (amt > req) - amt = req; + int buffered = ca_ringbuffer_buffered(p->buffer); + int requested = inNumFrames * p->packetSize; - if (amt) - read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt); - else - audio_pause(); - ioData->mBuffers[0].mDataByteSize = amt; + if (buffered > requested) + buffered = requested; + + if (buffered) { + ca_ringbuffer_read(p->buffer, + (unsigned char *)ioData->mBuffers[0].mData, + buffered); + } else { + audio_pause(ao); + } + + ioData->mBuffers[0].mDataByteSize = buffered; return noErr; } -static int control(int cmd, void *arg) +static int control(struct ao *ao, enum aocontrol cmd, void *arg) { + struct priv *p = ao->priv; ao_control_vol_t *control_vol; OSStatus err; Float32 vol; switch (cmd) { case AOCONTROL_GET_VOLUME: control_vol = (ao_control_vol_t *)arg; - if (ao->b_digital) { + if (p->b_digital) { // Digital output has no volume adjust. - int vol = ao->b_muted ? 0 : 100; + int vol = p->b_muted ? 0 : 100; *control_vol = (ao_control_vol_t) { .left = vol, .right = vol, }; return CONTROL_TRUE; } - err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, + err = AudioUnitGetParameter(p->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol); if (err == 0) { @@ -189,7 +194,7 @@ static int control(int cmd, void *arg) control_vol->left = control_vol->right = vol * 100.0 / 4.0; return CONTROL_TRUE; } else { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err); return CONTROL_FALSE; } @@ -197,7 +202,7 @@ static int control(int cmd, void *arg) case AOCONTROL_SET_VOLUME: control_vol = (ao_control_vol_t *)arg; - if (ao->b_digital) { + if (p->b_digital) { // Digital output can not set volume. Here we have to return true // to make mixer forget it. Else mixer will add a soft filter, // that's not we expected and the filter not support ac3 stream @@ -206,20 +211,20 @@ static int control(int cmd, void *arg) // Although not support set volume, but at least we support mute. // MPlayer set mute by set volume to zero, we handle it. if (control_vol->left == 0 && control_vol->right == 0) - ao->b_muted = 1; + p->b_muted = 1; else - ao->b_muted = 0; + p->b_muted = 0; return CONTROL_TRUE; } vol = (control_vol->left + control_vol->right) * 4.0 / 200.0; - err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, + err = AudioUnitSetParameter(p->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0); if (err == 0) { // printf("SET VOL=%f\n", vol); return CONTROL_TRUE; } else { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err); return CONTROL_FALSE; } @@ -235,7 +240,7 @@ static void print_format(int lev, const char *str, const AudioStreamBasicDescription *f) { uint32_t flags = (uint32_t) f->mFormatFlags; - ao_msg(MSGT_AO, lev, + ca_msg(MSGT_AO, lev, "%s %7.1fHz %" PRIu32 "bit [%c%c%c%c][%" PRIu32 "][%" PRIu32 "][%" PRIu32 "][%" PRIu32 "][%" PRIu32 "] %s %s %s%s%s%s\n", str, f->mSampleRate, f->mBitsPerChannel, (int)(f->mFormatID & 0xff000000) >> 24, @@ -368,7 +373,7 @@ static Boolean IsAudioPropertySettable(AudioObjectID id, static int AudioDeviceSupportsDigital(AudioDeviceID i_dev_id); static int AudioStreamSupportsDigital(AudioStreamID i_stream_id); -static int OpenSPDIF(void); +static int OpenSPDIF(struct ao *ao); static int AudioStreamChangeFormat(AudioStreamID i_stream_id, AudioStreamBasicDescription change_format); static OSStatus RenderCallbackSPDIF(AudioDeviceID inDevice, @@ -436,8 +441,12 @@ static void print_help(void) free(devids); } -static int init(int rate, const struct mp_chmap *channels, int format, int flags) +static int init(struct ao *ao, char *params) { + // int rate, int channels, int format, int flags) + struct priv *p = talloc_zero(ao, struct priv); + ao->priv = p; + AudioStreamBasicDescription inDesc; AudioComponentDescription desc; AudioComponent comp; @@ -463,24 +472,22 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags return 0; } - ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", - rate, ao_data.channels.num, af_fmt2str_short(format), flags); + ca_msg(MSGT_AO, MSGL_V, "init([%dHz][%dch][%s][%d])\n", + ao->samplerate, ao->channels.num, af_fmt2str_short(ao->format), 0); - ao = calloc(1, sizeof(ao_coreaudio_t)); + p->i_selected_dev = 0; + p->b_supports_digital = 0; + p->b_digital = 0; + p->b_muted = 0; + p->b_stream_format_changed = 0; + p->i_hog_pid = -1; + p->i_stream_id = 0; + p->i_stream_index = -1; + p->b_revert = 0; + p->b_changed_mixing = 0; - ao->i_selected_dev = 0; - ao->b_supports_digital = 0; - ao->b_digital = 0; - ao->b_muted = 0; - ao->b_stream_format_changed = 0; - ao->i_hog_pid = -1; - ao->i_stream_id = 0; - ao->i_stream_index = -1; - ao->b_revert = 0; - ao->b_changed_mixing = 0; - - global_ao->per_application_mixer = true; - global_ao->no_persistent_volume = true; + ao->per_application_mixer = true; + ao->no_persistent_volume = true; if (device_id == 0) { /* Find the ID of the default Device. */ @@ -488,60 +495,61 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags kAudioHardwarePropertyDefaultOutputDevice, sizeof(UInt32), &devid_def); if (err != noErr) { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err); goto err_out; } - } else + } else { devid_def = device_id; + } /* Retrieve the name of the device. */ err = GetAudioPropertyString(devid_def, kAudioObjectPropertyName, &psz_name); if (err != noErr) { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err); goto err_out; } - ao_msg(MSGT_AO, MSGL_V, + ca_msg(MSGT_AO, MSGL_V, "got audio output device ID: %" PRIu32 " Name: %s\n", devid_def, psz_name); /* Probe whether device support S/PDIF stream output if input is AC3 stream. */ - if (AF_FORMAT_IS_AC3(format)) { + if (AF_FORMAT_IS_AC3(ao->format)) { if (AudioDeviceSupportsDigital(devid_def)) - ao->b_supports_digital = 1; - ao_msg(MSGT_AO, MSGL_V, + p->b_supports_digital = 1; + ca_msg(MSGT_AO, MSGL_V, "probe default audio output device about support for digital s/pdif output: %d\n", - ao->b_supports_digital); + p->b_supports_digital); } free(psz_name); // Save selected device id - ao->i_selected_dev = devid_def; + p->i_selected_dev = devid_def; struct mp_chmap_sel chmap_sel = {0}; mp_chmap_sel_add_waveext(&chmap_sel); - if (!ao_chmap_sel_adjust(&ao_data, &chmap_sel, &ao_data.channels)) + if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels)) goto err_out; // Build Description for the input format - inDesc.mSampleRate = rate; + inDesc.mSampleRate = ao->samplerate; inDesc.mFormatID = - ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM; - inDesc.mChannelsPerFrame = ao_data.channels.num; - inDesc.mBitsPerChannel = af_fmt2bits(format); + p->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM; + inDesc.mChannelsPerFrame = ao->channels.num; + inDesc.mBitsPerChannel = af_fmt2bits(ao->format); - if ((format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F) { + if ((ao->format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F) { // float inDesc.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked; - } else if ((format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI) { + } else if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI) { // signed int inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; @@ -549,47 +557,47 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags // unsigned int inDesc.mFormatFlags = kAudioFormatFlagIsPacked; } - if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE) + if ((ao->format & AF_FORMAT_END_MASK) == AF_FORMAT_BE) inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian; inDesc.mFramesPerPacket = 1; - ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = + p->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket * - ao_data.channels.num * + ao->channels.num * (inDesc.mBitsPerChannel / 8); print_format(MSGL_V, "source:", &inDesc); - if (ao->b_supports_digital) { + if (p->b_supports_digital) { b_alive = 1; - err = GetAudioProperty(ao->i_selected_dev, + err = GetAudioProperty(p->i_selected_dev, kAudioDevicePropertyDeviceIsAlive, sizeof(UInt32), &b_alive); if (err != noErr) - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err); if (!b_alive) - ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n"); + ca_msg(MSGT_AO, MSGL_WARN, "device is not alive\n"); /* S/PDIF output need device in HogMode. */ - err = GetAudioProperty(ao->i_selected_dev, + err = GetAudioProperty(p->i_selected_dev, kAudioDevicePropertyHogMode, - sizeof(pid_t), &ao->i_hog_pid); + sizeof(pid_t), &p->i_hog_pid); if (err != noErr) { /* This is not a fatal error. Some drivers simply don't support this property. */ - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n", (char *)&err); - ao->i_hog_pid = -1; + p->i_hog_pid = -1; } - if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid()) { - ao_msg(MSGT_AO, MSGL_WARN, + if (p->i_hog_pid != -1 && p->i_hog_pid != getpid()) { + ca_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n"); goto err_out; } - ao->stream_format = inDesc; - return OpenSPDIF(); + p->stream_format = inDesc; + return OpenSPDIF(ao); } /* original analog output code */ @@ -603,106 +611,102 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags comp = AudioComponentFindNext(NULL, &desc); //Finds an component that meets the desc spec's if (comp == NULL) { - ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n"); + ca_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n"); goto err_out; } - err = AudioComponentInstanceNew(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component + err = AudioComponentInstanceNew(comp, &(p->theOutputUnit)); //gains access to the services provided by the component if (err) { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err); goto err_out; } // Initialize AudioUnit - err = AudioUnitInitialize(ao->theOutputUnit); + err = AudioUnitInitialize(p->theOutputUnit); if (err) { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err); goto err_out1; } size = sizeof(AudioStreamBasicDescription); - err = AudioUnitSetProperty(ao->theOutputUnit, + err = AudioUnitSetProperty(p->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size); if (err) { - ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", + ca_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err); goto err_out2; } size = sizeof(UInt32); - err = AudioUnitGetProperty(ao->theOutputUnit, + err = AudioUnitGetProperty(p->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size); if (err) { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err); goto err_out2; } //Set the Current Device to the Default Output Unit. - err = AudioUnitSetProperty(ao->theOutputUnit, + err = AudioUnitSetProperty(p->theOutputUnit, kAudioOutputUnitProperty_CurrentDevice, - kAudioUnitScope_Global, 0, &ao->i_selected_dev, - sizeof(ao->i_selected_dev)); + kAudioUnitScope_Global, 0, &p->i_selected_dev, + sizeof(p->i_selected_dev)); - ao->chunk_size = maxFrames; //*inDesc.mBytesPerFrame; + ao->samplerate = inDesc.mSampleRate; - ao_data.samplerate = inDesc.mSampleRate; - if (!ao_chmap_sel_get_def(&ao_data, &chmap_sel, &ao_data.channels, + if (!ao_chmap_sel_get_def(ao, &chmap_sel, &ao->channels, inDesc.mChannelsPerFrame)) goto err_out2; - ao_data.channels.num = inDesc.mChannelsPerFrame; - ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame; - ao_data.outburst = ao->chunk_size; - ao_data.buffersize = ao_data.bps; - ao->num_chunks = (ao_data.bps + ao->chunk_size - 1) / ao->chunk_size; - ao->buffer_len = ao->num_chunks * ao->chunk_size; - ao->buffer = av_fifo_alloc(ao->buffer_len); + ao->bps = ao->samplerate * inDesc.mBytesPerFrame; + ao->buffersize = ao->bps; + int chunk_size = maxFrames; //*inDesc.mBytesPerFrame; + int chunks = (ao->bps + chunk_size - 1) / chunk_size; + p->buffer = new_ca_ringbuffer(p, chunks, chunk_size); + ao->outburst = chunk_size; - ao_msg(MSGT_AO, MSGL_V, - "using %5d chunks of %d bytes (buffer len %d bytes)\n", - (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len); + ca_msg(MSGT_AO, MSGL_V, + "using %d chunks of %d bytes (buffer len %d bytes)\n", + p->buffer->chunks, p->buffer->chunk_size, p->buffer->len); renderCallback.inputProc = theRenderProc; - renderCallback.inputProcRefCon = 0; - err = AudioUnitSetProperty(ao->theOutputUnit, + renderCallback.inputProcRefCon = ao; + err = AudioUnitSetProperty(p->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct)); if (err) { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err); goto err_out2; } - reset(); + reset(ao); return CONTROL_OK; err_out2: - AudioUnitUninitialize(ao->theOutputUnit); + AudioUnitUninitialize(p->theOutputUnit); err_out1: - AudioComponentInstanceDispose(ao->theOutputUnit); + AudioComponentInstanceDispose(p->theOutputUnit); err_out: - av_fifo_free(ao->buffer); - free(ao); - ao = NULL; return CONTROL_FALSE; } /***************************************************************************** * Setup a encoded digital stream (SPDIF) *****************************************************************************/ -static int OpenSPDIF(void) +static int OpenSPDIF(struct ao *ao) { + struct priv *p = ao->priv; OSStatus err = noErr; UInt32 i_param_size, b_mix = 0; Boolean b_writeable = 0; @@ -711,18 +715,18 @@ static int OpenSPDIF(void) AudioObjectPropertyAddress property_address; /* Start doing the SPDIF setup process. */ - ao->b_digital = 1; + p->b_digital = 1; /* Hog the device. */ - ao->i_hog_pid = getpid(); + p->i_hog_pid = getpid(); - err = SetAudioProperty(ao->i_selected_dev, + err = SetAudioProperty(p->i_selected_dev, kAudioDevicePropertyHogMode, - sizeof(ao->i_hog_pid), &ao->i_hog_pid); + sizeof(p->i_hog_pid), &p->i_hog_pid); if (err != noErr) { - ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", + ca_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err); - ao->i_hog_pid = -1; + p->i_hog_pid = -1; goto err_out; } @@ -731,44 +735,44 @@ static int OpenSPDIF(void) property_address.mElement = kAudioObjectPropertyElementMaster; /* Set mixable to false if we are allowed to. */ - if (AudioObjectHasProperty(ao->i_selected_dev, &property_address)) { + if (AudioObjectHasProperty(p->i_selected_dev, &property_address)) { /* Set mixable to false if we are allowed to. */ - err = IsAudioPropertySettable(ao->i_selected_dev, + err = IsAudioPropertySettable(p->i_selected_dev, kAudioDevicePropertySupportsMixing, &b_writeable); - err = GetAudioProperty(ao->i_selected_dev, + err = GetAudioProperty(p->i_selected_dev, kAudioDevicePropertySupportsMixing, sizeof(UInt32), &b_mix); if (err == noErr && b_writeable) { b_mix = 0; - err = SetAudioProperty(ao->i_selected_dev, + err = SetAudioProperty(p->i_selected_dev, kAudioDevicePropertySupportsMixing, sizeof(UInt32), &b_mix); - ao->b_changed_mixing = 1; + p->b_changed_mixing = 1; } if (err != noErr) { - ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", + ca_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); goto err_out; } } /* Get a list of all the streams on this device. */ - i_param_size = GetAudioPropertyArray(ao->i_selected_dev, + i_param_size = GetAudioPropertyArray(p->i_selected_dev, kAudioDevicePropertyStreams, kAudioDevicePropertyScopeOutput, (void **)&p_streams); if (!i_param_size) { - ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n"); + ca_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n"); goto err_out; } i_streams = i_param_size / sizeof(AudioStreamID); - ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams); + ca_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams); - for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i) { + for (i = 0; i < i_streams && p->i_stream_index < 0; ++i) { /* Find a stream with a cac3 stream. */ AudioStreamRangedDescription *p_format_list = NULL; int i_formats = 0, j = 0, b_digital = 0; @@ -778,7 +782,7 @@ static int OpenSPDIF(void) (void **)&p_format_list); if (!i_param_size) { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "Could not get number of stream formats.\n"); continue; } @@ -802,23 +806,23 @@ static int OpenSPDIF(void) int i_current_rate_format = -1; int i_backup_rate_format = -1; - ao->i_stream_id = p_streams[i]; - ao->i_stream_index = i; + p->i_stream_id = p_streams[i]; + p->i_stream_index = i; - if (ao->b_revert == 0) { + if (p->b_revert == 0) { /* Retrieve the original format of this stream first if not done so already. */ - err = GetAudioProperty(ao->i_stream_id, + err = GetAudioProperty(p->i_stream_id, kAudioStreamPropertyPhysicalFormat, - sizeof(ao->sfmt_revert), - &ao->sfmt_revert); + sizeof(p->sfmt_revert), + &p->sfmt_revert); if (err != noErr) { - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "Could not retrieve the original stream format: [%4.4s]\n", (char *)&err); free(p_format_list); continue; } - ao->b_revert = 1; + p->b_revert = 1; } for (j = 0; j < i_formats; ++j) @@ -828,12 +832,12 @@ static int OpenSPDIF(void) kAudioFormat60958AC3 || p_format_list[j].mFormat.mFormatID == kAudioFormatAC3) { if (p_format_list[j].mFormat.mSampleRate == - ao->stream_format.mSampleRate) { + p->stream_format.mSampleRate) { i_requested_rate_format = j; break; } if (p_format_list[j].mFormat.mSampleRate == - ao->sfmt_revert.mSampleRate) + p->sfmt_revert.mSampleRate) i_current_rate_format = j; else if (i_backup_rate_format < 0 || p_format_list[j].mFormat.mSampleRate > @@ -843,40 +847,40 @@ static int OpenSPDIF(void) } if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */ - ao->stream_format = + p->stream_format = p_format_list[i_requested_rate_format].mFormat; else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */ - ao->stream_format = + p->stream_format = p_format_list[i_current_rate_format].mFormat; else - ao->stream_format = p_format_list[i_backup_rate_format].mFormat; + p->stream_format = p_format_list[i_backup_rate_format].mFormat; /* And if we have to, any digital format will be just fine (highest rate possible). */ } free(p_format_list); } free(p_streams); - if (ao->i_stream_index < 0) { - ao_msg(MSGT_AO, MSGL_WARN, + if (p->i_stream_index < 0) { + ca_msg(MSGT_AO, MSGL_WARN, "Cannot find any digital output stream format when OpenSPDIF().\n"); goto err_out; } - print_format(MSGL_V, "original stream format:", &ao->sfmt_revert); + print_format(MSGL_V, "original stream format:", &p->sfmt_revert); - if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format)) + if (!AudioStreamChangeFormat(p->i_stream_id, p->stream_format)) goto err_out; property_address.mSelector = kAudioDevicePropertyDeviceHasChanged; property_address.mScope = kAudioObjectPropertyScopeGlobal; property_address.mElement = kAudioObjectPropertyElementMaster; - err = AudioObjectAddPropertyListener(ao->i_selected_dev, + err = AudioObjectAddPropertyListener(p->i_selected_dev, &property_address, DeviceListener, NULL); if (err != noErr) - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err); @@ -884,80 +888,78 @@ static int OpenSPDIF(void) /* FIXME: If output stream is not native byte-order, we need change endian somewhere. */ /* Although there's no such case reported. */ #if BYTE_ORDER == BIG_ENDIAN - if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)) + if (!(p->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)) #else /* tell mplayer that we need a byteswap on AC3 streams, */ - if (ao->stream_format.mFormatID & kAudioFormat60958AC3) - ao_data.format = AF_FORMAT_AC3_LE; + if (p->stream_format.mFormatID & kAudioFormat60958AC3) + ao->format = AF_FORMAT_AC3_LE; - if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian) + if (p->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian) #endif - ao_msg(MSGT_AO, MSGL_WARN, + ca_msg(MSGT_AO, MSGL_WARN, "Output stream has non-native byte order, digital output may fail.\n"); - /* For ac3/dts, just use packet size 6144 bytes as chunk size. */ - ao->chunk_size = ao->stream_format.mBytesPerPacket; + ao->samplerate = p->stream_format.mSampleRate; + mp_chmap_from_channels(&ao->channels, p->stream_format.mChannelsPerFrame); + ao->bps = ao->samplerate * + (p->stream_format.mBytesPerPacket / + p->stream_format.mFramesPerPacket); - ao_data.samplerate = ao->stream_format.mSampleRate; - mp_chmap_from_channels(&ao_data.channels, ao->stream_format.mChannelsPerFrame); - ao_data.bps = ao_data.samplerate * - (ao->stream_format.mBytesPerPacket / - ao->stream_format.mFramesPerPacket); - ao_data.outburst = ao->chunk_size; - ao_data.buffersize = ao_data.bps; + /* For ac3/dts, just use packet size 6144 bytes as chunk size. */ + int chunk_size = p->stream_format.mBytesPerPacket; + int chunks = (ao->bps + chunk_size - 1) / chunk_size; + ao->outburst = chunk_size; + ao->buffersize = ao->bps; - ao->num_chunks = (ao_data.bps + ao->chunk_size - 1) / ao->chunk_size; - ao->buffer_len = ao->num_chunks * ao->chunk_size; - ao->buffer = av_fifo_alloc(ao->buffer_len); + p->buffer->chunks = (ao->bps + p->buffer->chunk_size - 1) / p->buffer->chunk_size; + p->buffer->len = p->buffer->chunks * p->buffer->chunk_size; + p->buffer = new_ca_ringbuffer(p, chunks, chunk_size); - ao_msg(MSGT_AO, MSGL_V, + ca_msg(MSGT_AO, MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", - (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len); + p->buffer->chunks, p->buffer->chunk_size, p->buffer->len); /* Create IOProc callback. */ - err = AudioDeviceCreateIOProcID(ao->i_selected_dev, + err = AudioDeviceCreateIOProcID(p->i_selected_dev, (AudioDeviceIOProc)RenderCallbackSPDIF, (void *)ao, - &ao->renderCallback); + &p->renderCallback); - if (err != noErr || ao->renderCallback == NULL) { - ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", + if (err != noErr || p->renderCallback == NULL) { + ca_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err); goto err_out1; } - reset(); + reset(ao); return CONTROL_TRUE; err_out1: - if (ao->b_revert) - AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert); + if (p->b_revert) + AudioStreamChangeFormat(p->i_stream_id, p->sfmt_revert); err_out: - if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != + if (p->b_changed_mixing && p->sfmt_revert.mFormatID != kAudioFormat60958AC3) { int b_mix = 1; - err = SetAudioProperty(ao->i_selected_dev, + err = SetAudioProperty(p->i_selected_dev, kAudioDevicePropertySupportsMixing, sizeof(int), &b_mix); if (err != noErr) - ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", + ca_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err); } - if (ao->i_hog_pid == getpid()) { - ao->i_hog_pid = -1; - err = SetAudioProperty(ao->i_selected_dev, + if (p->i_hog_pid == getpid()) { + p->i_hog_pid = -1; + err = SetAudioProperty(p->i_selected_dev, kAudioDevicePropertyHogMode, - sizeof(ao->i_hog_pid), &ao->i_hog_pid); + sizeof(p->i_hog_pid), &p->i_hog_pid); if (err != noErr) - ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", + ca_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err); } - av_fifo_free(ao->buffer); - free(ao); - ao = NULL; return CONTROL_FALSE; } @@ -978,7 +980,7 @@ static int AudioDeviceSupportsDigital(AudioDeviceID i_dev_id) (void **)&p_streams); if (!i_param_size) { - ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n"); + ca_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n"); return CONTROL_FALSE; } @@ -1008,7 +1010,7 @@ static int AudioStreamSupportsDigital(AudioStreamID i_stream_id) (void **)&p_format_list); if (!i_param_size) { - ao_msg(MSGT_AO, MSGL_WARN, "Could not get number of stream formats.\n"); + ca_msg(MSGT_AO, MSGL_WARN, "Could not get number of stream formats.\n"); return CONTROL_FALSE; } @@ -1053,7 +1055,7 @@ static int AudioStreamChangeFormat(AudioStreamID i_stream_id, StreamListener, (void *)&stream_format_changed); |